Not sure whether this is a problem with FireFly, with Asterisk, with both or just with me ;-)
I have: disallow=all allow=alaw in the general section of my sip.conf.
Using Ethereal on the PC running FireFly, I get the following results.
GS = grandstream BT100 initiating SIP call, FF = firefly receiving call
* -> FF: invite from GS to FF (PCMA) FF -> *: OK (PCMA) * -> FF: reinvite to GS IP (PCMA, G723, PCMU, G726-32, G729, iLBC) FF -> *: OK (iLBC) GS -> FF: RTP stream (unanswered by FF) [...RTP monologue continues...] [...hangup on the GS...] GS -> FF: RTCP goodbye * -> FF: reinvite to * ip (PCMA) FF -> *: OK (PCMA) * -> FF: BYE FF -> *: OK
Meanwhile on the *-console:
-- Attempting native bridge of SIP/grandstream-0f65 and SIP/willem-2cbe
Oct 18 15:15:39 WARNING[98311]: chan_sip.c:2804 process_sdp: No compatible codecs!
Oct 18 15:15:39 WARNING[98311]: chan_sip.c:2804 process_sdp: No compatible codecs!
Meanwhile on the FF PC with DebugView running: [368] Wrong input size for iLBC - requires 30ms frames [368] Stopping transmission due to send error [368] Finished reading
Apparently FF does not accept 20ms iLBC frames (which is the default on the GS phones).
So my questions:
* Why is * advertising codecs in the reinvite request which
shouldn't be used according to sip.conf? It advertises fine in the
initial invite.
* Why does FF answer "OK (iLBC)" upon the reinvite request, even
though I have turned off the iLBC codec in FF's configuration?
Probably a bug with FF?
* Why does * croak "no compatible codecs" when in fact both sides
have pcma/alaw enabled and even advertise them?Thanks!
Willem
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