Got too cute. Make AST_FORMAT_H100 be AST_FORMAT_H263.
_____ From: [email protected] [mailto:[email protected]] On Behalf Of michel freiha Sent: Thursday, January 29, 2009 2:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FAX Dear Danny, i got the following error during make [CC] rtp.c -> rtp.o rtp.c:1390:3: error: invalid preprocessing directive #[ rtp.c:1391: error: ‘AST_FORMAT_H100’ undeclared here (not in a function) rtp.c:1392: error: expected ‘}’ before ‘[’ token make[1]: *** [rtp.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-1.4.22.1/main' make: *** [main] Error 2 regards On Thu, Jan 29, 2009 at 9:48 PM, Danny Nicholas <[email protected]> wrote: Good new is "your'e getting somewhere". Bad new is – you have to modify rtp.c to allow this codec. You should be able to duplicate a line (around 1390) and change the value from [34] = {1, AST_FORMAT_H263}, To [100] = {1, AST_FORMAT_H100}, Then just do a make && make install on asterisk again. _____ From: [email protected] [mailto:[email protected]] On Behalf Of michel freiha Sent: Thursday, January 29, 2009 1:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FAX 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 <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 <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 _______________________________________________ -- 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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