When I make a call using sip, I get the line NOTICE[327696]: File rtp.c, Line 292 (ast_rtp_read): Unknown RTP codec 19 received Repeated many times on the console ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to ;bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls allow=gsm allow=ulaw allow=alaw [iconnect] type=friend username=******** password=**** host=sipauth.deltathree.com ;host=213.137.73.178 All I have been able to find about this topic is that 19 is supposed to be comfort noise (whatever that is) Any help is appreciated
What kind of UA are you using? ATA-186? Cisco 7960? If the former, set AudioMode: 0x00140014 and if the latter, set "enable_vad: "0" " Try that - it sets the Voice Auto Detect to "off". I don't know if that will solve the problem, since it seems to relate to Comfort Noise Generation, but my phones no longer produce buckets of codec errors with those settings. JT>When I make a call using sip, I get the line >NOTICE[327696]: File rtp.c, Line 292 (ast_rtp_read): Unknown RTP codec >19 received >Repeated many times on the console > >; SIP Configuration for Asterisk >; >[general] >port = 5060 ; Port to bind to >;bindaddr = 0.0.0.0 ; Address to bind to >context = outgoing ; Default for incoming calls >allow=gsm >allow=ulaw >allow=alaw > > >[iconnect] >type=friend >username=******** >password=**** >host=sipauth.deltathree.com >;host=213.137.73.178 > > > >All I have been able to find about this topic is that 19 is supposed to >be comfort noise (whatever that is) > >Any help is appreciated > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users
What is a UA? I am not using an ATA-186 or a Cisco 7960. The only Asterisk related hardware that I am using is TDM 400P and X100P. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of John Todd Sent: Wednesday, June 04, 2003 5:27 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] RTP codec error??? What kind of UA are you using? ATA-186? Cisco 7960? If the former, set AudioMode: 0x00140014 and if the latter, set "enable_vad: "0" " Try that - it sets the Voice Auto Detect to "off". I don't know if that will solve the problem, since it seems to relate to Comfort Noise Generation, but my phones no longer produce buckets of codec errors with those settings. JT>When I make a call using sip, I get the line >NOTICE[327696]: File rtp.c, Line 292 (ast_rtp_read): Unknown RTP codec >19 received >Repeated many times on the console > >; SIP Configuration for Asterisk >; >[general] >port = 5060 ; Port to bind to >;bindaddr = 0.0.0.0 ; Address to bind to >context = outgoing ; Default for incoming calls >allow=gsm >allow=ulaw >allow=alaw > > >[iconnect] >type=friend >username=******** >password=**** >host=sipauth.deltathree.com >;host=213.137.73.178 > > > >All I have been able to find about this topic is that 19 is supposed to >be comfort noise (whatever that is) > >Any help is appreciated > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
I updated the version of asterisk I was using and the problem seems to have been solved. Thanks for the help -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of John Todd Sent: Thursday, June 05, 2003 3:31 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] RTP codec error??? Sorry for my terminology assumptions. UA = User Agent, which is what the ATA-186 and 7960 are. Anything that normally is what the "end user" has on their desk or on their computer (in the case of a softphone) is considered a "UA". So, since you have both an ATA-186 and Cisco 7960, make the changes I describe below. If you don't understand what they are, take a look at the configuration guides for each piece of equipment, located on the Cisco website, or alternately use Google (which I find to be more useful at finding things than Cisco's terrible search interface.) JT>What is a UA? I am not using an ATA-186 or a Cisco 7960. The only >Asterisk related hardware that I am using is TDM 400P and X100P. > >-----Original Message----- >From: asterisk-users-admin@lists.digium.com >[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of John Todd >Sent: Wednesday, June 04, 2003 5:27 PM >To: asterisk-users@lists.digium.com >Subject: Re: [Asterisk-Users] RTP codec error??? > >What kind of UA are you using? ATA-186? Cisco 7960? If the former, >set AudioMode: 0x00140014 and if the latter, set "enable_vad: "0" " > >Try that - it sets the Voice Auto Detect to "off". I don't know if >that will solve the problem, since it seems to relate to Comfort >Noise Generation, but my phones no longer produce buckets of codec >errors with those settings. > >JT > > >>When I make a call using sip, I get the line >>NOTICE[327696]: File rtp.c, Line 292 (ast_rtp_read): Unknown RTP codec >>19 received >>Repeated many times on the console >> >>; SIP Configuration for Asterisk >>; >>[general] >>port = 5060 ; Port to bind to >>;bindaddr = 0.0.0.0 ; Address to bind to >>context = outgoing ; Default for incoming calls >>allow=gsm >>allow=ulaw >>allow=alaw >> >> >>[iconnect] >>type=friend >>username=******** >>password=**** >>host=sipauth.deltathree.com >>;host=213.137.73.178 >> >> >> >>All I have been able to find about this topic is that 19 is supposedto>>be comfort noise (whatever that is) >> >>Any help is appreciated >> >>_______________________________________________ >>Asterisk-Users mailing list >>Asterisk-Users@lists.digium.com >>http://lists.digium.com/mailman/listinfo/asterisk-users > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
Hi People I am wondering how in the sterisk world we provide the receptionist with the type of functionality she is used to from here traditional PABX. She can see who is on the phone, park a call until the person is free etc. She often has a panel with flashing extensions for incoming calls and lit extensions for those that are busy. Can this functionality be provided. Thanks in Advance Dave
Please search the archive and read the messages about threaded mail readers. Please search the archives, you are rehashing a thread that comes up from time to time. On Tue, 2003-07-15 at 22:55, Dave wrote:> Hi People > > I am wondering how in the sterisk world we provide the receptionist with the > type of functionality she is used to from here traditional PABX. > > She can see who is on the phone, park a call until the person is free etc. > > She often has a panel with flashing extensions for incoming calls and lit > extensions for those that are busy. > > Can this functionality be provided. > > Thanks in Advance > Dave > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users-- Steven Critchfield <critch@basesys.com>