[asterisk-users] Canreinvite=yes // native bridging // 2 sip channels with different Call-ID
I have 2 SIP-clients defined in my sip.conf : [GXP1200] type=friend context=intern host=dynamic username=GXP1200 secret=testpaswoord canreinvite=yes [BT201] type=friend context=intern host=dynamic username=BT201 secret=testpaswoord canreinvite=yes When I make a call from one to another this is displayed on the CLI : -- Executing [...@intern:1] Dial("SIP/GXP1200-093900c8", "SIP/BT201|30") in new stack -- Called BT201 -- SIP/BT201-09395070 is ringing -- SIP/BT201-09395070 answered SIP/GXP1200-093900c8 -- Native bridging SIP/GXP1200-093900c8 and SIP/BT201-09395070 >From voip-info.org I understand that 'canreinvite' means that the SIP-client will re-invite the other client, so that Asterisk is no longer in the path... This is indicated on the CLI with 'native bridging'. Then why are there 2 sip-channels with a different Call-ID ? The output shows that Asterisk is still in between ! asterisk*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 192.168.x.x GXP2020 4684b544470 00103/0 0x4 (ulaw) No Tx: ACK 192.168.x.x BT201 1212e00ffa1 00102/43234 0x4 (ulaw) No Tx: ACK 2 active SIP channels Is there something that I misunderstand here ?? Thanks for the feedback on this ! Greetingz, Jonas. ___ -- 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
Re: [asterisk-users] Canreinvite=yes // native bridging // 2 sip channels with different Call-ID
Asterisk still controls the signalling, but the audio path should be going through the phones directly. Fire up a tcpdump on the Asterisk server to varify this. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens Sent: Saturday, April 18, 2009 5:33 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Canreinvite=yes // native bridging // 2 sip channels with different Call-ID I have 2 SIP-clients defined in my sip.conf : [GXP1200] type=friend context=intern host=dynamic username=GXP1200 secret=testpaswoord canreinvite=yes [BT201] type=friend context=intern host=dynamic username=BT201 secret=testpaswoord canreinvite=yes When I make a call from one to another this is displayed on the CLI : -- Executing [...@intern:1] Dial("SIP/GXP1200-093900c8", "SIP/BT201|30") in new stack -- Called BT201 -- SIP/BT201-09395070 is ringing -- SIP/BT201-09395070 answered SIP/GXP1200-093900c8 -- Native bridging SIP/GXP1200-093900c8 and SIP/BT201-09395070 >From voip-info.org I understand that 'canreinvite' means that the SIP-client will re-invite the other client, so that Asterisk is no longer in the path... This is indicated on the CLI with 'native bridging'. Then why are there 2 sip-channels with a different Call-ID ? The output shows that Asterisk is still in between ! asterisk*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 192.168.x.x GXP2020 4684b544470 00103/0 0x4 (ulaw) No Tx: ACK 192.168.x.x BT201 1212e00ffa1 00102/43234 0x4 (ulaw) No Tx: ACK 2 active SIP channels Is there something that I misunderstand here ?? Thanks for the feedback on this ! Greetingz, Jonas. ___ -- 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
Re: [asterisk-users] Canreinvite=yes // native bridging // 2 sip channels with different Call-ID
14:38:01.229941 IP 192.168.4.240.sip > 192.168.4.248.sip: SIP, length: 889 14:38:01.230127 IP 192.168.4.248.sip > 192.168.4.240.sip: SIP, length: 515 14:38:01.251558 IP 192.168.4.240.sip > 192.168.4.248.sip: SIP, length: 497 14:38:01.271714 IP 192.168.4.240.sip > 192.168.4.248.sip: SIP, length: 1060 14:38:01.271904 IP 192.168.4.248.sip > 192.168.4.240.sip: SIP, length: 433 14:38:01.272133 IP 192.168.4.248.sip > 192.168.4.242.sip: SIP, length: 861 is what I see... only SIP, no RTP/UDP... I guess you're right... Thank you, Tom. On Sat, 2009-04-18 at 06:50 -0400, Tom Moore wrote: > Asterisk still controls the signalling, but the audio path should be > going through the phones directly. > Fire up a tcpdump on the Asterisk server to varify this. > > ___ -- 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