The VICIDIAL version is 2.0.2 and Asterisk is 1.2.13. Running on an Athlon 64 3000+ processor, nVidia chipset and 2 Gb of RAM. Four HFC-S cards are being used to connect to the ISDN (BRI) net, but only one card is actually connectet to the ISDN net (Due to the lack of lines.). Locally we are using pure SIP clients (ALaw).
Zaptel Version is 1.2.9.1.
Load average is about 0.10
But it seems that reportings are not working correctly and sometimes the the dialer doesn't hang up.
Hopper Level is at 10, and the dialer is set to manual dialing.
From the asterisk*CLI:
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-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing SetCallerPres("Local/8600051@SIP-6677,1", "prohib") in new stack
-- Executing Set("Local/8600051@SIP-6677,1", "DYNAMIC_FEATURES=automon") in new stack
-- Executing Dial("Local/8600051@SIP-6677,1", "Zap/g5/030XXXXXXX|55|WT") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g5/030XXXXXXX
-- Zap/1-1 is proceeding passing it to Local/8600051@SIP-6677,1
-- Channel 0/1, span 1 got hangup request
-- Channel 0/1, span 1 received AOC-E charging 0 units
Jan 29 18:44:55 WARNING[27103]: app_dial.c:733 wait_for_answer: Unable to forward voice
-- Hungup 'Zap/1-1'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("Local/8600051@SIP-6677,1", "") in new stack
== Spawn extension (SIP, 9030XXXXXXX, 5) exited non-zero on 'Local/8600051@SIP-6677,1'
-- Executing DeadAGI("Local/8600051@SIP-6677,1", "agi://127.0.0.1:4577/call_log") in new stack
ASTfastlog seems to die sometimes.
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asterisk ~ # screen -r
There are several suitable screens on:
4628.ASTupdate (Detached)
4630.ASTsend (Detached)
4632.ASTlisten (Detached)
4634.ASTVDauto (Detached)
4636.ASTVDremote (Detached)
4638.ASTVDadapt (Detached)
4583.ASTfastlog (Dead ???)
4650.ASTfastlog (Dead ???)
4640.ASTfastlog (Attached)
Remove dead screens with 'screen -wipe'.
Type "screen [-d] -r [pid.]tty.host" to resume one of them.
The dial extension in /etc/asterisk/extensions.conf:
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exten => _9X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9X.,n,SetCallerPres(prohib)
exten => _9X.,n,Set(DYNAMIC_FEATURES=automon)
exten => _9X.,n,Dial(${TRUNK5}/${EXTEN:1},55,WT)
exten => _9X.,n,Hangup()
Some more errors from the asterisk*CLI i get:
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== Spawn extension (SIP, 90554XXXXXXX, 5) exited non-zero on 'Local/8600051@SIP-a5fc,1'
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Jan 29 17:12:30 WARNING[4295]: chan_zap.c:8461 pri_fixup_principle: Call specified, but not found?
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Jan 29 17:50:55 WARNING[10482]: app_meetme.c:1555 conf_run: Unable to write frame to channel: Success
Jan 29 17:50:55 WARNING[10482]: app_meetme.c:1555 conf_run: Unable to write frame to channel: Success
Jan 29 17:50:55 WARNING[10482]: app_meetme.c:1555 conf_run: Unable to write frame to channel: Success
Jan 29 17:50:55 WARNING[10482]: app_meetme.c:1555 conf_run: Unable to write frame to channel: Success
Jan 29 17:50:55 WARNING[10482]: app_meetme.c:1555 conf_run: Unable to write frame to channel: Success
Jan 29 17:50:55 WARNING[10482]: app_meetme.c:1555 conf_run: Unable to write frame to channel: Success
Jan 29 17:50:55 WARNING[10482]: app_meetme.c:1555 conf_run: Unable to write frame to channel: Success
.....
Whats wrong?
Regards
Henrik