I hope someone can shed some insight on this.
I have some cisco phones, 7970 and 7940 models. After wrestling with them for a while, i got the SIP firmware on there, I got the config files TFTPed in, and I got them registering with asterisk.
All the 7940s work 100% fine.
So i bought a 7970. Set it up, firmware, config, worked 100% fine… for a week or two. Then it stopped receiving calls. I can MAKE calls all i want, the lines show as registered on the display and dont show red Xs or anything. All appears to be normal with the phone. if you call it, however, from internal or external, it goes straight to voice mail.
So i bought another 7970 thinking maybe this one was damaged somehow. Installed same everything, same problem.
if i am logged into the asterisk console, i see this when i try to call the 7970 phone
[Jun 23 05:29:17] WARNING: app_dial.c:1183 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination)
[Jun 23 05:29:17] WARNING: ast_expr2.fl:407 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected ‘=’, expecting $end; Input:
[Jun 23 05:29:17] WARNING: ast_expr2.fl:411 ast_yyerror: If you have questions, please refer to doc/channelvariables.txt in the asterisk source.
[Jun 23 05:29:22] NOTICE: chan_sip.c:15988 sip_poke_noanswer: Peer->call ‘511’ is true
[Jun 23 05:29:22] NOTICE: chan_sip.c:15988 sip_poke_noanswer: Peer->call ‘505’ is true
[Jun 23 05:29:22] NOTICE: chan_sip.c:15988 sip_poke_noanswer: Peer->call ‘504’ is true
Note that 504 and 505 are registered on the 7970
note that 511 is registered to an experimental blackberry sip client, it can call out, but not receive calls either. Im not too worried about that right now.
I have googled and looked in the channelvariables.txt file but no luck
Im sure some of you are going to ask to see config files, please let me know which parts are important for me to post.
Thanks in advance.