RE: [Asterisk-Users] context being ignored by inbound sip call
change context to context=remote in [general] in sip.conf you missing registration of peer :) turby -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of btb Sent: Thursday, February 23, 2006 4:10 AM To: Asterisk Non-Commercial Discussion Users Mailing List - Subject: [Asterisk-Users] context being ignored by inbound sip call hello- i was messing around with a did from ipkall.com, and asterisk seems to be ignoring the context specified in the sip config. in sip.conf, i've added: [7508] ;ipkall type = peer dtmfmode = rfc2833 context = remote callerid = ipkall incoming 7508 nat = no in extensions,conf, i have: [remote] exten = 7508,1,DISA(|internal) [internal] exten = 81,1,Dial(SIP/ion,20,tr) exten = 82,1,Dial(SCCP/82,20,tr) exten = 83,1,Dial(SIP/quark,20,tr) exten = 84,1,Dial(SIP/proton,20,tr) exten = 85,1,Dial(SIP/work1,20,tr) exten = 86,1,Dial(IAX2/work2,20,tr) yet when the call arrives, asterisk says: NOTICE[8100]: pbx.c:1731 pbx_extension_helper: Cannot find extension context 'default' what am i missing? thanks -ben ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] context being ignored by inbound sip call
Johnathan Corgan wrote: btb wrote: [7508] ;ipkall type = peer dtmfmode = rfc2833 context = remote callerid = ipkall incoming 7508 nat = no You've configured this entry as a peer, which is for dialing out, versus as a user, which is for incoming calls. Solution is to change to 'type=user'. If you really need a peer definition, you can use 'type=friend', which will cause * to create both a user and a peer entry for '7508' using the parameters listed. Some parameters are common to both peers and users so it saves space. Personally, I never use the 'type=friend' method, but rather maintain separate peer and user sections for outbound and inbound calls to/from other switches or endpoints. This helps _me_ keep things straight; others (probably most) prefer the combined 'type=friend' method, though. I would never recommend using a type=friend for a service provider connection. You need one peer for calling out and another for receiving calls, or at least add a host=hostname of provider's server to enable matching on IP on incoming calls. The problem here is, as you figured out Jonathan, that this peer section does not match the incoming call. Adding a host=hostname entry will help matching. /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] context being ignored by inbound sip call
Johnathan Corgan wrote: btb wrote: [7508] ;ipkall type = peer dtmfmode = rfc2833 context = remote callerid = ipkall incoming 7508 nat = no You've configured this entry as a peer, which is for dialing out, versus as a user, which is for incoming calls. Solution is to change to 'type=user'. If you really need a peer definition, you can use 'type=friend', which will cause * to create both a user and a peer entry for '7508' using the parameters listed. Some parameters are common to both peers and users so it saves space. Personally, I never use the 'type=friend' method, but rather maintain separate peer and user sections for outbound and inbound calls to/from other switches or endpoints. This helps _me_ keep things straight; others (probably most) prefer the combined 'type=friend' method, though. thanks jonathan- i originally had this entry as type=user, and switched to type=peer after finding the context was being ignored and reading that type=user may/is be(ing) phased out: http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer i've tried type=user again (as well as type=peer), with some additional parameters (mostly guesses, because i don't yet fully understand registration): [7508] ;ipkall type = peer host = dynamic dtmfmode = rfc2833 context = remote callerid = ipkall incoming 7508 nat = no insecure = very i gather the ideal method is to know the source ip and source port of the connection from my peer, and include that in the sip config? how can i make asterisk tell me where a connection is coming from? -ben ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] context being ignored by inbound sip call
On Feb 23, 2006, at 10.43, btb wrote: Johnathan Corgan wrote: btb wrote: [7508] ;ipkall type = peer dtmfmode = rfc2833 context = remote callerid = ipkall incoming 7508 nat = no You've configured this entry as a peer, which is for dialing out, versus as a user, which is for incoming calls. Solution is to change to 'type=user'. If you really need a peer definition, you can use 'type=friend', which will cause * to create both a user and a peer entry for '7508' using the parameters listed. Some parameters are common to both peers and users so it saves space. Personally, I never use the 'type=friend' method, but rather maintain separate peer and user sections for outbound and inbound calls to/ from other switches or endpoints. This helps _me_ keep things straight; others (probably most) prefer the combined 'type=friend' method, though. thanks jonathan- i originally had this entry as type=user, and switched to type=peer after finding the context was being ignored and reading that type=user may/is be(ing) phased out: http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer i've tried type=user again (as well as type=peer), with some additional parameters (mostly guesses, because i don't yet fully understand registration): [7508] ;ipkall type = peer host = dynamic dtmfmode = rfc2833 context = remote callerid = ipkall incoming 7508 nat = no insecure = very i gather the ideal method is to know the source ip and source port of the connection from my peer, and include that in the sip config? how can i make asterisk tell me where a connection is coming from? so, in answer to my own question, this ended up being what i needed in sip.conf: [ipkall] type = peer host = voiper.ipkall.com dtmfmode = rfc2833 context = remote callerid = ipkall incoming nat = no the key was the host parameter. as soon as i added that, matching occurred and the context was honored. thanks -ben ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] context being ignored by inbound sip call
hello- i was messing around with a did from ipkall.com, and asterisk seems to be ignoring the context specified in the sip config. in sip.conf, i've added: [7508] ;ipkall type = peer dtmfmode = rfc2833 context = remote callerid = ipkall incoming 7508 nat = no in extensions,conf, i have: [remote] exten = 7508,1,DISA(|internal) [internal] exten = 81,1,Dial(SIP/ion,20,tr) exten = 82,1,Dial(SCCP/82,20,tr) exten = 83,1,Dial(SIP/quark,20,tr) exten = 84,1,Dial(SIP/proton,20,tr) exten = 85,1,Dial(SIP/work1,20,tr) exten = 86,1,Dial(IAX2/work2,20,tr) yet when the call arrives, asterisk says: NOTICE[8100]: pbx.c:1731 pbx_extension_helper: Cannot find extension context 'default' what am i missing? thanks -ben ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] context being ignored by inbound sip call
btb wrote: [7508] ;ipkall type = peer dtmfmode = rfc2833 context = remote callerid = ipkall incoming 7508 nat = no You've configured this entry as a peer, which is for dialing out, versus as a user, which is for incoming calls. Solution is to change to 'type=user'. If you really need a peer definition, you can use 'type=friend', which will cause * to create both a user and a peer entry for '7508' using the parameters listed. Some parameters are common to both peers and users so it saves space. Personally, I never use the 'type=friend' method, but rather maintain separate peer and user sections for outbound and inbound calls to/from other switches or endpoints. This helps _me_ keep things straight; others (probably most) prefer the combined 'type=friend' method, though. -Johnathan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
