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Asterisk and T38 problem

October 16, 2015, 2:15 am
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Hello, I have managed to get the information that you asked. In the following link is a test call that the client made

http://www.filedropper.com/testcall

Statistics : Posted by _alx_ • on Fri Oct 16, 2015 2:15 am • Replies 12 • Views 856
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Asterisk and T38 problem

October 16, 2015, 2:15 am
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I believe 1.8 is past end of life.

You need to fix the bug. Whilst there is a bug in some versions of Asterisk that causes problem if the peer uses anything but port 101 for telephone events (RFC 2833), I don't think 127 is a likely port number for the peer to use for that, so I presume the bug lies in the peer. sip set debug on and debug level 5 output from chan_sip would be needed to confirm where blame lies.

Statistics : Posted by _alx_ • on Fri Oct 16, 2015 2:15 am • Replies 12 • Views 856
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Asterisk and T38 problem

October 16, 2015, 2:15 am
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Hello,

First post here! Image

I have an Asterisk box with 1.8.11, I use a sip trunk for my calls. I have also two dahdi extensions that my fax machines are connected to.

In my sip trunk I have enabled the t38 passthrough function.

When I try to send/receive a fax through the sip trunk, the procedure fails. If I turn on the sip and rtp debug I get the following error

NOTICE[22329] res_rtp_asterisk.c: Unknown RTP codec 127 received from '"public_IP_of_ my_sip_provider:29548'

Is there anything that I can try to avoid this message? Also is it right to enable T38 on my sip trunk even if I am using dahdi extensions to send and receive fax?

Statistics : Posted by _alx_ • on Fri Oct 16, 2015 2:15 am • Replies 12 • Views 856
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Func_odbc select special characters

November 10, 2015, 2:23 pm
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Func_odbc select special characters

November 10, 2015, 2:23 pm
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Hi all...not much an asterisk related question but...

i made in func_odbc e select query....but NOMINATIVO value lost all special characters such as quote and double quote..how can i escape these?

[IVRTAB1_11]
dsn=poiana
readsql= SELECT NOMINATIVO from tab1 WHERE CODICE_CONTRATTO=${codutente}

Many Thanks

Statistics : Posted by simone686 • on Tue Nov 10, 2015 4:23 pm • Replies 1 • Views 91
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Retransmisson

November 11, 2015, 3:30 am
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hello guys, I'm with a relay problem, where the customer has 200 branches on an internal network, and retransmits happens all day, retransmission of INVITE, CANCEL and even BYE

one example:
[2015-11-11 09:16:58] DEBUG[21120] chan_sip.c: Trying to put 'CANCEL sip:' onto UDP socket destined for 10.4.0.93:5060
[2015-11-11 09:16:59] VERBOSE[24011] chan_sip.c: Retransmitting #1 (no NAT) to 10.4.0.93:5060:

Asterisk 1.8.26.1

Statistics : Posted by diegocorp • on Wed Nov 11, 2015 5:30 am • Replies 0 • Views 17
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callerid asterisk-h323-avaya ipo500

November 11, 2015, 4:01 am
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Hi.
Sorry, I'm very bad speak english.
asterisk 12 - h323 - avaya ipo 500 . Call back and forth works fine.
But when you call asterisk-h323-avaya caller id only works number and no name.
Code: exten => _22XX,1,Noop(Now Caller ID is ${CALLERID(all)} that mean CID name = ${CALLERID(name)} and CID num = ${CALLERID(num)})
exten => _22XX,n,Dial(H323/${EXTEN}@192.168.13.13)    - 192.168.13.13 - ip avaya ipo
exten => _22XX,n,Hangup()

sip.conf
Code:     [21]
    context =  bells
    type = friend
    username = 21
    defaultuser = 21
    canreinvite = no
    directmedia = no
    nat = no
    dtmfmode = auto
    secret = ***
    qualify = 4000
    disallow = all
    allow = alaw
    allow = ulaw
    host = 10.10.24.42
    callerid = "T" <21>

debug
Code:     -- Executing [2099@bell:1] NoOp("SIP/21-0000002b", "Now Caller ID is "T" <21> that mean CID name = T and CID num = 21") in new stack
    -- Executing [2099@bell:2] Dial("SIP/21-0000002b", "H323/2099@192.168.13.13") in new stack
    -- Requested transfer capability: 0x00 - SPEECH
    -- Called H323/2099@192.168.13.13
    -- H323/192.168.13.13-27 is ringing

<--- Transmitting (no NAT) to 192.168.13.154:49945 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.13.154:49945;branch=z9hG4bK80f9856ed886e5118b3bd232d24fb09b;received=192.168.13.154;rport=49945
From: <sip:21@192.168.13.2>;tag=1512041172
To: <sip:2099@192.168.13.2>;tag=as6c1ddc45
Call-ID: 80F9856E-D886-E511-8B39-D232D24FB09B@192.168.13.154
CSeq: 20 INVITE
Server: Asterisk PBX 12.2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:2099@192.168.13.13:5060>
Content-Length: 0


Statistics : Posted by flatic • on Wed Nov 11, 2015 6:01 am • Replies 0 • Views 15
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[PJSIP] Incoming ITSP working at last! Is this a sane config

November 12, 2015, 7:07 am
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After 3-and-a-half long and frustrating days, I've finally figured out how to connect my ITSP (Voipfone) to an Asterisk install so that I can:

  • Play the correct message depending on the number being called.
  • Store the date/time/CID of the person calling and which group they called.
  • Not allow outgoing calls (message service only).

Note that with Voipfone, you can only have the calling number and any name set in the control panel.

The following config is designed to provide me with enough info as a learning basis, the idea being that anything from this point on is just dialplan - buy a number from Voipfone, give it a name in their control panel, and then just add a dialplan section with that name.

Does this look sane and safe for the time being?

pjsip.conf

Code: [simpletrans]
type = transport
protocol = UDP
bind = 0.0.0.0

[voipfone]
type = registration
retry_interval = 20
max_retries = 10
contact_user = 301XXXXX
transport = simpletrans
outbound_auth = voipfone-auth
server_uri = sip:301XXXXX@sip.voipfone.net:5060
;Note: IP in next line is MY Asterisk IP
client_uri = sip:301XXXXXN@46.101.37.X:5060

[voipfone-auth]
type = auth
auth_type = userpass
username = 301XXXXX
password = XXXXX

[voipfone-identify]
type = identify
endpoint = Elephant
; You cannot use FQDNs or hostnames. You must use IP addresses.
match = 195.189.173.27

[Elephant]
type = endpoint
transport=simpletrans
context = fromvoipfone
disallow = all
allow = alaw
outbound_auth = voipfone-auth

[acl]
type = acl
deny = 0.0.0.0/0.0.0.0
permit = 195.189.173.27


extensions.conf

Code: [fromvoipfone]
exten => _X.,1,Answer()
same => n,Set(VOLUME(TX)=8)
same => n,Wait(1)
same => n,DumpChan()
;same => n,SayNumber(${EXTEN})
same => n,Verbose(The caller ID all is ${CALLERID(all)})
same => n,Set(DB(${CALLERID(name)}/${CALLERID(num)})=${STRFTIME(${EPOCH},,%d%m%Y-%H:%M:%S)})
same => n,Playback(hello-world)
same => n,Goto(${CALLERID(name)},s,1)
;same = n,ControlPlayback(demo-instruct,5000,6,4,8,5)
same => n,Hangup()

[Chicken]
exten => s,1,Answer(500)
same => n,SayNumber(1)

[Elephant]
exten => s,1,Answer(500)
same => n,SayNumber(2)


Statistics : Posted by lardconcepts • on Thu Nov 12, 2015 9:07 am • Replies 0 • Views 20
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SIP trunk registers but incoming call fails with error 481

November 9, 2015, 8:37 am
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Thanks for the help here - finally resolved it. Will type up more detailed explanation when I've caught up on sleep!

Meantime, I've started a new clean thread with a working config viewtopic.php?f=1&t=96178

Statistics : Posted by lardconcepts • on Mon Nov 09, 2015 10:37 am • Replies 5 • Views 202
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SIP trunk registers but incoming call fails with error 481

November 9, 2015, 8:37 am
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Thanks for your help so far, and it's guided me to the correct bits of the wiki, but I am totally stumped. Just hitting midnight here and been at this since about 8am.

I just go round and round and round and round with google, trying to find example configs for other ITSPs that I might be able to adapt. And round and round and round and round with the wiki, going over it again and again and again with a fine tooth comb, checking the endpoints, identifiers and auth, looking at the logs, matching it to what should be happening.

https://wiki.asterisk.org/wiki/display/ ... istrations
https://wiki.asterisk.org/wiki/display/ ... gistration
https://wiki.asterisk.org/wiki/display/ ... _pjsip_acl
https://wiki.asterisk.org/wiki/display/ ... ion+Wizard
https://wiki.asterisk.org/wiki/display/ ... +res_pjsip

Going to try and sleep on it, but if anyone can shed any light meantime I'd be massively grateful!

Thanks.

Statistics : Posted by lardconcepts • on Mon Nov 09, 2015 10:37 am • Replies 5 • Views 202
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SIP trunk registers but incoming call fails with error 481

November 9, 2015, 8:37 am
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The 481 is a secondary fault. The primary problem is that the ITSP has cancelled, a transaction that was already complete. It has clearly cancelled it because it got a 401 insisting on its supplying authentication data, which it is not going to be able to do. It is not legal to cancel in response to 401, because 401 is a final response, and completes the transaction.

Also, you have either inconsistently redacted or the cancel doesn't even include the tags and ids for a call that ever existed.

Statistics : Posted by lardconcepts • on Mon Nov 09, 2015 10:37 am • Replies 5 • Views 202
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SIP trunk registers but incoming call fails with error 481

November 9, 2015, 8:37 am
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Thanks for your quick reply, David.

The thing is, though - registration IS successful. The two endpoints are connected together, it's just that a fraction of a second before the call bounces back to my SIP trunk supplier's default failover voicemail, it does actually hit my endpoint. From what Voipfone told me, there's no authentication issue here.

Besides, pjsip appears not to offer any option of "remotesecure" or "insecure".

I think the answer lies more in the error "SIP/2.0 481 Call/Transaction Does Not Exist" and a possible misconfig elsewhere, rather than authentication. But I could be wrong!

Statistics : Posted by lardconcepts • on Mon Nov 09, 2015 10:37 am • Replies 5 • Views 202
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SIP trunk registers but incoming call fails with error 481

November 9, 2015, 8:37 am
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You are asking them to authenticate themselves. ITSP's generally won't do that, so they abort the call.

On chan_sip, you would use remotesecret, rather than secret, or the older hack of insecure=invite, with secret. I'm not familiar with PJSIP.

Statistics : Posted by lardconcepts • on Mon Nov 09, 2015 10:37 am • Replies 5 • Views 202
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SIP trunk registers but incoming call fails with error 481

November 9, 2015, 8:37 am
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I am using a SIP trunk provider in the UK called Voipfone who provide extensions in the format of
accountID*extension, eg: 123456*200.

You "attach" a real phone number to that extension. ie: someone dialing 0123-456 would be attached to extension 200.

The Caller ID can be either in the form of the number calling or the number called.
I've tried both.

I also tried changing and commenting out from_user and from_domain because at this stage it doesn't seem to matter. I've only got one incoming line to play a test message.

I also tried the various NAT and port fixes currently as shown at the end of the endpoint config section.

I got in touch with the trunk provider who said everything looked good from their end, except that I appeared to have SIP ALG enabled - he said I might want to disable this.

This is running on a Ubuntu 15.10 Digitalocean Droplet with currently no firewall running AFAIK.
Nonetheless, I googled and applied modprobe -r nf_nat_sip but with no success.

pjsip config examples are rather a rare beast at the moment, so I'm going on a mix of the pjsip.sample, various wiki entries and posts in this forum. If anyone can spot anything wrong here I'd be hugely grateful!

extensions.conf

Code: [fromvoipfone]
exten => 200,1,Answer()
same => n,Wait(1)
same => n,Playback(hello-world)
same => n,Hangup()


pjsip.conf

Code: ;===============TRANSPORT

[simpletrans]
type = transport
protocol = UDP
bind = 0.0.0.0

;===============VOIPFONE

[voipfone]
type = registration
retry_interval = 20
max_retries = 10
contact_user = voipfone
expiration = 120
transport = simpletrans
outbound_auth = voipfone
client_uri = sip:3XXXXXX*200@sip.voipfone.net
server_uri = sip:sip.voipfone.net

[voipfone]
type = aor
contact = sip:3XXXXXX*200@sip.voipfone.net

[voipfone]
type = identify
endpoint = voipfone
match = sip.voipfone.net

[voipfone]
type = auth
auth_type = userpass
username = 3XXXXXX*200
password = XXXXXX

[voipfone]
type = endpoint
context = fromvoipfone
dtmf_mode = rfc4733
disallow = all
allow = alaw,gsm
;from_user = 3XXXXXX*200
;from_domain = sip.voipfone.net
auth = voipfone
direct_media=no
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes  ; necessary if endpoint does not know/register public ip:port
ice_support=yes
aors = voipfone


Log
Code: <--- Received SIP request (867 bytes) from UDP:195.189.173.27:5060 --->
INVITE sip:voipfone@X.X.X.X:5060 SIP/2.0
Via: SIP/2.0/UDP 195.189.173.27:5060;branch=XXXXXXX;rport
From: "01XXXXXXXXX" <sip:01XXXXXXXXX@195.189.173.27>;tag=XXXXXXX
To: <sip:voipfone@X.X.X.X:5060>
Contact: <sip:01XXXXXXXXX@195.189.173.27>
Call-ID: VFffdd4b638936ad3eafbd811f4bb292@voipfone
CSeq: 102 INVITE
User-Agent: Voipfone Sip Network
Date: Mon, 09 Nov 2015 16:10:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 337

v=0
o=root 26994 26994 IN IP4 195.189.173.27
s=session
c=IN IP4 195.189.173.27
t=0 0
m=audio 13002 RTP/AVP 8 2 97 3 110 101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

<--- Transmitting SIP response (533 bytes) to UDP:195.189.173.27:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 195.189.173.27:5060;rport=5060;received=195.189.173.27;branch=XXXXXXX
Call-ID: VFffdd4b638936ad3eafbd811f4bb292@voipfone
From: "01XXXXXXXXX" <sip:01XXXXXXXXX@195.189.173.27>;tag=XXXXXXX
To: <sip:voipfone@X.X.X.X>;tag=XXXXXXX
CSeq: 102 INVITE
WWW-Authenticate: Digest  realm="asterisk",nonce="XXXXXXX",opaque="XXXXXXX",algorithm=md5,qop="auth"
Server: Asterisk PBX 13.6.0
Content-Length:  0


<--- Received SIP request (411 bytes) from UDP:195.189.173.27:5060 --->
CANCEL sip:voipfone@X.X.X.X:5060 SIP/2.0
Via: SIP/2.0/UDP 195.189.173.27:5060;branch=z9hG4bK107fcf4f;rport
From: "01XXXXXXXXX" <sip:01XXXXXXXXX@195.189.173.27>;tag=VF7ac63d4bd8208e0621d7096b2824
To: <sip:voipfone@X.X.X.X:5060>
Contact: <sip:01XXXXXXXXX@195.189.173.27>
Call-ID: XXXXXXX@voipfone
CSeq: 102 CANCEL
User-Agent: Voipfone Sip Network
Content-Length: 0


[Nov  9 16:10:36] ERROR[811]: res_pjsip/pjsip_distributor.c:221 find_dialog: Could not find matching INVITE transaction for CANCEL request
<--- Transmitting SIP response (405 bytes) to UDP:195.189.173.27:5060 --->
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 195.189.173.27:5060;rport=5060;received=195.189.173.27;branch=XXXXXXX
Call-ID: VFffdd4b638936ad3eafbd811f4bb292@voipfone
From: "01XXXXXXXXX" <sip:01XXXXXXXXX@195.189.173.27>;tag=VF7ac63d4bd8208e0621d7096b2824
To: <sip:voipfone@X.X.X.X>;tag=z9hG4bK107fcf4f
CSeq: 102 CANCEL
Server: Asterisk PBX 13.6.0
Content-Length:  0


(BTW - yes, I know it's probably a security nightmare at the moment - there's no credit in the account so no calls can be made, it gets turned off when I'm not actively working on it, and I've got a console log open to watch for baddies!)

Statistics : Posted by lardconcepts • on Mon Nov 09, 2015 10:37 am • Replies 5 • Views 202
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WARNING message in log - Unable to run script

November 12, 2015, 7:28 am
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[Nov 12 08:51:32] NOTICE[20141]: pbx_spool.c:362 attempt_thread: Call completed to SIP/9548373029@voipms
-- Stopped music on hold on SIP/voipms-00000000
-- Executing [s@JoinConf:6] Set("SIP/voipms-00000000", "fhere=0") in new stack
-- Executing [s@JoinConf:7] ExecIf("SIP/voipms-00000000", "0?Playback(please_wait)") in new stack
-- Remote UNIX connection
-- Remote UNIX connection disconnected
-- Executing [s@JoinConf:8] ExecIf("SIP/voipms-00000000", "0?ExecIf(0?System(/var/lib/asterisk/agi-bin/call.sh 151112085013 3198009847))") in new stack
-- Executing [s@JoinConf:9] Set("SIP/voipms-00000000", "c=21") in new stack
-- Executing [s@JoinConf:10] EndWhile("SIP/voipms-00000000", "") in new stack
-- Executing [s@JoinConf:4] While("SIP/voipms-00000000", "1") in new stack
-- Executing [s@JoinConf:5] MusicOnHold("SIP/voipms-00000000", "default,3") in new stack
-- Started music on hold, class 'default', on SIP/voipms-00000000
-- Stopped music on hold on SIP/voipms-00000000
== Spawn extension (JoinConf, s, 5) exited non-zero on 'SIP/voipms-00000000'
-- Executing [h@JoinConf:1] System("SIP/voipms-00000000", "/var/lib/asterisk/agi-bin/uhangup.sh 151112085013") in new stack
[Nov 12 08:51:34] WARNING[19843]: app_system.c:134 system_exec_helper: Unable to execute '/var/lib/asterisk/agi-bin/uhangup.sh 151112085013'
== Spawn extension (JoinConf, h, 1) exited non-zero on 'SIP/voipms-00000000'
-- Executing [s@outgoing:1] Answer("SIP/voipms-00000005", "") in new stack
-- Executing [s@outgoing:2] ExecIf("SIP/voipms-00000005", "1?System(/var/lib/asterisk/agi-bin/missed.sh bbrown)") in new stack
-- Executing [s@outgoing:3] BackGround("SIP/voipms-00000005", "out_menu2") in new stack
-- <SIP/voipms-00000005> Playing 'out_menu2.ulaw' (language 'en')
-- Executing [1@outgoing:1] Goto("SIP/voipms-00000005", "joincall,s,1") in new stack
-- Goto (joincall,s,1)


the uhangup script just cleans up some files that other scripts create. Any idea why I wouldn't be able to run it?

Statistics : Posted by pistonhonda • on Thu Nov 12, 2015 9:28 am • Replies 0 • Views 17
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WARNING message in log - Unable to run script

November 12, 2015, 7:28 am
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This has been resolved. The script wasn't in the folder Image

Statistics : Posted by pistonhonda • on Thu Nov 12, 2015 9:28 am • Replies 0 • Views 17
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Ring Groups to Conference

November 12, 2015, 8:03 am
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Hi there..

As a newbie on asterisk..need some help to combine ring group to conference (or something like that).

1. I have 5 SIP extensions local/internal (2001 to 2005) up and running
2. And a conference code (50) working as well
3. A ring group (301).
with ring all, but as soon as one person answers, the other extensions in the group are dropped

Now the requirement is to keep ringing these extensions so that everybody that picks up the call lands in a conference with the caller.

Thank you

Statistics : Posted by xanalex • on Thu Nov 12, 2015 10:03 am • Replies 0 • Views 11
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[PJSIP] Requirement of "line=yes" in ITSP peering/incoming?

November 11, 2015, 2:49 pm
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A quick search suggests that it may be a proprietary extension to SIP, which, if implemented by the peer, allows you to have multiple registrations to the same peer and with either the same contact user, or with the peer overriding the contact user information (e.g. with incoming dialed digits), and still being able to distinguish between calls associated with different registrations. If so, it will be dependent on the peer supporting the feature.

Statistics : Posted by lardconcepts • on Wed Nov 11, 2015 4:49 pm • Replies 2 • Views 125
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[PJSIP] Requirement of "line=yes" in ITSP peering/incoming?

November 11, 2015, 2:49 pm
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What does "line" mean in terms of registration to an ITSP here the definition is given as

Quote:When enabled this option will cause a 'line' parameter to be added to the Contact header placed into the outgoing registration request. If the remote server sends a call this line parameter will be used to establish a relationship to the outbound registration, ultimately causing the configured endpoint to be used.


Isn't that exactly what I want? But none of the examples give this. For example, nothing in res_pjsip+Configuration+Examples nor in Endpoints and Location, A Match Made in Heaven nor in Asterisk 13 Configuration_res_pjsip so I'm a bit confused!

But it is mentioned in https://wiki.asterisk.org/wiki/display/ ... gistration

Quote:; If you would like to enable line support and have incoming calls related to this
; registration go to an endpoint automatically the "line" and "endpoint" options must
; be set. The "endpoint" option specifies what endpoint the incoming call should be
; associated with.


I was wondering if this was why incoming calls were getting cancelled, but even with and without these config settings, the call never gets to the dialplan.

Statistics : Posted by lardconcepts • on Wed Nov 11, 2015 4:49 pm • Replies 2 • Views 125
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Complete newbie need help in understanding how this works

October 19, 2015, 6:02 pm
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There are may ways to do whatever you like. If you need to place or receive a call from the telephone company using an RJ-11 standard phone jack, you may want to look at purchasing an FXO card. Digium makes a 4 port card that you can add modules to. I have an analog card with 1 FXS (phone side) and 1 FXO (phone company side) port. You'll need to configure dahdi to communicate with the Linux kernel, and chan_dahdi to communicate with Asterisk. On the inside, you could just as easily use a SIP-based phone connected to your network, or a soft(ware) phone installed on a computer or cell phone. If you go with either of those two, you won't need an FXS module.

In /etc/dahdi/system.conf, you'll need something like this...
loadzone = us
defaultzone=us
fxoks=1
fxsks=4

It defines the signalling for the card. FXS ports require FXO signalling and vice versa.

/etc/asterisk/chan_dahdi.conf will look something like this...
[channels]
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
callgroup=1
pickupgroup=1
callerid="Martin"
context=internal
signaling=fxo_ks
channel=>1

callerid=asreceived
context=inbound
signaling=fxs_ks
channel=>4

Your inbound context would just dial whatever internal extension you want, and you would dial out using something like this...
exten=> _NXXZXXXXXX,1,Dial(DAHDI/4/${EXTEN})

Statistics : Posted by rhymeguy • on Mon Oct 19, 2015 6:02 pm • Replies 2 • Views 387
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