[Asterisk-Users] Can audio streams go client to cleint with IAX?
With a service like http://www.freshtel.net/?show=home that uses IAX and has servers in Australia, is it possible for the audio streams to take a different path than the call setup and control? In other words can it work like SIP with canreinvite where the two endpoint negotiate audio streams between themselves rather than though the FreshTel server? 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] Can audio streams go client to cleint with IAX?
Reading this thread leads me to chance asking a somewhat broad question of the gurus: is there a place in VoIP for multicasting? Streaming scenarios, as well as conferencing, would seem to be ripe for that sort of integration, but I know nothing more about it beyond the fact that multicasting is oft-mentioned, but as far as I can tell (maybe kphone's vat hooks are an exception?) seldom implemented. . . Thanks in advance for any light that might be shed. B. ___ 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] Can audio streams go client to cleint with IAX?
yes, IAX does direct transfers - when both ends confirm they can see each other, the asterisk server tells them to talk directly. With the firefly network, we're seeing 90%+ connecting directly. Just to clarify, the audio doesn't separate from the call control. -Adam - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 1:59 AM Subject: [Asterisk-Users] Can audio streams go client to cleint with IAX? With a service like http://www.freshtel.net/?show=home that uses IAX and has servers in Australia, is it possible for the audio streams to take a different path than the call setup and control? In other words can it work like SIP with canreinvite where the two endpoint negotiate audio streams between themselves rather than though the FreshTel server? 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 ___ 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] Can audio streams go client to cleint with IAX?
Dear All, Now, it seems that both IAX and SIP can have the two endpoints communicate the media directly without the media stream passing through the asterisk, can we do the same with H323 too? TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Hart Sent: Monday, February 02, 2004 5:28 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Can audio streams go client to cleint with IAX? yes, IAX does direct transfers - when both ends confirm they can see each other, the asterisk server tells them to talk directly. With the firefly network, we're seeing 90%+ connecting directly. Just to clarify, the audio doesn't separate from the call control. -Adam - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 1:59 AM Subject: [Asterisk-Users] Can audio streams go client to cleint with IAX? With a service like http://www.freshtel.net/?show=home that uses IAX and has servers in Australia, is it possible for the audio streams to take a different path than the call setup and control? In other words can it work like SIP with canreinvite where the two endpoint negotiate audio streams between themselves rather than though the FreshTel server? 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 ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ 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] Can audio streams go client to cleint with IAX?
Sure, if you want to code it :) Wouldn't be that hard either. Actually what would be really sexy (if that's the word) is native transfers between H.323 and SIP given they both use the same RTP (correct me if I'm wrong on that) - Original Message - From: T. Chan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 9:56 AM Subject: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Dear All, Now, it seems that both IAX and SIP can have the two endpoints communicate the media directly without the media stream passing through the asterisk, can we do the same with H323 too? TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Hart Sent: Monday, February 02, 2004 5:28 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Can audio streams go client to cleint with IAX? yes, IAX does direct transfers - when both ends confirm they can see each other, the asterisk server tells them to talk directly. With the firefly network, we're seeing 90%+ connecting directly. Just to clarify, the audio doesn't separate from the call control. -Adam - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 1:59 AM Subject: [Asterisk-Users] Can audio streams go client to cleint with IAX? With a service like http://www.freshtel.net/?show=home that uses IAX and has servers in Australia, is it possible for the audio streams to take a different path than the call setup and control? In other words can it work like SIP with canreinvite where the two endpoint negotiate audio streams between themselves rather than though the FreshTel server? 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 ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ 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] Can audio streams go client to cleint with IAX?
Is it possible to make audio streams go client to client with H.323 ? (both client being H323) Thanks! -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de T. Chan Enviado el: lunes, 02 de febrero de 2004 23:56 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Dear All, Now, it seems that both IAX and SIP can have the two endpoints communicate the media directly without the media stream passing through the asterisk, can we do the same with H323 too? TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Hart Sent: Monday, February 02, 2004 5:28 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Can audio streams go client to cleint with IAX? yes, IAX does direct transfers - when both ends confirm they can see each other, the asterisk server tells them to talk directly. With the firefly network, we're seeing 90%+ connecting directly. Just to clarify, the audio doesn't separate from the call control. -Adam - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 1:59 AM Subject: [Asterisk-Users] Can audio streams go client to cleint with IAX? With a service like http://www.freshtel.net/?show=home that uses IAX and has servers in Australia, is it possible for the audio streams to take a different path than the call setup and control? In other words can it work like SIP with canreinvite where the two endpoint negotiate audio streams between themselves rather than though the FreshTel server? 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 ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ 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] Can audio streams go client to cleint with IAX?
If you define possible as is H.323 capable of it, then yes. If you define possible as is asterisk currently capable of it, then no. It wouldn't be too hard, look in channels/h323/ast_h323.cpp to see where Jeremy started on it. You just have to get the openh323 lib to initiate the transfer. - Original Message - From: Marc Fargas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 10:30 AM Subject: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Is it possible to make audio streams go client to client with H.323 ? (both client being H323) Thanks! -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de T. Chan Enviado el: lunes, 02 de febrero de 2004 23:56 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Dear All, Now, it seems that both IAX and SIP can have the two endpoints communicate the media directly without the media stream passing through the asterisk, can we do the same with H323 too? TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Hart Sent: Monday, February 02, 2004 5:28 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Can audio streams go client to cleint with IAX? yes, IAX does direct transfers - when both ends confirm they can see each other, the asterisk server tells them to talk directly. With the firefly network, we're seeing 90%+ connecting directly. Just to clarify, the audio doesn't separate from the call control. -Adam - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 1:59 AM Subject: [Asterisk-Users] Can audio streams go client to cleint with IAX? With a service like http://www.freshtel.net/?show=home that uses IAX and has servers in Australia, is it possible for the audio streams to take a different path than the call setup and control? In other words can it work like SIP with canreinvite where the two endpoint negotiate audio streams between themselves rather than though the FreshTel server? 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 ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ 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 ___ 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] Can audio streams go client to cleint with IAX?
Generally speaking, unless you're using an rtp proxy, the rtp audio should go client--client. H323 does the call setup and teardown and such, but the audio stream is usually direct. Jeremy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marc Fargas Sent: Monday, February 02, 2004 4:31 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Is it possible to make audio streams go client to client with H.323 ? (both client being H323) Thanks! -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de T. Chan Enviado el: lunes, 02 de febrero de 2004 23:56 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Dear All, Now, it seems that both IAX and SIP can have the two endpoints communicate the media directly without the media stream passing through the asterisk, can we do the same with H323 too? TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Hart Sent: Monday, February 02, 2004 5:28 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Can audio streams go client to cleint with IAX? yes, IAX does direct transfers - when both ends confirm they can see each other, the asterisk server tells them to talk directly. With the firefly network, we're seeing 90%+ connecting directly. Just to clarify, the audio doesn't separate from the call control. -Adam - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 1:59 AM Subject: [Asterisk-Users] Can audio streams go client to cleint with IAX? With a service like http://www.freshtel.net/?show=home that uses IAX and has servers in Australia, is it possible for the audio streams to take a different path than the call setup and control? In other words can it work like SIP with canreinvite where the two endpoint negotiate audio streams between themselves rather than though the FreshTel server? 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 ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ 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 ___ 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] Can audio streams go client to cleint with IAX?
Would that be something that Jeremy would work on? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Hart Sent: Monday, February 02, 2004 6:36 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Can audio streams go client to cleint with IAX? If you define possible as is H.323 capable of it, then yes. If you define possible as is asterisk currently capable of it, then no. It wouldn't be too hard, look in channels/h323/ast_h323.cpp to see where Jeremy started on it. You just have to get the openh323 lib to initiate the transfer. - Original Message - From: Marc Fargas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 10:30 AM Subject: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Is it possible to make audio streams go client to client with H.323 ? (both client being H323) Thanks! -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de T. Chan Enviado el: lunes, 02 de febrero de 2004 23:56 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Dear All, Now, it seems that both IAX and SIP can have the two endpoints communicate the media directly without the media stream passing through the asterisk, can we do the same with H323 too? TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Hart Sent: Monday, February 02, 2004 5:28 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Can audio streams go client to cleint with IAX? yes, IAX does direct transfers - when both ends confirm they can see each other, the asterisk server tells them to talk directly. With the firefly network, we're seeing 90%+ connecting directly. Just to clarify, the audio doesn't separate from the call control. -Adam - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 1:59 AM Subject: [Asterisk-Users] Can audio streams go client to cleint with IAX? With a service like http://www.freshtel.net/?show=home that uses IAX and has servers in Australia, is it possible for the audio streams to take a different path than the call setup and control? In other words can it work like SIP with canreinvite where the two endpoint negotiate audio streams between themselves rather than though the FreshTel server? 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 ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ 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 ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ 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] Can audio streams go client to cleint with IAX?
I think it would be fair to say Jeremy isn't really a fan of H.323 and, although he will fix bugs, (IMO) he won't be developing it anymore. You should give it a shot. - Original Message - From: T. Chan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 10:50 AM Subject: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Would that be something that Jeremy would work on? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Hart Sent: Monday, February 02, 2004 6:36 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Can audio streams go client to cleint with IAX? If you define possible as is H.323 capable of it, then yes. If you define possible as is asterisk currently capable of it, then no. It wouldn't be too hard, look in channels/h323/ast_h323.cpp to see where Jeremy started on it. You just have to get the openh323 lib to initiate the transfer. - Original Message - From: Marc Fargas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 10:30 AM Subject: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Is it possible to make audio streams go client to client with H.323 ? (both client being H323) Thanks! -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de T. Chan Enviado el: lunes, 02 de febrero de 2004 23:56 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Dear All, Now, it seems that both IAX and SIP can have the two endpoints communicate the media directly without the media stream passing through the asterisk, can we do the same with H323 too? TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Hart Sent: Monday, February 02, 2004 5:28 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Can audio streams go client to cleint with IAX? yes, IAX does direct transfers - when both ends confirm they can see each other, the asterisk server tells them to talk directly. With the firefly network, we're seeing 90%+ connecting directly. Just to clarify, the audio doesn't separate from the call control. -Adam - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 1:59 AM Subject: [Asterisk-Users] Can audio streams go client to cleint with IAX? With a service like http://www.freshtel.net/?show=home that uses IAX and has servers in Australia, is it possible for the audio streams to take a different path than the call setup and control? In other words can it work like SIP with canreinvite where the two endpoint negotiate audio streams between themselves rather than though the FreshTel server? 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 ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ 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 ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ Asterisk-Users mailing
RE: [Asterisk-Users] Can audio streams go client to cleint with IAX?
That doesn't accord to m tests. Actually I have gnugk as the gatekeeper then if I call directly both clients it goes fine but if they're connected trough asterisk sound is really really bad so it seems asterisk stays on the line and that audio is going between it. (both clients and asterisk are on the same network) Strange ugh? ;) -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Jeremy Jones Enviado el: martes, 03 de febrero de 2004 0:42 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Generally speaking, unless you're using an rtp proxy, the rtp audio should go client--client. H323 does the call setup and teardown and such, but the audio stream is usually direct. Jeremy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marc Fargas Sent: Monday, February 02, 2004 4:31 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Is it possible to make audio streams go client to client with H.323 ? (both client being H323) Thanks! -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de T. Chan Enviado el: lunes, 02 de febrero de 2004 23:56 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Dear All, Now, it seems that both IAX and SIP can have the two endpoints communicate the media directly without the media stream passing through the asterisk, can we do the same with H323 too? TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Hart Sent: Monday, February 02, 2004 5:28 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Can audio streams go client to cleint with IAX? yes, IAX does direct transfers - when both ends confirm they can see each other, the asterisk server tells them to talk directly. With the firefly network, we're seeing 90%+ connecting directly. Just to clarify, the audio doesn't separate from the call control. -Adam - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 1:59 AM Subject: [Asterisk-Users] Can audio streams go client to cleint with IAX? With a service like http://www.freshtel.net/?show=home that uses IAX and has servers in Australia, is it possible for the audio streams to take a different path than the call setup and control? In other words can it work like SIP with canreinvite where the two endpoint negotiate audio streams between themselves rather than though the FreshTel server? 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 ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ 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 ___ 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] Can audio streams go client to cleint with IAX?
Thanks a lot I didn't see tose posts, so I'll have to migarte to SIP as I can see.. Do I need any other software or only Asterisk and my FXO's and FXS's firmwares migrated to SIP ? Thanks a lot. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Adam Hart Enviado el: martes, 03 de febrero de 2004 0:52 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] Can audio streams go client to cleint with IAX? I think it would be fair to say Jeremy isn't really a fan of H.323 and, although he will fix bugs, (IMO) he won't be developing it anymore. You should give it a shot. - Original Message - From: T. Chan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 10:50 AM Subject: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Would that be something that Jeremy would work on? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Hart Sent: Monday, February 02, 2004 6:36 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Can audio streams go client to cleint with IAX? If you define possible as is H.323 capable of it, then yes. If you define possible as is asterisk currently capable of it, then no. It wouldn't be too hard, look in channels/h323/ast_h323.cpp to see where Jeremy started on it. You just have to get the openh323 lib to initiate the transfer. - Original Message - From: Marc Fargas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 10:30 AM Subject: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Is it possible to make audio streams go client to client with H.323 ? (both client being H323) Thanks! -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de T. Chan Enviado el: lunes, 02 de febrero de 2004 23:56 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Dear All, Now, it seems that both IAX and SIP can have the two endpoints communicate the media directly without the media stream passing through the asterisk, can we do the same with H323 too? TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Hart Sent: Monday, February 02, 2004 5:28 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Can audio streams go client to cleint with IAX? yes, IAX does direct transfers - when both ends confirm they can see each other, the asterisk server tells them to talk directly. With the firefly network, we're seeing 90%+ connecting directly. Just to clarify, the audio doesn't separate from the call control. -Adam - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 1:59 AM Subject: [Asterisk-Users] Can audio streams go client to cleint with IAX? With a service like http://www.freshtel.net/?show=home that uses IAX and has servers in Australia, is it possible for the audio streams to take a different path than the call setup and control? In other words can it work like SIP with canreinvite where the two endpoint negotiate audio streams between themselves rather than though the FreshTel server? 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 ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ 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 ___ 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
Re: [Asterisk-Users] Can audio streams go client to cleint with IAX?
what clients are you using? They probably won't support SIP - Original Message - From: Marc Fargas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 11:18 AM Subject: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Thanks a lot I didn't see tose posts, so I'll have to migarte to SIP as I can see.. Do I need any other software or only Asterisk and my FXO's and FXS's firmwares migrated to SIP ? Thanks a lot. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Adam Hart Enviado el: martes, 03 de febrero de 2004 0:52 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] Can audio streams go client to cleint with IAX? I think it would be fair to say Jeremy isn't really a fan of H.323 and, although he will fix bugs, (IMO) he won't be developing it anymore. You should give it a shot. - Original Message - From: T. Chan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 10:50 AM Subject: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Would that be something that Jeremy would work on? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Hart Sent: Monday, February 02, 2004 6:36 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Can audio streams go client to cleint with IAX? If you define possible as is H.323 capable of it, then yes. If you define possible as is asterisk currently capable of it, then no. It wouldn't be too hard, look in channels/h323/ast_h323.cpp to see where Jeremy started on it. You just have to get the openh323 lib to initiate the transfer. - Original Message - From: Marc Fargas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 10:30 AM Subject: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Is it possible to make audio streams go client to client with H.323 ? (both client being H323) Thanks! -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de T. Chan Enviado el: lunes, 02 de febrero de 2004 23:56 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Dear All, Now, it seems that both IAX and SIP can have the two endpoints communicate the media directly without the media stream passing through the asterisk, can we do the same with H323 too? TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Hart Sent: Monday, February 02, 2004 5:28 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Can audio streams go client to cleint with IAX? yes, IAX does direct transfers - when both ends confirm they can see each other, the asterisk server tells them to talk directly. With the firefly network, we're seeing 90%+ connecting directly. Just to clarify, the audio doesn't separate from the call control. -Adam - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 1:59 AM Subject: [Asterisk-Users] Can audio streams go client to cleint with IAX? With a service like http://www.freshtel.net/?show=home that uses IAX and has servers in Australia, is it possible for the audio streams to take a different path than the call setup and control? In other words can it work like SIP with canreinvite where the two endpoint negotiate audio streams between themselves rather than though the FreshTel server? 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 ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ 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
RE: [Asterisk-Users] Can audio streams go client to cleint with IAX?
FXS and FXO from http://micronet.info they have firmwares for SIP but I had problems when I tried it time ago. Continuing on the H.323 option I've being taking a look at my gatekeeper and it accepts a TransferCall source destination command on it's telnet interface. Is there anyway to make Asterisk connect to the telnet interface and Transfer the call throught it ? (the telnet interface doesn't require it to authenticate only to say TransferCall source e164 dest e164) I'll try upgrading SIP anyway I think ;) But I didn't succed last time (I think the problem was with the FXO itself not Asterisk) Also taking a look at the sourcecode of h323 channel there's a function about Forward call, any idea ? Thanks a lot for your help, Good Night! -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Adam Hart Enviado el: martes, 03 de febrero de 2004 1:33 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] Can audio streams go client to cleint with IAX? what clients are you using? They probably won't support SIP - Original Message - From: Marc Fargas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 11:18 AM Subject: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Thanks a lot I didn't see tose posts, so I'll have to migarte to SIP as I can see.. Do I need any other software or only Asterisk and my FXO's and FXS's firmwares migrated to SIP ? Thanks a lot. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Adam Hart Enviado el: martes, 03 de febrero de 2004 0:52 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] Can audio streams go client to cleint with IAX? I think it would be fair to say Jeremy isn't really a fan of H.323 and, although he will fix bugs, (IMO) he won't be developing it anymore. You should give it a shot. - Original Message - From: T. Chan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 10:50 AM Subject: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Would that be something that Jeremy would work on? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Hart Sent: Monday, February 02, 2004 6:36 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Can audio streams go client to cleint with IAX? If you define possible as is H.323 capable of it, then yes. If you define possible as is asterisk currently capable of it, then no. It wouldn't be too hard, look in channels/h323/ast_h323.cpp to see where Jeremy started on it. You just have to get the openh323 lib to initiate the transfer. - Original Message - From: Marc Fargas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 10:30 AM Subject: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Is it possible to make audio streams go client to client with H.323 ? (both client being H323) Thanks! -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de T. Chan Enviado el: lunes, 02 de febrero de 2004 23:56 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Dear All, Now, it seems that both IAX and SIP can have the two endpoints communicate the media directly without the media stream passing through the asterisk, can we do the same with H323 too? TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Hart Sent: Monday, February 02, 2004 5:28 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Can audio streams go client to cleint with IAX? yes, IAX does direct transfers - when both ends confirm they can see each other, the asterisk server tells them to talk directly. With the firefly network, we're seeing 90%+ connecting directly. Just to clarify, the audio doesn't separate from the call control. -Adam - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 1:59 AM Subject: [Asterisk-Users] Can audio streams go client to cleint with IAX? With a service like http://www.freshtel.net/?show=home that uses IAX and has servers in Australia, is it possible for the audio streams to take a different path than the call setup and control? In other words can it work like SIP with canreinvite where the two endpoint negotiate audio streams between themselves rather than though the FreshTel server? 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
Re: [Asterisk-Users] Can audio streams go client to cleint with IAX?
- Original Message - From: Adam Hart [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, February 02, 2004 5:27 PM Subject: Re: [Asterisk-Users] Can audio streams go client to cleint with IAX? yes, IAX does direct transfers - when both ends confirm they can see each other, the asterisk server tells them to talk directly. With the firefly network, we're seeing 90%+ connecting directly. Just to clarify, the audio doesn't separate from the call control. Thanks for the reply. Can you tell more about that last statement. If the audio doesn't serparate from the call control can the server keep track of how long the clients stay connected? Can it see DTMF that is sent between the clients and act upon it? Does IAX do this by default of do you have to set a parameter in IAX.conf. Thanks again ___ 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] Can audio streams go client to cleint with IAX?
yes, IAX does direct transfers - when both ends confirm they can see each other, the asterisk server tells them to talk directly. With the firefly network, we're seeing 90%+ connecting directly. Just to clarify, the audio doesn't separate from the call control. Thanks for the reply. Can you tell more about that last statement. If the audio doesn't serparate from the call control can the server keep track of how long the clients stay connected? Can it see DTMF that is sent between the clients and act upon it? Does IAX do this by default of do you have to set a parameter in IAX.conf. the server doesn't know, the call is completely between the clients from then on. Yep, IAX does it by default. ___ 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
