Hi... Being very new to A* myself I understand your fustrations with the manuals :)
It looks like you've made a typo in your extensions.conf quote "[sip] extern = 66,1,Dial(SIP/66) extern = 61,1,Dial(SIP/61) extern = 60,1,Dial(SIP/60) " it should be exten = 66,1,Dial(SIP/66) Hope that helps Jnn ----- Original Message ----- From: "Thomas Sparr" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, March 04, 2004 1:46 PM Subject: [Asterisk-Users] 2 Linphones communicating through Asterisk? > Hi all, > > I'm experimenting with the following setup: > An Asterisk server at 192.168.0.10. > 2 Linphones at 192.168.0.60 and 192.168.0.66. > The Linphones register themselves at the Asterisk as sip:[EMAIL PROTECTED] > and sip:[EMAIL PROTECTED] > If my understanding is correct they should be available on the Asterisk > as sip:[EMAIL PROTECTED] and sip:[EMAIL PROTECTED] > However, if I try to call sip:[EMAIL PROTECTED] from the other Linphone > the Asterisk debug says: > > Looking for 66 in sip > Transmitting (no NAT): > SIP/2.0 404 Not Found > > Are there any merciful soul on this list who can point me in the rigth > direction? > If your answer are RTFM, please tell me which FM to R. > Asterisk sip debug follow below. > Also attaching config files for Asterisk and Linphone I have messed > with. All others are from make samples in asterisk. > versions: Linphone 0.12.1, libosip 0.9.7, asterisk 2 days old from cvs. > > Regards > > Thomas > > > Sip read: > REGISTER sip:192.168.0.10 SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK3085308593 > From: <sip:[EMAIL PROTECTED]>;tag=680816676 > To: <sip:[EMAIL PROTECTED]>;tag=680816676 > Call-ID: [EMAIL PROTECTED] > CSeq: 0 REGISTER > Contact: <sip:[EMAIL PROTECTED]> > max-forwards: 10 > expires: 3600 > user-agent: oSIP/Linphone-0.12.1 > Content-Length: 0 > > > 11 headers, 0 lines > Using latest request as basis request > Sending to 192.168.0.60 : 5060 (non-NAT) > Transmitting (no NAT): > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK3085308593 > From: <sip:[EMAIL PROTECTED]>;tag=680816676 > To: <sip:[EMAIL PROTECTED]>;tag=680816676 > Call-ID: [EMAIL PROTECTED] > CSeq: 0 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:[EMAIL PROTECTED]> > Content-Length: 0 > > > to 192.168.0.60:5060 > -- Registered SIP '60' at 192.168.0.60 port 5060 expires 3600 > Transmitting (no NAT): > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK3085308593 > From: <sip:[EMAIL PROTECTED]>;tag=680816676 > To: <sip:[EMAIL PROTECTED]>;tag=680816676 > Call-ID: [EMAIL PROTECTED] > CSeq: 0 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Expires: 3600 > Contact: <sip:[EMAIL PROTECTED]>;expires=3600 > Date: Thu, 04 Mar 2004 13:45:14 GMT > Content-Length: 0 > > Sip read: > INVITE sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK962844352 > From: <sip:[EMAIL PROTECTED]>;tag=4243659372;tag=4075507534 > To: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > CSeq: 20 INVITE > Contact: <sip:[EMAIL PROTECTED]> > max-forwards: 10 > user-agent: oSIP/Linphone-0.12.1 > Content-Type: application/sdp > Content-Length: 367 > > v=0 > o=60 123456 654321 IN IP4 192.168.0.60 > s=A conversation > c=IN IP4 192.168.0.60 > t=0 0 > m=audio 7078 RTP/AVP 0 8 3 110 111 115 101 > b=AS:110 20 > b=AS:111 28 > a=rtpmap:0 PCMU/8000/1 > a=rtpmap:8 PCMA/8000/1 > a=rtpmap:3 GSM/8000/1 > a=rtpmap:110 speex/8000/1 > a=rtpmap:111 speex/16000/1 > a=rtpmap:115 1015/8000/1 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-11 > > 11 headers, 16 lines > Using latest request as basis request > Sending to 192.168.0.60 : 5060 (non-NAT) > Found audio format UNKN > Found audio format ALAW > Found audio format UNKN > Found audio format UNKN > Found audio format UNKN > Found audio format UNKN > Found audio format UNKN > Found description format PCMU > Found description format PCMA > Found description format GSM > Found description format speex > Found description format speex > Found description format 1015 > Found description format telephone-event > Capabilities: us - 12, them - 526/0, combined - 12 > Non-codec capabilities: us - 1, them - 1, combined - 1 > Looking for 66 in sip > Transmitting (no NAT): > SIP/2.0 404 Not Found > Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK962844352 > From: <sip:[EMAIL PROTECTED]>;tag=4243659372;tag=4075507534 > To: <sip:[EMAIL PROTECTED]>;tag=as7e281fb9 > Call-ID: [EMAIL PROTECTED] > CSeq: 20 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:[EMAIL PROTECTED]> > Content-Length: 0 > > > to 192.168.0.60:5060 > > > Sip read: > ACK sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK962844352 > From: <sip:[EMAIL PROTECTED]>;tag=4243659372;tag=4075507534 > To: <sip:[EMAIL PROTECTED]>;tag=as7e281fb9 > Call-ID: [EMAIL PROTECTED] > CSeq: 20 ACK > Content-Length: 0 > > > > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
