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: 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" <sip:58...@100.x.y.z>;tag=as00522e07 To: <sip:20100324...@195.x.y.z> Contact: <sip:58...@100.x.y.z> 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" <sip:58...@100.x.y.z>;tag=as00522e07 To: <sip:20100324...@195.x.y.z>;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" <sip:58...@100.x.y.z>;tag=as00522e07 To: <sip:20100324...@195.x.y.z>;tag=gK02b3c8db Call-ID: 7f169cce7003bb01365f72ce2a3aa...@100.x.y.z CSeq: 102 INVITE Contact: <sip:20100324...@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:58...@100.x.y.z>;tag=as00522e07 To: <sip:20100324...@195.x.y.z>;tag=gK02b3c8db Call-ID: 7f169cce7003bb01365f72ce2a3aa...@100.x.y.z CSeq: 102 INVITE Contact: <sip:20100324...@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-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 >> >> >> >> >> >> _________________________________________________________________ >> 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
-- _____________________________________________________________________ -- 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