[asterisk-users] issue after install dahdi

John Novack jnovack at stromberg-carlson.org
Mon Oct 21 13:48:03 CDT 2013


A VERY OLD and beyond EOF version.
If you MUST, due to some driver issue, use Asterisk 1.4, then please use 1.4.44
Otherwise I suggest you move to something more current, either version 1.8.current or beyond.
Also, CLI says 1.4.43, your message says 1.4.32 ???

Some examination of chan_dahdi and your dialplan would help someone give you some assistance.
Is this a fresh install, or one that has been working for years?

What Digium card?

John Novack

Salaheddine Elharit wrote:
> i need your help regarding some issue related to the outband calls
>
> i have installed asterisk 1.4.32 with dahdi and i have 1 card diguim with 2 ports
> when i try to call my phone number all time i receive message  busy number
>
> this error just with g1.
>
> with g2 there is no problem i can call without issue
>
> can anyone see the CLI and tell me what is the problem
>
> thanks and regards
>
>   == Parsing '/etc/asterisk/asterisk.conf': Found
>   == Parsing '/etc/asterisk/extconfig.conf': Found
> Connected to Asterisk 1.4.43.18495-AheevaCCS-3.2.12 currently running on SRVRADI                                                        O (pid = 4147)
> Verbosity is at least 3
>     -- Executing [0661049303 at agents:1] Set("SIP/223-00000021", "CALLERID(number)          =520460587") in new stack
>     -- Executing [0661049303 at agents:2] Dial("SIP/223-00000021", "DAHDI/g1/066104          9303|30") in new stack
>     -- Requested transfer capability: 0x00 - SPEECH
>     -- Called g1/0661049303
>     -- Moving call (DAHDI/3-1) from channel 3 to 2.
> [Oct 21 17:43:27] WARNING[4264]: chan_dahdi.c:9438 pri_fixup_principle: Can't mo                                                          ve call (DAHDI/3-1) from channel 3 to 2.  It is already in use.
> [Oct 21 17:43:27] WARNING[4264]: chan_dahdi.c:9558 pri_find_fixup_principle: Spa                                                          n 1: PRI requested channel 1/2 is not available.
>     -- Hungup 'DAHDI/3-1'
>   == Everyone is busy/congested at this time (1:0/0/1)
>     -- Executing [0661049303 at agents:3] Hangup("SIP/223-00000021", "") in new sta          ck
>   == Spawn extension (agents, 0661049303, 3) exited non-zero on 'SIP/223-0000002                                                        1'
>     -- Executing [h at agents:1] GotoIf("SIP/223-00000021", "0?3:2") in new stack
>     -- Goto (agents,h,2)
>     -- Executing [h at agents:2] AHEventsProxy("SIP/223-00000021", "MSG_TYPE_TERMIN                  ATE_CALL::::1382377407") in new stack
>  AHEventsProxy: Channel [SIP/223-00000021]. Data [MSG_TYPE_TERMINATE_CALL::::138  2377407]
>     -- chan is SIP/223-00000021
>  AHEventsProxy: Send To CtiServer: socket:[89]. message:[41,1382377407^^^^stcrpb  x^~]
>     -- Executing [h at agents:3] Hangup("SIP/223-00000021", "") in new stack
>   == Spawn extension (agents, h, 3) exited non-zero on 'SIP/223-00000021'
>     -- SIP/224-00000020 is ringing
> SRVRADIO*CLI>
> Disconnected from Asterisk server
> Executing last minute cleanups
>
>
>
>
>

-- 

Dog is my Co-pilot

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131021/54fefa1b/attachment.html>


More information about the asterisk-users mailing list