[asterisk-users] Strange Invite issue
David White
David.White at watchguard.com
Fri Apr 30 17:02:05 CDT 2010
I don't know in your particular case, but if I call a PSTN endpoint via my provider, the SIP signaling is different than if I'm calling a remote SIP endpoint. This is because PSTN gateways have to make decisions (about codecs, eg) independently of the remote endpoints.
In other words, remote SIP endpoints generate their own SDPs, which your provider forwards to you. Gateways often have to generate their own. Those SDPs will necessarily be different.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com on behalf of Tarek Sawah
Sent: Fri 4/30/2010 2:49 PM
To: Asterisk Users
Subject: Re: [asterisk-users] Strange Invite issue
then why is it happening on a few destinations on that particular provider?
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> Date: Fri, 30 Apr 2010 13:09:05 -0700
> From: David.White at watchguard.com
> To: asterisk-users at lists.digium.com; asterisk-users at 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-----
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> From: asterisk-users-bounces at lists.digium.com on behalf of Tarek Sawah
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> Sent: Fri 4/30/2010 12:11 PM
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> To: Asterisk Users
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> Subject: Re: [asterisk-users] Strange Invite issue
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> 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
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> [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
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> [K -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
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> -- 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
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> Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport
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> From: "58169" ;tag=as00522e07
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> To:
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> Contact:
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> Call-ID: 7f169cce7003bb01365f72ce2a3aaeba at 100.X.Y.Z
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> CSeq: 102 INVITE
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> User-Agent: Asterisk PBX
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> Max-Forwards: 70
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> Date: Fri, 30 Apr 2010 18:52:23 GMT
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> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
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> Supported: replaces
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> Content-Type: application/sdp
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> Content-Length: 267
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> v=0
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> o=root 12516 12516 IN IP4 100.X.Y.Z
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> s=session
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> c=IN IP4 100.X.Y.Z
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> t=0 0
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> m=audio 13984 RTP/AVP 18 101
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> a=rtpmap:18 G729/8000
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> a=fmtp:18 annexb=no
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> a=rtpmap:101 telephone-event/8000
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> a=fmtp:101 0-16
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> a=silenceSupp:off - - - -
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> a=ptime:20
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> a=sendrecv
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> --- -- Called PROVIDER1/20100324519
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> [K
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> <--- SIP read from 195.X.Y.Z:5060 --->SIP/2.0 100 Trying
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> Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060
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> From: "58169" ;tag=as00522e07
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> To: ;tag=gK02b3c8db
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> Call-ID: 7f169cce7003bb01365f72ce2a3aaeba at 100.X.Y.Z
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> CSeq: 102 INVITE
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> Content-Length: 0
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> <------------->
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> [K --- (7 headers 0 lines) ---
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> [K
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> <--- SIP read from 195.X.Y.Z:5060 --->SIP/2.0 180 Ringing
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> Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060
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> From: "58169" ;tag=as00522e07
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> To: ;tag=gK02b3c8db
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> Call-ID: 7f169cce7003bb01365f72ce2a3aaeba at 100.X.Y.Z
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> CSeq: 102 INVITE
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> Contact:
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> Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
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> Content-Length: 260
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> Content-Disposition: session; handling=required
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> Content-Type: application/sdp
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> v=0
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> o=Sonus_UAC 10183 6645 IN IP4 195.X.Y.Z
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> s=SIP Media Capabilities
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> c=IN IP4 195.219.240.5
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> t=0 0
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> m=audio 15846 RTP/AVP 18 101
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> a=rtpmap:18 G729/8000
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> a=fmtp:18 annexb=no
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> a=rtpmap:101 telephone-event/8000
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> a=fmtp:101 0-15
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> a=sendonly
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> a=maxptime:20
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> <------------->
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> [K --- (11 headers 12 lines) ---
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> [K Found RTP audio format 18
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> [K Found RTP audio format 101
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> [K Peer audio RTP is at port 195.219.240.5:15846
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> [K Found audio description format G729 for ID 18
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> [K Found audio description format telephone-event for ID 101
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> [K Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
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> [K Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
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> [K Peer audio RTP is at port 195.219.240.5:15846
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> [K -- SIP/PROVIDER1-1fd586a0 is ringing
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> [K -- Call on SIP/PROVIDER1-1fd586a0 placed on hold
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> [K -- Started music on hold, class 'default', on SIP/58169-ac47fda0
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> [K -- SIP/PROVIDER1-1fd586a0 is making progress passing it to SIP/58169-ac47fda0
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> [K sip show channels
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> 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
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> [K
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> <--- SIP read from 195.X.Y.Z:5060 --->SIP/2.0 180 Ringing
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> Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport=5060
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> From: "58169" ;tag=as00522e07
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> To: ;tag=gK02b3c8db
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> Call-ID: 7f169cce7003bb01365f72ce2a3aaeba at 100.X.Y.Z
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> CSeq: 102 INVITE
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> Contact:
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> Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
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> Content-Length: 0
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> <------------->
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> [K --- (9 headers 0 lines) ---
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> [K -- SIP/PROVIDER1-1fd586a0 is ringing
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> -- Tarek Sawah
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> Integrated Digital Systems
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> CCNA, MCSE, RHCE, VoIP
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> USA: +1 347 562 2308
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>> 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
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>>
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>> Can you post a sip debug
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>>
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>> Tarek Sawah wrote:
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>>> 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..
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>>> this is happening only with this provide although i have 3 other providers i route calls through..
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>>> can anyone explain what is going on?
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>>>
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>>> --
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>>> Tarek Sawah / Integrated Digital Systems / CCNA, MCSE, RHCE, VoIP / +1 347 562 2308
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>>>
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>>>
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>>>
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>>>
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>>>
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>>> _________________________________________________________________
>
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>>
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>>
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>> --
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