Hi All,
I am running SAIL 3.1.1-9 on SME 8.0 with 9 SIP trunks and for as long as I can remember everything has worked great until this afternoon
I tried to call out and the number I called went straight to a busy tone, so I tried again and again, still the same
So I then tried a different number and got the same result
So I tried to call in on my mobile and it worked OK. Went into SAIL admin panel and all my trunks are showing as not connected (big red X) but showing the IP address as if they are connected
So I then used X-Lite to connect one of my SIP trunks and it dialed out OK
I deleted it from X-Lite and still not working with SAIL. So now I am beat, bottom line is I can receive calls but not make them, my routes haven't changed and my external IP address is still correct. Can anyone please shed some light on it for me and point me in the right direction. Thanks for looking.
Kind regards,
Del
Asterisk console output when I dial out:
[root@sark-pbx ~]# asterisk -rvvvvv
Asterisk 1.8.5.0, Copyright (C) 1999 - 2011 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for detail s.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
== Parsing '/etc/asterisk/asterisk.conf': == Found
== Parsing '/etc/asterisk/extconfig.conf': == Found
Connected to Asterisk 1.8.5.0 currently running on sark-pbx (pid = 2781)
Verbosity is at least 5
> doing dnsmgr_lookup for 'sipgate.co.uk'
> ast_get_srv: SRV lookup for '_sip._udp.sipgate.co.uk' mapped to host sipgate.co.uk, port 5060
> doing dnsmgr_lookup for 'sipgate.co.uk'
> ast_get_srv: SRV lookup for '_sip._udp.sipgate.co.uk' mapped to host sipgate.co.uk, port 5060
> doing dnsmgr_lookup for 'sipgate.co.uk'
> ast_get_srv: SRV lookup for '_sip._udp.sipgate.co.uk' mapped to host sipgate.co.uk, port 5060
> doing dnsmgr_lookup for 'sipgate.co.uk'
> ast_get_srv: SRV lookup for '_sip._udp.sipgate.co.uk' mapped to host sipgate.co.uk, port 5060
> doing dnsmgr_lookup for 'sipgate.co.uk'
> ast_get_srv: SRV lookup for '_sip._udp.sipgate.co.uk' mapped to host sipgate.co.uk, port 5060
> doing dnsmgr_lookup for 'sipgate.co.uk'
> ast_get_srv: SRV lookup for '_sip._udp.sipgate.co.uk' mapped to host sipgate.co.uk, port 5060
> doing dnsmgr_lookup for 'sipgate.co.uk'
> ast_get_srv: SRV lookup for '_sip._udp.sipgate.co.uk' mapped to host sipgate.co.uk, port 5060
> doing dnsmgr_lookup for 'sipgate.co.uk'
> ast_get_srv: SRV lookup for '_sip._udp.sipgate.co.uk' mapped to host sipgate.co.uk, port 5060
> doing dnsmgr_lookup for 'sipgate.co.uk'
> ast_get_srv: SRV lookup for '_sip._udp.sipgate.co.uk' mapped to host sipgate.co.uk, port 5060
> doing dnsmgr_lookup for 'sipgate.co.uk'
> ast_get_srv: SRV lookup for '_sip._udp.sipgate.co.uk' mapped to host sipgate.co.uk, port 5060
> doing dnsmgr_lookup for 'sip.voipcheap.co.uk'
> doing dnsmgr_lookup for 'sip.voipcheap.co.uk'
> doing dnsmgr_lookup for 'sipgw.voicenetwork.ca'
> ast_get_srv: SRV lookup for '_sip._udp.sipgw.voicenetwork.ca' mapped to host sipgw.voicenetwork.ca, port 5060
> doing dnsmgr_lookup for 'sipgw.voicenetwork.ca'
> ast_get_srv: SRV lookup for '_sip._udp.sipgw.voicenetwork.ca' mapped to host sipgw.voicenetwork.ca, port 5060
> doing dnsmgr_lookup for 'did.voicenetwork.ca'
> doing dnsmgr_lookup for 'did.voicenetwork.ca'
== Using SIP RTP CoS mark 5
-- Executing [07907339916@internal:1] AGI("SIP/402-00000036", "sarkhpe,OutCos,07907339916") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
-- <SIP/402-00000036>AGI Script sarkhpe completed, returning 0
-- Executing [07907339916@402closedcos:1] AGI("SIP/402-00000036", "sarkhpe,OutCluster,07907339916") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
-- AGI Script Executing Application: (Set) Options: (CDR(accountcode)=default)
-- <SIP/402-00000036>AGI Script sarkhpe completed, returning 0
-- Executing [07907339916@qrxvtmny:1] AGI("SIP/402-00000036", "sarkhpe,OutRoute,outbound") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
-- AGI Script Executing Application: (Set) Options: (__filename=1343077839-07907339916-402.wav)
-- AGI Script Executing Application: (MixMonitor) Options: (/var/spool/asterisk/monitor/1343077839-07907339916-402.wav,,/bin/sh /opt/sark/scripts/selmix mixmon /var/spool/asterisk/monitor/1343077839-07907339916-402.wav)
== Begin MixMonitor Recording SIP/402-00000036
-- AGI Script Executing Application: (Set) Options: (TIMEOUT(absolute)=14400)
Channel will hangup at 2012-07-24 02:10:39.937 BST.
-- AGI Script Executing Application: (Dial) Options: (SIP/07907339916@peer8022,,T)
[Jul 23 22:10:39] WARNING[8738]: app_dial.c:2196 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- AGI Script Executing Application: (Playback) Options: (beep)
-- <SIP/402-00000036> Playing 'beep.gsm' (language 'en-gb')
[Jul 23 22:10:39] NOTICE[8738]: channel.c:4128 __ast_read: Dropping incompatible voice frame on SIP/402-00000036 of format alaw since our native format has changed to 0x4 (ulaw)
-- AGI Script Executing Application: (Playtones) Options: (congestion)
-- AGI Script Executing Application: (Congestion) Options: ()
-- <SIP/402-00000036>AGI Script sarkhpe completed, returning 4
== Spawn extension (qrxvtmny, 07907339916, 1) exited non-zero on 'SIP/402-00000036'
-- Executing [h@qrxvtmny:1] Hangup("SIP/402-00000036", "") in new stack
== Spawn extension (qrxvtmny, h, 1) exited non-zero on 'SIP/402-00000036'
== MixMonitor close filestream
== Executing [/bin/sh /opt/sark/scripts/selmix mixmon /var/spool/asterisk/monitor/1343077839-07907339916-402.wav]
== End MixMonitor Recording SIP/402-00000036
sark-pbx*CLI>
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