-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Armand A. Verstappen Sent: Sunday, August 10, 2003 5:29 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Iconnecthere
Hi Andrew, On Sun, 2003-08-10 at 19:39, Andrew Joakimsen wrote: > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Armand A. > Verstappen > Sent: Sunday, August 10, 2003 4:57 AM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Iconnecthere > > On Sun, 2003-08-10 at 10:18, Andrew Joakimsen wrote: > > On Sun, 2003-08-10 at 09:02, Andrew Joakimsen wrote: > > > Does anyone have Asterisk working with Iconnect here for incoming > > and/or > > > outgoing calls? > > > > have a look at: > > > > > http://www.loligo.com/asterisk/example-configs.2003-04-24/extensions.con > > f > > > > there's a section in there dealing with Iconnect > > > That helps a lot. But now I get this message when I try to dial any > > number > > > > NOTICE[5126]: File pbx.c, Line 1089 (pbx_extension_helper): Cannot > find > > extension context 'default' > > You get this notice doing what? dialing in, dialing out? At any rate, it > looks like you've halfway implemented the examples, sip.conf having > context=default, but no context [default] in extensions.conf. > > When I try to dial out, but there is a [default] section in my > extensions.conf Hmm... okay. We're going to need a little more context here. What kind of device/software are you calling from? sip / h323 / zap / quicknet / ... ? What's the related config setup like (so sip.conf h323.conf/oh323.conf zap.conf phone.conf ...), and what are your extensions set up like (extensions.conf). wkr I seem to have my configuration working except for outgoing and incoming calls for the rest of the world. For now I am concerned more about outgoing calls than anything else. Whenever I try to make an outgoing call I get these messages from the sip debug in the console s=session c=IN IP4 64.36.104.203 t=0 0 m=audio 6620 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 (no NAT) to 4.42.235.170:5060 -- Called [EMAIL PROTECTED] Sip read: SIP/2.0 404 Not Found 4 Via: SIP/2.0/UDP 64.36.104.203:5060;branch=z9hG4bK37d8c90a From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as220b2c68 To: <sip:[EMAIL PROTECTED]>;tag=1m6lkhivci11cjdooja30ex45 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Content-Length: 0 Notice in particular the From line. Now notice a working session from eStara softphone: v=0 o=eStara 22079953 22079953 IN IP4 64.36.104.202 s=eStara c=IN IP4 64.36.104.202 t=0 0 m=audio 8014 RTP/AVP 0 4 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.36.104.202:5060 From: Anonymous <sip:[EMAIL PROTECTED]>;tag=1d436a9 To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 22079953 INVITE Content-Length: 0 Notice how the from is different, my SIP service will not accept calls unless the proper from name is configured, how can I configure this? Here are the relevant sections from my sip.conf file [general] port = 5060 ; Port to bind to context = from-sip ; Default for incoming calls maxexpirey=13600 ; Max length of incoming registration we allow defaultexpirey=3600 ; Default length of incoming/outoing registration register => 17862324057:[EMAIL PROTECTED]/5500 [packet8.net] type=friend username=17862324057 secret=xxxxxxxxxxx host=packet8.net context=demo Thanks again for all the help you have provided me with. _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
