I have this problem too. If i call out (not only with capi) the codec of my choice is used (ALAW for internal phone another for outgoing). For an incomming call alaw is used, even if i disable alaw in globals.
So my bandwith is highly consumed and i can not do anything. nico Lars Boegild Thomsen wrote: > Actually I've played around with the last issue quite a lot and this is > indeed getting weirder. > > Let me try to describe the problem. > > sip.conf is configured with: > > disallow = all > allow = gsm > allow = ulaw > allow = alaw > > Snom phone is configured to use GSM as default codec but with "Offer > Answer/Full" option set. > > If I place a call FROM the Snom phone to an external number (going out of > the CAPI/Fritz/ISDN interface) everything works beautifully - and "sip > show channels" show that the Snom phone is using GSM. > > If a call come IN on the Capi interface and is routed to the phone there > is the described pulsating sound heard on the Snom phone alone and "sip > show > channels" report that ALAW is being used as codec. How come the choice of > codec is different? AFAIK when gsm is first in sip.conf this should be > the preferred codec. > > I haven't tried to roll back to an earlier Snom image (using 2.05d) but > this > problem is definitely a new one. Using an Asterisk CVS-HEAD as of today. > > So - I am not sure exactly where this bug is. As far as I can see there > might be two problems - one that the codec of my choice is not the one > being > used. Second the pulsating noice when using ALAW (which should work fine > too). > > Any ideas? > > Regards, > > Lars.... > >> -----Original Message----- >> From: [EMAIL PROTECTED] >> [mailto:[EMAIL PROTECTED] Behalf Of Lars Boegild >> Thomsen >> Sent: 18 May 2004 12:00 >> To: [EMAIL PROTECTED] Digium. Com >> Subject: [Asterisk-Users] Problems w. chan_capi + ztdummy >> >> >> Hi Everybody >> >> I've got a weird problem. I am running one Asterisk system on a dual >> processor box. This box mostly do VoIP only but it has a Fritz PCI ISDN >> card installed with latest drivers. Dialing out through the ISDN >> cards from >> an internal Snom phone works fine and so does dialing in. Except - if I >> load the ztdummy module (for IAX channels) the capi drivers starts acting >> up. It is hard to describe the sound but it breaks up so badly that it >> is impossible to understand the voice prompts and they also start playing >> extremely slow (demo congrats alone takes more than 30 seconds >> before going >> to the next prompt in the standard demo setup). >> >> I am nearly updating this particular box every day and within the last >> couple of days something else has happened. When dialing OUT on the ISDN >> card everything works fine. When someone dial IN through the card and >> connect to the internal Snom phone there is a pulsating background noice >> that can only be heard on the VoIP phone. From outside (the ISDN) things >> sound perfect - from inside you can still hear what is being said - but >> there is that pulsating quite high noice. >> >> Any ideas? >> >> Regards, >> >> Lars... >> >> -- >> Lars Boegild Thomsen >> Technical Director >> JustIT Sdn. Bhd. >> Cell Phone (MY): +60 (16) 323 1999 >> ICQ: 6478559 >> Yahoo Chat: [EMAIL PROTECTED] >> MSN Chat: [EMAIL PROTECTED] >> http://www.justit.ws >> Phone: +1 (360) 515 3551 (US) +45 8692 1951 (DK) +60 (3) 2057 2646 (MY) >> Fax : +60 (3) 2057 2647 (MY) >> >> _______________________________________________ >> 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
