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
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
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=" 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
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
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
