Cannot make an external call

November 28, 2014, 7:23 pm

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Debug enabled in logger.conf, core set verbose 5, core set debug 5 and sip set debug on.

However, this will produce a lot of output, so if you can see an obvious primary failure with lower levels, go for that.

Your sip show peers shows a lot of unregistered devices, but the logging isn't of a sufficient level to tell whether you are trying to access one of those, and, if so, which.

Statistics : Posted by cafe6 • on Fri Nov 28, 2014 9:23 pm • Replies 7 • Views 480

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Cannot make an external call

November 28, 2014, 7:23 pm

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I'm using number 117
Below are logs when i type command
asterisk -r:

Code: [root@localhost ~]# asterisk -r
Asterisk 11.13.0, Copyright (C) 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 11.13.0 currently running on localhost (pid = 3425)
[2014-12-03 07:46:48] WARNING[3812][C-0000000e]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2014-12-03 07:46:52] WARNING[3812][C-0000000e]: channel.c:4860 ast_prod: Prodding channel 'SIP/117-0000000e' failed

SIP show peers:

Code: localhost*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
100 (Unspecified) D No No A 0 UNKNOWN
101/101 192.168.0.101 D No No A 5062 OK (52 ms)
102/102 192.168.0.102 D No No A 5062 OK (57 ms)
103/103 192.168.0.103 D No No A 5062 OK (58 ms)
104/104 192.168.0.104 D No No A 5062 OK (53 ms)
105/105 192.168.0.105 D No No A 5062 OK (56 ms)
106 (Unspecified) D No No A 0 UNKNOWN
107 (Unspecified) D No No A 0 UNKNOWN
108/108 192.168.0.108 D No No A 5062 OK (161 ms)
109/109 192.168.0.109 D No No A 5062 OK (52 ms)
110/110 192.168.0.110 D No No A 5062 OK (57 ms)
111/111 192.168.0.111 D No No A 5062 OK (56 ms)
112/112 192.168.0.112 D No No A 5060 OK (15 ms)
113/113 192.168.0.162 D No No A 5062 OK (57 ms)
114/114 192.168.0.114 D No No A 5062 OK (190 ms)
115 (Unspecified) D No No A 0 UNKNOWN
116/116 192.168.0.116 D No No A 5062 OK (56 ms)
117/117 192.168.0.117 D No No A 5060 OK (15 ms)
118 (Unspecified) D No No A 0 UNKNOWN
119 (Unspecified) D No No A 0 UNKNOWN
120/120 192.168.0.120 D No No A 5062 OK (54 ms)
121/121 192.168.0.121 D No No A 5062 OK (55 ms)
122/122 192.168.0.176 D No No A 5062 OK (56 ms)
123/123 192.168.0.123 D No No A 5062 OK (58 ms)
222 (Unspecified) D No No A 0 UNKNOWN
401/401 private ip D No No A 5062 OK (57 ms)
402 (Unspecified) D No No A 0 UNKNOWN
501 (Unspecified) D No No A 0 UNKNOWN
789/789 (Unspecified) D No No A 0 UNKNOWN
987/987 private ip D No No A 39372 OK (13 ms)
trunk private ip Yes Yes 5060 OK (80 ms)
31 sip peers [Monitored: 21 online, 10 offline Unmonitored: 0 online, 0 offline]
localhost*CLI>

Statistics : Posted by cafe6 • on Fri Nov 28, 2014 9:23 pm • Replies 7 • Views 480

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Cannot make an external call

November 28, 2014, 7:23 pm

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It appears that your connection to your SIP provider is lagging out every now and then. Try increasing the qualify or disabling it altogether. Also you may want to run "sip show peers" when this problem occurs to check on the trunk status.

Statistics : Posted by cafe6 • on Fri Nov 28, 2014 9:23 pm • Replies 7 • Views 480

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Cannot make an external call

November 28, 2014, 7:23 pm

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I install Elastix 2.5.0 Stable 32bit: http://www.elastix.org/index.php/en/downloads.html
I still do not found reason for problem. Please help me.
I can not call external phone number. But, if external phone number call me (or other internal phone number), then i can call out external.

Statistics : Posted by cafe6 • on Fri Nov 28, 2014 9:23 pm • Replies 7 • Views 480

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Cannot make an external call

November 28, 2014, 7:23 pm

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The problem lies in sip.conf (which includes all files included from it by FreePBX and possibly users.conf. Given the behaviour, my guess is you have qualify=yes, but something is discarding the OPTION requests, but it could be that you some how have the peer configured as dynamic.

Please try and prune down FreePBX traces to the minimum, as we are not FreePBX experts here and have to trawl through the whole lot to find anything relevant.

Statistics : Posted by cafe6 • on Fri Nov 28, 2014 9:23 pm • Replies 7 • Views 480

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Cannot make an external call

November 28, 2014, 7:23 pm

≫ Next: SIP --> Packets being sent to different IP

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I have a problem but our supplier still not find out the reason
- All the contact internal still work normal
- All the resource of main Voip still work normal
Problem is I cannot make an external call (supplier cannot receive our signal)
Only when there have a call from someone outside by this public number, we can call out right away.
This problem happen frequently, and not follow any rule and time.
I already restarted sever VOIP, moderm but still not get result.
Hope we can receive your support soon.
Below is log everytime problem happen

Code: [root@localhost ~]# asterisk -vvvr
Asterisk 11.13.0, Copyright (C) 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 11.13.0 currently running on localhost (pid = 12470)
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [19001570@from-internal:1] Macro("SIP/117-00000066", "user-callerid,LIMIT,EXTERNAL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/117-00000066", "TOUCH_MONITOR=1417222744.102") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/117-00000066", "AMPUSER=117") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/117-00000066", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/117-00000066", "1?Set(REALCALLERIDNUM=117)") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/117-00000066", "AMPUSER=117") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/117-00000066", "0?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/117-00000066", "AMPUSERCIDNAME=ROOM CS") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/117-00000066", "0?report") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/117-00000066", "AMPUSERCID=117") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/117-00000066", "__DIAL_OPTIONS=tr") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/117-00000066", "CALLERID(all)="ROOM CS" <117>") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("SIP/117-00000066", "0?limit") in new stack
-- Executing [s@macro-user-callerid:13] ExecIf("SIP/117-00000066", "1?Set(GROUP(concurrency_limit)=117)") in new stack
-- Executing [s@macro-user-callerid:14] ExecIf("SIP/117-00000066", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:15] GotoIf("SIP/117-00000066", "1?continue") in new stack
-- Goto (macro-user-callerid,s,28)
-- Executing [s@macro-user-callerid:28] Set("SIP/117-00000066", "CALLERID(number)=117") in new stack
-- Executing [s@macro-user-callerid:29] Set("SIP/117-00000066", "CALLERID(name)=ROOM CS") in new stack
-- Executing [s@macro-user-callerid:30] Set("SIP/117-00000066", "CDR(cnum)=117") in new stack
-- Executing [s@macro-user-callerid:31] Set("SIP/117-00000066", "CDR(cnam)=ROOM CS") in new stack
-- Executing [s@macro-user-callerid:32] Set("SIP/117-00000066", "CHANNEL(language)=en") in new stack
-- Executing [19001570@from-internal:2] Set("SIP/117-00000066", "MOHCLASS=default") in new stack
-- Executing [19001570@from-internal:3] Set("SIP/117-00000066", "_NODEST=") in new stack
-- Executing [19001570@from-internal:4] Gosub("SIP/117-00000066", "sub-record-check,s,1(out,19001570,)") in new stack
-- Executing [s@sub-record-check:1] Set("SIP/117-00000066", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [s@sub-record-check:2] GotoIf("SIP/117-00000066", "1?check") in new stack
-- Goto (sub-record-check,s,7)
-- Executing [s@sub-record-check:7] Set("SIP/117-00000066", "__MON_FMT=wav") in new stack
-- Executing [s@sub-record-check:8] GotoIf("SIP/117-00000066", "1?next") in new stack
-- Goto (sub-record-check,s,11)
-- Executing [s@sub-record-check:11] ExecIf("SIP/117-00000066", "0?Return()") in new stack
-- Executing [s@sub-record-check:12] ExecIf("SIP/117-00000066", "0?Set(__REC_POLICY_MODE=)") in new stack
-- Executing [s@sub-record-check:13] GotoIf("SIP/117-00000066", "0?out,1") in new stack
-- Executing [s@sub-record-check:14] Set("SIP/117-00000066", "__REC_STATUS=INITIALIZED") in new stack
-- Executing [s@sub-record-check:15] Set("SIP/117-00000066", "NOW=1417222744") in new stack
-- Executing [s@sub-record-check:16] Set("SIP/117-00000066", "__DAY=29") in new stack
-- Executing [s@sub-record-check:17] Set("SIP/117-00000066", "__MONTH=11") in new stack
-- Executing [s@sub-record-check:18] Set("SIP/117-00000066", "__YEAR=2014") in new stack
-- Executing [s@sub-record-check:19] Set("SIP/117-00000066", "__TIMESTR=20141129-075904") in new stack
-- Executing [s@sub-record-check:20] Set("SIP/117-00000066", "__FROMEXTEN=117") in new stack
-- Executing [s@sub-record-check:21] Set("SIP/117-00000066", "__CALLFILENAME=out-19001570-117-20141129-075904-1417222744.102") in new stack
-- Executing [s@sub-record-check:22] Goto("SIP/117-00000066", "out,1") in new stack
-- Goto (sub-record-check,out,1)
-- Executing [out@sub-record-check:1] ExecIf("SIP/117-00000066", "1?Set(__REC_POLICY_MODE=dontcare)") in new stack
-- Executing [out@sub-record-check:2] GosubIf("SIP/117-00000066", "0?record,1(exten,19001570,117)") in new stack
-- Executing [out@sub-record-check:3] Return("SIP/117-00000066", "") in new stack
-- Executing [19001570@from-internal:5] Macro("SIP/117-00000066", "dialout-trunk,2,19001570,,off") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/117-00000066", "DIAL_TRUNK=2") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/117-00000066", "0?sub-pincheck,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/117-00000066", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/117-00000066", "DIAL_NUMBER=19001570") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/117-00000066", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/117-00000066", "OUTBOUND_GROUP=OUT_2") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/117-00000066", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/117-00000066", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/117-00000066", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/117-00000066", "outbound-callerid,2") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/117-00000066", "0?Set(CALLERPRES()=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/117-00000066", "0?Set(REALCALLERIDNUM=117)") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/117-00000066", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/117-00000066", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/117-00000066", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/117-00000066", "TRUNKOUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/117-00000066", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,14)
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/117-00000066", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/117-00000066", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:16] ExecIf("SIP/117-00000066", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:17] ExecIf("SIP/117-00000066", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
-- Executing [s@macro-outbound-callerid:18] Set("SIP/117-00000066", "CDR(outbound_cnum)=117") in new stack
-- Executing [s@macro-outbound-callerid:19] Set("SIP/117-00000066", "CDR(outbound_cnam)=ROOM CS") in new stack
-- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/117-00000066", "0?sub-flp-2,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/117-00000066", "OUTNUM=19001570") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/117-00000066", "custom=SIP/out") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/117-00000066", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
-- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/117-00000066", "0?Set(DIAL_TRUNK_OPTIONS=M(confirm))") in new stack
-- Executing [s@macro-dialout-trunk:17] Macro("SIP/117-00000066", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/117-00000066", "") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/117-00000066", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/117-00000066", "1?Set(CONNECTEDLINE(num,i)=19001570)") in new stack
-- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/117-00000066", "1?Set(CONNECTEDLINE(name,i)=CID:117)") in new stack
-- Executing [s@macro-dialout-trunk:21] GotoIf("SIP/117-00000066", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:22] Dial("SIP/117-00000066", "SIP/out/19001570,300,") in new stack
[2014-11-29 07:59:04] WARNING[14854][C-0000005f]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-trunk:23] NoOp("SIP/117-00000066", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 20") in new stack
-- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/117-00000066", "0?continue,1:s-CHANUNAVAIL,1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/117-00000066", "RC=20") in new stack
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/117-00000066", "20,1") in new stack
-- Goto (macro-dialout-trunk,20,1)
-- Executing [20@macro-dialout-trunk:1] Goto("SIP/117-00000066", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/117-00000066", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 20 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:2] Set("SIP/117-00000066", "CALLERID(number)=117") in new stack
-- Executing [19001570@from-internal:6] Macro("SIP/117-00000066", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/117-00000066", "") in new stack
-- Executing [s@macro-outisbusy:2] GotoIf("SIP/117-00000066", "0?emergency,1") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("SIP/117-00000066", "0?intracompany,1") in new stack
-- Executing [s@macro-outisbusy:4] Playback("SIP/117-00000066", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
-- <SIP/117-00000066> Playing 'all-circuits-busy-now.gsm' (language 'en')
-- <SIP/117-00000066> Playing 'pls-try-call-later.gsm' (language 'en')
-- Executing [s@macro-outisbusy:5] Congestion("SIP/117-00000066", "20") in new stack
[2014-11-29 07:59:08] WARNING[14854][C-0000005f]: channel.c:4860 ast_prod: Prodding channel 'SIP/117-00000066' failed
== Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/117-00000066' in macro 'outisbusy'
== Spawn extension (from-internal, 19001570, 6) exited non-zero on 'SIP/117-00000066'
-- Executing [h@from-internal:1] Hangup("SIP/117-00000066", "") in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/117-00000066'
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [0909358543@from-internal:1] Macro("SIP/112-00000067", "user-callerid,LIMIT,EXTERNAL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/112-00000067", "TOUCH_MONITOR=1417222761.103") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/112-00000067", "AMPUSER=112") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/112-00000067", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/112-00000067", "1?Set(REALCALLERIDNUM=112)") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/112-00000067", "AMPUSER=112") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/112-00000067", "0?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/112-00000067", "AMPUSERCIDNAME=ROOM CS CNC") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/112-00000067", "0?report") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/112-00000067", "AMPUSERCID=112") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/112-00000067", "__DIAL_OPTIONS=tr") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/112-00000067", "CALLERID(all)="ROOM CS CNC" <112>") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("SIP/112-00000067", "0?limit") in new stack
-- Executing [s@macro-user-callerid:13] ExecIf("SIP/112-00000067", "1?Set(GROUP(concurrency_limit)=112)") in new stack
-- Executing [s@macro-user-callerid:14] ExecIf("SIP/112-00000067", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:15] GotoIf("SIP/112-00000067", "1?continue") in new stack
-- Goto (macro-user-callerid,s,28)
-- Executing [s@macro-user-callerid:28] Set("SIP/112-00000067", "CALLERID(number)=112") in new stack
-- Executing [s@macro-user-callerid:29] Set("SIP/112-00000067", "CALLERID(name)=ROOM CS CNC") in new stack
-- Executing [s@macro-user-callerid:30] Set("SIP/112-00000067", "CDR(cnum)=112") in new stack
-- Executing [s@macro-user-callerid:31] Set("SIP/112-00000067", "CDR(cnam)=ROOM CS CNC") in new stack
-- Executing [s@macro-user-callerid:32] Set("SIP/112-00000067", "CHANNEL(language)=en") in new stack
-- Executing [0909358543@from-internal:2] Set("SIP/112-00000067", "MOHCLASS=default") in new stack
-- Executing [0909358543@from-internal:3] Set("SIP/112-00000067", "_NODEST=") in new stack
-- Executing [0909358543@from-internal:4] Gosub("SIP/112-00000067", "sub-record-check,s,1(out,0909358543,)") in new stack
-- Executing [s@sub-record-check:1] Set("SIP/112-00000067", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [s@sub-record-check:2] GotoIf("SIP/112-00000067", "1?check") in new stack
-- Goto (sub-record-check,s,7)
-- Executing [s@sub-record-check:7] Set("SIP/112-00000067", "__MON_FMT=wav") in new stack
-- Executing [s@sub-record-check:8] GotoIf("SIP/112-00000067", "1?next") in new stack
-- Goto (sub-record-check,s,11)
-- Executing [s@sub-record-check:11] ExecIf("SIP/112-00000067", "0?Return()") in new stack
-- Executing [s@sub-record-check:12] ExecIf("SIP/112-00000067", "0?Set(__REC_POLICY_MODE=)") in new stack
-- Executing [s@sub-record-check:13] GotoIf("SIP/112-00000067", "0?out,1") in new stack
-- Executing [s@sub-record-check:14] Set("SIP/112-00000067", "__REC_STATUS=INITIALIZED") in new stack
-- Executing [s@sub-record-check:15] Set("SIP/112-00000067", "NOW=1417222761") in new stack
-- Executing [s@sub-record-check:16] Set("SIP/112-00000067", "__DAY=29") in new stack
-- Executing [s@sub-record-check:17] Set("SIP/112-00000067", "__MONTH=11") in new stack
-- Executing [s@sub-record-check:18] Set("SIP/112-00000067", "__YEAR=2014") in new stack
-- Executing [s@sub-record-check:19] Set("SIP/112-00000067", "__TIMESTR=20141129-075921") in new stack
-- Executing [s@sub-record-check:20] Set("SIP/112-00000067", "__FROMEXTEN=112") in new stack
-- Executing [s@sub-record-check:21] Set("SIP/112-00000067", "__CALLFILENAME=out-0909358543-112-20141129-075921-1417222761.103") in new stack
-- Executing [s@sub-record-check:22] Goto("SIP/112-00000067", "out,1") in new stack
-- Goto (sub-record-check,out,1)
-- Executing [out@sub-record-check:1] ExecIf("SIP/112-00000067", "1?Set(__REC_POLICY_MODE=dontcare)") in new stack
-- Executing [out@sub-record-check:2] GosubIf("SIP/112-00000067", "0?record,1(exten,0909358543,112)") in new stack
-- Executing [out@sub-record-check:3] Return("SIP/112-00000067", "") in new stack
-- Executing [0909358543@from-internal:5] Macro("SIP/112-00000067", "dialout-trunk,2,0909358543,,off") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/112-00000067", "DIAL_TRUNK=2") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/112-00000067", "0?sub-pincheck,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/112-00000067", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/112-00000067", "DIAL_NUMBER=0909358543") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/112-00000067", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/112-00000067", "OUTBOUND_GROUP=OUT_2") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/112-00000067", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/112-00000067", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/112-00000067", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/112-00000067", "outbound-callerid,2") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/112-00000067", "0?Set(CALLERPRES()=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/112-00000067", "0?Set(REALCALLERIDNUM=112)") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/112-00000067", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/112-00000067", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/112-00000067", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/112-00000067", "TRUNKOUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/112-00000067", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,14)
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/112-00000067", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/112-00000067", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:16] ExecIf("SIP/112-00000067", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:17] ExecIf("SIP/112-00000067", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
-- Executing [s@macro-outbound-callerid:18] Set("SIP/112-00000067", "CDR(outbound_cnum)=112") in new stack
-- Executing [s@macro-outbound-callerid:19] Set("SIP/112-00000067", "CDR(outbound_cnam)=ROOM CS CNC") in new stack
-- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/112-00000067", "0?sub-flp-2,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/112-00000067", "OUTNUM=0909358543") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/112-00000067", "custom=SIP/out") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/112-00000067", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
-- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/112-00000067", "0?Set(DIAL_TRUNK_OPTIONS=M(confirm))") in new stack
-- Executing [s@macro-dialout-trunk:17] Macro("SIP/112-00000067", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/112-00000067", "") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/112-00000067", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:19] ExecIf("SIP/112-00000067", "1?Set(CONNECTEDLINE(num,i)=0909358543)") in new stack
-- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/112-00000067", "1?Set(CONNECTEDLINE(name,i)=CID:112)") in new stack
-- Executing [s@macro-dialout-trunk:21] GotoIf("SIP/112-00000067", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:22] Dial("SIP/112-00000067", "SIP/out/0909358543,300,") in new stack
[2014-11-29 07:59:21] WARNING[14855][C-00000060]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-trunk:23] NoOp("SIP/112-00000067", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 20") in new stack
-- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/112-00000067", "0?continue,1:s-CHANUNAVAIL,1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/112-00000067", "RC=20") in new stack
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/112-00000067", "20,1") in new stack
-- Goto (macro-dialout-trunk,20,1)
-- Executing [20@macro-dialout-trunk:1] Goto("SIP/112-00000067", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/112-00000067", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 20 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:2] Set("SIP/112-00000067", "CALLERID(number)=112") in new stack
-- Executing [0909358543@from-internal:6] Macro("SIP/112-00000067", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/112-00000067", "") in new stack
-- Executing [s@macro-outisbusy:2] GotoIf("SIP/112-00000067", "0?emergency,1") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("SIP/112-00000067", "0?intracompany,1") in new stack
-- Executing [s@macro-outisbusy:4] Playback("SIP/112-00000067", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
-- <SIP/112-00000067> Playing 'all-circuits-busy-now.gsm' (language 'en')
-- <SIP/112-00000067> Playing 'pls-try-call-later.gsm' (language 'en')
-- Executing [s@macro-outisbusy:5] Congestion("SIP/112-00000067", "20") in new stack
[2014-11-29 07:59:25] WARNING[14855][C-00000060]: channel.c:4860 ast_prod: Prodding channel 'SIP/112-00000067' failed
== Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/112-00000067' in macro 'outisbusy'
== Spawn extension (from-internal, 0909358543, 6) exited non-zero on 'SIP/112-00000067'
-- Executing [h@from-internal:1] Hangup("SIP/112-00000067", "") in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/112-00000067'
localhost*CLI>

Statistics : Posted by cafe6 • on Fri Nov 28, 2014 9:23 pm • Replies 7 • Views 480

↧

SIP --> Packets being sent to different IP

December 8, 2014, 12:56 am

≫ Next: Video Stream before answering the call

≪ Previous: Cannot make an external call

The problem that I am having is that when I try to connect my SIP client on my cell phone to my asterisk server it registers fine. However when trying to monitor the RTP ports that are being sent from the server to the phone. I keep getting an IP that I don't recognize in my logs. This leads to a problem hearing calls from any of the extensions or outbound calls.

Help please!!!

Here is a copy of my sip.conf:

Code: [104]
defaultuser=104
secret=edited
type=friend
callerid="Boys Room"
host=dynamic
context=users
qualify=no                ; NAT keep-alive
dtmfmode=rfc2833          ; either RFC2833 or INFO for the BudgeTone
directmediadeny=0.0.0.0/0    ; Use directmediapermit and directmediadeny to restrict
mailbox=105@default

[105]
defaultuser=105
secret=edited
type=friend
callerid="Kitchen"
host=dynamic
context=users
qualify=no                ; NAT keep-alive
dtmfmode=rfc2833          ; either RFC2833 or INFO for the BudgeTone
directmediadeny=0.0.0.0/0    ; Use directmediapermit and directmediadeny to restrict
mailbox=105@default

[110]
defaultuser=110
secret=edited
type=friend
callerid="Dads Cell"
host=dynamic
context=users
qualify=no                ; NAT keep-alive
;directmediadeny=0.0.0.0/0    ; Use directmediapermit and directmediadeny to restrict
mailbox=105@default
nat=no

[OOMA]
defaultuser=OOMA
fromuser=OOMA
secret=edited
type=friend
disallow=all
allow=ulaw
allow=alaw
allow=gsm
context=pstn-in
qualify=yes
host=dynamic

Here is a copy of the verbose (verbose 3) from the asterisk CLI:

Code: =~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2014.12.08 01:40:18 =~=~=~=~=~=~=~=~=~=~=~=
login as: root
DD-WRT v24-sp2 std (c) 2014 NewMedia-NET GmbH
Release: 01/10/14 (SVN revision: 23320)
root@192.168.1.1's password:
==========================================================

____ ___ __ ______ _____ ____ _ _
| _ \| _ \ \ \ / / _ \_ _| __ _|___ \| || |
|| | || ||____\ \ /\ / /| |_) || | \ \ / / __) | || |_
||_| ||_||_____\ V V / | _ < | | \ V / / __/|__ _|
|___/|___/ \_/\_/ |_| \_\|_| \_/ |_____| |_|

DD-WRT v24-sp2
http://www.dd-wrt.com

==========================================================

BusyBox v1.22.0 (2014-01-10 03:53:14 CET) built-in shell (ash)
Enter 'help' for a list of built-in commands.

[01;31mroot@DD-WRT [01;34m~ $ [00masterisk -0[Jr
Asterisk 1.8.30.0, Copyright (C) 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.8.30.0 currently running on DD-WRT (pid = 3273)
DD-WRT*CLI>
[0KVerbosity is at least 5

[KDD-WRT*CLI> core show help

DD-WRT*CLI>
[0K ! Execute a shell command
aoc set debug enable cli debugging of AOC messages
cc cancel Kill a CC transaction
cc report status Reports CC stats
cdr show status Display the CDR status
cel show status Display the CEL status
channel request hangup Request a hangup on a given channel
cli check permissions Try a permissions config for a user
cli reload permissions Reload CLI permissions config
cli show permissions Show CLI permissions
config list Show all files that have loaded a configuration file
config reload Force a reload on modules using a particular configuration file
core abort shutdown Cancel a running shutdown
core ping taskprocessor Ping a named task processor
core reload Global reload
core restart gracefully Restart Asterisk gracefully
core restart now Restart Asterisk immediately
core restart when convenient Restart Asterisk at empty call volume
core set debug channel Enable/disable debugging on a channel
core set {debug|verbose} Set level of debug/verbose chattiness
core show applications [like|d Shows registered dialplan applications
core show application Describe a specific dialplan application
core show calls [uptime] Display information on calls
core show channels [concise|ve Display information on channels
core show channel Display information on a specific channel
core show channeltypes List available channel types
core show channeltype Give more details on that channel type
core show codecs [audio|video| Displays a list of codecs
core show codec Shows a specific codec
core show config mappings Display config mappings (file names to config engines)
core show file formats Displays file formats
core show functions [like] Shows registered dialplan functions
core show function Describe a specific dialplan function
core show help Display help list, or specific help on a command
core show hints Show dialplan hints
core show hint Show dialplan hint
core show image formats Displays image formats
core show license Show the license(s) for this copy of Asterisk
core show switches Show alternative switches
core show taskprocessors List instantiated task processors and statistics
core show translation Display translation matrix
core show uptime [seconds] Show uptime information
core show version Display version info
core show warranty Show the warranty (if any) for this copy of Asterisk
core stop gracefully Gracefully shut down Asterisk
core stop now Shut down Asterisk immediately
core stop when convenient Shut down Asterisk at empty call volume
core waitfullybooted Wait for Asterisk to be fully booted
data get Data API get
data show providers Show data providers
database del Removes database key/value
database deltree Removes database keytree/values
database get Gets database value
database put Adds/updates database value
database show Shows database contents
database showkey Shows database contents
dialplan add extension Add new extension into context
dialplan add ignorepat Add new ignore pattern
dialplan add include Include context in other context
dialplan debug Show fast extension pattern matching data structures
dialplan reload Reload extensions and *only* extensions
dialplan remove extension Remove a specified extension
dialplan remove ignorepat Remove ignore pattern from context
dialplan remove include Remove a specified include from context
dialplan save Save current dialplan into a file
dialplan set chanvar Set a channel variable
dialplan set extenpatternmatch Use the Old extension pattern matching algorithm.
dialplan set extenpatternmatch Use the New extension pattern matching algorithm.
dialplan set global Set global dialplan variable
dialplan show chanvar Show channel variables
dialplan show globals Show global dialplan variables
dialplan show Show dialplan
dnsmgr refresh Performs an immediate refresh
dnsmgr reload Reloads the DNS manager configuration
dnsmgr status Display the DNS manager status

[KDD-WRT*CLI>
[0K event dump cache Dump the internal event cache (for debugging)
features reload Reloads configured features
features show Lists configured features
group show channels Display active channels with group(s)
http show status Display HTTP server status
indication add Add the given indication to the country
indication remove Remove the given indication from the country
indication show Display a list of all countries/indications
logger mute Toggle logging output to a console
logger reload Reopens the log files
logger rotate Rotates and reopens the log files
logger set level {DEBUG|NOTICE Enables/Disables a specific logging level for this console
logger show channels List configured log channels
manager reload Reload manager configurations
manager set debug [on|off] Show, enable, disable debugging of the manager code
manager show command Show a manager interface command
manager show commands List manager interface commands
manager show connected List connected manager interface users
manager show eventq List manager interface queued events
manager show settings Show manager global settings
manager show users List configured manager users
manager show user Display information on a specific manager user
mixmonitor {start|stop} Execute a MixMonitor command
module load Load a module by name
module reload Reload configuration for a module
module show [like] List modules and info
module unload Unload a module by name
no debug channel Disable debugging on channel(s)
parkedcalls show List currently parked calls
rtcp set debug {on|off|ip} Enable/Disable RTCP debugging
rtcp set stats {on|off} Enable/Disable RTCP stats
rtp set debug {on|off|ip} Enable/Disable RTP debugging
say load [new|old] Set or show the say mode
sip notify Send a notify packet to a SIP peer
sip prune realtime [peer|all] Prune cached Realtime users/peers
sip qualify peer Send an OPTIONS packet to a peer
sip reload Reload SIP configuration
sip set debug {on|off|ip|peer} Enable/Disable SIP debugging
sip set history {on|off} Enable/Disable SIP history
sip show {channels|subscriptio List active SIP channels or subscriptions
sip show channelstats List statistics for active SIP channels
sip show channel Show detailed SIP channel info
sip show domains List our local SIP domains
sip show history Show SIP dialog history
sip show inuse List all inuse/limits
sip show mwi Show MWI subscriptions
sip show objects List all SIP object allocations
sip show peers List defined SIP peers
sip show peer Show details on specific SIP peer
sip show registry List SIP registration status
sip show sched Present a report on the status of the scheduler queue
sip show settings Show SIP global settings
sip show tcp List TCP Connections
sip show users List defined SIP users
sip show user Show details on specific SIP user
sip unregister Unregister (force expiration) a SIP peer from the registry
stun set debug {on|off} Enable/Disable STUN debugging
timing test Run a timing test
udptl set debug {on|off|ip} Enable/Disable UDPTL debugging
voicemail reload Reload voicemail configuration
voicemail show users List defined voicemail boxes
voicemail show zones List zone message formats

[KDD-WRT*CLI> core set bv[K[Kverbose 5[K34[K

DD-WRT*CLI>
[0KVerbosity was 5 and is now 3

[KDD-WRT*CLI> sip set debug on

DD-WRT*CLI>
[0KSIP Debugging enabled

[KDD-WRT*CLI> r
[0K
<--- SIP read from UDP:192.168.1.132:5060 --->
ªªªª
<------------->

[KDD-WRT*CLI> rtp set debug on

DD-WRT*CLI>
[0KRTP Debugging Enabled

[KDD-WRT*CLI> sip show peers

DD-WRT*CLI>
[0KName/username Host Dyn Forcerport ACL Port Status
103/103 192.168.1.132 D N 5060 Unmonitored
104/104 192.168.1.146 D N 5060 Unmonitored
105/105 192.168.1.144 D N 5060 Unmonitored
110/110 66.87.65.6 D N 7167 Unmonitored
OOMA/OOMA 192.168.1.115 D N 5061 OK (8 ms)
5 sip peers [Monitored: 1 online, 0 offline Unmonitored: 4 online, 0 offline]

[KDD-WRT*CLI>
[0KReliably Transmitting (NAT) to 192.168.1.115:5061:
OPTIONS sip:OOMA@192.168.1.115:5061 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK313540e8;rport

Max-Forwards: 70

From: "asterisk" <sip:OOMA@192.168.1.1>;tag=as00411e3a

To: <sip:OOMA@192.168.1.115:5061>

Contact: <sip:OOMA@192.168.1.1:5060>

Call-ID: 18596f122e1579cb4ebfc0b6028f0d4e@192.168.1.1:5060

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX 1.8.30.0

Date: Mon, 08 Dec 2014 01:41:26 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Length: 0

---

[KDD-WRT*CLI>
[0K
<--- SIP read from UDP:192.168.1.115:5061 --->
SIP/2.0 200 OK
To: <sip:OOMA@192.168.1.115:5061>;tag=db64ee9e1a9ce5cbi1
From: "asterisk" <sip:OOMA@192.168.1.1>;tag=as00411e3a
Call-ID: 18596f122e1579cb4ebfc0b6028f0d4e@192.168.1.1:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK313540e8
Server: Linksys/SPA3102-5.1.7(GW)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '18596f122e1579cb4ebfc0b6028f0d4e@192.168.1.1:5060' Method: OPTIONS

[KDD-WRT*CLI>
[0K
<--- SIP read from UDP:66.87.65.6:7167 --->

<------------->

[KDD-WRT*CLI> sip [K[K[K
[0K
<--- SIP read from UDP:66.87.65.6:7167 --->

<------------->

[KDD-WRT*CLI>
[0K
<--- SIP read from UDP:66.87.65.6:7167 --->
INVITE sip:444@50.123.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 29.152.33.6:39705;rport;branch=z9hG4bKPj7G.mmZov1F4rxIM3OBUxTbWkhYlRWple
Max-Forwards: 70
From: "YouKnowWho" <sip:110@50.123.xxx.xxx>;tag=pBlcOp6BiUyfsOg4CszLxfgzN5yTK2Wk
To: <sip:444@50.123.xxx.xxx>
Contact: <sip:110@66.87.65.6:7167;ob>
Call-ID: BlZxnc9dIb7Djhdgp0eMMFtPWkKDUVp4
CSeq: 5310 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_hltespr-19/r2457
Content-Type: application/sdp
Content-Length: 358

v=0
o=- 3627016916 3627016916 IN IP4 29.152.33.6
s=pjmedia
c=IN IP4 29.152.33.6
t=0 0
m=audio 4004 RTP/AVP 96 3 0 8 101
c=IN IP4 29.152.33.6
a=rtcp:4005 IN IP4 29.152.33.6
a=sendrecv
a=rtpmap:96 SILK/8000
a=fmtp:96 useinbandfec=0
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
--- (15 headers 16 lines) ---
Sending to 66.87.65.6:7167 (NAT)
Using INVITE request as basis request - BlZxnc9dIb7Djhdgp0eMMFtPWkKDUVp4
Found peer '110' for '110' from 66.87.65.6:7167

<--- Reliably Transmitting (NAT) to 66.87.65.6:7167 --->
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 29.152.33.6:39705;branch=z9hG4bKPj7G.mmZov1F4rxIM3OBUxTbWkhYlRWple;received=66.87.65.6;rport=7167

From: "YouKnowWho" <sip:110@50.123.xxx.xxx>;tag=pBlcOp6BiUyfsOg4CszLxfgzN5yTK2Wk

To: <sip:444@50.123.xxx.xxx>;tag=as41ae53d5

Call-ID: BlZxnc9dIb7Djhdgp0eMMFtPWkKDUVp4

CSeq: 5310 INVITE

Server: Asterisk PBX 1.8.30.0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="03637f2e"

Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'BlZxnc9dIb7Djhdgp0eMMFtPWkKDUVp4' in 32000 ms (Method: INVITE)

[KDD-WRT*CLI>
[0K
<--- SIP read from UDP:66.87.65.6:7167 --->
ACK sip:444@50.123.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 29.152.33.6:39705;rport;branch=z9hG4bKPj7G.mmZov1F4rxIM3OBUxTbWkhYlRWple
Max-Forwards: 70
From: "YouKnowWho" <sip:110@50.123.xxx.xxx>;tag=pBlcOp6BiUyfsOg4CszLxfgzN5yTK2Wk
To: <sip:444@50.123.xxx.xxx>;tag=as41ae53d5
Call-ID: BlZxnc9dIb7Djhdgp0eMMFtPWkKDUVp4
CSeq: 5310 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

[KDD-WRT*CLI>
[0K
<--- SIP read from UDP:66.87.65.6:7167 --->
INVITE sip:444@50.123.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 29.152.33.6:39705;rport;branch=z9hG4bKPjRZ8mecihqO5GV.tij-jCbk5AMTm5Y3Rn
Max-Forwards: 70
From: "YouKnowWho" <sip:110@50.123.xxx.xxx>;tag=pBlcOp6BiUyfsOg4CszLxfgzN5yTK2Wk
To: <sip:444@50.123.xxx.xxx>
Contact: <sip:110@66.87.65.6:7167;ob>
Call-ID: BlZxnc9dIb7Djhdgp0eMMFtPWkKDUVp4
CSeq: 5311 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_hltespr-19/r2457
Authorization: Digest username="110", realm="asterisk", nonce="03637f2e", uri="sip:444@50.123.xxx.xxx", response="972550c9ee5fb3bdfb92ad07a576281b", algorithm=MD5
Content-Type: application/sdp
Content-Length: 358

v=0
o=- 3627016916 3627016916 IN IP4 29.152.33.6
s=pjmedia
c=IN IP4 29.152.33.6
t=0 0
m=audio 4004 RTP/AVP 96 3 0 8 101
c=IN IP4 29.152.33.6
a=rtcp:4005 IN IP4 29.152.33.6
a=sendrecv
a=rtpmap:96 SILK/8000
a=fmtp:96 useinbandfec=0
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->

[KDD-WRT*CLI>
[0K--- (16 headers 16 lines) ---
Sending to 66.87.65.6:7167 (NAT)
Using INVITE request as basis request - BlZxnc9dIb7Djhdgp0eMMFtPWkKDUVp4

[KDD-WRT*CLI>
[0KFound peer '110' for '110' from 66.87.65.6:7167

[KDD-WRT*CLI>
[0K == Using SIP RTP CoS mark 5

[KDD-WRT*CLI>
[0KFound RTP audio format 96

[KDD-WRT*CLI>
[0KFound RTP audio format 3

[KDD-WRT*CLI>
[0KFound RTP audio format 0

[KDD-WRT*CLI>
[0KFound RTP audio format 8

[KDD-WRT*CLI>
[0KFound RTP audio format 101

[KDD-WRT*CLI>
[0KFound unknown media description format SILK for ID 96

[KDD-WRT*CLI>
[0KFound audio description format GSM for ID 3

[KDD-WRT*CLI>
[0KFound audio description format PCMU for ID 0

[KDD-WRT*CLI>
[0KFound audio description format PCMA for ID 8

[KDD-WRT*CLI>
[0KFound audio description format telephone-event for ID 101

[KDD-WRT*CLI>
[0KCapabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)

[KDD-WRT*CLI>
[0KNon-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)

[KDD-WRT*CLI>
[0KPeer audio RTP is at port 29.152.33.6:4004

[KDD-WRT*CLI>
[0KLooking for 444 in users (domain 50.123.xxx.xxx)

[KDD-WRT*CLI>
[0Klist_route: hop: <sip:110@66.87.65.6:7167;ob>

[KDD-WRT*CLI>
[0K
<--- Transmitting (NAT) to 66.87.65.6:7167 --->
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 29.152.33.6:39705;branch=z9hG4bKPjRZ8mecihqO5GV.tij-jCbk5AMTm5Y3Rn;received=66.87.65.6;rport=7167

From: "YouKnowWho" <sip:110@50.123.xxx.xxx>;tag=pBlcOp6BiUyfsOg4CszLxfgzN5yTK2Wk

To: <sip:444@50.123.xxx.xxx>

Call-ID: BlZxnc9dIb7Djhdgp0eMMFtPWkKDUVp4

CSeq: 5311 INVITE

Server: Asterisk PBX 1.8.30.0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: <sip:444@50.123.xxx.xxx:5060>

Content-Length: 0

<------------>

[KDD-WRT*CLI>
[0K -- Executing [444@users:1] [1;36mVoiceMailMain[0m("[1;35mSIP/110-00000021[0m", "[1;35mdefault[0m") in new stack

[KDD-WRT*CLI>
[0KAudio is at 12010

[KDD-WRT*CLI>
[0KAdding codec 0x2 (gsm) to SDP

[KDD-WRT*CLI>
[0KAdding codec 0x4 (ulaw) to SDP

[KDD-WRT*CLI>
[0KAdding codec 0x8 (alaw) to SDP

[KDD-WRT*CLI>
[0KAdding non-codec 0x1 (telephone-event) to SDP

[KDD-WRT*CLI>
[0K
<--- Reliably Transmitting (NAT) to 66.87.65.6:7167 --->
SIP/2.0 200 OK

Via: SIP/2.0/UDP 29.152.33.6:39705;branch=z9hG4bKPjRZ8mecihqO5GV.tij-jCbk5AMTm5Y3Rn;received=66.87.65.6;rport=7167

From: "YouKnowWho" <sip:110@50.123.xxx.xxx>;tag=pBlcOp6BiUyfsOg4CszLxfgzN5yTK2Wk

To: <sip:444@50.123.xxx.xxx>;tag=as16fb6bec

Call-ID: BlZxnc9dIb7Djhdgp0eMMFtPWkKDUVp4

CSeq: 5311 INVITE

Server: Asterisk PBX 1.8.30.0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: <sip:444@50.123.xxx.xxx:5060>

Content-Type: application/sdp

Require: timer

Content-Length: 311

v=0

o=root 153541190 153541190 IN IP4 50.123.xxx.xxx

s=Asterisk PBX 1.8.30.0

c=IN IP4 50.123.xxx.xxx

t=0 0

m=audio 12010 RTP/AVP 3 0 8 101

a=rtpmap:3 GSM/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

<------------>

[KDD-WRT*CLI>
[0K
<--- SIP read from UDP:66.87.65.6:7167 --->
ACK sip:444@50.123.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP 29.152.33.6:39705;rport;branch=z9hG4bKPjtABjaU6cCglUvxW7jQVsX3FIM0swyy1Y
Max-Forwards: 70
From: "YouKnowWho" <sip:110@50.123.xxx.xxx>;tag=pBlcOp6BiUyfsOg4CszLxfgzN5yTK2Wk
To: <sip:444@50.123.xxx.xxx>;tag=as16fb6bec
Call-ID: BlZxnc9dIb7Djhdgp0eMMFtPWkKDUVp4
CSeq: 5311 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

[KDD-WRT*CLI>
[0K
<--- SIP read from UDP:66.87.65.6:7167 --->
INVITE sip:444@50.123.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP 29.152.33.6:39705;rport;branch=z9hG4bKPjdHQbcYv.dWpTPDlzhaxP23-gr50SSy7o
Max-Forwards: 70
From: "YouKnowWho" <sip:110@50.123.xxx.xxx>;tag=pBlcOp6BiUyfsOg4CszLxfgzN5yTK2Wk
To: <sip:444@50.123.xxx.xxx>;tag=as16fb6bec
Contact: <sip:110@66.87.65.6:7167;ob>
Call-ID: BlZxnc9dIb7Djhdgp0eMMFtPWkKDUVp4
CSeq: 5312 INVITE
Authorization: Digest username="110", realm="asterisk", nonce="03637f2e", uri="sip:444@50.123.xxx.xxx", response="972550c9ee5fb3bdfb92ad07a576281b", algorithm=MD5
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 258

v=0
o=- 3627016916 3627016917 IN IP4 29.152.33.6
s=pjmedia
c=IN IP4 29.152.33.6
t=0 0
m=audio 4004 RTP/AVP 3 101
c=IN IP4 29.152.33.6
a=rtcp:4005 IN IP4 29.152.33.6
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->

[KDD-WRT*CLI>
[0K--- (15 headers 12 lines) ---

[KDD-WRT*CLI>
[0KSending to 66.87.65.6:7167 (NAT)

[KDD-WRT*CLI>
[0KFound RTP audio format 3

[KDD-WRT*CLI>
[0KFound RTP audio format 101

[KDD-WRT*CLI>
[0KFound audio description format GSM for ID 3

[KDD-WRT*CLI>
[0KFound audio description format telephone-event for ID 101

[KDD-WRT*CLI>
[0KCapabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x2 (gsm)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm)

[KDD-WRT*CLI>
[0KNon-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)

[KDD-WRT*CLI>
[0KPeer audio RTP is at port 29.152.33.6:4004

[KDD-WRT*CLI>
[0K
<--- Transmitting (NAT) to 66.87.65.6:7167 --->
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 29.152.33.6:39705;branch=z9hG4bKPjdHQbcYv.dWpTPDlzhaxP23-gr50SSy7o;received=66.87.65.6;rport=7167

From: "YouKnowWho" <sip:110@50.123.xxx.xxx>;tag=pBlcOp6BiUyfsOg4CszLxfgzN5yTK2Wk

To: <sip:444@50.123.xxx.xxx>;tag=as16fb6bec

Call-ID: BlZxnc9dIb7Djhdgp0eMMFtPWkKDUVp4

CSeq: 5312 INVITE

Server: Asterisk PBX 1.8.30.0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: <sip:444@50.123.xxx.xxx:5060>

Content-Length: 0

<------------>

[KDD-WRT*CLI>
[0KAudio is at 12010

[KDD-WRT*CLI>
[0KAdding codec 0x2 (gsm) to SDP

[KDD-WRT*CLI>
[0KAdding non-codec 0x1 (telephone-event) to SDP

[KDD-WRT*CLI>
[0K
<--- Reliably Transmitting (NAT) to 66.87.65.6:7167 --->
SIP/2.0 200 OK

Via: SIP/2.0/UDP 29.152.33.6:39705;branch=z9hG4bKPjdHQbcYv.dWpTPDlzhaxP23-gr50SSy7o;received=66.87.65.6;rport=7167

From: "YouKnowWho" <sip:110@50.123.xxx.xxx>;tag=pBlcOp6BiUyfsOg4CszLxfgzN5yTK2Wk

To: <sip:444@50.123.xxx.xxx>;tag=as16fb6bec

Call-ID: BlZxnc9dIb7Djhdgp0eMMFtPWkKDUVp4

CSeq: 5312 INVITE

Server: Asterisk PBX 1.8.30.0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: <sip:444@50.123.xxx.xxx:5060>

Content-Type: application/sdp

Require: timer

Content-Length: 263

v=0

o=root 153541190 153541191 IN IP4 50.123.xxx.xxx

s=Asterisk PBX 1.8.30.0

c=IN IP4 50.123.xxx.xxx

t=0 0

m=audio 12010 RTP/AVP 3 101

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

<------------>

[KDD-WRT*CLI>
[0K
<--- SIP read from UDP:66.87.65.6:7167 --->
ACK sip:444@50.123.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP 29.152.33.6:39705;rport;branch=z9hG4bKPjXpRoKyviYeGYeOu7IDZ7tMCMXmQZ63y2
Max-Forwards: 70
From: "YouKnowWho" <sip:110@50.123.xxx.xxx>;tag=pBlcOp6BiUyfsOg4CszLxfgzN5yTK2Wk
To: <sip:444@50.123.xxx.xxx>;tag=as16fb6bec
Call-ID: BlZxnc9dIb7Djhdgp0eMMFtPWkKDUVp4
CSeq: 5312 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058332, ts 000160, len 000033)
-- <SIP/110-00000021> Playing 'vm-login.gsm' (language 'en')

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058333, ts 000320, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058334, ts 000480, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058335, ts 000640, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058336, ts 000800, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058337, ts 000960, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058338, ts 001120, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058339, ts 001280, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058340, ts 001440, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058341, ts 001600, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058342, ts 001760, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058343, ts 001920, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058344, ts 002080, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058345, ts 002240, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058346, ts 002400, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058347, ts 002560, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058348, ts 002720, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058349, ts 002880, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058350, ts 003040, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058351, ts 003200, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058352, ts 003360, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058353, ts 003520, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058354, ts 003680, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058355, ts 003840, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058356, ts 004000, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058357, ts 004160, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058358, ts 004320, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058359, ts 004480, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058360, ts 004640, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058361, ts 004800, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058362, ts 004960, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058363, ts 005120, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058364, ts 005280, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058365, ts 005440, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058366, ts 005600, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058367, ts 005760, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058368, ts 005920, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058369, ts 006080, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058370, ts 006240, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058371, ts 006400, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058372, ts 006560, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058373, ts 006720, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058374, ts 006880, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058375, ts 007040, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058376, ts 007200, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058377, ts 007360, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058378, ts 007520, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058379, ts 007680, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058380, ts 007840, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058381, ts 008000, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058382, ts 008160, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058383, ts 008320, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058384, ts 008480, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058385, ts 008640, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058386, ts 008800, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058387, ts 008960, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058388, ts 009120, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058389, ts 009280, len 000033)

[KDD-WRT*CLI>
[0KSent RTP packet to 29.152.33.6:4004 (type 03, seq 058390, ts 009440, len 000033)

[KDD-WRT*CLI>
[0K
<--- SIP read from UDP:66.87.65.6:7167 --->
BYE sip:444@50.123.xxx.xxx:5060 SIP/2.0
Via: SIP/2.0/UDP 29.152.33.6:39705;rport;branch=z9hG4bKPjf.7HTfMU45GXZT1Swo67bqiV3ecs9lGS
Max-Forwards: 70
From: "YouKnowWho" <sip:110@50.123.xxx.xxx>;tag=pBlcOp6BiUyfsOg4CszLxfgzN5yTK2Wk
To: <sip:444@50.123.xxx.xxx>;tag=as16fb6bec
Call-ID: BlZxnc9dIb7Djhdgp0eMMFtPWkKDUVp4
CSeq: 5313 BYE
User-Agent: CSipSimple_hltespr-19/r2457
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 66.87.65.6:7167 (NAT)

[KDD-WRT*CLI>
[0KScheduling destruction of SIP dialog 'BlZxnc9dIb7Djhdgp0eMMFtPWkKDUVp4' in 32000 ms (Method: BYE)

<--- Transmitting (NAT) to 66.87.65.6:7167 --->
SIP/2.0 200 OK

Via: SIP/2.0/UDP 29.152.33.6:39705;branch=z9hG4bKPjf.7HTfMU45GXZT1Swo67bqiV3ecs9lGS;received=66.87.65.6;rport=7167

From: "YouKnowWho" <sip:110@50.123.xxx.xxx>;tag=pBlcOp6BiUyfsOg4CszLxfgzN5yTK2Wk

To: <sip:444@50.123.xxx.xxx>;tag=as16fb6bec

Call-ID: BlZxnc9dIb7Djhdgp0eMMFtPWkKDUVp4

CSeq: 5313 BYE

Server: Asterisk PBX 1.8.30.0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Length: 0

<------------>

[KDD-WRT*CLI>
[0K[Dec 8 01:42:14] [1;31mWARNING[0m[14188]: [1;37mfile.c[0m:[1;37m767[0m [1;37mast_readaudio_callback[0m: Failed to write frame

[KDD-WRT*CLI>
[0K[Dec 8 01:42:14] [1;31mWARNING[0m[14188]: [1;37mapp_voicemail.c[0m:[1;37m9834[0m [1;37mvm_authenticate[0m: Couldn't read username

[KDD-WRT*CLI>
[0K
<--- SIP read from UDP:192.168.1.144:5060 --->
REGISTER sip:192.168.1.1 SIP/2.0
Via:SIP/2.0/UDP 192.168.1.144:5060;rport;branch=z9hG4bKa30bc6c4
From:<sip:105@192.168.1.1>;tag=7c8634-307-60cd0359
To:<sip:105@192.168.1.1>
Contact:<sip:105@192.168.1.144:5060;transport=UDP>;+sip.instance="<urn:uuid:d5256240-5c8b-3a24-990c-6813ac91d433>"
Call-ID:86340000-192e8b5@192.168.1.144
CSeq:84 REGISTER
Expires: 180
User-Agent:Mitel-5320-SIP-Phone 05.02.02.07 08000F6DAA6A
Supported:path,outbound,eventlist,gruu
Max-Forwards:70
Content-Length:0

<------------->

[KDD-WRT*CLI>
[0K--- (12 headers 0 lines) ---

[KDD-WRT*CLI>
[0KSending to 192.168.1.144:5060 (NAT)

[KDD-WRT*CLI>
[0K
<--- Transmitting (NAT) to 192.168.1.144:5060 --->
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 192.168.1.144:5060;branch=z9hG4bKa30bc6c4;received=192.168.1.144;rport=5060

From: <sip:105@192.168.1.1>;tag=7c8634-307-60cd0359

To: <sip:105@192.168.1.1>;tag=as1e419572

Call-ID: 86340000-192e8b5@192.168.1.144

CSeq: 84 REGISTER

Server: Asterisk PBX 1.8.30.0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="146520a8"

Content-Length: 0

<------------>

[KDD-WRT*CLI>
[0KScheduling destruction of SIP dialog '86340000-192e8b5@192.168.1.144' in 32000 ms (Method: REGISTER)

[KDD-WRT*CLI>
[0K
<--- SIP read from UDP:192.168.1.144:5060 --->
REGISTER sip:192.168.1.1 SIP/2.0
Via:SIP/2.0/UDP 192.168.1.144:5060;rport;branch=z9hG4bKa30bc6c5
From:<sip:105@192.168.1.1>;tag=7c8634-307-60cd0359
To:<sip:105@192.168.1.1>
Contact:<sip:105@192.168.1.144:5060;transport=UDP>;+sip.instance="<urn:uuid:d5256240-5c8b-3a24-990c-6813ac91d433>"
Call-ID:86340000-192e8b5@192.168.1.144
CSeq:85 REGISTER
Expires: 180
User-Agent:Mitel-5320-SIP-Phone 05.02.02.07 08000F6DAA6A
Authorization: Digest username="105", realm="asterisk", nonce="146520a8", uri="sip:192.168.1.1", response="36599a1e6a3e80267678ae5151b0caea", opaque="", algorithm=MD5
Supported:path,outbound,eventlist,gruu
Max-Forwards:70
Content-Length:0

<------------->

[KDD-WRT*CLI>
[0K--- (13 headers 0 lines) ---

[KDD-WRT*CLI>
[0KSending to 192.168.1.144:5060 (NAT)

[KDD-WRT*CLI>
[0K
<--- Transmitting (NAT) to 192.168.1.144:5060 --->
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.144:5060;branch=z9hG4bKa30bc6c5;received=192.168.1.144;rport=5060

From: <sip:105@192.168.1.1>;tag=7c8634-307-60cd0359

To: <sip:105@192.168.1.1>;tag=as1e419572

Call-ID: 86340000-192e8b5@192.168.1.144

CSeq: 85 REGISTER

Server: Asterisk PBX 1.8.30.0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Expires: 180

Contact: <sip:105@192.168.1.144:5060;transport=UDP>;expires=180

Date: Mon, 08 Dec 2014 01:42:21 GMT

Content-Length: 0

<------------>

[KDD-WRT*CLI>
[0KReliably Transmitting (NAT) to 192.168.1.144:5060:
NOTIFY sip:105@192.168.1.144:5060;transport=UDP SIP/2.0

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK45b75450;rport

Max-Forwards: 70

Route: <sip:105@192.168.1.144:5060;transport=UDP>

From: "asterisk" <sip:asterisk@192.168.1.1>;tag=as35a601e4

To: <sip:105@192.168.1.144:5060;transport=UDP>;tag=7c865a-28-66a7d8be

Contact: <sip:asterisk@192.168.1.1:5060>

Call-ID: 865a0000-3ac7d5c2@192.168.1.144

CSeq: 185 NOTIFY

User-Agent: Asterisk PBX 1.8.30.0

E
[KDD-WRT*CLI>
[0Kvent: message-summary

Content-Type: application/simple-message-summary

Subscription-State: active

Content-Length: 91

Messages-Waiting: no

Message-Account: sip:asterisk@192.168.1.1

Voice-Message: 0/0 (0/0)

---

[KDD-WRT*CLI>
[0KScheduling destruction of SIP dialog '86340000-192e8b5@192.168.1.144' in 32000 ms (Method: REGISTER)

[KDD-WRT*CLI>
[0K
<--- SIP read from UDP:192.168.1.144:5060 --->
SIP/2.0 200 OK
Via:SIP/2.0/UDP 192.168.1.1:5060;rport=5060;received=192.168.1.1;branch=z9hG4bK45b75450
From:"asterisk" <sip:asterisk@192.168.1.1>;tag=as35a601e4
To:<sip:105@192.168.1.144:5060;transport=UDP>;tag=7c865a-28-66a7d8be
CSeq:185 NOTIFY
User-Agent:Mitel-5320-SIP-Phone 05.02.02.07 08000F6DAA6A
Call-ID:865a0000-3ac7d5c2@192.168.1.144
Contact:<sip:105@192.168.1.144:5060;transport=UDP>;+sip.instance="<urn:uuid:d5256240-5c8b-3a24-990c-6813ac91d433>"
Allow:SUBSCRIBE,NOTIFY
Supported:path,outbound,eventlist,gruu
Content-Length:0

<------------->
--- (11 headers 0 lines) ---

[KDD-WRT*CLI>
[0KReliably Transmitting (NAT) to 192.168.1.115:5061:
OPTIONS sip:OOMA@192.168.1.115:5061 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK326923b1;rport

Max-Forwards: 70

From: "asterisk" <sip:OOMA@192.168.1.1>;tag=as0e30455e

To: <sip:OOMA@192.168.1.115:5061>

Contact: <sip:OOMA@192.168.1.1:5060>

Call-ID: 4bb3b215471586fb065b9a3b30408a0b@192.168.1.1:5060

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX 1.8.30.0

Date: Mon, 08 Dec 2014 01:42:26 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Length: 0

---

[KDD-WRT*CLI>
[0K
<--- SIP read from UDP:192.168.1.115:5061 --->
SIP/2.0 200 OK
To: <sip:OOMA@192.168.1.115:5061>;tag=db64ee9e1a9ce5cbi1
From: "asterisk" <sip:OOMA@192.168.1.1>;tag=as0e30455e
Call-ID: 4bb3b215471586fb065b9a3b30408a0b@192.168.1.1:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK326923b1
Server: Linksys/SPA3102-5.1.7(GW)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
--- (10 headers 0 lines) ---

[KDD-WRT*CLI>
[0KReally destroying SIP dialog '4bb3b215471586fb065b9a3b30408a0b@192.168.1.1:5060' Method: OPTIONS

[KDD-WRT*CLI>
[0K
<--- SIP read from UDP:192.168.1.132:5060 --->
ªªªª
<------------->

[KDD-WRT*CLI> sip show peers[13Grtp set debug on[Kff

DD-WRT*CLI>
[0KRTP Debugging Disabled

[KDD-WRT*CLI> rtp set debug off[13Gsip show peers[K
[0K
<--- SIP read from UDP:192.168.1.146:5060 --->
REGISTER sip:192.168.1.1 SIP/2.0
Via:SIP/2.0/UDP 192.168.1.146:5060;rport;branch=z9hG4bKa30ec539
From:<sip:104@192.168.1.1>;tag=7c8636-3ac-23f9bc5b
To:<sip:104@192.168.1.1>
Contact:<sip:104@192.168.1.146:5060;transport=UDP>;+sip.instance="<urn:uuid:7fe792de-bbf7-3ce7-9b78-c1484a0fdeea>"
Call-ID:86360000-1f3cb623@192.168.1.146
CSeq:84 REGISTER
Expires: 180
User-Agent:Mitel-5320-SIP-Phone 05.02.02.07 08000F6DA47E
Supported:path,outbound,eventlist,gruu
Max-Forwards:70
Content-Length:0

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.1.146:5060 (NAT)

<--- Transmitting (NAT) to 192.168.1.146:5060 --->
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 192.168.1.146:5060;branch=z9hG4bKa30ec539;received=192.168.1.146;rport=5060

From: <sip:104@192.168.1.1>;tag=7c8636-3ac-23f9bc5b

To: <sip:104@192.168.1.1>;tag=as454ebd64

Call-ID: 86360000-1f3cb623@192.168.1.146

CSeq: 84 REGISTER

Server: Asterisk PBX 1.8.30.0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="018f1a8b"

Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '86360000-1f3cb623@192.168.1.146' in 32000 ms (Method: REGISTER)

[KDD-WRT*CLI> sip show peers
[0K
<--- SIP read from UDP:192.168.1.146:5060 --->
REGISTER sip:192.168.1.1 SIP/2.0
Via:SIP/2.0/UDP 192.168.1.146:5060;rport;branch=z9hG4bKa30ec53a
From:<sip:104@192.168.1.1>;tag=7c8636-3ac-23f9bc5b
To:<sip:104@192.168.1.1>
Contact:<sip:104@192.168.1.146:5060;transport=UDP>;+sip.instance="<urn:uuid:7fe792de-bbf7-3ce7-9b78-c1484a0fdeea>"
Call-ID:86360000-1f3cb623@192.168.1.146
CSeq:85 REGISTER
Expires: 180
User-Agent:Mitel-5320-SIP-Phone 05.02.02.07 08000F6DA47E
Authorization: Digest username="104", realm="asterisk", nonce="018f1a8b", uri="sip:192.168.1.1", response="188de183047efe6b79ef04600f87b403", opaque="", algorithm=MD5
Supported:path,outbound,eventlist,gruu
Max-Forwards:70
Content-Length:0

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.1.146:5060 (NAT)

<--- Transmitting (NAT) to 192.168.1.146:5060 --->
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.146:5060;branch=z9hG4bKa30ec53a;received=192.168.1.146;rport=5060

From: <sip:104@192.168.1.1>;tag=7c8636-3ac-23f9bc5b

To: <sip:104@192.168.1.1>;tag=as454ebd64

Call-ID: 86360000-1f3cb623@192.168.1.146

CSeq: 85 REGISTER

Server: Asterisk PBX 1.8.30.0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Expires: 180

Contact: <sip:104@192.168.1.146:5060;transport=UDP>;expires=180

Date: Mon, 08 Dec 2014 01:42:45 GMT

Content-Length: 0

<------------>

[KDD-WRT*CLI> sip show peers
[0KReliably Transmitting (NAT) to 192.168.1.146:5060:
NOTIFY sip:104@192.168.1.146:5060;transport=UDP SIP/2.0

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK14f5c28b;rport

Max-Forwards: 70

Route: <sip:104@192.168.1.146:5060;transport=UDP>

From: "asterisk" <sip:asterisk@192.168.1.1>;tag=as44514617

To: <sip:104@192.168.1.146:5060;transport=UDP>;tag=7c865d-82-1652954b

Contact: <sip:asterisk@192.168.1.1:5060>

Call-ID: 865d0000-614ff84f@192.168.1.146

CSeq: 185 NOTIFY

User-Agent: Asterisk PBX 1.8.30.0

E
[KDD-WRT*CLI> sip show peers
[0Kvent: message-summary

Content-Type: application/simple-message-summary

Subscription-State: active

Content-Length: 91

Messages-Waiting: no

Message-Account: sip:asterisk@192.168.1.1

Voice-Message: 0/0 (0/0)

---

[KDD-WRT*CLI> sip show peers
[0KScheduling destruction of SIP dialog '86360000-1f3cb623@192.168.1.146' in 32000 ms (Method: REGISTER)

[KDD-WRT*CLI> sip show peers
[0K
<--- SIP read from UDP:192.168.1.146:5060 --->
SIP/2.0 200 OK
Via:SIP/2.0/UDP 192.168.1.1:5060;rport=5060;received=192.168.1.1;branch=z9hG4bK14f5c28b
From:"asterisk" <sip:asterisk@192.168.1.1>;tag=as44514617
To:<sip:104@192.168.1.146:5060;transport=UDP>;tag=7c865d-82-1652954b
CSeq:185 NOTIFY
User-Agent:Mitel-5320-SIP-Phone 05.02.02.07 08000F6DA47E
Call-ID:865d0000-614ff84f@192.168.1.146
Contact:<sip:104@192.168.1.146:5060;transport=UDP>;+sip.instance="<urn:uuid:7fe792de-bbf7-3ce7-9b78-c1484a0fdeea>"
Allow:SUBSCRIBE,NOTIFY
Supported:path,outbound,eventlist,gruu
Content-Length:0

<------------->

[KDD-WRT*CLI> sip show peers
[0K--- (11 headers 0 lines) ---

[KDD-WRT*CLI> sip show peers
[0KReally destroying SIP dialog 'BlZxnc9dIb7Djhdgp0eMMFtPWkKDUVp4' Method: BYE

[KDD-WRT*CLI> sip show peers[13Grtp set debug on[13Gsi[29G[13Gcore set verbose 3[13Gsip set debug on[K[K[K
[0K
<--- SIP read from UDP:66.87.65.6:7167 --->

<------------->

[KDD-WRT*CLI> sip set debug off

DD-WRT*CLI>
[0KSIP Debugging Disabled

[KDD-WRT*CLI> sip set debug off[13Grt[30G[13Gsip show peers[K[13Grtp set debug on[13Gsi[29G[13Gcore set verbose 3[K5

DD-WRT*CLI>
[0KVerbosity was 3 and is now 5

[KDD-WRT*CLI> core set verbose 5[13Gsip set debug off[K[13Grt[30G[13Gsip show peers[K

DD-WRT*CLI>
[0KName/username Host Dyn Forcerport ACL Port Status
103/103 192.168.1.132 D N 5060 Unmonitored
104/104 192.168.1.146 D N 5060 Unmonitored
105/105 192.168.1.144 D N 5060 Unmonitored
110/110 66.87.65.6 D N 7167 Unmonitored
OOMA/OOMA 192.168.1.115 D N 5061 OK (9 ms)
5 sip peers [Monitored: 1 online, 0 offline Unmonitored: 4 online, 0 offline]

[KDD-WRT*CLI> exci[K[Kit

[01;31mroot@DD-WRT [01;34m~ $ [00mexit

Statistics : Posted by bridges86406 • on Mon Dec 08, 2014 2:56 am • Replies 0 • Views 10

↧

Video Stream before answering the call

December 8, 2014, 1:41 am

≫ Next: Bridge configuration in Asterisk 13

≪ Previous: SIP --> Packets being sent to different IP

Hi folks,

i am programming a door intercom system with asterisk and raspberry. The backend works fine.
But I have a question with asterisk and video calls.
Usually, the video stream starts after answering the call. Is it possible to start the video stream to all clients direct after starting the call? Because I want to see, who is standing at the door, before I answer the call and speak with him.

Thank's a lot.

Statistics : Posted by moonwalker05 • on Mon Dec 08, 2014 3:41 am • Replies 0 • Views 5

↧

Bridge configuration in Asterisk 13

December 8, 2014, 1:42 am

≫ Next: Sending Faxes with resolution Superfine and Ultrafine

≪ Previous: Video Stream before answering the call

I've recently upgraded from Asterisk 11 to Asterisk 13.

Most of it went smoothly thanks to the documentation detailing how to upgrade to 12 and then how to upgrade to 13.

The only thing that didn't work correctly was Music On Hold. Eventually I tracked this down to using bridge_softmix instead of bridge_simple.

What I'm asking is, does anyone have any explanation as to why MOH would not work with bridge_softmix? Asterisk 11 had been working for at least a year with bridge_softmix and the MOH was fine. With the same configuration (almost) Asterisk 13 insists I use bridge_simple otherwise I see no messages on the CLI about hold music starting or stopping. Unloading bridge_softmix and then loading bridge_simple fixes the issue.

Also does anyone have any documentation on what bridges I should be using? I can't seem to find anything in the upgrade documentation that says "MOH will no longer work in softmix, you should use simple". This has me concerned that I've done something wrong elsewhere in my config that is causing softmix to not work correctly.

Statistics : Posted by lizardloop • on Mon Dec 08, 2014 3:42 am • Replies 0 • Views 6

↧

Sending Faxes with resolution Superfine and Ultrafine

August 7, 2013, 8:06 am

≫ Next: Sending Faxes with resolution Superfine and Ultrafine

≪ Previous: Bridge configuration in Asterisk 13

Although it might send the fax with more data, the actual image information is the same.
I am now trying to find out what you actually need to do to improve image quality
Nevermind .... I was not paying attention the solution above works fine. It's still a fax....

Statistics : Posted by oli004 • on Wed Aug 07, 2013 8:06 am • Replies 4 • Views 2668

↧

Sending Faxes with resolution Superfine and Ultrafine

August 7, 2013, 8:06 am

≫ Next: Sending Faxes with resolution Superfine and Ultrafine

≪ Previous: Sending Faxes with resolution Superfine and Ultrafine

Hi

Asterisk Version: 10.12.2
FFA Version: 10.1_1.3.1

I would like to send faxes via FFA's SendFax() application with higher resolutions e.g.:
- 300x300 DPI (Superfine)
- 400x400 DPI (Ultrafine)

Does anyone know how to do this?

I already tried to change the ghostscript command, which generated the tiff file, but that did not seem to do the trick:
gs -q -r400x400 -dNOPAUSE -dBATCH -sDEVICE=tiffg4 -sPAPERSIZE=letter -sOutputFile=fax.tif input.pdf

Trying to send this file resulted in a segfault and a forced restart of asterisk:
[Aug 7 15:51:14] ERROR[31069]: res_fax_digium.c:2145 dgm_fax_start: FAX handle 0: failed to queue document 'fax.tif'
[Aug 7 15:51:14] ERROR[31069]: res_fax.c:1380 generic_fax_exec: channel 'channel' FAX session '0' failure, reason: 'failed to start FAX session' (INIT_ERROR)
j16344*CLI>
Disconnected from Asterisk server
Executing last minute cleanups

My questions:
- Is faxing in a different resolution supported by FFA?
- If yes, in which format do I have to generate the TIFF-file?
- Or is there a fundamental error in my thinking?

Thanks

Greetings

Oli4

Statistics : Posted by oli004 • on Wed Aug 07, 2013 8:06 am • Replies 4 • Views 2668

↧

Sending Faxes with resolution Superfine and Ultrafine

August 7, 2013, 8:06 am

≫ Next: Sending Faxes with resolution Superfine and Ultrafine

≪ Previous: Sending Faxes with resolution Superfine and Ultrafine

Nice find

Statistics : Posted by oli004 • on Wed Aug 07, 2013 8:06 am • Replies 4 • Views 2668

↧

Sending Faxes with resolution Superfine and Ultrafine

August 7, 2013, 8:06 am

≫ Next: Stilhaus Kitchens Reviews

≪ Previous: Sending Faxes with resolution Superfine and Ultrafine

The trick was to add the tiff's size to the gs command (via the -g flag):

For 300 DPI:
gs -r300x300 -g2592x3507 -q -dNOPAUSE -dBATCH -sDEVICE=tiffg4 -sPAPERSIZE=a4 -sOutputFile=fax.tif input.pdf

For 400 DPI:
gs -r400x400 -g3456x4676 -q -dNOPAUSE -dBATCH -sDEVICE=tiffg4 -sPAPERSIZE=a4 -sOutputFile=fax.tif input.pdf

Have a nice day.

Statistics : Posted by oli004 • on Wed Aug 07, 2013 8:06 am • Replies 4 • Views 2668

↧

Stilhaus Kitchens Reviews

December 8, 2014, 11:28 pm

≫ Next: File sharing with asterisk

≪ Previous: Sending Faxes with resolution Superfine and Ultrafine

Stilhaus Kitchens did a proper job and I couldn't thank the fitters Paul and Luke enough. They really pulled out all the stops to get the kitchen fitted on time.

Statistics : Posted by axdogfyx • on Tue Dec 09, 2014 1:28 am • Replies 0 • Views 24

↧

File sharing with asterisk

December 3, 2014, 6:45 am

≫ Next: File sharing with asterisk

≪ Previous: Stilhaus Kitchens Reviews

Mhf,
so you are telling me that there is no file transfer support in asterisk ?

Thanks

Statistics : Posted by yluom • on Wed Dec 03, 2014 8:45 am • Replies 2 • Views 174

↧

File sharing with asterisk

December 3, 2014, 6:45 am

≫ Next: File sharing with asterisk

≪ Previous: File sharing with asterisk

There does not appear to be MSRP support in the latest version of Asterisk.

Statistics : Posted by yluom • on Wed Dec 03, 2014 8:45 am • Replies 2 • Views 174

↧

File sharing with asterisk

December 3, 2014, 6:45 am

≫ Next: Kamailio & Asterisk as SBC

≪ Previous: File sharing with asterisk

Hi all.
I just set up an asterisk server on my debian box.
After some fighting with asterisk's config, I finally succeeded to make two android phones to call each other. Even video call is working ! So it's pretty cool.
My configuration is really basic. I'm using imsDroid softphone on the android phones.
Now, I need to be able to do file transfer (file sharing) between those two phones.
imsDroid have this built-in capability. But when I try to send a file (in this case, an image), I get this error on asterisk:

Code: WARNING[19405]: chan_sip.c:9084 process_sdp: Unsupported SDP media type in offer: message 34515 TCP/MSRP *
WARNING[19405]: chan_sip.c:9177 process_sdp: Failing due to no acceptable offer found

I've been searching all day on google and forums, I've read many papers but I can't find anything to resolve this issue.
I bet I'm missing something.
Do I need to add a codec for this to work ? How can I debug this ?
Please consider that I'm a beginner with asterisk and with the SIP protocol.

Thanks in advance

Have a great day !

Statistics : Posted by yluom • on Wed Dec 03, 2014 8:45 am • Replies 2 • Views 174

↧

Kamailio & Asterisk as SBC

December 9, 2014, 1:58 am

≫ Next: SIP trunk between two asterisk boxes 11.7 an 1.8 failed

≪ Previous: File sharing with asterisk

Hi

We study the possibility to integrate Asterisk as SBC and as voice and conferencing solutions integrated to kamailio.

The purpose of projet is to implement a VoIP secure solution with Kamailio as core IMS network.

In this project VoIP call will be established with zRTP. SBC is transparent for zRTP call, zRTP is establish between 2 end point (terminals). However for conferencing service we need to establish zRTP session with conferencing server (MRF).

Question related to voice and conferencing solutions :
In case of using Asterisk conferencing solution, is it compliant to ZRTP ? if not have you roadmap related zRTP integration ?
An other way is to custom conferencing service for our need. According to your knowledge is it possible ? (can you assist our developpement team to this customization ?)

Question related to SBC :
In case of using Asterisk SBC solution, the major problem is the failed over mechanism in case of Asterisk SBC malfunctions : all on-going calls are ended, immediately.
Is there failed over mecanism between SBCs Asterisk ? (or is it suported by Kamailio core Network ?)

Thanks you in advance for your answer

Statistics : Posted by ludovic • on Tue Dec 09, 2014 3:58 am • Replies 0 • Views 14

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SIP trunk between two asterisk boxes 11.7 an 1.8 failed

December 9, 2014, 2:16 am

≫ Next: Yealink phones with Switchvox- DND

≪ Previous: Kamailio & Asterisk as SBC

Hi;
I try to establish connection between two asterisk boxes. Both are in the same subnet (LAN) one is based on Asterisk 11.7 and CentOS server the second on is Asterisk 1.8 and it works on Debian server.
There is no firewalls etc between servers. Both servers has disabled IPv6 support.

I have configured users on both sides and try to register each other.
I can not get authentication on both sides, the SIP messages look strange for me.

Could somebody look at this SIP messages (from tcpdump) and tell me if they are normal or there is some bug in the communication.
Any help will be much appreciated

Code: 11:10:58.252838 IP (tos 0x60, ttl 64, id 41559, offset 0, flags [none], proto: UDP (17), length: 524) 10.96.7.254.sip > 10.96.7.253.sip: SIP, length: 496
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.96.7.253:5060;bran\000\000\000\000\000\000!\000\000\000\000\000\000\000host 10.96.7.253\000\000\000\000\000\000\000\000!\000\000\000\000\000\000\000`\336\236\017\000\000\000\000\240\333\236\017\000\000\000\000 \000\000\000\000\000\000\0001\000\000\000\000\000\000\000\020\322\236\017\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\021\000\000\000\020\000\000\000\001\000\000\000\000\000\000\0001\000\000\000\000\000\000\000_,\322\0251\000\000\000\360\232\312\374\214+\000\000\220\333\236\017\000\000\000\0008J\365\0251\000\000\0000\000\000\000\000\000\000\0001\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\301\006\000\000\000\000\000\000P\320\266V\377\377\377\377\320*\250\231\377\377\377\377`\027\014\233\377\377\377\377\360\332\325\233\377\377\377\377\220\256\331\234\377\377\377\377\220\265\244\235\377\377\377\377\220\220\271\236\377\377\377\377\220\227\204\237\377\377\377\377\000\266\232\240\377\377\377\377\000\275e\241\377\377\377\377`|}\246\377\377\377\377\020\336v\310\377\377\377\377\020K\347\314\377\377\377\377\220\027\251\315\377\377\377\377\020C\242\316\377\377\377\377\0204\222\317\377\377\377\377`\251\200\320\377\377\377\377\000\272\204\320\377\377\377\377p\222\225\321\377\377\377\377`\273\212\322\377\377\377\377p\377b\323\377\377\377\377\220#K\324\377\377\377\377\020\255^\325\377\377\377\377\020\264)\326\377\377\377\377\020\032,\327\377\377\377\377\020\226\011\330\377\377\377\377\220\301\002\331\377\377\377\377\020x\351\331
11:11:00.812317 arp who-has 10.96.7.253 tell 10.96.7.254
11:11:00.812420 arp reply 10.96.7.253 is-at 00:0c:29:fd:2a:8c
11:11:03.524284 IP (tos 0x0, ttl 64, id 14417, offset 0, flags [none], proto: UDP (17), length: 515) 10.96.7.253.sip > 10.96.7.254.sip: SIP, length: 487
OPTIONS sip:10.96.7.254 SIP/2.0
Via: SIP/2.0/UDP 10.9\000\000\000\000\000\000!\000\000\000\000\000\000\000host 10.96.7.253\000\000\000\000\000\000\000\000!\000\000\000\000\000\000\000`\336\236\017\000\000\000\000\240\333\236\017\000\000\000\000 \000\000\000\000\000\000\0001\000\000\000\000\000\000\000\020\322\236\017\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\021\000\000\000\020\000\000\000\001\000\000\000\000\000\000\0001\000\000\000\000\000\000\000_,\322\0251\000\000\000\360\232\312\374\214+\000\000\220\333\236\017\000\000\000\0008J\365\0251\000\000\0000\000\000\000\000\000\000\0001\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\301\006\000\000\000\000\000\000P\320\266V\377\377\377\377\320*\250\231\377\377\377\377`\027\014\233\377\377\377\377\360\332\325\233\377\377\377\377\220\256\331\234\377\377\377\377\220\265\244\235\377\377\377\377\220\220\271\236\377\377\377\377\220\227\204\237\377\377\377\377\000\266\232\240\377\377\377\377\000\275e\241\377\377\377\377`|}\246\377\377\377\377\020\336v\310\377\377\377\377\020K\347\314\377\377\377\377\220\027\251\315\377\377\377\377\020C\242\316\377\377\377\377\0204\222\317\377\377\377\377`\251\200\320\377\377\377\377\000\272\204\320\377\377\377\377p\222\225\321\377\377\377\377`\273\212\322\377\377\377\377p\377b\323\377\377\377\377\220#K\324\377\377\377\377\020\255^\325\377\377\377\377\020\264)\326\377\377\377\377\020\032,\327\377\377\377\377\020\226\011\330\377\377\377\377\220\301\002
11:11:03.524483 IP (tos 0x60, ttl 64, id 41560, offset 0, flags [none], proto: UDP (17), length: 524) 10.96.7.254.sip > 10.96.7.253.sip: SIP, length: 496
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.96.7.253:5060;bran\000\000\000\000\000\000!\000\000\000\000\000\000\000host 10.96.7.253\000\000\000\000\000\000\000\000!\000\000\000\000\000\000\000`\336\236\017\000\000\000\000\240\333\236\017\000\000\000\000 \000\000\000\000\000\000\0001\000\000\000\000\000\000\000\020\322\236\017\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\021\000\000\000\020\000\000\000\001\000\000\000\000\000\000\0001\000\000\000\000\000\000\000_,\322\0251\000\000\000\360\232\312\374\214+\000\000\220\333\236\017\000\000\000\0008J\365\0251\000\000\0000\000\000\000\000\000\000\0001\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\301\006\000\000\000\000\000\000P\320\266V\377\377\377\377\320*\250\231\377\377\377\377`\027\014\233\377\377\377\377\360\332\325\233\377\377\377\377\220\256\331\234\377\377\377\377\220\265\244\235\377\377\377\377\220\220\271\236\377\377\377\377\220\227\204\237\377\377\377\377\000\266\232\240\377\377\377\377\000\275e\241\377\377\377\377`|}\246\377\377\377\377\020\336v\310\377\377\377\377\020K\347\314\377\377\377\377\220\027\251\315\377\377\377\377\020C\242\316\377\377\377\377\0204\222\317\377\377\377\377`\251\200\320\377\377\377\377\000\272\204\320\377\377\377\377p\222\225\321\377\377\377\377`\273\212\322\377\377\377\377p\377b\323\377\377\377\377\220#K\324\377\377\377\377\020\255^\325\377\377\377\377\020\264)\326\377\377\377\377\020\032,\327\377\377\377\377\020\226\011\330\377\377\377\377\220\301\002\331\377\377\377\377\020x\351\331
11:11:08.531321 arp who-has 10.96.7.254 tell 10.96.7.253
11:11:08.531332 arp reply 10.96.7.254 is-at 00:0c:29:e7:70:18
11:11:22.280515 IP (tos 0x0, ttl 64, id 14418, offset 0, flags [none], proto: UDP (17), length: 515) 10.96.7.253.sip > 10.96.7.254.sip: SIP, length: 487
OPTIONS sip:10.96.7.254 SIP/2.0
Via: SIP/2.0/UDP 10.9\000\000\000\000\000\000!\000\000\000\000\000\000\000host 10.96.7.253\000\000\000\000\000\000\000\000!\000\000\000\000\000\000\000`\336\236\017\000\000\000\000\240\333\236\017\000\000\000\000 \000\000\000\000\000\000\0001\000\000\000\000\000\000\000\020\322\236\017\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\021\000\000\000\020\000\000\000\001\000\000\000\000\000\000\0001\000\000\000\000\000\000\000_,\322\0251\000\000\000\360\232\312\374\214+\000\000\220\333\236\017\000\000\000\0008J\365\0251\000\000\0000\000\000\000\000\000\000\0001\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\301\006\000\000\000\000\000\000P\320\266V\377\377\377\377\320*\250\231\377\377\377\377`\027\014\233\377\377\377\377\360\332\325\233\377\377\377\377\220\256\331\234\377\377\377\377\220\265\244\235\377\377\377\377\220\220\271\236\377\377\377\377\220\227\204\237\377\377\377\377\000\266\232\240\377\377\377\377\000\275e\241\377\377\377\377`|}\246\377\377\377\377\020\336v\310\377\377\377\377\020K\347\314\377\377\377\377\220\027\251\315\377\377\377\377\020C\242\316\377\377\377\377\0204\222\317\377\377\377\377`\251\200\320\377\377\377\377\000\272\204\320\377\377\377\377p\222\225\321\377\377\377\377`\273\212\322\377\377\377\377p\377b\323\377\377\377\377\220#K\324\377\377\377\377\020\255^\325\377\377\377\377\020\264)\326\377\377\377\377\020\032,\327\377\377\377\377\020\226\011\330\377\377\377\377\220\301\002
11:11:22.280838 IP (tos 0x60, ttl 64, id 41561, offset 0, flags [none], proto: UDP (17), length: 524) 10.96.7.254.sip > 10.96.7.253.sip: SIP, length: 496
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.96.7.253:5060;bran\000\000\000\000\000\000!\000\000\000\000\000\000\000host 10.96.7.253\000\000\000\000\000\000\000\000!\000\000\000\000\000\000\000`\336\236\017\000\000\000\000\240\333\236\017\000\000\000\000 \000\000\000\000\000\000\0001\000\000\000\000\000\000\000\020\322\236\017\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\021\000\000\000\020\000\000\000\001\000\000\000\000\000\000\0001\000\000\000\000\000\000\000_,\322\0251\000\000\000\360\232\312\374\214+\000\000\220\333\236\017\000\000\000\0008J\365\0251\000\000\0000\000\000\000\000\000\000\0001\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\000\301\006\000\000\000\000\000\000P\320\266V\377\377\377\377\320*\250\231\377\377\377\377`\027\014\233\377\377\377\377\360\332\325\233\377\377\377\377\220\256\331\234\377\377\377\377\220\265\244\235\377\377\377\377\220\220\271\236\377\377\377\377\220\227\204\237\377\377\377\377\000\266\232\240\377\377\377\377\000\275e\241\377\377\377\377`|}\246\377\377\377\377\020\336v\310\377\377\377\377\020K\347\314\377\377\377\377\220\027\251\315\377\377\377\377\020C\242\316\377\377\377\377\0204\222\317\377\377\377\377`\251\200\320\377\377\377\377\000\272\204\320\377\377\377\377p\222\225\321\377\377\377\377`\273\212\322\377\377\377\377p\377b\323\377\377\377\377\220#K\324\377\377\377\377\020\255^\325\377\377\377\377\020\264)\326\377\377\377\377\020\032,\327\377\377\377\377\020\226\011\330\377\377\377\377\220\301\002\331\377\377\377\377\020x\351\331

Statistics : Posted by witekprytek • on Tue Dec 09, 2014 4:16 am • Replies 0 • Views 17

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Yealink phones with Switchvox- DND

December 10, 2014, 11:28 am

≫ Next: Yealink phones with Switchvox- DND

≪ Previous: SIP trunk between two asterisk boxes 11.7 an 1.8 failed

I'm finding a little inconsistency with the Yealink T46G phone, which we're so fond of using, as it integrates with Switchvox (SMB 360, Rls 5.10)....specifically the DND button functionality.

It seems to perform the DND properly (i.e., calls are sent to voicemail) but having the phone's DND status show on other Yealink phones doesn't seem to work properly..realizing that in most cases, the DND status of a BLF-monitored extension normally shows the same indication as a 'Busy' status (whether that be solid greeen or solid red, depending on how the Yealink phone's BLF mode is set, 0~3) but I'm finding there is no change in reference to the monitored phone engaging the DND button.

I've tried leaving the 'DND On' and 'DND Off' fields blank on the Yealink phone, as well as setting them to the Asterisk codes of *78 for 'DND On', and *79 for 'DND Off', but neither produces the BLF light change that would indicate the status.

Any help would be appreciated.

Statistics : Posted by phoneguy797 • on Wed Dec 10, 2014 1:28 pm • Replies 1 • Views 28

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