There should be no noticeable difference between slin, ulaw and alaw so what you have is fine. The problem must be elsewhere.
Vieri wrote: > --- On Thu, 5/13/10, Gareth Blades <list-aster...@skycomuk.com> wrote: > >> Show the details on the active >> channels when using both methods and >> check what codecs are being used. > > The audio codecs are different: > > Type: SIP > State: Up (6) > Rings: 0 > NativeFormats: 0x4 (ulaw) > WriteFormat: 0x40 (slin) > ReadFormat: 0x40 (slin) > WriteTranscode: Yes > ReadTranscode: Yes > > Type: IAX2 > State: Up (6) > Rings: 0 > NativeFormats: 0x8 (alaw) > WriteFormat: 0x8 (alaw) > ReadFormat: 0x8 (alaw) > WriteTranscode: No > ReadTranscode: No > > By the way, I have this in iax.conf: > > [interboxIAX2] > deny=all > allow=ulaw > allow=gsm > type=friend > host=192.168.250.111 > secret=mysecret > auth=plaintext > requirecalltoken=no > qualify=yes > context=mycontext > trunk=yes > username=interbox > > Shouldn't the channel details report ulaw instead of alaw? > > Also, if I change [interboxIAX2] and replace ulaw with alaw, the result is > the same (I still experience bad audio quality). > > Maybe I should try slin but how do I "force it"? > >> Vieri wrote: >>> Hi, >>> >>> I have an audio quality problem regarding IAX2. I have >> 2 Asterisk servers interconnected via 2 LAN trunks at 1Gbps >> (no nat, no firewall). >>> One trunk is SIP and the other IAX2. >>> Normally, I use IAX2 but have noticed easily >> reproducible audio quality problems (voice in/out is OK but >> there's a "third" noise overlapping with a "scratchy sound" >> as if it were some kind of interference). >>> So lately I setup calls to go through the SIP trunk >> and audio quality is OK (no "third overlapping noise"). >>> This is happening between Asterisk 1.4.31 and a >> 1.2.40. >>> I'm wondering if there's something I can tweak in IAX2 >> to eliminate this artifact. >>> Could the IAX2 jitter buffer between 1.2 and 1.4 be an >> issue (I believe it's enabled by default)? >>> Thanks, >>> >>> Vieri >>> >>> >>> >>> >>> >> >> -- >> _____________________________________________________________________ >> -- 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 >> > > > > -- _____________________________________________________________________ -- 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