When you start asterisk with -vvvvvvvvvvgc, asterisk will show a lot of messages on the CLI.
When something goes wrong, asterisk will most likely print something about it, we will try to explain the most common problems here.
When you see messages such as these:
Aug 24 18:04:24 WARNING[14287]: Maximum retries exceeded on call 0063031240@10.1.1.110 for seqno 2 (Non-critical Response)
Look if it starts with a NOTICE, WARNING or ERROR, then click on the corresponding link:
Notices are normally not so bad, but very often asterisk might start flooding your console with messages so bad that it will affect your I/O (if you are also logging to disk) or CPU. (especially with zaptel messages).
Warnings are very often problems that will make sure this one call does not go through, or has no audio.
Errors are critical, and the functionality of the complete asterisk is in danger.
For specific solutions for notice's, warnings or errors please click on the right link: (please note that there are thousands, so this is a work in progress, help us when you can!).
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Kevin Merck (kevin dot merck at es3inc dot com) 04 January 2008 23:11:49 What file would produce the following error? I am new to asterisk and the errors link in not working.
Jan 4 15:32:47 ERROR[2337]: chan_sip.c:11029 handle_request_subscribe: Got SUBSCRIBE for extension 102@es3 from 10.0.0.72, but there is no hint for that extension
thameur (ismailthameur at yahoo dot fr) 16 June 2007 23:02:11 WARNING[...] can not find own IP adress, SIP disabled.
please help me.
Pasquale (clark-able at hotmail dot com) 07 March 2007 11:04:46 -- Executing [23@maincall:1] Answer("SIP/24-081d2bd8", "") in new stack
-- Executing [23@maincall:2] Dial("SIP/24-081d2bd8", "SIP/23|5|r") in new stack
-- Called 23
-- SIP/23-081d7f90 is ringing
-- Nobody picked up in 5000 ms
-- Executing [23@maincall:3] GotoIfTime("SIP/24-081d2bd8", "13:00-16:10|*|*|*?6") in new stack
-- Executing [23@maincall:4] GotoIf("SIP/24-081d2bd8", "0?6") in new stack
-- Executing [23@maincall:5] GotoIf("SIP/24-081d2bd8", "0?7") in new stack
-- Executing [23@maincall:6] VoiceMail("SIP/24-081d2bd8", "u1515@default") in new stack
[Mar 7 11:01:09] WARNING[4166]: app_voicemail.c:6794 vm_exec: Prefixing the mailbox with an option is deprecated ('u1515@default').
[Mar 7 11:01:09] WARNING[4166]: app_voicemail.c:6795 vm_exec: Please move all leading options to the second argument.
-- <SIP/24-081d2bd8> Playing 'vm-theperson' (language 'en')
-- <SIP/24-081d2bd8> Playing 'digits/1' (language 'en')
-- <SIP/24-081d2bd8> Playing 'digits/5' (language 'en')
-- <SIP/24-081d2bd8> Playing 'digits/1' (language 'en')
-- <SIP/24-081d2bd8> Playing 'digits/5' (language 'en')
-- <SIP/24-081d2bd8> Playing 'vm-isunavail' (language 'en')
-- <SIP/24-081d2bd8> Playing 'vm-intro' (language 'en')
[Mar 7 11:01:20] WARNING[4166]: app_voicemail.c:2500 inboxcount: IMAP user not set for mailbox 1515
[Mar 7 11:01:20] NOTICE[4166]: app_voicemail.c:3002 leave_voicemail: Can not leave voicemail, unable to count messages
== Spawn extension (maincall, 23, 6) exited non-zero on 'SIP/24-081d2bd8'
Hallo,
I've got these problems!
Anyone know how to resolve all?
thanks a lot!
farhan altaf (farhanaltaf83 at yahoo dot com) 18 January 2007 19:24:29 I just want to implement a simple PBX on the LAN using asterisk & X-lite soft phones.i have installed latest asterisk on Red Hat AS 4.0.
On installation of asterisk there were no configuration files in the /etc/asterisk folder.
I have written sip.conf and extensions.conf but after running asterisk i found this
" WARNING[3197]: loader.c:721 load_modules: No 'modules.conf' found, no modules will be loaded."
and the Xlite clients are not being authenticated :( if ane1 can tell me y???
do i need ane other .conf file or i m doing some blunder..
farhan altaf (farhanaltaf83 at yahoo dot com) 18 January 2007 19:24:20 I just want to implement a simple PBX on the LAN using asterisk & X-lite soft phones.i have installed latest asterisk on Red Hat AS 4.0.
On installation of asterisk there were no configuration files in the /etc/asterisk folder.
I have written sip.conf and extensions.conf but after running asterisk i found this
" WARNING[3197]: loader.c:721 load_modules: No 'modules.conf' found, no modules will be loaded."
and the Xlite clients are not being authenticated :( if ane1 can tell me y???
do i need ane other .conf file or i m doing some blunder..
christian chambi (ccchristian at hotmail dot com) 20 April 2006 19:00:25 Asterisk Ready.
*CLI> Apr 20 13:03:49 NOTICE[27837]: chan_sip.c:11389 sip_poke_noanswer: Peer 'int2' is now UNREACHABLE! Last qualify: 0
*CLI>
Fred (fmyers at landair dot com) 28 December 2005 17:32:47 NOTICE[5139]: cdr.c:1185 do_reload: CDR simple logging enabled