I still have: Client 1 ---------Asterisk1--Asterisk2 Client 2
When client1 do a call, asterisk1 forward to asterisk2, asterisk2 forward to Asterisk1 At this moment, asterisk1 say : 404Not found But I have insecure=very This is the sip debug at that moment: <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP://192.168.1.151:5060 ---> INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.151:5060;branch=z9hG4bK2b4a242e;rport Max-Forwards: 70 From: "103" <sip:[EMAIL PROTECTED]>;tag=as636875d3 To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.1 Date: Thu, 04 Dec 2008 14:55:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 289 v=0 o=root 1545198644 1545198644 IN IP4 192.168.1.151 s=Asterisk PBX 1.6.0.1 c=IN IP4 192.168.1.151 t=0 0 m=audio 12272 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (14 headers 13 lines) --- == Using SIP RTP CoS mark 5 Sending to 192.168.1.151 : 5060 (NAT) Using INVITE request as basis request - [EMAIL PROTECTED] No user '103' in SIP users list Found peer 'media' for '103' from 192.168.1.151:5060 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.151:12272 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.151:12272 Looking for 33170725012 in media (domain 192.168.1.153) <--- Reliably Transmitting (no NAT) to 192.168.1.151:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.151:5060;branch=z9hG4bK2b4a242e;received=192.168.1.151;rport=5060 From: "103" <sip:[EMAIL PROTECTED]>;tag=as636875d3 To: <sip:[EMAIL PROTECTED]>;tag=as242de969 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 Have you an idea why ? -----Message d'origine----- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de BERGANZ François Envoyé : jeudi 4 décembre 2008 09:15 À : 'Asterisk Users Mailing List - Non-Commercial Discussion' Objet : Re: [asterisk-users] canreinvite=yes problem Now, I have : Client 1 ---------Asterisk1--Asterisk2 Client 2 I need that sip sign go to Asterisk2 But RTP go to Asterisk1 and no more. Where have I to insert canreinvite ? Thank you -----Message d'origine----- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Eric "ManxPower" Wieling Envoyé : mercredi 3 décembre 2008 19:25 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] canreinvite=yes problem canreinvite=yes should work as long as 1) there is no NAT involved anywhere in the call path, 2) All legs of the call are using the same codec, 3) you do not have the t/T/w/W (and maybe a few other) options to the Dial line. Remember the only way you can really tell if a reinvite happens is by looking at the RTP audio. The SIP signaling will not and has never had a "reinvite" feature for signaling. Why did you post the same message at :23, :28, and :35 mins past the hour? If you need immediate support you should contact Digium support and pay for a service contract. BERGANZ François wrote: > I need to test canreinvite=yes with 2softphones and 1 asterisk. > I want to have that : > http://www.panoramisk.com/wp-content/uploads/2007/05/asterisk-call-flow-outb > ridge.png > But I have that http://www.zimagez.com/zimage/canreinvite.php > Canreinvite=yes work for all phones or just asterisk?... -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
