408 from Sip Error No calls From Dialer Congestion Error

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408 from Sip Error No calls From Dialer Congestion Error

Postby shanjay86 » Sat Oct 08, 2011 4:05 pm

Goautodial CE 2.0
BUILD: 100527-2211
Asterisk 1.4.27.1-1
Form Iso
Eyebeam as softphone
Single server
no hardware or addon

I have been using goautodial at more then 3-4 locations over a year Never faced a problem but at one of my centers till yday it was working fine all of a sudden i started getting this issue..


Cant get call from Agent interface or / Eyebeam

Manual dial from EyeBeam error : called failed 408 "Temporarily Unavailable"
manual dial from Agent interface: "DIAL ALERT:

Call Rejected: CHANUNAVAIL
Cause: 20 - Subscriber absent."


Asterisk -r code
Code: Select all
[Oct  8 16:52:36]     -- Executing [918006249897@default:1] AGI("SIP/cc100-00000007", "agi://127.0.0.1:4577/call_log") in new stack
[Oct  8 16:52:36]     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct  8 16:52:36]     -- Executing [918006249897@default:2] Dial("SIP/cc100-00000007", "SIP/goautodial/18006249897||tTor") in new stack
[Oct  8 16:52:36] WARNING[6162]: app_dial.c:1296 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Oct  8 16:52:36]   == Everyone is busy/congested at this time (1:0/0/1)
[Oct  8 16:52:36]     -- Executing [918006249897@default:3] Hangup("SIP/cc100-00000007", "") in new stack
[Oct  8 16:52:36]   == Spawn extension (default, 918006249897, 3) exited non-zero on 'SIP/cc100-00000007'
[Oct  8 16:52:36]     -- Executing [h@default:1] DeadAGI("SIP/cc100-00000007", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Oct  8 16:52:36]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL---------- completed, returning 0



Dial plan if needed:

Code: Select all
register => Username:Pass@Domain:5060/Username

Account Entry
[goautodial]
disallow=all
allow=ulaw
allow=g729
type=friend
accountcode=goautodial
username=XXXXX
secret=XXXXX
host=myDomain
dtmfmode=rfc2833
fromuser=052510344
qualify=yes
insecure=very

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${SIPgoautodial}/${EXTEN:1},,tTor)
exten => _91NXXNXXXXXX,3,Hangup


Any help i would be thankful other wise only option i have to format dialer and start over.

I am able to call using eyebeam bypassing my dialer from domain.i know sip is working Tried more then one SIP account.
shanjay86
 
Posts: 20
Joined: Sat Oct 08, 2011 3:40 pm

Postby shanjay86 » Sun Oct 09, 2011 2:10 pm

44 views no help... :roll:
shanjay86
 
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Postby noworldorder » Sun Oct 09, 2011 4:24 pm

sorry - if I had the skill I would assist. I had your exact problem and found no guidance so I reinstalled.
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Postby shanjay86 » Mon Oct 10, 2011 4:41 am

noworldorder wrote:sorry - if I had the skill I would assist. I had your exact problem and found no guidance so I reinstalled.


Thanks i think only option i have will wait till end of day
shanjay86
 
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Postby maykelsoft » Mon Oct 10, 2011 10:43 am

have you tried to reboot the server? please be sure that the SIP peer going to provider is connected.
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Postby shanjay86 » Tue Oct 11, 2011 9:15 am

maykelsoft wrote:have you tried to reboot the server? please be sure that the SIP peer going to provider is connected.


Yeh tried it over 300 times in 2 days.. well i have formatted and restarted from fresh and everything is working fine now...
shanjay86
 
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Joined: Sat Oct 08, 2011 3:40 pm

Postby shanjay86 » Thu Oct 13, 2011 3:40 pm

again started facing same issue.... :( anyone get any info on this?!?
shanjay86
 
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Postby gardo » Thu Oct 13, 2011 4:25 pm

http://goautodial.com
Empowering the next generation contact centers
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Postby shanjay86 » Fri Oct 14, 2011 7:15 am

gardo wrote:This might help:



Have you checked with your VoIP carrier?


yes i did.. when i dial from eyebeam... using just my sip account goes thru... but when goin thru dialerr... i get tht error...
shanjay86
 
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Hi san

Postby hunter2009 » Sat Oct 15, 2011 6:10 am

Check the sip registred or Not and check with Sip debug and traceroute you will find it :D
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with warm Regards
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Skype Id : hunter200924
Thank you
Share your experience & Ideas here so that it will be use full for All
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sip status

Postby striker » Mon Oct 17, 2011 1:26 pm

check whether ur sip trunk is registered properly

in asterisk cli type sip show peers
and sip show registry

check the internet connectivity to your server.
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