then why is it happening on a few destinations on that particular provider?
________________________________
> Date: Fri, 30 Apr 2010 13:09:05 -0700
> From: david.wh...@watchguard.com
> To: asterisk-users@lists.digium.com; asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Strange Invite issue
>
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> in the SIP/2.0 180 Ringing, the SDP shows:
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> a=sendonly
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> this is "hold" by rfc 3264. then when the other end picks up, a new SDP is
> probably sent with
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> a=sendrecv
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> I believe your server is acting correctly.
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>
> -----Original Message-----
>
> From: asterisk-users-boun...@lists.digium.com on behalf of Tarek Sawah
>
> Sent: Fri 4/30/2010 12:11 PM
>
> To: Asterisk Users
>
> Subject: Re: [asterisk-users] Strange Invite issue
>
>
>
>
>
> 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 [0020100324...@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:20100324...@195.x.y.z SIP/2.0
>
> Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport
>
> From: "58169" ;tag=as00522e07
>
> To:
>
> Contact:
>
> Call-ID: 7f169cce7003bb01365f72ce2a3aa...@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" ;tag=as00522e07
>
> To: ;tag=gK02b3c8db
>
> Call-ID: 7f169cce7003bb01365f72ce2a3aa...@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" ;tag=as00522e07
>
> To: ;tag=gK02b3c8db
>
> Call-ID: 7f169cce7003bb01365f72ce2a3aa...@100.x.y.z
>
> CSeq: 102 INVITE
>
> Contact:
>
> 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" ;tag=as00522e07
>
> To: ;tag=gK02b3c8db
>
> Call-ID: 7f169cce7003bb01365f72ce2a3aa...@100.x.y.z
>
> CSeq: 102 INVITE
>
> Contact:
>
> 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
>
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>
>
>
> -- Tarek Sawah
>
>
>
> Integrated Digital Systems
>
>
>
> CCNA, MCSE, RHCE, VoIP
>
>
>
>
>
> USA: +1 347 562 2308
>
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>
>> Date: Thu, 29 Apr 2010 16:52:24 +0100
>
>> From: list-aster...@skycomuk.com
>
>> To: asterisk-users@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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>>
>
>>
>
>> --
>
>> _____________________________________________________________________
>
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