You need to determine what codecs are expected (sip set debug on from CLI). Commenting out the disallow=all lets * use any available codecs, but may slow down the process or cause undesirable results by using/accounting for unneeded or unwanted codecs.
_____ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 3:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FAX Dear Sir, What do you mean by manual fax? I need to offer the ability for each extension to use voice and FAX...MAybe the voice will use G729 and the FAX ulaw for the same extension...If I configure the device in a manner that use ulaw for FAX and G729 for voice then this should work smoothly with an extension where G729,ulaw, alaw are allowed? Regards On Wed, Jan 28, 2009 at 11:17 PM, Danny Nicholas <da...@debsinc.com> wrote: The codecs should only be needed for a "manual" fax, where a voice interaction might be expected or anticipated. _____ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 3:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FAX Dear Sir, If I commant all codecs including disallow=all, then which codec should I define on the extensions from where I'm trying to send FAX? Regards On Wed, Jan 28, 2009 at 10:29 PM, Danny Nicholas <da...@debsinc.com> wrote: >From your sip.conf, you are only allowing ulaw and alaw codes. I'd try adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FAX Dear SIr, please find attached my sip.conf file Regards On Wed, Jan 28, 2009 at 5:37 PM, Danny Nicholas <da...@debsinc.com> wrote: Show us your sip.conf _____ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] FAX Hi all, When trying to send a FAX I got the following error: Executing [003228949...@micho:1] Dial("SIP/028949469-08466918", "SIP/003228949...@80.169.210.181|60") in new stack [Jan 28 17:19:20] WARNING[10040]: chan_sip.c:3039 sip_call: No audio format found to offer. Cancelling call to 003228949469 -- Couldn't call 0032234534...@1.1.1.1.1 Where I should define the codec other than the extension in order to succeed the call? Regards _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com <http://www.api-digital.com/> -- 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 <http://www.api-digital.com/> -- 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 <http://www.api-digital.com/> -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
