-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Well, a bit of a newbie using Asterisk and trying to get some Sipura SPA-841 phones to talk to the Asterisk server. Not doing something right apparently.... If I turn on debug for the extension in the CLI (sip debug ip xxx.xxx.xxx.xxx) I see frequent 5 packet attempts by the server to contact the phone, but seems to always be failing. The status for the extension in the phone shows either Disabled (or on another extension Failed). The sip.conf entry for example is: [302] type=friend username=302 secret=mysecret mailbox=302 context=ext-local host=10.0.0.2 dtmfmode=rfc2833 canreinvite=yes qualify=no and I've tried a number of permutations on that setup and some 'examples' of SIP configs I've found on the net. Same results: Retransmitting #1 (no NAT): NOTIFY sip:302@10.0.0.2 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK2b963e11 From: "Unknown" <sip:Unknown@10.0.0.1>;tag=as149adc4e To: <sip:302@10.0.0.2> Contact: <sip:Unknown@10.0.0.1> Call-ID: 185e61bf1fc8d52012e4234e07fa0fbc@10.0.0.1 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 43 Messages-Waiting: yes Voice-Message: 1/0 to 10.0.0.2:5060 Checked pinging of the IP from that machine and all is fine and all traffic to/from that IP space is opened on the server. Any ideas or suggestions/examples of configs? Chris -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFCCqq7G4PxJjbMvv0RApm4AJ94ZvivrgJEEH3HILw3wMY0m36DJACgktFV 5UTErcJULAqNuOjfE4zCFZw=r8SO -----END PGP SIGNATURE-----
Giovanni Powell
2005-Feb-10 07:07 UTC
[Asterisk-Users] Asterisk and Sipura SPA-841 SIP phones
Nothing to do with your question, but by any chance, when you plugged the phone into the wall did you hear a dialtone or is this something generated by asterisk
Nicolas Bougues
2005-Feb-10 08:00 UTC
[Asterisk-Users] Asterisk and Sipura SPA-841 SIP phones
On Thu, Feb 10, 2005 at 09:07:58AM -0500, Giovanni Powell wrote:> Nothing to do with your question, but by any chance, when you plugged > the phone into the wall did you hear a dialtone or is this something > generated by asteriskOn a SIP phone, the dial tone is locally generated. The Sipura will only generate a dial tone if registrered. BTW, you can easily check on the Sipura web interface that the dial tones are parametered there. -- Nicolas Bougues Axialys Interactive