[Asterisk-Users] Sip and autonegotiating codecs

[Tod Detre]

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I've got a problem with asterisk and some sip clients auto negotiating
codecs. I have a bunch of soft phones (xlite) that support
gsm,speex,alaw,ulaw, and ilbc. I have a quintum analog to sip gateway
that only has g729,ulaw,alaw. This isn't a problem if I have all of
the client using ulaw by default (deny all, allow ulaw;speex;gsm;etc).
But if I put speex first (deny all, allow speex;ulaw;etc) I can make
calls from soft phone to soft phone, but if I try to call the quintum
box, I get a "couldn't translate between speex and ulaw".

Is it possible to have asterisk/xlite talk speex if xlite is talk to
another xlite or to asterisk itself, but if it calls the quintum box
xlite will use ulaw?


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Regards,
Tod Detre
Technical Lead
Global Information Technology
CampusEAI Consortium
1940 East 6th Street, 11th Floor
Cleveland, OH 44114
Tel:  216.589.9626
Fax:  216.589.9639 www.campuseai.org

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