i install freepbx 2.6 asterisk 1.6 on centos 5.4,and add the sip extension, when i try to call the sip phone to sip phone, its declined please help me
Check that the phones are registered.
If they are, check why the calls are declined by looking in /var/log/asterisk/full, or by running asterisk -r (make sure to enable SIP debug).
This is my Full file info
Note from one of the Forum Admins: Incredibly long log removed for clarity
Guide me what the mistake i have done,
I have removed the log as it was incredibly long and as SkykingOH pointed out, did not contain any useful information.
my sip phones are registered properly with my pbx, i have check the phone status page its registered with my pbx, phone registration led also green status, still i cant call one sip ext to another.
when i press *60 its calling 192.168.1.60 ip address
Ok I will bite, what is 192.168.1.60 ?
Really need the whole picture to help you.
Why don’t you post the output of “sip show peers” for us. A small fragment of the log when you place a call with “core set verbose 10” , “core set debug 10” and “sip set debug” all on. Don’t forget to turn those commands off or your logs will be huge.
Post the output using the
[code][/code] tags so we can read it.
see my details
sip show peers
Name/username Host Dyn Nat ACL port Status 200/200 192.168.1.170 D N 5060 ok (86 ms) 201/201 192.168.1.169 D N 5060 ok (86 ms) 2 sip peers [Monitored : 2 online, 0 offline unmonitored:0 online, 0 offline]
core set verbose
verbosity is at least 10
core set debug
Core debug was 0 and is now 10
sip set debug
No such command 'sip set debug' (type 'help sip set debug' for other possible commands)
What about the log fragment (thanks for using the code tags) ???
log Fragment ?
Yes, a log fragment. Either captured from the console or cut from the log file.
With no other traffic on the server originate a call to the failing extension. Only post the log, with the code tags, for the time period of the call attempt.
call attempt log
[Dec 21 15:05:30] DEBUG chan_sip.c: AST hangup cause 16 (no match found in SIP) [Dec 21 15:05:30] DEBUG app_queue.c: Device 'SIP/201' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Dec 21 15:05:30] DEBUG chan_sip.c: Trying to put 'SIP/2.0 603' onto UDP socket destined for 192.168.1.169:5060 [Dec 21 15:05:30] DEBUG devicestate.c: No provider found, checking channel drivers for SIP - 201 [Dec 21 15:05:30] DEBUG chan_sip.c: Checking device state for peer 201 [Dec 21 15:05:30] DEBUG devicestate.c: Changing state for SIP/201 - state 1 (Not in use) [Dec 21 15:05:30] DEBUG devicestate.c: device 'SIP/201' state '1' [Dec 21 15:05:30] DEBUG app_queue.c: Device 'SIP/201' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Dec 21 15:05:30] DEBUG chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Dec 21 15:05:30] DEBUG chan_sip.c: Stopping retransmission on '[email protected]' of Response 2959: Match Found [Dec 21 15:05:37] DEBUG chan_sip.c: Auto destroying SIP dialog '[email protected]' [Dec 21 15:05:37] DEBUG chan_sip.c: Destroying SIP dialog [email protected] [Dec 21 15:05:37] DEBUG chan_sip.c: Invalid SIP message - rejected , no callid, len 336 [Dec 21 15:05:38] DEBUG chan_sip.c: Invalid SIP message - rejected , no callid, len 336
That is a bit too little log, and you don’t have SIP debug on.
The invalid SIP messages is odd. What kind of phones are they?
I did not know it was possible to include that much log and not include the relevant portion of the log.
You have to participate in the trouble resolution process. Posting an unsanitized log to the forum is not the way to get help. If you do feel the need to post long logs use pastebib.ca and then simply leave a link.
You also need to answer the questions asked of you. Are the phones registered?
Can you reach the Asterisk server (try *60 for time).