re-posting the question.
-----------
use case:
when some one in my pbx calls 100.200, I have translations well defined, Media
also (media via asterisk) --Works.
when some one calls bob, or for any names I am adding Domain and call is been
sent to the other party -- Works, no media...
For the cases when it is talking to the external work,
I want Astersik not to do anything with the SDP/payload.
I want it to send as it is to the external proxy.
How can I achieve this? so that the SDP/payload will not be modified for users
talking to the external world.
I want media for those external devices to come Directly to the users in my
pbx. (with out going t asterisk)
2) also related question is can I have the xml payload in the originator and
call is routed via PBX to the Target.
The xml payload also must be carried to the target.
is it possible....
This will really help me as I was held up with this :(
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Aditya Kumar wrote:> re-posting the question. > ----------- > use case: > when some one in my pbx calls 100.200, I have translations well defined, > Media also (media via asterisk) --Works. > when some one calls bob, or for any names I am adding Domain and call is > been sent to the other party -- Works, no media... > > For the cases when it is talking to the external work, > I want Astersik not to do anything with the SDP/payload. > I want it to send as it is to the external proxy. > > How can I achieve this? so that the SDP/payload will not be modified for > users talking to the external world. > I want media for those external devices to come Directly to the users > in my pbx. (with out going t asterisk) > > 2) also related question is can I have the xml payload in the originator > and call is routed via PBX to the Target. > The xml payload also must be carried to the target. > is it possible.... > > This will really help me as I was held up with this :(Neither of these are possible; Asterisk is a B2BUA, not a proxy, and as such the outgoing INVITE is a *different* session from the incoming one. That means that Asterisk has to be able to understand the SDP content that arrives so it can forward media between the two sessions. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kfleming at digium.com Check us out at www.digium.com & www.asterisk.org
Thanks a lot Kevin for the reply ________________________________ From: Kevin P. Fleming <kpfleming at digium.com> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Sent: Thu, April 29, 2010 5:43:15 AM Subject: Re: [asterisk-users] No change in payload. (SDP) Aditya Kumar wrote:> re-posting the question. > ----------- > use case: > when some one in my pbx calls 100.200, I have translations well defined, > Media also (media via asterisk) --Works. > when some one calls bob, or for any names I am adding Domain and call is > been sent to the other party -- Works, no media... > > For the cases when it is talking to the external work, > I want Astersik not to do anything with the SDP/payload. > I want it to send as it is to the external proxy. > > How can I achieve this? so that the SDP/payload will not be modified for > users talking to the external world. > I want media for those external devices to come Directly to the users > in my pbx. (with out going t asterisk) > > 2) also related question is can I have the xml payload in the originator > and call is routed via PBX to the Target. > The xml payload also must be carried to the target. > is it possible.... > > This will really help me as I was held up with this :(Neither of these are possible; Asterisk is a B2BUA, not a proxy, and as such the outgoing INVITE is a *different* session from the incoming one. That means that Asterisk has to be able to understand the SDP content that arrives so it can forward media between the two sessions. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kfleming at digium.com Check us out at www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100430/1461025a/attachment.htm