Thanks for the response. I have disallow=all and allow=alaw in sip.conf for my SIP user. Any other idea? --AM
On Tue, Jun 28, 2011 at 4:23 PM, Fellipe Paes <[email protected]>wrote: > Hello! > > In your sip.conf use alaw as your first codec option and see what happens. > Best regards, > > Fellipe Paes > > ------------------------------ > Date: Tue, 28 Jun 2011 15:29:11 +0530 > From: [email protected] > To: [email protected] > Subject: [asterisk-users] Asked to transmit frame type slin, while native > formats is 0x8 (alaw) > > > > Asterisk 1.8.3.2 > > I have been getting this warning constantly on CLI in a call scenario where > I use local channels to connect SIP with PSTN. > I use callfile and local channel to first call a PSTN number and if > answered, use local channel to call SIP phone with music on hold enabled in > Dial string. > If I call PSTN from SIP directly or vice versa I don't see this warning > coming. > On SIP I have allowed only one codec(alaw). > > [Jun 28 15:05:00] WARNING[31016] chan_sip.c: Asked to transmit frame type > slin, while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw) > > > I also tried to yes/no option transcode_via_sln in asterisk.conf without > any success. > Any idea? > Thanks, > --AM > > -- _____________________________________________________________________ -- > 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 >
-- _____________________________________________________________________ -- 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
