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(166986000:61000:3)) -- Limit Data for this call: timelimit = 166986000 play_warning = 61000 play_to_caller = yes play_to_callee = no warning_freq = 3 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 00102/0 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
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: 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(166986000:61000:3)) -- Limit Data for this call: timelimit = 166986000 play_warning = 61000 play_to_caller = yes play_to_callee = no warning_freq = 3 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 00102/0 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
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: david.wh...@watchguard.com To: asterisk-users@lists.digium.com; asterisk-users@lists.digium.com 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: 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(166986000:61000:3)) -- Limit Data for this call: timelimit = 166986000 play_warning = 61000 play_to_caller = yes play_to_callee = no warning_freq = 3 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 00102/0 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
Re: [asterisk-users] Strange Invite issue
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-boun...@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? 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 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(166986000:61000:3)) -- Limit Data for this call: timelimit = 166986000 play_warning = 61000 play_to_caller = yes play_to_callee = no warning_freq = 3 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
[asterisk-users] Strange Invite issue
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
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
