[asterisk-users] canreinvite yes or no for PBX
Hey Guys! I have a stupid question about canreinvite. We are using asterisk 1.8.3.2 as a PBX we don't have NAT or firewall thing in between asterisk and phone. so i should use conreinvite=no right ? what is the default value of conreinvite in asterisk 1.8.3.2 ? i meant yes or no ? -S -- _ -- 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
[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
[asterisk-users] Canreinvite after media connection
Howdy, Is it possible to send a reinvite after the media has connected? Scenario: Inbound call hits asterisk ivr then is sent out to an extension using the dial command. We have to carry the rtp streams in this case as asterisk cant send the reinvite after the ivr has stopped playing the message as we already connected the call. Question: Any way around this or is there a better way we can do it? Cheers, Taff ___ -- 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
[asterisk-users] canreinvite per route
Can I activate/deactive the canreinvite SIP flag on the dial plan? The idea is to allow reinvite only for exten - exten calls, and not for outbound calls ___ -- 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 per route
Have canreinvite set for your internal extens. You can also have canreinvite enabled by default for all and use one or more of the 't','T','h','H','w','W' or 'L' options set in your dial commands which will override the canreinvite option and not send re-invites. cheers - Ben --- On Sat, 1/17/09, Gabriel Ortiz Lour ortiz.ad...@gmail.com wrote: From: Gabriel Ortiz Lour ortiz.ad...@gmail.com Subject: [asterisk-users] canreinvite per route To: asterisk-users@lists.digium.com Date: Saturday, January 17, 2009, 10:06 PM Can I activate/deactive the canreinvite SIP flag on the dial plan? The idea is to allow reinvite only for exten - exten calls, and not for outbound calls ___ -- 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
[asterisk-users] canreinvite question
Is it possible to allow reinvites to/from specific devices? For example; exten 2001 and 2002 can reinvite to each other, but not 2003 and 2004 exten 2003 and 2004 can reinvite to each other, but not 2001 and 2002 Can that be done? Devices 2001 2002 are behind one firewall, and 2003 2004 are behind another. Tim ___ -- 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 question
In the sip.conf [2001] ... Canreinvite=yes [2002] ... Canreinvite=no Cordialement, BERGANZ François http://www.acropolistelecom.net Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. -Message d'origine- De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Tim Johnson Envoyé : jeudi 18 décembre 2008 19:49 À : asterisk-users@lists.digium.com Objet : [asterisk-users] canreinvite question Is it possible to allow reinvites to/from specific devices? For example; exten 2001 and 2002 can reinvite to each other, but not 2003 and 2004 exten 2003 and 2004 can reinvite to each other, but not 2001 and 2002 Can that be done? Devices 2001 2002 are behind one firewall, and 2003 2004 are behind another. Tim ___ -- 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
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
Re: [asterisk-users] canreinvite=yes problem
On 3 Dec 2008, at 17:38, BERGANZ François wrote: Someone have a solution for me ? De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] De la part de BERGANZ François Envoyé : mercredi 3 décembre 2008 18:24 À : asterisk-users@lists.digium.com Objet : [asterisk-users] canreinvite=yes problem Hello, 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-outbridge.png But I have that http://www.zimagez.com/zimage/canreinvite.php Canreinvite=yes work for all phones or just asterisk?... Can you help me? Thank you Yes. 1. POST ONCE 2. If no one replies within 20 mins, don't start chasing 3. If its that important pay for support 4. Read documentation ___ -- 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 problem
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
Re: [asterisk-users] canreinvite=yes problem
Reinvites will happen by default. Post your sip.conf [general] and the peers in sip.conf masking only the passwords. Also paste the part of extensions.conf that you use to Dial. BERGANZ François wrote: 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
[asterisk-users] canreinvite=yes problem
Hello, 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?... Can you help me? Thank you ___ -- 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
[asterisk-users] canreinvite=yes --problems
Hello, 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?... Can you help me? Thank you ___ -- 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 problem
Someone have a solution for me ? De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de BERGANZ François Envoyé : mercredi 3 décembre 2008 18:24 À : asterisk-users@lists.digium.com Objet : [asterisk-users] canreinvite=yes problem Hello, 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?... Can you help me? Thank you ___ -- 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 problem
Hello, canreinvite, don't work with all softphone or hardphone. Regards On Wed, Dec 3, 2008 at 12:38 PM, BERGANZ François [EMAIL PROTECTED] wrote: Someone have a solution for me ? *De :* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *De la part de* BERGANZ François *Envoyé :* mercredi 3 décembre 2008 18:24 *À :* asterisk-users@lists.digium.com *Objet :* [asterisk-users] canreinvite=yes problem Hello, 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-outbridge.png But I have that http://www.zimagez.com/zimage/canreinvite.php Canreinvite=yes work for all phones or just asterisk?... Can you help me? Thank you ___ -- 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
Re: [asterisk-users] canreinvite=yes problem
On Wed, Dec 03, 2008 at 06:23:32PM +0100, BERGANZ François wrote: Hello, 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-outbridge.png But I have that http://www.zimagez.com/zimage/canreinvite.php Canreinvite=yes work for all phones or just asterisk?... I believe canreinvite=yes is the default option unless you set it to canreinvite=no I would leave it set to yes unless there is some reason to change it, for example the phone is behind NAT, or transfers etc don't work correctly without it being set to no. If it's still not doing the right thing, then it's worth also checking the nat= option There are also other settings which can cause asterisk to stay in the media path, as BOTH sip devices need canreinvite=yes, otherwise it will stay in the media path. Specifying certain options on the Dial() cmd may also cause it to stay in the media path. Rob ___ -- 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 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
[asterisk-users] canreinvite question
Hi All, if i do this setup: |---[ext 100] |--[router/nat gw]--| | |---[ext 101] | [asterisk]--[internet]---| | | |---[ext 200] |--[router/nat gw]--| |---[ext 201] If i set, canreinvite=yes on all ext, assuming all ip phones have the same codec, if 100 calls 101, or vice versa will rtp still go thru asterisk? and same scenario for 200 to 202 or vice versa. what if 100 call 200 or 201? or 200 calls 100 or 100? will rtp still go thru asterisk? thank you regards, Ron ___ -- 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 question
On Fri, 22 Feb 2008 18:50:16 +0800, Ron [EMAIL PROTECTED] wrote: If i set, canreinvite=yes on all ext, assuming all ip phones have the same codec, if 100 calls 101, or vice versa will rtp still go thru asterisk? and same scenario for 200 to 202 or vice versa. ... and I'd like to add to this question: If the phones have the option Enable NAT, I expected them to be able to talk to each other directly, but they didn't, and I had to set them to canreinvite=no in sip.conf. Why is that? ___ -- 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
[asterisk-users] canreinvite option - gona have problems?
Hi list, can anyone tell me how problematic it is setting canreinvite=yes ? I know if its to avoid issues with bad implementatins of SIP on other devices then maybe you cant give a black and white answer, but any constructive comments welcome! Reason being I think I have to set this to yes to enable mediaproxy RTP proxy on my OpenSER box to interoperate correctly with Asterisk, Thanks Andy.___ -- 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
Dear C F; So in that case, if I placed canrenvite=yes for both endpoint, it is not condition that traffic will be directly via the endpoint while signaling via Asterisk as still Asterisk should detect whethor it is necessary to stay in the path or not? Please advise. How can I know that the traffic went directly between the endpoints and did not go via the asterisk? Regards Bilal Ghayad Mobile: 009659849460 - By default assuming you have no global setting otherwise, if asterisk doesnt see a need to stay in the path then it wont. hence if it has to transcode between different codecs, capture DTMF or different protocols it will stay in the path. On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; If I need traffic to be directly between the endpoints, then I have to set the canreinvite = yes? If I did not configure the canrenvite at all, then by default it will pass the traffic via Asterisk and not directly between the endpoints? What if one endpoint was SIP and configured with canreinvite=yes while other endpoint was IAX2 and configured with canreinvite=yes, then they can send traffic to each other directly or it will be via Asterisk? Regards Bilal Check out the hottest 2008 models today at Yahoo! Autos. http://autos.yahoo.com/new_cars.html ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
Don't know about IAX. As for SIP, You will know what ip address and port the audios should be transmitted to by looking at the sdp session. Just goto the * console and enable sip debug. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad Sent: Tuesday, September 11, 2007 10:14 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] canreinvite Dear C F; So in that case, if I placed canrenvite=yes for both endpoint, it is not condition that traffic will be directly via the endpoint while signaling via Asterisk as still Asterisk should detect whethor it is necessary to stay in the path or not? Please advise. How can I know that the traffic went directly between the endpoints and did not go via the asterisk? Regards Bilal Ghayad Mobile: 009659849460 - By default assuming you have no global setting otherwise, if asterisk doesnt see a need to stay in the path then it wont. hence if it has to transcode between different codecs, capture DTMF or different protocols it will stay in the path. On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; If I need traffic to be directly between the endpoints, then I have to set the canreinvite = yes? If I did not configure the canrenvite at all, then by default it will pass the traffic via Asterisk and not directly between the endpoints? What if one endpoint was SIP and configured with canreinvite=yes while other endpoint was IAX2 and configured with canreinvite=yes, then they can send traffic to each other directly or it will be via Asterisk? Regards Bilal Check out the hottest 2008 models today at Yahoo! Autos. http://autos.yahoo.com/new_cars.html ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
How can I know that the traffic went directly between the endpoints and did not go via the asterisk? I'm sure there are many ways to do this one way would be to do rtp debug on the cli and watch for media packets another would be to do tcpdump on the command line and watch for packets there. Regards Bilal Ghayad Mobile: 009659849460 - By default assuming you have no global setting otherwise, if asterisk doesnt see a need to stay in the path then it wont. hence if it has to transcode between different codecs, capture DTMF or different protocols it will stay in the path. On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; If I need traffic to be directly between the endpoints, then I have to set the canreinvite = yes? If I did not configure the canrenvite at all, then by default it will pass the traffic via Asterisk and not directly between the endpoints? What if one endpoint was SIP and configured with canreinvite=yes while other endpoint was IAX2 and configured with canreinvite=yes, then they can send traffic to each other directly or it will be via Asterisk? Regards Bilal Check out the hottest 2008 models today at Yahoo! Autos. http://autos.yahoo.com/new_cars.html ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
The others answered correctly personal I like using rtp debug. As for making sure in the DialPlan that the RTP goes end to end without asterisk. 1. Make sure they both use the same codec and protocol. 2. Don't put any options in app_dial, like tTwW or anything else that will force asterisk to stay in the stream to listen for DTMF. On 9/11/07, bilal ghayyad [EMAIL PROTECTED] wrote: Dear C F; So in that case, if I placed canrenvite=yes for both endpoint, it is not condition that traffic will be directly via the endpoint while signaling via Asterisk as still Asterisk should detect whethor it is necessary to stay in the path or not? Please advise. How can I know that the traffic went directly between the endpoints and did not go via the asterisk? Regards Bilal Ghayad Mobile: 009659849460 - By default assuming you have no global setting otherwise, if asterisk doesnt see a need to stay in the path then it wont. hence if it has to transcode between different codecs, capture DTMF or different protocols it will stay in the path. On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; If I need traffic to be directly between the endpoints, then I have to set the canreinvite = yes? If I did not configure the canrenvite at all, then by default it will pass the traffic via Asterisk and not directly between the endpoints? What if one endpoint was SIP and configured with canreinvite=yes while other endpoint was IAX2 and configured with canreinvite=yes, then they can send traffic to each other directly or it will be via Asterisk? Regards Bilal Check out the hottest 2008 models today at Yahoo! Autos. http://autos.yahoo.com/new_cars.html ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
[asterisk-users] canreinvite
Hi List; If I need traffic to be directly between the endpoints, then I have to set the canreinvite = yes? If I did not configure the canrenvite at all, then by default it will pass the traffic via Asterisk and not directly between the endpoints? What if one endpoint was SIP and configured with canreinvite=yes while other endpoint was IAX2 and configured with canreinvite=yes, then they can send traffic to each other directly or it will be via Asterisk? Regards Bilal Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search http://search.yahoo.com/search?fr=oni_on_mailp=graduation+giftscs=bz ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
By default assuming you have no global setting otherwise, if asterisk doesnt see a need to stay in the path then it wont. hence if it has to transcode between different codecs, capture DTMF or different protocols it will stay in the path. On 9/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; If I need traffic to be directly between the endpoints, then I have to set the canreinvite = yes? If I did not configure the canrenvite at all, then by default it will pass the traffic via Asterisk and not directly between the endpoints? What if one endpoint was SIP and configured with canreinvite=yes while other endpoint was IAX2 and configured with canreinvite=yes, then they can send traffic to each other directly or it will be via Asterisk? Regards Bilal Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search http://search.yahoo.com/search?fr=oni_on_mailp=graduation+giftscs=bz ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
[asterisk-users] canreinvite problems
Hi, I've been working on migrating my asterisk from zap to sip (due to compatibility issues between my TDM400P and my Hauppauge PVR500). I've purchased a Linksys SPA-3102 and a Siemens Gigaset SL75 WLAN (wireless SIP phone). I managed to get it all working with my asterisk 1.4.0 installation, but I'm seeing some interesting things with the canreinvite option that I can't explain, even after reading: http://www.voip-info.org/wiki-Asterisk+sip+canreinvite My setup: - asterisk server with: - eth0 = 192.168.254.254 (internal network) - eth1 = Internet IP-address - ZAP/1 (FXO) not used - ZAP/2 (FXS) not used - ZAP/3 and ZAP/4 (FXS) with DECT phones - SPA-3102: - WAN interface configured with DHCP, it gets 192.168.254.104(internal network) - LAN interface is not being used - Line1: DECT phone - PSTN: is connected to the PSTN - Siemens SL75 WLAN: 192.168.254.105 - Laptop (192.168.254.125) with an Eyebeam and idefisk softphone All the SIP endpoints are connected to the internal network, there should be no NAT issues. In all situations I'm able to dial the other phone and make it ring. From the ZAP endpoints to the SIP endpoints (and vice versa) I get sound. Same applies to the IAX2 client (idefisk). When I have 2 SIP endpoints that both aren't configured with canreinvite=no then I get no sound. Conclusion: all media needs to go through the asterisk server in order to get sound. Questions: 1. Are all of my SIP endpoints incompatible with the canreinvite=yes option? 2. Is there a list of SIP endpoints that are known to work with canreinvite=yes? 3. Are other people also experiencing this? with kind regards, Stefan van der Eijk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] canreinvite problems
Stefan, When I have 2 SIP endpoints that both aren't configured with canreinvite=no then I get no sound. The Sipura 3102 definitely works fine with canreinvite=yes and I never really had a problem with any of the Sipura devices in this respect, especially when there is no NAT involved. However, the default Auto NetService Private IP Ranges: includes 192.168.0.0-192.168.255.255, so your 192.168.254.0/24 network would be considered a LAN address by the 3102 and hence the traffic would go out the LAN interface (not WAN). Change this setting by removing this range. It's on the Admin Advanced LAN Setup tab. If that doesn't help, then you need to check what traffic is being sent. Since all devices are on the same internal network I assume they can see each other. You need to look at the Invite (and ReInvite) messages sent and received and see if the IP addresses for RTP listed there make sense. Then I suggest you use tcpdump to see what traffic is sent by each device, and where. If you have a switched network environment this will be a bit tricky as your * box won't see this traffic, so you may want to use a hub for this test (just temporarily) or if available set up port mirroring to sniff the traffic. Good luck and keep us posted. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] canreinvite problems
On 2/10/07, Luki [EMAIL PROTECTED] wrote: Stefan, When I have 2 SIP endpoints that both aren't configured with canreinvite=no then I get no sound. The Sipura 3102 definitely works fine with canreinvite=yes and I never really had a problem with any of the Sipura devices in this respect, especially when there is no NAT involved. However, the default Auto NetService Private IP Ranges: includes 192.168.0.0-192.168.255.255, so your 192.168.254.0/24 network would be considered a LAN address by the 3102 and hence the traffic would go out the LAN interface (not WAN). Change this setting by removing this range. It's on the Admin Advanced LAN Setup tab. If that doesn't help, then you need to check what traffic is being sent. Since all devices are on the same internal network I assume they can see each other. You need to look at the Invite (and ReInvite) messages sent and received and see if the IP addresses for RTP listed there make sense. Then I suggest you use tcpdump to see what traffic is sent by each device, and where. If you have a switched network environment this will be a bit tricky as your * box won't see this traffic, so you may want to use a hub for this test (just temporarily) or if available set up port mirroring to sniff the traffic. Good luck and keep us posted. Luki, I just configured the wlan phone and my eyebeam endpoints with canreinvite=yes (which should put the sipura out of the picture). Calling the wlan phone from eyebeam: no sound gets through. Putting a canreinvite=no in either one of the configurations (for the wlan01 or the eyebeam) forces the media through the asterisk and sound gets through. I'll get wireshark running on my laptop so I can post the SIP conversations here. regards, Stefan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] canreinvite=yes and RTP dropping in and out
Hey guys,I'm having yet another strange problem. I've recently set canreinvite=yes, allowing the RTP streams to avoid our * server. Now, a few people are experience one way audio drops on internal calls. External calls are working fine (they re-invite directly to a Cisco router). Sometimes, if you wait 20 seconds or more, the stream will resume. Flipping the person on and off hold won't resume the stream. We're using 7960 phones. Enabled_vad is set to 0 (disabled). It doesn't seem to happen all of the time. There are no sip messages being exchanged when the stream stops or restarts.Any suggestions?Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] canreinvite=yes and RTP dropping in and out
- Original Message - From: Gary Richardson [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Wed, 02 Aug 2006 13:54:04 -0300 Subject: [asterisk-users] canreinvite=yes and RTP dropping in and out Hey guys, I'm having yet another strange problem. I've recently set canreinvite=yes, allowing the RTP streams to avoid our * server. Now, a few people are experience one way audio drops on internal calls. External calls are working fine (they re-invite directly to a Cisco router). Sometimes, if you wait 20 seconds or more, the stream will resume. Flipping the person on and off hold won't resume the stream. We're using 7960 phones. Enabled_vad is set to 0 (disabled). It doesn't seem to happen all of the time. There are no sip messages being exchanged when the stream stops or restarts. Any suggestions? If the audio is going directly there's not too much you can do to examine it. There may be software out there to sniff the data on your network and examine the RTP stream, maybe even see when it drops out (if it really does drop out, ie: stream actually stops). I know there's some Windows software out there capable of this as I picked a copy up while at Spring VON but you might need to look around. OH - can you also send a sip debug with the reinvites? I'm just curious to see the RTP information in the SDP. Thanks. Joshua Colp Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] canreinvite=yes and RTP dropping in and out
My next attempt at this is going to be putting a hub in between the path to the switch. I'm hoping to be able to sniff the packets to see what's going on.Also, using the network status page on the hard phones, the transmit and receive counters for the direction of the channel slows way down as if almost no data is being transmitted. How do I send a sip debug?Thanks.On 8/2/06, Joshua Colp [EMAIL PROTECTED] wrote: - Original Message -From: Gary Richardson[mailto:[EMAIL PROTECTED]]To: Asterisk Users Mailing List -Non-Commercial Discussion [mailto: asterisk-users@lists.digium.com]Sent:Wed, 02 Aug 2006 13:54:04 -0300Subject: [asterisk-users] canreinvite=yesand RTP dropping in and out Hey guys, I'm having yet another strange problem. I've recently set canreinvite=yes, allowing the RTP streams to avoid our * server. Now, a few people are experience one way audio drops on internal calls. External calls are working fine (they re-invite directly to a Cisco router). Sometimes, if you wait 20 seconds or more, the stream will resume. Flipping the person on and off hold won't resume the stream. We're using 7960 phones. Enabled_vad is set to 0 (disabled). It doesn't seem to happen all of the time. There are no sip messages being exchanged when the stream stops or restarts. Any suggestions?If the audio is going directly there's not too much you can do to examine it. There may be software out there to sniff the data on your network and examine the RTP stream, maybe even see when it drops out (if it really does drop out, ie: stream actually stops). I know there's some Windows software out there capable of this as I picked a copy up while at Spring VON but you might need to look around. OH - can you also send a sip debug with the reinvites? I'm just curious to see the RTP information in the SDP. Thanks.Joshua ColpDigium___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] canreinvite=yes and RTP dropping in and out
- Original Message - From: Gary Richardson [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Wed, 02 Aug 2006 14:34:31 -0300 Subject: Re: [asterisk-users] canreinvite=yes and RTP dropping in and out My next attempt at this is going to be putting a hub in between the path to the switch. I'm hoping to be able to sniff the packets to see what's going on. Also, using the network status page on the hard phones, the transmit and receive counters for the direction of the channel slows way down as if almost no data is being transmitted. How do I send a sip debug? Actually since this happens randomly I doubt that will help. Is there any other traffic on the network too? Never know... or a faulty switch? Grasping at random things but nothing really comes to mind. Thanks. Joshua Colp Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Canreinvite and remotely registered devices
We have a customer who would like to do RTP directly between SIP devices. The devices are not registered directly to Asterisk, but to SER on another machine. It seems in this case canreinvite = yes is never used. Does anyone know of a way of persuading Asterisk to issue re-invites in this case? -- Alistair Cunningham, Integrics Ltd, +44 20 799 39 799 http://integrics.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Canreinvite and remotely registered devices
On Mon, 2006-07-31 at 11:10 +0100, Alistair Cunningham wrote: We have a customer who would like to do RTP directly between SIP devices. The devices are not registered directly to Asterisk, but to SER on another machine. It seems in this case canreinvite = yes is never used. Does anyone know of a way of persuading Asterisk to issue re-invites in this case? Although not clear from your posting I assume that the call between the two phones is setup through the Asterisk server. Asterisk will not let go if you have ie the T or t option in your Dial statement. Remove those for starters. If Asterisk is not involved at all I guess you need to find out what the equivalent of canreinvite=yes is in SER country. Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Canreinvite and remotely registered devices
Patrick wrote: It seems in this case canreinvite = yes is never used. Does anyone know of a way of persuading Asterisk to issue re-invites in this case? Although not clear from your posting I assume that the call between the two phones is setup through the Asterisk server. Asterisk will not let go if you have ie the T or t option in your Dial statement. Remove those for starters. If Asterisk is not involved at all I guess you need to find out what the equivalent of canreinvite=yes is in SER country. Patrick, Yes, Asterisk is handling the call setup for billing purposes. There is no t or T in the dial. SER does not have an equivalent of canreinvite as it is a SIP proxy not an end point. Alistair Cunningham, Integrics Ltd, +44 20 799 39 799 http://integrics.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Canreinvite and remotely registered devices
- Original Message - From: Alistair Cunningham [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Mon, 31 Jul 2006 07:10:43 -0300 Subject: [asterisk-users] Canreinvite and remotely registered devices We have a customer who would like to do RTP directly between SIP devices. The devices are not registered directly to Asterisk, but to SER on another machine. It seems in this case canreinvite = yes is never used. Does anyone know of a way of persuading Asterisk to issue re-invites in this case? What do you mean by not used? Even if going through SER it should still be used. -- Alistair Cunningham, Integrics Ltd, +44 20 799 39 799 http://integrics.com/ Joshua Colp Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Canreinvite and remotely registered devices
Joshua Colp wrote: We have a customer who would like to do RTP directly between SIP devices. The devices are not registered directly to Asterisk, but to SER on another machine. It seems in this case canreinvite = yes is never used. Does anyone know of a way of persuading Asterisk to issue re-invites in this case? What do you mean by not used? Even if going through SER it should still be used. Joshua, That's what I would have expected, but Asterisk is not issuing re-invites. One thing I should have mentioned is that Asterisk, SER, and the phones are all on an RFC1918 network (but there is no NAT between them). Maybe Asterisk is seeing that: 1. The phones are offering an SDP address that is in RFC1918. 2. The SDP and SIP headers are not the address the SIP is coming from, as SER is relaying them. and therefore incorrectly concluding that there's NAT between it and the phones? Alistair Cunningham, Integrics Ltd, +44 20 799 39 799 http://integrics.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Canreinvite
How can I check if SIP re-invite is really working ? I'm trying it with two grandstream gxp2000. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Canreinvite
- Original Message - From: Giordano Grandis [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Fri, 28 Jul 2006 07:01:08 -0300 Subject: [asterisk-users] Canreinvite How can I check if SIP re-invite is really working ? If you do a sip debug you should see two INVITEs to each side after the call is established with the IP address of the GXP2000 in the SDP. You can also run rtp debug to see if the RTP audio stream is running through Asterisk. I'm trying it with two grandstream gxp2000. Thanks Joshua Colp Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [asterisk-users] Canreinvite
Ok, thanks, also if i do not have rtp debug (i'm using asterisk 1.0.9) Hi -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Joshua Colp Inviato: venerdì 28 luglio 2006 12.54 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] Canreinvite - Original Message - From: Giordano Grandis [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Fri, 28 Jul 2006 07:01:08 -0300 Subject: [asterisk-users] Canreinvite How can I check if SIP re-invite is really working ? If you do a sip debug you should see two INVITEs to each side after the call is established with the IP address of the GXP2000 in the SDP. You can also run rtp debug to see if the RTP audio stream is running through Asterisk. I'm trying it with two grandstream gxp2000. Thanks Joshua Colp Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Canreinvite
I will try your suggestion and I will let you know. Thank you On 6/18/06, Philippe Lindheimer [EMAIL PROTECTED] wrote:How have you confirmed that they did not reinvite? The channels are still controlled by Asterisk (sip signalling), it is the rtp streams that go direct. You can do a sip show channel 146b518a4cd on the specific channel to see where the rtp streams are going. Or ... if this is the only active channel on the box, just do a rtp debug. If the rtp stream is going through asterisk, it will be very obvious. If not, you won't see a constant flow of rtp debug messages going on. pFrom: Il Neofita [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comDate: Sun, 18 Jun 2006 05:01:20 -0400Subject: Re: [Asterisk-Users] Canreinvite This is the dial in extensionsexten = _40001,1,Dial(SIP/40001,30) exten = _40002,1,Dial(SIP/40002,30) From: Il Neofita [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sun, 18 Jun 2006 05:22:35 -0400Subject: Re: [Asterisk-Users] Canreinvite cosa vedo a console -- Executing Dial(SIP/40001-3760, SIP/40002|30) in new stack -- Called 40002 -- SIP/40002-4753 is ringing -- SIP/40002-4753 answered SIP/40001-3760 -- Attempting native bridge of SIP/40001-3760 and SIP/40002-4753srvlinux*CLI sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message82.X2.XX3.X3 40002 146b518a4cd 00103/0 alaw No Tx: ACK 82.X2.XX3.X3 40001 CBD1DB85-8B 00102/30987 alaw No Tx: ACK2 active SIP channels Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail Beta. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Canreinvite
This is the dial in extensionsexten = _40001,1,Dial(SIP/40001,30) exten = _40002,1,Dial(SIP/40002,30) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Canreinvite
cosa vedo a console -- Executing Dial(SIP/40001-3760, SIP/40002|30) in new stack -- Called 40002 -- SIP/40002-4753 is ringing -- SIP/40002-4753 answered SIP/40001-3760 -- Attempting native bridge of SIP/40001-3760 and SIP/40002-4753srvlinux*CLI sip show channelsPeer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message82.X2.XX3.X3 40002 146b518a4cd 00103/0 alaw No Tx: ACK 82.X2.XX3.X3 40001 CBD1DB85-8B 00102/30987 alaw No Tx: ACK2 active SIP channels ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Canreinvite
How have you confirmed that they did not reinvite? The channels are still controlled by Asterisk (sip signalling), it is the rtp streams that go direct. You can do a sip show channel 146b518a4cd on the specific channel to see where the rtp streams are going. Or ... if this is the only active channel on the box, just do a rtp debug. If the rtp stream is going through asterisk, it will be very obvious. If not, you won't see a constant flow of rtp debug messages going on.pFrom: "Il Neofita" [EMAIL PROTECTED]To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.comDate: Sun, 18 Jun 2006 05:01:20 -0400Subject: Re: [Asterisk-Users] Canreinvite This is the dial in extensionsexten = _40001,1,Dial(SIP/40001,30) exten =
_40002,1,Dial(SIP/40002,30) From: "Il Neofita" [EMAIL PROTECTED]To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.comDate: Sun, 18 Jun 2006 05:22:35 -0400Subject: Re: [Asterisk-Users] Canreinvite cosa vedo a console -- Executing Dial("SIP/40001-3760", "SIP/40002|30") in new stack -- Called 40002 -- SIP/40002-4753 is ringing -- SIP/40002-4753 answered SIP/40001-3760 -- Attempting native bridge of SIP/40001-3760 and SIP/40002-4753srvlinux*CLI sip show channelsPeer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message82.X2.XX3.X3
40002 146b518a4cd 00103/0 alaw No Tx: ACK 82.X2.XX3.X3 40001 CBD1DB85-8B 00102/30987 alaw No Tx: ACK2 active SIP channels
Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail Beta.___
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[Asterisk-Users] Canreinvite
I put canreinvite=yes in my sip, for a sipura 3000 and a xlite, however, if I call the traffic still go throw the asterisk. How come? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Canreinvite
Il Neofita wrote: I put canreinvite=yes in my sip, for a sipura 3000 and a xlite, however, if I call the traffic still go throw the asterisk. How come? Are you using the same codecs on the SPA3000 and the xlite? If no then there's your reason. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://www.linuxha.com/ Main site http://linuxha.blogspot.com/My HA Blog http://home.comcast.net/~ncherry/ Backup site ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Canreinvite
What does your dial command look like? On 6/17/06, Il Neofita [EMAIL PROTECTED] wrote: I put canreinvite=yes in my sip, for a sipura 3000 and a xlite, however, if I call the traffic still go throw the asterisk. How come? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] canreinvite=no and codecs.
Hi, folks. If i use canreivite=no option in my sip.conf for users is this mean that i need to load 729 and 723 codecs for thos UA that want to transmit it? Or this is just traffic redirection feature? How this option reflect on server load? -- = Best regards, Nikolay Pavlov. --- = ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] canreinvite, bandwidth, dial option
I just read: Certain options to the Dial() statement require that Asterisk is in the media path, and consequently Asterisk will not let go of it: /t/, ''T, h, H, w, W or L (with multiple arguments). Probably there are more. I had in my memory that r, R, m would also prevent a reinvite. Can anybody say something on that? Below is a list of all options. o *t*: Allow the /called/ user to transfer the call by hitting # o *T*: Allow the /calling/ user to transfer the call by hitting # o *r*: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. Really, you almost certainly do not want to use this. Asterisk will generate ring tones automatically where it is appropriate to do so. r makes it go the next step and additionally generate ring tones where it is probably not appropriate to do so. o *R*: Indicate ringing to the calling party when the called party indicates ringing, pass no audio until answered. This is available only if you are using kapejod's bristuff http://www.voip-info.org/wiki/index.php?page=Asterisk+zaphfc. o *m*: Provide Music on Hold to the calling party until the called channel answers. This is mutually exclusive with option 'r', obviously. Use m(class) to specify a class for the music on hold. o *n*: (Asterisk 1.1 and later) July 2005 bug 752 http://bugs.digium.com/view.php?id=752 was included in CVS (Asterisk 1.1) and enhances the privacy manager considerably. As part of this patch, the 'n' flag to Dial got changed to be used as part of the privacy features, instead of being the 'dont jump to +101' flag. That flag is now 'j'. o *o*: Restore the Asterisk v1.0 CallerId behaviour (send the original caller's ID) in Asterisk v1.2 (default: send this extension's number) o *j*: Asterisk 1.2 and later: Jump to priority n+101 if all of the requested channels were busy (just like behaviour in Asterisk 1.0.x) o *M(*/x/*)*: Executes the macro (x) upon connect of the call (i.e. when the called party answers) o *h*: Allow the callee to hang up by dialing *** o *H*: Allow the caller to hang up by dialing *** o *C*: Reset the CDR (Call Detail Record) for this call. This is like using the NoCDR http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+NoCDR command o *P(*/x/*)*: Use the PrivacyManager http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+PrivacyManager, using /x/ as the database (/x/ is optional) o *g*: When the called party hangs up, exit to execute more commands in the current context. o *G(context^exten^pri)*: If the call is answered, transfer both parties to the specified context and extension. The calling party is transferred to priority x, and the called party to priority x+1. This allows the dialplan to distinguish between the calling and called legs of the call (new in v1.2). o *A(*/x/*)*: Play an announcement (/x/.gsm) to the called party. o *S(*/n/*)*: Hangup the call /n/ seconds AFTER called party picks up. o *d*: This flag trumps the 'H' flag and intercepts any dtmf while waiting for the call to be answered and returns that value on the spot. This allows you to dial a 1-digit exit extension while waiting for the call to be answered - see also RetryDial http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+RetryDial o *D(*/digits/*)*: After the called party answers, send /digits/ as a DTMF stream, then connect the call to the originating channel. o *L(*x[:y][:z]*)*: Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional. The following special variables are optional for limit calls: (pasted from app_dial.c) + *LIMIT_PLAYAUDIO_CALLER* - yes|no (default yes) - Play sounds to the caller. + *LIMIT_PLAYAUDIO_CALLEE* - yes|no - Play sounds to the callee. + *LIMIT_TIMEOUT_FILE* - File to play when time is up. + *LIMIT_CONNECT_FILE* - File to play when call begins. + *LIMIT_WARNING_FILE* - File to play as warning if 'y' is defined. If *LIMIT_WARNING_FILE* is not defined, then the default
[Asterisk-Users] canreinvite=yes
Hi all iam working with * just started can some one explain me canreinvite=yes when should i use the above options I would like to use my * server for authentication and directly talk SIP user to SIP user with out consuming my * bandwidth, is that correct Does any one know, which provider support this option ram ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] canreinvite=yes
canreinvite = yes tells the phones to try and talk to each other and leave Asterisk out of the mix. The important word here is TRY. There are lots of reasons that it might not quite work, and there was a big discussion on the list about it a little while ago. PaulH On Thu, 2006-03-02 at 01:55 +0530, ram wrote: Hi all iam working with * just started can some one explain me canreinvite=yes when should i use the above options I would like to use my * server for authentication and directly talk SIP user to SIP user with out consuming my * bandwidth, is that correct Does any one know, which provider support this option ram ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] canreinvite=yes
Hi thanks, would mind pointing to me that let me check and see is that discussion will help me ram On 3/2/06, Paul Hales [EMAIL PROTECTED] wrote: canreinvite = yes tells the phones to try and talk to each other andleave Asterisk out of the mix. The important word here is TRY.There are lots of reasons that it might not quite work, and there was abig discussion on the list about it a little while ago.PaulHOn Thu, 2006-03-02 at 01:55 +0530, ram wrote: Hi all iam working with * just started can some one explain me canreinvite=yes when should i use the above options I would like to use my * server for authentication and directly talk SIP user to SIP user with out consuming my * bandwidth, is that correct Does any one know, which provider support this option ram ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] canreinvite always =no * no matter what we try :-(
Hello and thanks for replying! Steve, The mission is to actually get a reinvite to work on the lan. There isn't anything special to get this working... normally. I trust you verified the traffic flow with a network monitor tool (tcpdump?), Actully ethereal, It is encouraging to hear that it does not take anything special. I've tried what seems to be a simple arrangement, no nat two phones on the lan same codec, lack of canreinvite line and also tried canreinvite=yes I am not using a global nat=yes statement. also tried nat=no on each phone just in case of a default option. correct? Does SIP debug give you any info (i.e., does it match the right peer) -- you don't show if you allow reinvites globally? What about the nat= setting? I've not set nat= or canreinvite= globally just on each phone I can certainly try that but having specific settings on the phones seems to almost guarantee I know where I stand with those two :-) I've not torn apart the sip debug on this yet as I am quite new to SIP but will do so if need be. Was just trying the simple approach first. Couple pointers I can give you to get you excited: 1) Reinvites work quite reliably, I use them between the PTSN gateway and the end user's ATA, all the way across the Internet -- nicely reduces latency. 2) If you use RFC2833 for DTMF you can issue an reinvite and still use t/T for transfer. NOTE that you have to modify the source to make asterisk reinvite even when it needs to listen to DTMFs. I give no guarantees how well it will work for you but it does work. See AST_BRIDGE_DTMF_CHANNEL_0 | AST_BRIDGE_DTMF_CHANNEL_1 in rtp.c. 3) Reinvites *can* work even if both ends are behind NAT. It really depends on the NATing router and the ATA. Sipura's and good NAT routers work, but I would not call it reliable -- it's really pushing it a bit... Yep I will eventually go there but right now still just trying to get it to work for a test on the lan and have not seen it fly yet. asterisk always creates a 'native bridge' and seems to hold on for dear life so far as I have seen :-) So if you really want to see why your Reinvites do not work, then you probably will have to make your hands dirty and analyze where ast_rtp_bridge() in rtp.c bails out. But since you are on a LAN it makes the situation a lot easier. Yep! Still new at this but enjoy getting hands dirty. Thanks for your time! Steve To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] canreinvite always =no * no matter what we try :-(
Actully ethereal OK... Try canreinvite=yes in the [general] section; this makes it the default setting for all peers unless specified otherwise. Do the same for nat=no in [general] to rule out all NAT'ing related issues. You don't have tT in your Dial() statement, that's good. You say you verified that no transcoding is needed (i.e. both ends use the same codec). Well, then it should work! Once you get it to work, you can individualize the accounts and no longer use a global setting. But that's down the line. asterisk always creates a 'native bridge' and seems to hold on for dear life so far as I have seen :-) It says Attempting Native Bridge but it doesn't tell if you if it succeeded or not; there was once a notice saying the the bridge could not be established (failed?) but it caused even more confusion. You could add some statements to the ast_rtp_bridge() code in rtp.c and give yourself some feedback -- succeeded, failed because X / Y / Z. Hope that helps... --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] canreinvite always =no * no matter what we try :-(
been testing with a rather simple setup. The mission is to actually get a reinvite to work on the lan. I am trying with two sipura phones G.711 codec forced on both both on the lan no nat no fancy options suchs as tT or H No matter what we do asterisk hangs on to the media path, how in the world do I get a reinvite to work where the media path is actually handled by the two phones on the lan? Any pointers greatly appreciated! Steve Pretty simple extensions, on lan no nat sip.conf [4785] type=friend username=4785 secret=test host=dynamic canreinvite=yes [4786] type=friend username=4786 secret=tesst host=dynamic canreinvite=yes extensions.conf exten = 4785,1,Dial(SIP/4785,66) exten = 4785,3,hangup exten = 4786,1,Dial(SIP/4786,66) exten = 4786,3,hangup ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] canreinvite always =no * no matter what we try :-(
please turn on all the debug, warning, error etc messages in the console, see logger.conf, then type sip peer peer1 debug and sip peer peer2 debug to see the SIP messages. How are you testing if asterisk is in the media path? Regards On 1/23/06, Steve Gladden [EMAIL PROTECTED] wrote: been testing with a rather simple setup. The mission is to actually get a reinvite to work on the lan. I am trying with two sipura phones G.711 codec forced on both both on the lan no nat no fancy options suchs as tT or H No matter what we do asterisk hangs on to the media path, how in the world do I get a reinvite to work where the media path is actually handled by the two phones on the lan? Any pointers greatly appreciated! Steve Pretty simple extensions, on lan no nat sip.conf [4785] type=friend username=4785 secret=test host=dynamic canreinvite=yes [4786] type=friend username=4786 secret=tesst host=dynamic canreinvite=yes extensions.conf exten = 4785,1,Dial(SIP/4785,66) exten = 4785,3,hangup exten = 4786,1,Dial(SIP/4786,66) exten = 4786,3,hangup ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] canreinvite always =no * no matter what we try :-(
Steve, The mission is to actually get a reinvite to work on the lan. There isn't anything special to get this working... normally. I trust you verified the traffic flow with a network monitor tool (tcpdump?), correct? Does SIP debug give you any info (i.e., does it match the right peer) -- you don't show if you allow reinvites globally? What about the nat= setting? Couple pointers I can give you to get you excited: 1) Reinvites work quite reliably, I use them between the PTSN gateway and the end user's ATA, all the way across the Internet -- nicely reduces latency. 2) If you use RFC2833 for DTMF you can issue an reinvite and still use t/T for transfer. NOTE that you have to modify the source to make asterisk reinvite even when it needs to listen to DTMFs. I give no guarantees how well it will work for you but it does work. See AST_BRIDGE_DTMF_CHANNEL_0 | AST_BRIDGE_DTMF_CHANNEL_1 in rtp.c. 3) Reinvites *can* work even if both ends are behind NAT. It really depends on the NATing router and the ATA. Sipura's and good NAT routers work, but I would not call it reliable -- it's really pushing it a bit... So if you really want to see why your Reinvites do not work, then you probably will have to make your hands dirty and analyze where ast_rtp_bridge() in rtp.c bails out. But since you are on a LAN it makes the situation a lot easier. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] canreinvite always =no * no matter what we try :-(
How are you testing if asterisk is in the media path? Two ways: One phone on a hub with ethereal on a laptop and watching the rtp packets, pretty obvious that asterisk is staying in the media path. and that the rtp i not coming from the other phone. Way two, in the middle of an active/established call unplugging the ethernet cable from the asterisk box audio instantly dies on both phones when this occurs. plug asterisk box back into it's ethernet termination audio comes right back. Seems odd that these reinvites are supposed to magically occur (from what I gather) and it only happens when the sun is shining and everything is just right... I'd like a way to force it or KNOW that it should be occuring versus just expecting it to 'possibly' occur automatically if all conditions are met and automatically detected. Or maybe I have this all worng :-) Thanks! Steve please turn on all the debug, warning, error etc messages in the console, see logger.conf, then type sip peer peer1 debug and sip peer peer2 debug to see the SIP messages. How are you testing if asterisk is in the media path? Regards On 1/23/06, Steve Gladden [EMAIL PROTECTED] wrote: been testing with a rather simple setup. The mission is to actually get a reinvite to work on the lan. I am trying with two sipura phones G.711 codec forced on both both on the lan no nat no fancy options suchs as tT or H No matter what we do asterisk hangs on to the media path, how in the world do I get a reinvite to work where the media path is actually handled by the two phones on the lan? Any pointers greatly appreciated! Steve Pretty simple extensions, on lan no nat sip.conf [4785] type=friend username=4785 secret=test host=dynamic canreinvite=yes [4786] type=friend username=4786 secret=tesst host=dynamic canreinvite=yes extensions.conf exten = 4785,1,Dial(SIP/4785,66) exten = 4785,3,hangup exten = 4786,1,Dial(SIP/4786,66) exten = 4786,3,hangup ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] canreinvite always =no * no matter what we try :-(
On Mon, 23 Jan 2006, Steve Gladden wrote: been testing with a rather simple setup. The mission is to actually get a reinvite to work on the lan. I am trying with two sipura phones G.711 codec forced on both both on the lan no nat no fancy options suchs as tT or H No matter what we do asterisk hangs on to the media path, how in the world do I get a reinvite to work where the media path is actually handled by the two phones on the lan? Any pointers greatly appreciated! Remove from your Dial command all options that require Asterisk to hear the media stream. (T, t etc) Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] canreinvite=yes
Hi, Just one question. The documentation I have seen says that the RTP audio stream is routed directly(if allowed ...), but never anything about video streams? Is this just because documents are pre 1.2 or is it true that audio can go directly, but video must pass through Asterisk? Anyone? Does anyone have experience with H263 on the 1.2.rc1 version? I think there is a bug, and will trace and submit it to Bugzilla..?? Trond ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] canreinvite=yes
Trond Andersen wrote: Just one question. The documentation I have seen says that the RTP audio stream is routed directly(if allowed ...), but never anything about video streams? Is this just because documents are pre 1.2 or is it true that audio can go directly, but video must pass through Asterisk? All RTP streams are handled identically, regardless of their content. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] canreinvite = yes with PAP2
Has anyone made this work? For me everything is fine until I switch canreinvite form no to yes. What happens is that asterisk hangs on attempting native bridge ... from what I understand attempting native bridge means that the RTP is routed through asterisk (just without any codec translation) But it shouldn't do that ... right? ... canreinvite is set to yes ... What's the best way to deal with this issue? I've also read that the only way to get the following situation ... UA --- NAT --- Internet --- NAT --- UA ... to work without passing the media path through asterisk is to use SER together with asterisk. Is that still true or was that because I was reading stuff from back in 2003? Some other discussions mention that canreinvite will simply not work with certain UAs .. is PAP2 one of those? .. Couple of other discussions that I've seen conclude that passing media stream UA-to-UA is just not practical when NAT is involved and is best to be avoided all together ... I'd like to make it work because it seems like a great way to save expensive server bandwidth. But if it will cause more trouble than it's worth then I will probably pass the media path through Asterisk and live with the fact that it will eat up my bandwidth. Also, IAX is superior when dealing with NATs , does it also handle UA-to-UA in NATed environment smoothly? What would be a good PAP2 alternative that uses IAX? This is my sip.conf: [1001] username=1001 type=friend secret= qualify=yes port=5060 nat=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=yes callerid=Test1 1001 ... My PAP2 is configured with: STUN=yes STUN=stun.xten.net NAT Keepalive = 15 Outbound proxy = blank Proxy = IP of asterisk Any suggestions? Thank you, Tomas ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] canreinvite=no being ignored?
Am I reading the data below incorrectly, or does it appear that even though I have the directive canreinvite=no set for the two asterisk boxes, they are trying to do a reinvite (which fails) anyway? Is this expected behaviour in this situation? If so, how can I prevent this? Lots of output Using CVS Head from 2005-08-28, I have two asterisk boxen, one (box A) has a sip ua (2608) attached which is generating a call, the other machine (box B) has the final destination. Sip config for the phone on box A (via Realtime): pbx3*CLI sip show peer 2608 * Name : 2608 Secret : Set MD5Secret: Not set Context : assigned-device Language : AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : -1 Inc. limit : 0 Outg. limit : 0 Dynamic : Yes Callerid : Expire : 386 Expiry : 900 Insecure : no Nat : RFC3581 ACL : No CanReinvite : No PromiscRedir : No User=Phone : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 192.168.10.32 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 2608 SIP Options : (none) Codecs : 0x4 (ulaw) Codec Order : (ulaw) Status : OK (16 ms) Useragent: Sipura/SPA841-0.9.1 Reg. Contact : sip:[EMAIL PROTECTED]:5060 Sip config for Box B on box A: pbx3*CLI sip show peer pbx1 * Name : boxb Secret : Not set MD5Secret: Not set Context : inter-system-inbound-main Language : AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : -1 Inc. limit : 0 Outg. limit : 0 Dynamic : No Callerid : Expire : -1 Expiry : 900 Insecure : no Nat : RFC3581 ACL : No CanReinvite : No PromiscRedir : No User=Phone : No DTMFmode : rfc2833 LastMsg : 0 ToHost : cut for public display Addr-IP : cut for public display Port 5060 Defaddr-IP : 0.0.0.0 Port 0 Def. Username: SIP Options : (none) Codecs : 0x4 (ulaw) Codec Order : (ulaw) Status : Unmonitored Useragent: Reg. Contact : Sip config for Box A on Box B pbx1*CLI sip show peer pbx3 pbx1*CLI * Name : boxa Secret : Not set MD5Secret: Not set Context : inter-system-inbound-main Language : AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : -1 Inc. limit : 0 Outg. limit : 0 Dynamic : No Callerid : Expire : -1 Expiry : 900 Insecure : no Nat : No ACL : No CanReinvite : No PromiscRedir : No User=Phone : No DTMFmode : rfc2833 LastMsg : 0 ToHost : cut for public display Addr-IP : cut for public display Port 5060 Defaddr-IP : 0.0.0.0 Port 0 Def. Username: SIP Options : (none) Codecs : 0x4 (ulaw) Codec Order : (ulaw) Status : Unmonitored Useragent: Reg. Contact : Dial command as appears on boxa -- Executing Dial(SIP/2608-8049, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] Aug 31 02:01:29 NOTICE[10496]: chan_sip.c:9028 handle_response: Failed to authenticate on INVITE to '2608 sip:2608@boxa-ip-here;tag=as4124f74a' -- SIP/boxb-ae96 is circuit-busy SIP Debug as it appears on boxb from the call above -- SIP read from boxa-ip-address:5060: INVITE sip:c1#1234@boxb-ip-address SIP/2.0 Via: SIP/2.0/UDP boxa-ip-address:5060;branch=z9hG4bK1d216175;rport From: 2608 sip:[EMAIL PROTECTED];tag=as4124f74a To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: 3f1250096c1a12b0259689006888f106@boxb-ip-address CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 31 Aug 2005 09:01:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 214 v=0 =root 10496 10496 IN IP4 boxa-ip-address s=session c=IN IP4 boxa-ip-address t=0 0 m=audio 19014 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (12 headers 10 lines)--- Using INVITE request as basis request - 3f1250096c1a12b0259689006888f106@boxa-ip-address Sending to boxa-ip-address : 5060 (non-NAT)
[Asterisk-Users] canreinvite in sip.conf
Hi, Im using asterisk 1.0.6 and I would let media path be connected directly between the phones without going through Asterisk. I have to it with an AtCom320 (with pa168s chip). I just saw and tryied to do what this page http://www.voip-info.org/tiki-index.php?page=Asterisk:%20Letting%20SIP%20clients%20connect%20directly says. Before going on (with sniffer eth traffic between * and two phones) Id like to known if it can works. Does anyone just did it? Thanks in advance Gio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] canreinvite=yes not working with sipura device.
I'm trying to get canreinvite=yes to work. I would like asterisk to release the line and let the 2 ports on the sipura device to talk to each other directly. Is there a setting I need to activate on the sipura device, or is there something else I need to do? It's possible that it is a nat problem as the sip device is behind a firewall, but it works fine otherwise. Any suggestions? Thanks, Jon. p.s. it's a sipura3000, but it should be the same for any sip device. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] canreinvite=yes not working with sipura device.
I'm trying to get canreinvite=yes to work. As the name says, this setting allows reinvites but does not force them. I just ran into the same issue last week. Here the caveats: Reinvites will only happen when both ends use the same codes, there is no t or T option in the dial command when making the call, and I think when both ends are behind the same NAT or not NAT'ed... but don't quote me on the last one. For the Sipura 3000 the last one is not an issue since it's the same physical device for both ports. Try sip debug and look for line like Found Peer or Found User on an incoming call to make sure the right one is used, unless you have canreinvite turned on globally. --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Canreinvite issue
On Apr 7, 2005 8:36 PM, kaiser [EMAIL PROTECTED] wrote: Hi , all: Anyone try sip channel with canreinvite=yes? sometimes we see a new INVITE will be send to UA immediately after user hangup the call. It makes the phone ring again after hangup. Anyone know what happen? It not always, maybe 2-5% only. But it make user crazy. Thanks... So that's what causes that. Had it happen a few times with my Sipura 2000. Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Canreinvite issue
Hi , all: Anyone try sip channel with canreinvite=yes? sometimes we see a new INVITE will be send to UA immediately after user hangup the call. It makes the phone ring again after hangup. Anyone know what happen? It not always, maybe 2-5% only. But it make user crazy. Thanks... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Canreinvite=???
The KEY thing you are missing is that IAX does NOT use RTP for audio. IAX uses IAX for audio and IAX for signaling. You CANNOT reinvite between a SIP/RTP endpoint and an IAX endpoint. Since SIP, H323, MGCP, and SCCP/Skinny all use RTP for audio, in theory you could do RTP reinvites between these protocols. I have no idea if Asterisk supports this or not. On Sat, 2004-09-18 at 00:52, Carlos Arnt wrote: Looking at this explanation : When SIP initiates the call, the INVITE message contains the information on where to send the media streams. Asterisk uses itself as the end-points of media streams when setting up the call. Once the call has been accepted, Asterisk sends another (re)INVITE message to the clients with the information necessary to have the two clients send the media streams directly to each other. So if i really understand this using this option i can make the RTP packets flow from one device to another when they connect leaving only the SIP to asterisk . So for example if then I put my Grandstream with a real ip address and use * with a real ip address i can make my calls from nufone flow direct to my grandstream leaving my * bandwidth free . Like this : Grandstream begin call SIP--- Asterisk | | - Nufone. Open RTP Channel Grandstream Real IP -- Nufone IP Right ?? If i'm right , i try this and with tcpdump see the even with everyone using real ip's, the RTP still going over asterisk using my bandwidth . (Note, I force grandstream to use the same codec then Nufone, G729) Can someone give-me some light ?? ;) Can i make this ??? Use asterisk only to begin the call and let the RTP flow over the client and nufone network ?? Thanks alot ! Carlos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Canreinvite=???
Hi, everyone ! Looking at this explanation : When SIP initiates the call, the INVITE message contains the information on where to send the media streams. Asterisk uses itself as the end-points of media streams when setting up the call. Once the call has been accepted, Asterisk sends another (re)INVITE message to the clients with the information necessary to have the two clients send the media streams directly to each other. So if i really understand this using this option i can make the RTP packets flow from one device to another when they connect leaving only the SIP to asterisk . So for example if then I put my Grandstream with a real ip address and use * with a real ip address i can make my calls from nufone flow direct to my grandstream leaving my * bandwidth free . Like this : Grandstream begin call SIP--- Asterisk | | - Nufone. Open RTP Channel Grandstream Real IP -- Nufone IP Right ?? If i'm right , i try this and with tcpdump see the even with everyone using real ip's, the RTP still going over asterisk using my bandwidth . (Note, I force grandstream to use the same codec then Nufone, G729) Can someone give-me some light ?? ;) Can i make this ??? Use asterisk only to begin the call and let the RTP flow over the client and nufone network ?? Thanks alot ! Carlos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Users Canreinvite=[yes|no] explained (new subject)
Senad Jordanovic wrote: brian wrote: That's the only way to make it work. Devices behind nat, on same network, can call each other ONLY if canreinvite is set to no? Is that what you are saying? Canreinvite=yes *only* works if all devices are on the same side of the NAT, the outside or the inside. If one device is on a different side of a NAT device than another phone or Asterisk, you can't allow re-invites for that device. Let me explain: Phone A - Asterisk - phone B Phone A calls phone B. The first thing you have to remember is that Asterisk is *not* a SIP proxy, it's a SIP PBX. So phone A sends an invite to Asterisk. Asterisk starts a *new* SIP dialog and sends an invite to Phone B. If Phone B accepts the call (answers), it sends an 200 OK sip message to Asterisk. Asterisk sends a 200 OK SIP message to phone A. We now have *two* SIP calls, or dialogs, that are bridged through Asterisk. The natural behaviour for the Asterisk SIP channel in this case is to get a feeling of Hey, these two phones are using me just for transport of bits between themselves. Why not get out of the loop and let them exchange these bits directly with each other? I can use the spare CPU cycles for something more meaningful than shipping these bits... So, in order to get out of this meaningless position, Asterisk checks if the phones are compatible. If they support the same codec, if they can talk to each other. If *one* of them have canreinvite=no or something else that stops a direct audio relationship from phone A to B, Asterisk stays in the middle of things, shipping bits between the phones (the audio stream). If they are compatible, Asterisk sends a new SIP INVITE to both of the phones, redirecting the media streams for the current calls to each other. This is called a Re-INVITE, an INVITE where we change media for an existing SIP dialog, not trying to start a new call. This works perfectly well if all phones are compatible and we have the network configured so that they can talk full duplex to each other. If one is behind a NAT, they can't. If that's the case, we'll only get one audio stream, from the phone on the inside to the one on the outside. That's why we want to set canreinvite=no to make sure Asterisk doesn't even try to re-INVITE. -- Executive Summary: canreinvite=no in a [peer] configuration makes Asterisk stay in the middle of the media stream. This is the safe setting, but remember it makes your CPU always handle the media stream. Regards, /Olle PS. Yes, the Asteriskdocs.org project is allowed to use this text if they find it meaningful :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [Asterisk-Users Canreinvite=[yes|no] explained (new subject)
Based on my post yesterday, and the call trace I have, if Asterisk were to make a decision a little differently when sending the the ReINVITEs to phone B in your example (lets say Phone A is the one behind NAT) media might work both directions. In the trace I posted, asterisk first send a reinvite with the private IP of phone A, but then it sent a second reinvite with the visible IP of phone A, which I think would have made the call work in some NAT environments, but then it sent a *3rd* reinvite to phone B, back to the private IP of phone A, breaking the audio from phone B to phone A. What I think happened was when phone A send the 200 OK for its reinvite, Asterisk saw the SDP info from that packet and triggered the 3rd reinvite to phone B, but since nat=1 was on, it should have ignored that SDP, or at least sent the visible IP in the 3rd reinvite and not the private IP. In case you don't have it handy, the call trace I am referring to is here: http://www.cheapnet.net/~mike/asterisk_excel_with_reinvite.log In this log, only the 192.x network is nat'd. 10.x and 172.x have straight routing between them. 10.10.11.77 is the visible IP to 192.168.222.197 according to 10.x and 172.x. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] canreinvite and transcoding
Does anyone know if it is possible to force a extension to not allow transcoding? If you spec canreinvite=yes the cal may still transcoded if the parties do not choose a the same code on each end. In my situation it is better that the call fail than have it transcoded. Also, I see some limited reference to canreinvite=update. Does this command exist and if so what does it do. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] canreinvite=yes in sip.conf still go through asterisk
Hi, Thank you for the information. There are ts in Dial command in extensions.conf. When I deleted these ts, each sip phones were directly communicating. I just wrote these ts from the examples. Does these t and T are used for transfer(blind/consaltation) from called user and calling user, respectively? If we don't have these 't' and 'T', can't we do transfer? Regards, Zen Girish Gopinath [EMAIL PROTECTED] wrote : Zen, I am trying to confirm the command 'canreinvite=yes' in sip.conf using grandstream BT101/2s and snom100s. In either case, no description nor 'canreinvite=yes', media stream always go through *. Do I need another settings for confirming sip clients directly communicate each other? Do you have a Dial statement that has t or T in it? This will force the media stream to pass through Asterisk. Regards, Girish _ Contact brides grooms FREE! http://www.shaadi.com/ptnr.php?ptnr=hmltag Only on www.shaadi.com. Register now! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] canreinvite=yes in sip.conf still go through asterisk
t and T are for # transfers. Other types of transfer are done in other ways. Zap FLASH transfers are set in /etc/asterisk/zapata.conf. I don't know how you enable/disable SIP or other types of transfers. On Thu, 2004-03-04 at 06:51, Zen Kato wrote: Hi, Thank you for the information. There are ts in Dial command in extensions.conf. When I deleted these ts, each sip phones were directly communicating. I just wrote these ts from the examples. Does these t and T are used for transfer(blind/consaltation) from called user and calling user, respectively? If we don't have these 't' and 'T', can't we do transfer? Regards, Zen Girish Gopinath [EMAIL PROTECTED] wrote : Zen, I am trying to confirm the command 'canreinvite=yes' in sip.conf using grandstream BT101/2s and snom100s. In either case, no description nor 'canreinvite=yes', media stream always go through *. Do I need another settings for confirming sip clients directly communicate each other? Do you have a Dial statement that has t or T in it? This will force the media stream to pass through Asterisk. Regards, Girish _ Contact brides grooms FREE! http://www.shaadi.com/ptnr.php?ptnr=hmltag Only on www.shaadi.com. Register now! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- For Asterisk PBX related documentation go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section also see http://www.voip-info.org/wiki-Asterisk also see my site at http://www.fnords.org/~eric/asterisk/ BTEL Consulting ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] canreinvite=yes in sip.conf still go through asterisk
Hi, I am trying to confirm the command 'canreinvite=yes' in sip.conf using grandstream BT101/2s and snom100s. In either case, no description nor 'canreinvite=yes', media stream always go through *. Do I need another settings for confirming sip clients directly communicate each other? -- Zen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] canreinvite=yes in sip.conf still go through asterisk
Zen, I am trying to confirm the command 'canreinvite=yes' in sip.conf using grandstream BT101/2s and snom100s. In either case, no description nor 'canreinvite=yes', media stream always go through *. Do I need another settings for confirming sip clients directly communicate each other? Do you have a Dial statement that has t or T in it? This will force the media stream to pass through Asterisk. Regards, Girish _ Contact brides grooms FREE! http://www.shaadi.com/ptnr.php?ptnr=hmltag Only on www.shaadi.com. Register now! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] canreinvite and codec negotations...
Okay, now on to my problem.. I have people who will be using ulaw, and I have people who will be using g729.. I want to set it up so that canreinivte will work.. I have a single cisco gateway.. Asterisks isn't handling the negotation between the 2 devices very well.. For example.. [gateway] type=friend host=1.2.3.4 canreinvite=yes qualify=200 dtmfmode=rfc2833 context=default disallow=all allow=ulaw allow=g729 [123] type=friend secret=abc nat=yes host=dynamic canreinvite=yes qualify=200 context=default dtmfmode=rfc2833 mailbox=2199 callerid=Joe Blow 123-456-7890 disallow=all allow=g729 [321] type=friend secret=abc nat=yes host=dynamic canreinvite=yes qualify=200 context=default dtmfmode=rfc2833 mailbox=321 callerid=Joe Blow 321-456-7890 disallow=all allow=ulaw Okay, in this configs, gateway would be my cisco 26xx gateway.. ext 123 would be a g729 customer.. and 321 would be a ulaw customer. When someone calls ext 123, the cisco send the call to asterisk, it *WILL* use ulaw... then it will initiate a call to g729, well... Now we have a codec mismatch, and canreinvite won't work... EVEN though gateway can do g729.. ext 321 won't have any problems.. It'll work fine for them.. What can we do to get this to work like it should? Thank, Billy +--+ | Billy HuddlestonSenior System Administrator | | Net-Express http://www.nxs.net | | 114 Sherway Rd. Voice: 865-691-2011 | | Knoxville, TN 37922 Fax: 865-691-9894 | | [EMAIL PROTECTED]| +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
