Hi all, I have an interesting problem with Asterisk 1.4.11 - 3 SIP phones:
[phone1] allow=g722 allow=alaw .... [phone2] allow=alaw allow=g722 .... [phone3] allow=alaw Now, when I try to call: 1. phone1 calling phone2, I got through, using G.722 codec 2. phone2 calling phone1, I get through, using Alaw 3. phone3 calling phone1 or phone2, OK using Alaw But: 4. phone1 calling phone3 fails: [Sep 20 10:14:32] WARNING[30706]: chan_sip.c:2963 sip_call: No audio format found to offer. Cancelling call to phone3 Any ideas what could be wrong? Many thanks for any suggestion.... Ondrej The information contained in this e-mail and in any attachments is confidential and is designated solely for the attention of the intended recipient(s). If you are not an intended recipient, you must not use, disclose, copy, distribute or retain this e-mail or any part thereof. If you have received this e-mail in error, please notify the sender by return e-mail and delete all copies of this e-mail from your computer system(s). Please direct any additional queries to: [EMAIL PROTECTED] Thank You. _______________________________________________ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
