[asterisk-users] Strange Invite issue
Tarek Sawah
tareksawah at hotmail.com
Fri Apr 30 16:49:16 CDT 2010
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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>> 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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