I'm getting now the below notice: rtp.c: Unknown RTP codec 100 received from 'GW address'
On Thu, Jan 29, 2009 at 9:18 PM, michel freiha <[email protected]> wrote: > Do you mean call limit on the extension or on the outgoing gateway? Kindly > note that my outbound dialpeer has meeb defined as follow: > > [outbound] > exten => _X.,1,Dial(SIP/${ext...@outbound_gw,60) > Regards > > > On Thu, Jan 29, 2009 at 8:58 PM, Danny Nicholas <[email protected]> wrote: > >> Doesn't matter – the call-limit is important because 1 call can actually >> be 2-N hops. >> >> >> ------------------------------ >> >> *From:* [email protected] [mailto: >> [email protected]] *On Behalf Of *michel freiha >> *Sent:* Thursday, January 29, 2009 12:45 PM >> >> *To:* Asterisk Users Mailing List - Non-Commercial Discussion >> *Subject:* Re: [asterisk-users] FAX >> >> >> >> Dear Danny, >> >> >> >> This is the only call on asterisk...:) >> >> >> >> Regards >> >> On Thu, Jan 29, 2009 at 8:35 PM, Danny Nicholas <[email protected]> >> wrote: >> >> Try increasing (or adding) call-limit on sip.conf. >> >> >> ------------------------------ >> >> *From:* [email protected] [mailto: >> [email protected]] *On Behalf Of *michel freiha >> *Sent:* Thursday, January 29, 2009 12:27 PM >> >> >> *To:* Asterisk Users Mailing List - Non-Commercial Discussion >> *Subject:* Re: [asterisk-users] FAX >> >> >> >> Dear Sir, >> >> >> >> When trying to send a FAX with T.38I got the following error message >> >> >> >> >> [Jan 29 20:16:22] WARNING[19890] chan_sip.c: Maximum retries exceeded on >> transmission [email protected] for seqno 102 (Critical >> Response) -- See doc/sip-retransmit.txt. >> [Jan 29 20:16:22] WARNING[19890] chan_sip.c: Hanging up call >> [email protected] - no reply to our critical packet (see >> doc/sip-retransmit.txt). >> >> >> >> >> >> Regards >> >> On Thu, Jan 29, 2009 at 12:04 AM, michel freiha <[email protected]> >> wrote: >> >> Dear Danny, >> >> >> >> Thanks a lot for the help...I'll try and let you know >> >> >> >> Regards >> >> On Wed, Jan 28, 2009 at 11:56 PM, Danny Nicholas <[email protected]> >> wrote: >> >> 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:* [email protected] [mailto: >> [email protected]] *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 <[email protected]> >> wrote: >> >> The codecs should only be needed for a "manual" fax, where a voice >> interaction might be expected or anticipated. >> >> >> ------------------------------ >> >> *From:* [email protected] [mailto: >> [email protected]] *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 <[email protected]> >> 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:* [email protected] [mailto: >> [email protected]] *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 <[email protected]> >> wrote: >> >> Show us your sip.conf >> >> >> ------------------------------ >> >> *From:* [email protected] [mailto: >> [email protected]] *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/ >> [email protected]|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 [email protected] >> >> 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 -- >> >> 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 >> >> >> >> >> _______________________________________________ >> -- 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 >> >> >> >> >> _______________________________________________ >> -- 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 >> >> >> >> >> >> >> _______________________________________________ >> -- 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 >> >> >> >> _______________________________________________ >> -- 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 >> > >
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