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: [email protected] 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? ________________________________ > Date: Fri, 30 Apr 2010 13:09:05 -0700 > From: [email protected] > To: [email protected]; [email protected] > Subject: Re: [asterisk-users] Strange Invite issue > > > > > > > > > > > > > > > > > in the SIP/2.0 180 Ringing, the SDP shows: > > > > a=sendonly > > > > this is "hold" by rfc 3264. then when the other end picks up, a new SDP is > probably sent with > > > > a=sendrecv > > > > I believe your server is acting correctly. > > > > -----Original Message----- > > From: [email protected] 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:[email protected] SIP/2.0 > > Via: SIP/2.0/UDP 100.X.Y.Z:5060;branch=z9hG4bK667c26ed;rport > > From: "58169" ;tag=as00522e07 > > To: > > Contact: > > Call-ID: [email protected] > > 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: [email protected] > > 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: [email protected] > > 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: [email protected] > > 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 > > > > > > > > > > > > -- Tarek Sawah > > > > Integrated Digital Systems > > > > CCNA, MCSE, RHCE, VoIP > > > > > > USA: +1 347 562 2308 > > > > > > > > > > > > > >> Date: Thu, 29 Apr 2010 16:52:24 +0100 > >> From: [email protected] > >> To: [email protected] > >> 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 > >>> > >>> > >>> > >>> > >>> > >>> _________________________________________________________________ > >>> Hotmail has tools for the New Busy. Search, chat and e-mail from your inbox. > >>> http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_1 > >> > >> > >> -- > >> _____________________________________________________________________ > >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >> New to Asterisk? Join us for a live introductory webinar every Thurs: > >> http://www.asterisk.org/hello > >> > >> asterisk-users mailing list > >> To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _________________________________________________________________ > > The New Busy is not the too busy. Combine all your e-mail accounts with > Hotmail. > > http://www.windowslive.com/campaign/thenewbusy?tile=multiaccount&ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4 > > -- > > _____________________________________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > New to Asterisk? Join us for a live introductory webinar every Thurs: > > http://www.asterisk.org/hello > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > Call 347 562 2308Phone to call with Connect _________________________________________________________________ Hotmail has tools for the New Busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_1 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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