Re: [Asterisk-Users] SIP conections, with RTP not going trough Asterisk
Hi, sorry to bother again. But I still cannot make it work. I made all acounts have canreinvite=yes, but found no option in Dial aplication to make the phones exchange RTP directly between them. Can anyone tell me wich option should I look at? I am stuck with this (probably simple) problem for almost a whole week. Thanks for any help. Ronald Wiplinger wrote: Tiago Stein D`Agostini wrote: Hi, Ie been looking for some time how to use asterisk to initiate SIP connections between 2 IP phones, but afetr initiated the communication making the RTP go directly from one telephone to the other, without passing by asterisk. Unfortunately I found no explanations of how to do it. Does anyone care to give a pointer to any explanation about how to do it? canreinvite=yes and look at the options for dial() Thanks in advance bye Ronald Wiplinger ___ --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] SIP conections, with RTP not going trough Asterisk
On 17/04/06, Tiago Stein D`Agostini [EMAIL PROTECTED] wrote: Hi, sorry to bother again. But I still cannot make it work. I made all acounts have canreinvite=yes, but found no option in Dial aplication to make the phones exchange RTP directly between them. Can anyone tell me wich option should I look at? I am stuck with this (probably simple) problem for almost a whole week. You're trying too hard - unless you tell it not to, the Dial application will do what you're asking. As Olle said, this is the default. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --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] SIP conections, with RTP not going trough Asterisk
Tiago Stein D`Agostini wrote: Hi, sorry to bother again. But I still cannot make it work. I made all acounts have canreinvite=yes, but found no option in Dial aplication to make the phones exchange RTP directly between them. Can anyone tell me wich option should I look at? I am stuck with this (probably simple) problem for almost a whole week. What is exactly your dial command? bye Ronald Wiplinger Thanks for any help. Ronald Wiplinger wrote: Tiago Stein D`Agostini wrote: Hi, Ie been looking for some time how to use asterisk to initiate SIP connections between 2 IP phones, but afetr initiated the communication making the RTP go directly from one telephone to the other, without passing by asterisk. Unfortunately I found no explanations of how to do it. Does anyone care to give a pointer to any explanation about how to do it? canreinvite=yes and look at the options for dial() Thanks in advance bye Ronald Wiplinger ___ --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 --- avast! Antivirus: Inbound message clean. Virus Database (VPS): 0615-3, 2006/04/14 Tested on: 2006/4/17 ¤U¤È 07:19:19 avast! - copyright (c) 1988-2006 ALWIL Software. http://www.avast.com -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com http://voip.elmit.com http://e-paper.elmit.com Tel. (M) +886.939.775.516 (O) +886.2.2835.7765 (ENUM) or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ --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] SIP conections, with RTP not going trough Asterisk
Tiago Stein D`Agostini wrote: Hi, sorry to bother again. But I still cannot make it work. I made all acounts have canreinvite=yes, but found no option in Dial aplication to make the phones exchange RTP directly between them. Can anyone tell me wich option should I look at? I am stuck with this (probably simple) problem for almost a whole week. The canreinvite=yes is required, however your Dial statements used to complete calls between the sip devices cannot use several of the options including t, T, etc. If you remove all options from the Dial statement, restart asterisk, and place a test call, those sip phones that can see each other will auto-negotiate rtp directly between them. If they cannot see each other (eg, nat or firewalls involved), they will not auto-negotiate direct rtp. There is no option for you to specify to forced direct rtp. ___ --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] SIP conections, with RTP not going trough Asterisk
So, is there any other option that prevents that from happening? Something that I might have turned on and makes Dial work trough asterisk? I already even removed asterisk completelyu from system and reinstalled it to be fresh new... still all RTP goes trough Asterisk machine. And the server really can't handle many connections this way. Thanks for the help. Peter Bowyer wrote: On 17/04/06, Tiago Stein D`Agostini [EMAIL PROTECTED] wrote: Hi, sorry to bother again. But I still cannot make it work. I made all acounts have canreinvite=yes, but found no option in Dial aplication to make the phones exchange RTP directly between them. Can anyone tell me wich option should I look at? I am stuck with this (probably simple) problem for almost a whole week. You're trying too hard - unless you tell it not to, the Dial application will do what you're asking. As Olle said, this is the default. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --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] SIP conections, with RTP not going trough Asterisk
Hi Ronald! Please check if the following points are NOT activated. * is not using direct phone to phone RTP streams if: -) either of the clients is configured with canreinvite=no -) the clients cannot agree on a common set of codecs and * needs to perform codec conversion -) either of the clients is configured with nat=yes -) * needs to listen to DTMF tones during the call (for transfers or any other features) Hope this helps, Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tiago Stein D`Agostini Sent: Montag, 17. April 2006 13:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP conections,with RTP not going trough Asterisk Hi, sorry to bother again. But I still cannot make it work. I made all acounts have canreinvite=yes, but found no option in Dial aplication to make the phones exchange RTP directly between them. Can anyone tell me wich option should I look at? I am stuck with this (probably simple) problem for almost a whole week. Thanks for any help. Ronald Wiplinger wrote: Tiago Stein D`Agostini wrote: Hi, Ie been looking for some time how to use asterisk to initiate SIP connections between 2 IP phones, but afetr initiated the communication making the RTP go directly from one telephone to the other, without passing by asterisk. Unfortunately I found no explanations of how to do it. Does anyone care to give a pointer to any explanation about how to do it? canreinvite=yes and look at the options for dial() Thanks in advance bye Ronald Wiplinger ___ --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] SIP conections, with RTP not going trough Asterisk
Thanks, that was the problem, I had the t option on the Dial application. Nor that I removed them it works. Thank you. Rich Adamson wrote: Tiago Stein D`Agostini wrote: Hi, sorry to bother again. But I still cannot make it work. I made all acounts have canreinvite=yes, but found no option in Dial aplication to make the phones exchange RTP directly between them. Can anyone tell me wich option should I look at? I am stuck with this (probably simple) problem for almost a whole week. The canreinvite=yes is required, however your Dial statements used to complete calls between the sip devices cannot use several of the options including t, T, etc. If you remove all options from the Dial statement, restart asterisk, and place a test call, those sip phones that can see each other will auto-negotiate rtp directly between them. If they cannot see each other (eg, nat or firewalls involved), they will not auto-negotiate direct rtp. There is no option for you to specify to forced direct rtp. ___ --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] SIP conections, with RTP not going trough Asterisk
On 17/04/06, Tiago Stein D`Agostini [EMAIL PROTECTED] wrote: So, is there any other option that prevents that from happening? Something that I might have turned on and makes Dial work trough asterisk? I already even removed asterisk completelyu from system and reinstalled it to be fresh new... still all RTP goes trough Asterisk machine. And the server really can't handle many connections this way. What options are you using? Post an extract of your dialplan and sip.conf. And how are you determining that the RTP is going through Asterisk? Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --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] SIP conections, with RTP not going trough Asterisk
On 4/17/06, Alex Mosburger [EMAIL PROTECTED] wrote: -) * needs to listen to DTMF tones during the call (for transfers or any other features) Does this mean you cannot do any blind or attended transfer? or only the # transfer option (asterisk built-in, from features.conf) doesn't work? cheers ___ --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] SIP conections, with RTP not going trough Asterisk
Hi, Ie been looking for some time how to use asterisk to initiate SIP connections between 2 IP phones, but afetr initiated the communication making the RTP go directly from one telephone to the other, without passing by asterisk. Unfortunately I found no explanations of how to do it. Does anyone care to give a pointer to any explanation about how to do it? Thanks in advance ___ --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] SIP conections, with RTP not going trough Asterisk
Tiago Stein D`Agostini wrote: Hi, Ie been looking for some time how to use asterisk to initiate SIP connections between 2 IP phones, but afetr initiated the communication making the RTP go directly from one telephone to the other, without passing by asterisk. Unfortunately I found no explanations of how to do it. Does anyone care to give a pointer to any explanation about how to do it? canreinvite=yes and look at the options for dial() Thanks in advance bye Ronald Wiplinger ___ --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] SIP conections, with RTP not going trough Asterisk
12 apr 2006 kl. 14.58 skrev Ronald Wiplinger: Tiago Stein D`Agostini wrote: Hi, Ie been looking for some time how to use asterisk to initiate SIP connections between 2 IP phones, but afetr initiated the communication making the RTP go directly from one telephone to the other, without passing by asterisk. Unfortunately I found no explanations of how to do it. Does anyone care to give a pointer to any explanation about how to do it? canreinvite=yes and look at the options for dial() Thanks in advance Actually, it's the default mode. Just connect your phones to Asterisk on the same LAN, and Asterisk will get out of the media path. /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --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
