search for: packet2packet

...private addressing, so "canreinvite=no" is set on both the SIP trunk and handset configurations so I can get around the nasty NAT issues. One odd behaviour I am seeing is certain destinations are resulting in different SIP codes being sent back to Asterisk, which then initiates unwanted Packet2Packet bridging. http://www.pastebin.ca/849961 <--- Working call. Line 280 shows a "SIP/2.0 183 Session Progress" and the RTP stream works as intended. I hung the call up before being answered in case you were wondering where the answer part of the debug occurs http://www.pastebin.ca/849965...

I'm just starting to upgrade some boxes from 1.2.x to 1.4.1 as well as trying to get some of the RTP traffic offloaded from the network. I think I'm misunderstanding what the console messages mean when it says "Packet2Packet Bridding SIP/blah to SIP/blah". I though that meant that it had successfully (re)INVITED and the media was no longer going through my Asterisk box, but ethereal says different. I'm not having much luck finding any information on this on the wiki or google. Can someone point me in...

...5166-08e78508", "DYNAMIC_FEATURES=automon") in new stack -- Executing [s at macro-stdexten:3] Dial("SIP/5166-08e78508", "SIP/5143|20|wW") in new stack -- Called 5143 -- SIP/5143-08e744f8 is ringing -- SIP/5143-08e744f8 answered SIP/5166-08e78508 -- Packet2Packet bridging SIP/5166-08e78508 and SIP/5143-08e744f8 -- Packet2Packet bridging SIP/5166-08e78508 and SIP/5143-08e744f8 -- Packet2Packet bridging SIP/5166-08e78508 and SIP/5143-08e744f8 -- Packet2Packet bridging SIP/5166-08e78508 and SIP/5143-08e744f8 -- Packet2Packet bridging SIP/5166-0...

...t;, "SIP/0825387205 at sipoperator") in new stack -- Called 0825387205 at sipoperator -- SIP/sipoperator-28fed000 is making progress passing it to SIP/toto.fr-28fdf000 -- SIP/sipoperator-28fed000 is ringing -- SIP/sipoperator-28fed000 answered SIP/toto.fr-28fdf000 -- Packet2Packet bridging SIP/toto.fr-28fdf000 and SIP/sipoperator-28fed000 (((*****AUDIO IS CUT DURING 2 TO 3 SECONDS*****))) == Spawn extension (incoming_clients, 0825387205, 1) exited non-zero on 'SIP/toto.fr-28fdf000' Native Bridging it's same problem. it's sip module bug ?? When capturin...

Hi everybody Hope I picked the right mailing list. If not, please tell me. We've got a problem with call forwardings. It's exactly the same problem as described in bug 12708, which is resolved by now. Situation: Caller -> asterisk -> call forward to mobile (packet2packet bridge) Quote from original bug reporter: 'One issue that we have noticed repeatedly is that there is a large delay between when a call is answered and when voice traffic actually flows. The delay is also asymmetrical and of the scope of about 2 seconds. This is very noticeable as calling...

Dear concern, I want to configure packet2packet bridging in asterisk. How could I do this any of the tutorial or instructions will help ? I found the setting the canreinvite=yes will do the stuff but it is not working I am using asterisk 12.3 version I am very new to asterisk please help me in doing the same. Thanks in advance. -- Regards...

Hi, I am new to asterisk I want to configure my asterisk server such that it only establishes the call rest the audio must bypass the server and transmitted directly to the peer In my config file I did changes which are below canreinvite=yes nat=force_rtp dirtectmedia=yes directsetup=yes I am using asterisk version 12.3 -- Regards Sameer Rathod 8109413462 -------------- next part

...on (intern, 51, 1) exited non-zero on 'SIP/BT201-088f93e0' -- Executing [52 at intern:1] Dial("SIP/GXP1200-088f93e0", "SIP/BT201| 30") in new stack -- Called BT201 -- SIP/BT201-088faa00 is ringing -- SIP/BT201-088faa00 answered SIP/GXP1200-088f93e0 -- Packet2Packet bridging SIP/GXP1200-088f93e0 and SIP/BT201-088faa00 == Spawn extension (intern, 52, 1) exited non-zero on 'SIP/GXP1200-088f93e0' Why is there this native bridging ? Does this mean that Asterisk is no longer in the middle of it ? Also : there is no audio at all ! Just when I put down th...

...>-------- Original message --------</div><div>From: Sameer Rathod <sameer at hostnsoft.com> </div><div>Date:30/06/2014 9:23 PM (GMT+02:00) </div><div>To: asterisk-users at lists.digium.com </div><div>Subject: [asterisk-users] Fwd: Regarding packet2packet bridging </div><div> </div> Dear concern, I want to configure packet2packet bridging in asterisk. How could I do this any of the tutorial or instructions will help ? I found the setting the canreinvite=yes will do the stuff but it is not working I am using asterisk 12.3 vers...

..."FNAME=callrec_110_1272534191.4_GWTEST_1272534191") in new stack -- Executing [110 at internal:4] Dial("SIP/100-00000004", "SIP/110|20") in new stack -- Called 110 -- SIP/110-00000005 is ringing -- SIP/110-00000005 answered SIP/100-00000004 -- Packet2Packet bridging SIP/100-00000004 and SIP/110-00000005 -- Packet2Packet bridging SIP/100-00000004 and SIP/110-00000005 -- Executing [s at macro-recstart:1] Set("SIP/100-00000004", "FNAME=callrec_110_1272534191.4_GWTEST_1272534203") in new stack -- Executing [s at macro-r...

...When I make or receive a call with either of these extensions, it's possible to start and stop recording by pressing "*1". However, this only works if I press the two keys in quick succession; if I'm not fast enough, all I see is lots of the following console output: -- Packet2Packet bridging SIP/1000-081cffb0 and SIP/2000-08241270 -- Packet2Packet bridging SIP/1000-081cffb0 and SIP/2000-08241270 -- Packet2Packet bridging SIP/1000-081cffb0 and SIP/2000-08241270 -- Packet2Packet bridging SIP/1000-081cffb0 and SIP/2000-08241270 In other words, unless I'm also mon...

...to mp3. I am trying to reduce disk space and want my calls to be recorded in mp3 Instead of wav. Sent from Samsung Mobile -------- Original message -------- From: Sameer Rathod Date:30/06/2014 9:23 PM (GMT+02:00) To: asterisk-users at lists.digium.com Subject: [asterisk-users] Fwd: Regarding packet2packet bridging Dear concern, I want to configure packet2packet bridging in asterisk. How could I do this any of the tutorial or instructions will help ? I found the setting the canreinvite=yes will do the stuff but it is not working I am using asterisk 12.3 version I am very new to asterisk ple...

...ot;SIP/666,30,m") in new stack == Using SIP RTP CoS mark 5 -- Called 666 -- Started music on hold, class 'default', on SIP/555-b7a80948 -- SIP/666-089cb090 is ringing -- SIP/666-089cb090 answered SIP/555-b7a80948 -- Stopped music on hold on SIP/555-b7a80948 -- Packet2Packet bridging SIP/555-b7a80948 and SIP/666-089cb090 -- Executing [h at incoming_vpbx:1] NoOp("SIP/555-b7a80948", "***************") in new stack Thanking you... ---Asterisk User -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium...

...t;SIP/0824537518 at 10.65.138.102|40|L(3600000)") in new stack -- Setting call duration limit to 3600 seconds. -- Called 0824537518 at 10.65.138.102 -- Call on SIP/10.65.138.105-0a67bbd8 left from hold -- SIP/10.65.138.105-0a67bbd8 answered SIP/sipCloverCSC-b7eba8a8 -- Packet2Packet bridging SIP/sipClCSC-b7eba8a8 and SIP/10.65.138.105-0a67bbd8 [Jun 15 09:21:48] WARNING[5306]: channel.c:768 ast_channel_alloc: Channel allocation failed: Can't create alert pipe! [Jun 15 09:21:48] WARNING[5306]: chan_sip.c:3783 sip_new: Unable to allocate AST channel structure for SIP channe...

....138.102|40|L(3600000)") in new stack > -- Setting call duration limit to 3600 seconds. > -- Called 0824537518 at 10.65.138.102 > -- Call on SIP/10.65.138.105-0a67bbd8 left from hold > -- SIP/10.65.138.105-0a67bbd8 answered SIP/sipCloverCSC-b7eba8a8 > -- Packet2Packet bridging SIP/sipClCSC-b7eba8a8 and > SIP/10.65.138.105-0a67bbd8 > [Jun 15 09:21:48] WARNING[5306]: channel.c:768 ast_channel_alloc: Channel > allocation > failed: Can't create alert pipe! > [Jun 15 09:21:48] WARNING[5306]: chan_sip.c:3783 sip_new: Unable to allocate > AST cha...

...30 This is the only line that prints on the console... Typically I get a few lines like: -- Executing [33 at smvoice-sip:1] Dial("SIP/404-18afe560", "SIP/bt610tmm/1044") in new stack -- Called bt610tmm/1044 -- SIP/bt610tmm-b4046c70 answered SIP/404-18afe560 -- Packet2Packet bridging SIP/404-18afe560 and SIP/bt610tmm-b4046c70 == Spawn extension (smvoice-sip, 33, 1) exited non-zero on 'SIP/404-18afe560' Note that both call attempts are from the same phone 404. How can I find out why the first situation above is not showing me dialplan messages like case nu...

...to mp3. I am trying to reduce disk space and want my calls to be recorded in mp3 Instead of wav. Sent from Samsung Mobile -------- Original message -------- From: Sameer Rathod Date:30/06/2014 9:23 PM (GMT+02:00) To: asterisk-users at lists.digium.com Subject: [asterisk-users] Fwd: Regarding packet2packet bridging Dear concern, I want to configure packet2packet bridging in asterisk. How could I do this any of the tutorial or instructions will help ? I found the setting the canreinvite=yes will do the stuff but it is not working I am using asterisk 12.3 version...

...w have a single context (entryocginternal) where I have "include => parkedcalls". The log below shows me calling from one internal extension to another, then picking up, then parking the call. -- SIP/239-0915d5c8 is ringing -- SIP/239-0915d5c8 answered SIP/233-0915bf40 -- Packet2Packet bridging SIP/233-0915bf40 and SIP/239-0915d5c8 -- Started music on hold, class 'default', on SIP/239-0915d5c8 == Spawn extension (macro-dialinternal, s, 7) exited non-zero on 'SIPPeer/SIP/233-0915bf40<ZOMBIE>' in macro 'dialinternal' == Spawn extension (macro-d...

...working fine. But X-Lite softphones can't answer phone calls, and when one of them calls on of the Linksys phones they "connect" but neither party can hear hear the other. I noticed that the Linksys phones are connected via Native bridging while the X-Lite ones are connected via Packet2Packet bridging. Also, on the X-Lite phones there is a about a 30 second lag between when the X-Lite client hits dial/call and when the called party starts ringing. ::Asterisk setup:: Asterisk 1.4.4 Zaptel 1.4.3 (only ztdummy compiled) Asterisk Addons 1.4.1 CentOS 5 VMWare Workstation 6 ::sip.conf::...

...ct: <sip:412563105 at 164.77.171.XXX> Call-ID: 5b7470690d6cd476346bc1113f609a4b at 164.77.171.XXX CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- Call on SIP/nyphone-081a7768 left from hold -- SIP/nyphone-081a7768 answered SIP/2563105-0819cf80 -- Packet2Packet bridging SIP/2563105-0819cf80 and SIP/nyphone-081a7768 vaca*CLI> <--- SIP read from 72.55.143.XXX:5060 ---> SIP/2.0 183 Session Progress CSeq: 103 INVITE Via: SIP/2.0/UDP 164.77.171.XXX:5060;branch=z9hG4bK3cb0e5aa;rport From: "2563105" <sip:412563105 at 164.77.171.XXX>;ta...