[asterisk-users] Strange Invite issue
Tarek Sawah
tareksawah at hotmail.com
Fri Apr 30 14:11:57 CDT 2010
Before posting let me mention that this doesn't happen with ALL destination on this provider.. some destination doesn't face this problem .. but this is a sample call
[K -- Executing [0020100324519 at a2billing:1] [1;36;40mDeadAGI[0;37;40m("[1;35;40mSIP/58169-ac47fda0[0;37;40m", "[1;35;40ma2billing.php|1[0;37;40m") in new stack
[K -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
-- AGI Script Executing Application: (Dial) Options: (SIP/PROVIDER1/20100324519|60|HL(166986000:61000:30000)) -- Limit Data for this call: > timelimit = 166986000 > play_warning = 61000 > play_to_caller = yes > play_to_callee = no > warning_freq = 30000 > start_sound = (null) > warning_sound = timeleft > end_sound = (null)Audio is at 100.X.Y.Z port 13984Adding codec 0x100 (g729) to SDPAdding non-codec 0x1 (telephone-event) to SDPReliably Transmitting (no NAT) to 195.X.Y.Z:5060:INVITE sip:20100324519 at 195.X.Y.Z SIP/2.0
Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport
From: "58169" <sip:58169 at 100.X.Y.Z>;tag=as00522e07
To: <sip:20100324519 at 195.X.Y.Z>
Contact: <sip:58169 at 100.X.Y.Z>
Call-ID: 7f169cce7003bb01365f72ce2a3aaeba at 100.X.Y.Z
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 30 Apr 2010 18:52:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 267
v=0
o=root 12516 12516 IN IP4 100.X.Y.Z
s=session
c=IN IP4 100.X.Y.Z
t=0 0
m=audio 13984 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
--- -- Called PROVIDER1/20100324519
[K
<--- SIP read from 195.X.Y.Z:5060 --->SIP/2.0 100 Trying
Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060
From: "58169" <sip:58169 at 100.X.Y.Z>;tag=as00522e07
To: <sip:20100324519 at 195.X.Y.Z>;tag=gK02b3c8db
Call-ID: 7f169cce7003bb01365f72ce2a3aaeba at 100.X.Y.Z
CSeq: 102 INVITE
Content-Length: 0
<------------->
[K --- (7 headers 0 lines) ---
[K
<--- SIP read from 195.X.Y.Z:5060 --->SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060
From: "58169" <sip:58169 at 100.X.Y.Z>;tag=as00522e07
To: <sip:20100324519 at 195.X.Y.Z>;tag=gK02b3c8db
Call-ID: 7f169cce7003bb01365f72ce2a3aaeba at 100.X.Y.Z
CSeq: 102 INVITE
Contact: <sip:20100324519 at 195.X.Y.Z:5060>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Content-Length: 260
Content-Disposition: session; handling=required
Content-Type: application/sdp
v=0
o=Sonus_UAC 10183 6645 IN IP4 195.X.Y.Z
s=SIP Media Capabilities
c=IN IP4 195.219.240.5
t=0 0
m=audio 15846 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendonly
a=maxptime:20
<------------->
[K --- (11 headers 12 lines) ---
[K Found RTP audio format 18
[K Found RTP audio format 101
[K Peer audio RTP is at port 195.219.240.5:15846
[K Found audio description format G729 for ID 18
[K Found audio description format telephone-event for ID 101
[K Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
[K Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[K Peer audio RTP is at port 195.219.240.5:15846
[K -- SIP/PROVIDER1-1fd586a0 is ringing
[K -- Call on SIP/PROVIDER1-1fd586a0 placed on hold
[K -- Started music on hold, class 'default', on SIP/58169-ac47fda0
[K -- SIP/PROVIDER1-1fd586a0 is making progress passing it to SIP/58169-ac47fda0
[K sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 195.X.Y.Z 2010032451 7f169cce700 00102/00000 0x100 (g729) Yes Init: INVITE 78.184.197.119 58169 AC8455D8edd 00101/160518 0x4 (ulaw) No Rx: INVITE 2 active SIP channels
[K
<--- SIP read from 195.X.Y.Z:5060 --->SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060
From: "58169" <sip:58169 at 100.X.Y.Z>;tag=as00522e07
To: <sip:20100324519 at 195.X.Y.Z>;tag=gK02b3c8db
Call-ID: 7f169cce7003bb01365f72ce2a3aaeba at 100.X.Y.Z
CSeq: 102 INVITE
Contact: <sip:20100324519 at 195.X.Y.Z:5060>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Content-Length: 0
<------------->
[K --- (9 headers 0 lines) ---
[K -- SIP/PROVIDER1-1fd586a0 is ringing
-- Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
USA: +1 347 562 2308
> Date: Thu, 29 Apr 2010 16:52:24 +0100
> From: list-asterisk at skycomuk.com
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] Strange Invite issue
>
> Can you post a sip debug
>
> Tarek Sawah wrote:
>> Greetings List.
>> I'm facing a strange issue with one of my providers.. after sending an INVITE request my server places the call on hold.. until the call is answered..
>> this is happening only with this provide although i have 3 other providers i route calls through..
>> can anyone explain what is going on?
>>
>> --
>> Tarek Sawah / Integrated Digital Systems / CCNA, MCSE, RHCE, VoIP / +1 347 562 2308
>>
>>
>>
>>
>>
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