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Custom Pickup Buttons

June 15, 2014, 6:11 am
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Hi,

Wa want to use 2 functions keys on a SNOM device. I only know it's possible to use *8.

So the company has 3 sites. All sites are divided in departments.

- The first pickup button should pickup only calls from the same department in 1 site.
- The second button should pickup any extension for 1 site.

Could somebody give me a starting point please.

Tnx in advance!

Statistics : Posted by quastenk • on Sun Jun 15, 2014 6:11 am • Replies 0 • Views 38
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Poor sound quality after Read

June 16, 2014, 8:08 am
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exten => s,n,Playback(invalid)
exten => s,n,Read(TgExtn,sayService,4)
exten => s,n,Playback(invalid)

First playback is OK, sayService message is OK, second playback has very poor quality...
Image Why?
What is wrong and where to look a problem?

Note 1: invalid = standard Asterisk sound file invalid.gsm
Note 2: The same happen with other sound files

Statistics : Posted by jumpow • on Mon Jun 16, 2014 8:08 am • Replies 1 • Views 60
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API switchvox.backups.add JSON optional directories

April 29, 2014, 2:06 pm
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Update : Digium confirmed that they were seeing the same issue and are currently looking into the issue.

Statistics : Posted by linkn3echo • on Tue Apr 29, 2014 2:06 pm • Replies 2 • Views 355
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SIP Trunks Wont Register Untill Local IP Change

June 16, 2014, 2:00 pm
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Hi,

I have installed latest version of Elastix few days back, and now am facing a problem where the Asterisk has intermittent loss of internet (5-8 secs max) and the trunks go offline. After the internet is back online the trunks dont seem to register no matter what I do. I have SIP 2 trunks, Voipfone & Vonage UK.

Both the trunks get unreachable and nothing seems to work, I have found the following commands to no use in this situation...

CLI> reload
CLI> sip reload
# amportal restart

I have also tried to disable both the trunks through gui then reload and then enable and then reload, but that also doesnt seem to work.

I then have to change the local IP of my Asterisk from 192.168.1.106 to 192.168.1.150 reboot the machine and both trunks get online automatically.

When I face this issue whilst sitting on IP 192.168.1.150 I simply jump back to 192.168.1.106 and reboot and everythings fine again.

My Asterisk is connected to Draytek 2830, I have made sure that all I have all ports from 5000-6000 pointing to my Asterisk local IP also these ports are not blocked. Everything seems fine till the trunks go down.

Also I have another machine running Elastix in VM environment (for testing purposes) this problem doesnt seem to come on that particular machine at all, tho it is connected to the same router.

Any thoughts?

Regards,

Statistics : Posted by jeet • on Mon Jun 16, 2014 2:00 pm • Replies 0 • Views 38
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PBX Seems to be Declining (603) All SIP Calls

June 16, 2014, 3:04 pm
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The parameter of Dial is invalid; it is missing a technology/ prefix. That is either totally invalid, or will be interpreted as Local/. I'd have to check the code to be sure.

Statistics : Posted by orangeman555 • on Mon Jun 16, 2014 3:04 pm • Replies 1 • Views 58
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PBX Seems to be Declining (603) All SIP Calls

June 16, 2014, 3:04 pm
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For a simple home communication system, I've set up some very simple SIP / Extensions. Go easy on me, I'm very new to this system.

For now the only way I've gotten them to work (in testing) is to take down the firewall. Still, I seem to be getting **instant** 603's with every try from every phone. Why am I getting 603's seemingly by default?

When I make a call, this is what it reports:

Code:    <--- SIP read from UDP:192.168.1.8:5060 --->
    INVITE sip:103@192.168.1.6 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.8:5060;rport;branch=z9hG4bKPj43e20551-6dbc-411b-919a-ab117c06ae05
    Max-Forwards: 70
    From: <sip:0000FFFF004@192.168.1.6>;tag=12e33b02-c7c1-4b83-9ba4-f08c426ece25
    To: <sip:103@192.168.1.6>
    Contact: <sip:0000FFFF004@192.168.1.8:5060>
    Call-ID: 16962f1e-d2e0-4987-af02-dc765ffa793f
    CSeq: 6702 INVITE
    Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE
    upported: replaces, 100rel
    Content-Type: application/sdp
    Content-Length: 361
   
    v=0
    o=dinosaur 3611940779 0 IN IP4 192.168.1.8
    s=sflphone
    c=IN IP4 192.168.1.8
    t=0 0
    m=audio 37600 RTP/AVP 0 3 8 9 110 111 112
    a=rtpmap:0 PCMU/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:9 G722/8000
    a=rtpmap:110 speex/8000
    a=rtpmap:111 speex/16000
    a=rtpmap:112 speex/32000
    a=sendrecv
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    <------------->
    --- (12 headers 16 lines) ---
    Sending to 192.168.1.8:5060 (NAT)
    Using INVITE request as basis request - 16962f1e-d2e0-4987-af02-dc765ffa793f
    Found peer '0000FFFF004' for '0000FFFF004' from 192.168.1.8:5060
   
    <--- Reliably Transmitting (NAT) to 192.168.1.8:5060 --->
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bKPj43e20551-6dbc-411b-919a-ab117c06ae05;received=192.168.1.8;rport=5060
    From: <sip:0000FFFF004@192.168.1.6>;tag=12e33b02-c7c1-4b83-9ba4-f08c426ece25
    To: <sip:103@192.168.1.6>;tag=as69cdb064
    Call-ID: 16962f1e-d2e0-4987-af02-dc765ffa793f
    CSeq: 6702 INVITE
    Server: Asterisk PBX SVN-branch-1.8-r416150
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5572b5df"
    Content-Length: 0
   
   
    <------------>
    Scheduling destruction of SIP dialog '16962f1e-d2e0-4987-af02-dc765ffa793f' in 32000 ms (Method: INVITE)
   
    <--- SIP read from UDP:192.168.1.8:5060 --->
    ACK sip:103@192.168.1.6 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.8:5060;rport;branch=z9hG4bKPj43e20551-6dbc-411b-919a-ab117c06ae05
    Max-Forwards: 70
    From: <sip:0000FFFF004@192.168.1.6>;tag=12e33b02-c7c1-4b83-9ba4-f08c426ece25
    To: <sip:103@192.168.1.6>;tag=as69cdb064
    Call-ID: 16962f1e-d2e0-4987-af02-dc765ffa793f
    CSeq: 6702 ACK
    Content-Length: 0
   
    <------------->
    --- (8 headers 0 lines) ---
   
    <--- SIP read from UDP:192.168.1.8:5060 --->
    INVITE sip:103@192.168.1.6 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.8:5060;rport;branch=z9hG4bKPj7504ac55-fcbe-470a-a5c5-2174a0699d0d
    Max-Forwards: 70
    From: <sip:0000FFFF004@192.168.1.6>;tag=12e33b02-c7c1-4b83-9ba4-f08c426ece25
    To: <sip:103@192.168.1.6>
    Contact: <sip:0000FFFF004@192.168.1.8:5060>
    Call-ID: 16962f1e-d2e0-4987-af02-dc765ffa793f
    CSeq: 6703 INVITE
    Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE
    upported: replaces, 100rel
    Authorization: Digest username="0000FFFF004", realm="asterisk", nonce="5572b5df", uri="sip:103@192.168.1.6", response="44810c7fbf0d8a99e34ea07b5e62ee79", algorithm=MD5
    Content-Type: application/sdp
    Content-Length: 361
   
    v=0
    o=dinosaur 3611940779 0 IN IP4 192.168.1.8
    s=sflphone
    c=IN IP4 192.168.1.8
    t=0 0
    m=audio 37600 RTP/AVP 0 3 8 9 110 111 112
    a=rtpmap:0 PCMU/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:9 G722/8000
    a=rtpmap:110 speex/8000
    a=rtpmap:111 speex/16000
    a=rtpmap:112 speex/32000
    a=sendrecv
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    <------------->
    --- (13 headers 16 lines) ---
    Sending to 192.168.1.8:5060 (NAT)
    Using INVITE request as basis request - 16962f1e-d2e0-4987-af02-dc765ffa793f
    Found peer '0000FFFF004' for '0000FFFF004' from 192.168.1.8:5060
      == Using SIP RTP CoS mark 5
    Found RTP audio format 0
    Found RTP audio format 3
    Found RTP audio format 8
    Found RTP audio format 9
    Found RTP audio format 110
    Found RTP audio format 111
    Found RTP audio format 112
    Found audio description format PCMU for ID 0
    Found audio description format GSM for ID 3
    Found audio description format PCMA for ID 8
    Found audio description format G722 for ID 9
    Found audio description format speex for ID 110
    Found audio description format speex for ID 111
    Found unknown media description format speex for ID 112
    Found audio description format telephone-event for ID 101
    Capabilities: us - 0x100e (gsm|ulaw|alaw|g722), peer - audio=0x20000120e (gsm|ulaw|alaw|speex|speex16|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100e (gsm|ulaw|alaw|g722)
    Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
    Peer audio RTP is at port 192.168.1.8:37600
    Looking for 103 in LocalSets (domain 192.168.1.6)
    list_route: hop: <sip:0000FFFF004@192.168.1.8:5060>
   
    <--- Transmitting (NAT) to 192.168.1.8:5060 --->
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bKPj7504ac55-fcbe-470a-a5c5-2174a0699d0d;received=192.168.1.8;rport=5060
    From: <sip:0000FFFF004@192.168.1.6>;tag=12e33b02-c7c1-4b83-9ba4-f08c426ece25
    To: <sip:103@192.168.1.6>
    Call-ID: 16962f1e-d2e0-4987-af02-dc765ffa793f
    CSeq: 6703 INVITE
    Server: Asterisk PBX SVN-branch-1.8-r416150
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Contact: <sip:103@192.168.1.6:5060>
    Content-Length: 0
   
   
    <------------>
        -- Executing [103@LocalSets:1] Dial("SIP/0000FFFF004-0000001a", "0000FFFF005") in new stack
      == Spawn extension (LocalSets, 103, 1) exited non-zero on 'SIP/0000FFFF004-0000001a'
    Scheduling destruction of SIP dialog '16962f1e-d2e0-4987-af02-dc765ffa793f' in 32000 ms (Method: INVITE)
   
    <--- Reliably Transmitting (NAT) to 192.168.1.8:5060 --->
    SIP/2.0 603 Declined
    Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bKPj7504ac55-fcbe-470a-a5c5-2174a0699d0d;received=192.168.1.8;rport=5060
    From: <sip:0000FFFF004@192.168.1.6>;tag=12e33b02-c7c1-4b83-9ba4-f08c426ece25
    To: <sip:103@192.168.1.6>;tag=as165ecdc9
    Call-ID: 16962f1e-d2e0-4987-af02-dc765ffa793f
    CSeq: 6703 INVITE
    Server: Asterisk PBX SVN-branch-1.8-r416150
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Content-Length: 0
   
   
    <------------>
   
    <--- SIP read from UDP:192.168.1.8:5060 --->
    ACK sip:103@192.168.1.6 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.8:5060;rport;branch=z9hG4bKPj7504ac55-fcbe-470a-a5c5-2174a0699d0d
    Max-Forwards: 70
    From: <sip:0000FFFF004@192.168.1.6>;tag=12e33b02-c7c1-4b83-9ba4-f08c426ece25
    To: <sip:103@192.168.1.6>;tag=as165ecdc9
    Call-ID: 16962f1e-d2e0-4987-af02-dc765ffa793f
    CSeq: 6703 ACK
    Content-Length: 0
   
    <------------->
    --- (8 headers 0 lines) ---
   
    <--- SIP read from UDP:192.168.1.5:63992 --->
   
    <------------->
    Really destroying SIP dialog 'cb7123d1-4244-4673-a200-dc851e1c8415' Method: REGISTER


The phones themselves are not set to decline calls, so I can only assume its happening somewhere in Asterisk.

Statistics : Posted by orangeman555 • on Mon Jun 16, 2014 3:04 pm • Replies 1 • Views 58
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How to redirect SIP Register request from asterisk?

June 16, 2014, 4:51 am
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david55 wrote:Asterisk is not a SIP proxy.

So does that mean i can't achieve what i am trying? I am using asterisk as my SIP Server I want to forward the REGISTER request from few users to some other server.

Statistics : Posted by suhas_s • on Mon Jun 16, 2014 4:51 am • Replies 2 • Views 63
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How to redirect SIP Register request from asterisk?

June 16, 2014, 4:51 am
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Asterisk is not a SIP proxy.

Statistics : Posted by suhas_s • on Mon Jun 16, 2014 4:51 am • Replies 2 • Views 63
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How to redirect SIP Register request from asterisk?

June 16, 2014, 4:51 am
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Hi All,
I wanted to redirect SIP REGISTER request from asterisk server to some other open server. Is there any way I can achieve this? I went through many options of sip.conf couldn't find anything useful. Please help regarding the same.

Thanks in advance!

Statistics : Posted by suhas_s • on Mon Jun 16, 2014 4:51 am • Replies 2 • Views 63
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Memory leak problem...

June 11, 2014, 11:56 pm
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The memory used figure displayed by top should increase until it is close to the physical memory present. if it doesn't, your systems is grossly over-provisioned with memory.

You haven't provided the logging needed to tackle the second part of the question.

Statistics : Posted by leunge • on Wed Jun 11, 2014 11:56 pm • Replies 16 • Views 410
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Memory leak problem...

June 11, 2014, 11:56 pm
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I have another question!! can you help me? T.T
I show memory percentage increase and suddenly down ( using 'top')
what it means?? T.T ...

And at that time the call(using asterisk server) suddenly ended.. it is my problem... the call suddenly ended.....

Statistics : Posted by leunge • on Wed Jun 11, 2014 11:56 pm • Replies 16 • Views 410
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Memory leak problem...

June 11, 2014, 11:56 pm
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OH thanks T.T .. I was stupid...
It is test server.. main server I didn't patch ....
main server down every 3 hours and the percentage of memory ( I use top ) increases.
so I want to know why memory increase ..T.T...
then I will do the patch thanks~!

Statistics : Posted by leunge • on Wed Jun 11, 2014 11:56 pm • Replies 16 • Views 410
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Memory leak problem...

June 11, 2014, 11:56 pm
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The size isn't changing, so you don't have a leak. RSS is an indicator of the memory access pattern, not the total memory used. It will probably go up due to increasing fragmentation.

Statistics : Posted by leunge • on Wed Jun 11, 2014 11:56 pm • Replies 16 • Views 410
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Memory leak problem...

June 11, 2014, 11:56 pm
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Hello I have a memory problem .... but I can't do anything.. I searched and use this patch
https://issues.asterisk.org/jira/browse/ASTERISK-23616

but.. every call rss memory increase... is it ok???
how I can fix it? T.T

oh sorry... I uploaded twice!

# ls -l /proc/756/fd | wc -l
Quote:37
29
27
37
27
37
29
37
29



# ps -eo rss,size,vsize,pmem,pcpu,time,cmd --sort -rss | head -n 11 |grep asterisk
Quote:29100 3132608 3627272 0.4 4.6 00:00:00 /usr/sbin/asterisk -c
29260 3132608 3627272 0.4 3.7 00:00:00 /usr/sbin/asterisk -c
29364 3198640 3695440 0.4 2.8 00:00:00 /usr/sbin/asterisk -c
29648 3198640 3695440 0.4 2.5 00:00:00 /usr/sbin/asterisk -c
29676 3198640 3695440 0.4 2.2 00:00:00 /usr/sbin/asterisk -c
29620 3198640 3695440 0.4 2.0 00:00:00 /usr/sbin/asterisk -c
29916 3198640 3695440 0.4 1.7 00:00:00 /usr/sbin/asterisk -c
29860 3198640 3695440 0.4 1.6 00:00:00 /usr/sbin/asterisk -c
29860 3198640 3695440 0.4 1.6 00:00:00 /usr/sbin/asterisk -c
30072 3198640 3695440 0.4 1.4 00:00:01 /usr/sbin/asterisk -c
30072 3198640 3695440 0.4 1.4 00:00:01 /usr/sbin/asterisk -c
30072 3198640 3695440 0.4 1.4 00:00:01 /usr/sbin/asterisk -c
30232 3198640 3695440 0.4 1.3 00:00:01 /usr/sbin/asterisk -c
30176 3198640 3695440 0.4 1.3 00:00:01 /usr/sbin/asterisk -c
30176 3198640 3695440 0.4 1.3 00:00:01 /usr/sbin/asterisk -c


# valgrind --tool=memcheck --leak-check=yes --show-reachable=yes /usr/sbin/asterisk -c
Quote:==1504== Memcheck, a memory error detector
==1504== Copyright (C) 2002-2012, and GNU GPL'd, by Julian Seward et al.
==1504== Using Valgrind-3.8.1 and LibVEX; rerun with -h for copyright info
==1504== Command: /usr/sbin/asterisk -c
==1504==
Privilege escalation protection disabled!
See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
Asterisk already running on /var/run/asterisk/asterisk.ctl. Use 'asterisk -r' to connect.
==1504==
==1504== HEAP SUMMARY:
==1504== in use at exit: 381 bytes in 3 blocks
==1504== total heap usage: 141 allocs, 138 frees, 10,143 bytes allocated
==1504==
==1504== 8 bytes in 1 blocks are still reachable in loss record 1 of 3
==1504== at 0x4C2677B: calloc (vg_replace_malloc.c:593)
==1504== by 0x4F2C5E: ast_read_threadstorage_callid (utils.h:513)
==1504== by 0x4F21F9: ast_log (logger.c:1568)
==1504== by 0x513388: pbx_live_dangerously (pbx.c:4179)
==1504== by 0x4444E9: main (asterisk.c:3504)
==1504==
==1504== 93 bytes in 1 blocks are still reachable in loss record 2 of 3
==1504== at 0x4C2677B: calloc (vg_replace_malloc.c:593)
==1504== by 0x49960C: cfmtime_new (utils.h:513)
==1504== by 0x49E77D: config_text_file_load (config.c:1656)
==1504== by 0x49C703: ast_config_internal_load (config.c:2589)
==1504== by 0x49D0EA: ast_config_load2 (config.c:2610)
==1504== by 0x4436DA: main (asterisk.c:3274)
==1504==
==1504== 280 bytes in 1 blocks are still reachable in loss record 3 of 3
==1504== at 0x4C2677B: calloc (vg_replace_malloc.c:593)
==1504== by 0x4F1E56: ast_log_full (utils.h:513)
==1504== by 0x4F2247: ast_log (logger.c:1574)
==1504== by 0x513388: pbx_live_dangerously (pbx.c:4179)
==1504== by 0x4444E9: main (asterisk.c:3504)
==1504==
==1504== LEAK SUMMARY:
==1504== definitely lost: 0 bytes in 0 blocks
==1504== indirectly lost: 0 bytes in 0 blocks
==1504== possibly lost: 0 bytes in 0 blocks
==1504== still reachable: 381 bytes in 3 blocks
==1504== suppressed: 0 bytes in 0 blocks
==1504==
==1504== For counts of detected and suppressed errors, rerun with: -v
==1504== ERROR SUMMARY: 0 errors from 0 contexts (suppressed: 6 from 6)



Quote:*CLI> ==13035== Thread 48:
==13035== Use of uninitialised value of size 8
==13035== at 0x1EB98A27: pj_crc32_update (in /usr/lib64/asterisk/modules/res_rtp_asterisk.so)
==13035== by 0x1EB98AEA: pj_crc32_calc (in /usr/lib64/asterisk/modules/res_rtp_asterisk.so)
==13035== by 0x1EB8D641: pj_stun_msg_encode (in /usr/lib64/asterisk/modules/res_rtp_asterisk.so)
==13035== by 0x1EB8FEB0: send_response (in /usr/lib64/asterisk/modules/res_rtp_asterisk.so)
==13035== by 0x1EB9003B: authenticate_req (in /usr/lib64/asterisk/modules/res_rtp_asterisk.so)
==13035== by 0x1EB90369: on_incoming_request (in /usr/lib64/asterisk/modules/res_rtp_asterisk.so)
==13035== by 0x1EB9086D: pj_stun_session_on_rx_pkt (in /usr/lib64/asterisk/modules/res_rtp_asterisk.so)
==13035== by 0x1EB89A6A: pj_ice_sess_on_rx_pkt (in /usr/lib64/asterisk/modules/res_rtp_asterisk.so)
==13035== by 0x1EB7A2B9: __rtp_recvfrom.clone.3 (res_rtp_asterisk.c:1606)
==13035== by 0x1EB7F04E: ast_rtp_read (res_rtp_asterisk.c:1638)
==13035== by 0x223E855D: sip_read (chan_sip.c:8326)
==13035== by 0x4854F7: __ast_read (channel.c:4054)
==13035==
==13035== Syscall param socketcall.sendto(msg) points to uninitialised byte(s)
==13035== at 0x511AB63: ??? (in /lib64/libc-2.12.so)
==13035== by 0x1EBA9202: pj_sock_sendto (in /usr/lib64/asterisk/modules/res_rtp_asterisk.so)
==13035== by 0x1EB7617C: ast_rtp_on_ice_tx_pkt (res_rtp_asterisk.c:1152)
==13035== by 0x1EB8807C: on_stun_send_msg (in /usr/lib64/asterisk/modules/res_rtp_asterisk.so)
==13035== by 0x1EB8FF26: send_response (in /usr/lib64/asterisk/modules/res_rtp_asterisk.so)
==13035== by 0x1EB9003B: authenticate_req (in /usr/lib64/asterisk/modules/res_rtp_asterisk.so)
==13035== by 0x1EB90369: on_incoming_request (in /usr/lib64/asterisk/modules/res_rtp_asterisk.so)
==13035== by 0x1EB9086D: pj_stun_session_on_rx_pkt (in /usr/lib64/asterisk/modules/res_rtp_asterisk.so)
==13035== by 0x1EB89A6A: pj_ice_sess_on_rx_pkt (in /usr/lib64/asterisk/modules/res_rtp_asterisk.so)
==13035== by 0x1EB7A2B9: __rtp_recvfrom.clone.3 (res_rtp_asterisk.c:1606)
==13035== by 0x1EB7F04E: ast_rtp_read (res_rtp_asterisk.c:1638)
==13035== by 0x223E855D: sip_read (chan_sip.c:8326)
==13035== Address 0x26e9f95f is 79 bytes inside a block of size 1,000 alloc'd
==13035== at 0x4C279EE: malloc (vg_replace_malloc.c:270)
==13035== by 0x1EBA8838: default_block_alloc (in /usr/lib64/asterisk/modules/res_rtp_asterisk.so)
==13035== by 0x1EBAF04B: pj_pool_create_block (in /usr/lib64/asterisk/modules/res_rtp_asterisk.so)
==13035== by 0x1EBAF1F7: pj_pool_allocate_find (in /usr/lib64/asterisk/modules/res_rtp_asterisk.so)
==13035== by 0x1EBAEEE0: pj_pool_alloc (in /usr/lib64/asterisk/modules/res_rtp_asterisk.so)
==13035== by 0x1EB8FE7E: send_response (in /usr/lib64/asterisk/modules/res_rtp_asterisk.so)
==13035== by 0x1EB9003B: authenticate_req (in /usr/lib64/asterisk/modules/res_rtp_asterisk.so)
==13035== by 0x1EB90369: on_incoming_request (in /usr/lib64/asterisk/modules/res_rtp_asterisk.so)
==13035== by 0x1EB9086D: pj_stun_session_on_rx_pkt (in /usr/lib64/asterisk/modules/res_rtp_asterisk.so)
==13035== by 0x1EB89A6A: pj_ice_sess_on_rx_pkt (in /usr/lib64/asterisk/modules/res_rtp_asterisk.so)
==13035== by 0x1EB7A2B9: __rtp_recvfrom.clone.3 (res_rtp_asterisk.c:1606)
==13035== by 0x1EB7F04E: ast_rtp_read (res_rtp_asterisk.c:1638)
==13035==


Statistics : Posted by leunge • on Wed Jun 11, 2014 11:56 pm • Replies 16 • Views 410
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Troubleshooting advice needed - finger pointing by various

June 17, 2014, 7:29 pm
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Help,

Background - we are a long time user of Switchvox, currently on Switchvox 5.8.5 with Polycom phones.

We are also a longtime user of Junction Networks VOIP for additional and 800 lines.

We are no longer able to connect to Junction Networks as we could in the past - since switching to AT&T for internet and phones (though we did connect for about 1 week).

We previously used Cox cable for internet access, along with Bellsouth for POTS lines - and have switched to AT&T - consolidating the POTS lines plus the internet access to one account.

We have used Junction for additional VOIP lines for at least 3 years without issue.

In Switchvox - we cannot connect to Junction Networks. Junction have told us they can see the incoming traffic, but when they return the response, we are not accepting the traffic back.

When we get Cisco involved for the firewall/router, they have said that their test shows that our Switchvox is sending out malformed packets.

When we talk to Switchvox - they tell us using their tool that their system is fine - and not returning malformed packets.

Subsequent to switching to AT&T - we initially could not connect - but then magically, we did make a connection for about a week -then it disconnected again and never did reconnect.

We have been working on this for weeks on and off - and have never been able to reconnect to Junction Networks.

Our internal tech engineer has even connected our Switchvox directly to the Internet using the AT&T device, getting it a public IP address - and it still will not connect.

When we run the SIP Provider Diagnosis in Switchvox, all tests pass except the last - "SIP Response of VOIP Provider Host Test" - which fails "Your voip provider...is not responding to SIP OPTIONS requests."

I would appreciate any advice on how to troubleshoot this type of issue. I honestly believe that AT&T is blocking something, but have no idea how to prove what might be getting blocked.

Thanks in advance!

Statistics : Posted by coecpa • on Tue Jun 17, 2014 7:29 pm • Replies 0 • Views 17
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Search

Retrieve voicemail remotely using main line

June 17, 2014, 11:25 am
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Ah, just found something interesting. When prompted, if I enter the mailbox number and passcode at normal speed, it's only getting some of the digits (hence why it's puking on the login). However, when I enter the mailbox number and passcode very slowly, it picks up on all the digits and logs in successfully.

Is there a setting to tweak this?

Statistics : Posted by isgur513 • on Tue Jun 17, 2014 11:25 am • Replies 3 • Views 71
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Retrieve voicemail remotely using main line

June 17, 2014, 11:25 am
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This is a sip trunk coming into the Asterisk box. The trunk allows ulaw,alaw and dtmfmode is set to rfc2833. Like I said, it's strange that it works from local extensions but not over the trunk.

When watching the asterisk CLI during the call, I don't even see any errors or activity when pressing the digits.

-Matt

Statistics : Posted by isgur513 • on Tue Jun 17, 2014 11:25 am • Replies 3 • Views 71
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Retrieve voicemail remotely using main line

June 17, 2014, 11:25 am
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What transcoding and protocol changes happen between the PSTN and Asterisk. If any VoIP is involved, which DTMF mode is used, and do both sides agree?

Statistics : Posted by isgur513 • on Tue Jun 17, 2014 11:25 am • Replies 3 • Views 71
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Retrieve voicemail remotely using main line

June 17, 2014, 11:25 am
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I am trying to retrieve voicemail messages remotely by calling into the main phone line, waiting to hear the voicemail prompt, hitting * (which brings me to VoiceMailMain) and then typing the voicemail box/pin.

This is what I currently have. It gets me all the way to VoiceMailMain where I put in the voicemail box and the PIN and then returns a "passcode incorrect" message. The PIN is definitely correct and works when calling from internal extensions.

Code:
[from-pstn]
exten => 5555555555,1,Dial(SIP/Phone1,30,tr)
exten => 5555555555,2,VoiceMail(5555555555,s,u)
exten => a,1,VoiceMailMain
exten => a,n,Hangup()



Is there a better, more correct way of achieving this?

Statistics : Posted by isgur513 • on Tue Jun 17, 2014 11:25 am • Replies 3 • Views 71
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Event Trigger Timeout or Async

August 1, 2012, 1:19 pm
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Just found out the hard way: No it has not been implemented.

Statistics : Posted by wasabi • on Wed Aug 01, 2012 1:19 pm • Replies 4 • Views 1763
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