trixter http://www.0xdecafbad.com
2005-Sep-25 23:22 UTC
[Asterisk-Users] change codec based on callerid (sip/iax)
I have been asked if asterisk can change codecs dynamically based on the calling party's caller id. I couldnt find anything, and dont know that this is something that asterisk can do, but it occurs to me that possibly with a reinvite it can be done, however I dont think you can issue those from the dialplan or agi. The only solution I can think of on this is to use something like ser (www.iptel.org/ser) in between the asterisk box and forward effectivly to a different account on the asterisk box based on caller id (ie ser makes a choice which account to use). codecs then would be negotiated normally at connect time. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050926/97181602/attachment.pgp
Michael D Schelin
2005-Sep-26 10:33 UTC
[Asterisk-Users] change codec based on callerid (sip/iax)
This can be done by modifying the source code. trixter http://www.0xdecafbad.com wrote:>I have been asked if asterisk can change codecs dynamically based on the >calling party's caller id. I couldnt find anything, and dont know that >this is something that asterisk can do, but it occurs to me that >possibly with a reinvite it can be done, however I dont think you can >issue those from the dialplan or agi. > >The only solution I can think of on this is to use something like ser >(www.iptel.org/ser) in between the asterisk box and forward effectivly >to a different account on the asterisk box based on caller id (ie ser >makes a choice which account to use). codecs then would be negotiated >normally at connect time. > > > > >------------------------------------------------------------------------ > >_______________________________________________ >--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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050926/e58a7e98/attachment.htm
trixter http://www.0xdecafbad.com
2005-Sep-26 12:50 UTC
[Asterisk-Users] change codec based on callerid (sip/iax)
On Mon, 2005-09-26 at 10:33 -0700, Michael D Schelin wrote:> This can be done by modifying the source code. >how helpful. If I modify it enough it will be 100% identical to windows xp, anything can be done by modifying any code. That however doesnt answer my question with anything that isnt obvious, such as is there a way to do it without a modification? Would the ser idea work (which may be better as the call volume would likely exceed asterisks ability to process calls anyway? -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050926/8252c67d/attachment.pgp