Asterisk Man
2011-Jun-28 09:59 UTC
[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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110628/d741ba6e/attachment.htm>
Fellipe Paes
2011-Jun-28 10:53 UTC
[asterisk-users] Asked to transmit frame type slin, while native formats is 0x8 (alaw)
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: theasteriskman at gmail.com To: asterisk-users at lists.digium.com 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110628/51b80145/attachment.htm>