[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
[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
[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
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
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=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
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" ;tag=as636875d3 To: Contact: 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" ;tag=as636875d3 To: ;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 ___
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 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 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 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
[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 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
[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 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
Re: [Asterisk-Users] canreinvite=yes in sip.conf still go through asterisk
Hi, Thank you for the information. There are "t"s in Dial command in extensions.conf. When I deleted these "t"s, each sip phones were directly communicating. I just wrote these "t"s 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 "t"s in Dial command in > extensions.conf. When I deleted these "t"s, each sip phones were > directly communicating. I just wrote these "t"s 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
Re: [Asterisk-Users] canreinvite=yes in sip.conf still go through asterisk
Hi Zen, From: Zen Kato <[EMAIL PROTECTED]> 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? 'T' and 't' are used for transfer using # The 'T' allows the calling user to transfer the call. 't' allows the called user to transfer the call. Andy Powell's guide to Asterisk http://www.automated.it/guidetoasterisk.htm has these details, It is simple, and contains some good basic things about Asterisk. Regards, Girish 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 > _ Skin is in! Bollywood is sizzling. http://server1.msn.co.in/slideshow/striptease/ Check out these hot pics! ___ 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 :-)
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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
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=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.
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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.
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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
