Require Vicidial issues to be sorted asap

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Require Vicidial issues to be sorted asap

Postby sasb » Wed Aug 25, 2010 7:01 am

Hi i have a vicidial box with 18 agents on it, the problem ihave is that as we currently do not have computers delivered until next week i require the agents to dial from the handsets manually without logging into vicidial, however when they make a call and connects they speak with customer for about a minute and the call disconnects...

Anybody have any clues?

Thanks

Shahid
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Postby williamconley » Wed Aug 25, 2010 8:10 am

when you post, please post your entire configuration including (but not limited to) your installation method and OS with kernel or version, vicidial version and build, asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box.

Similar to This:
Vicibox X.X from .iso | Vicidial X.X.X Build XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

this IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)
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Postby boybawang » Wed Aug 25, 2010 8:19 am

could be your internet connection, pm me if you want paid support since your posting this on the job and consultation thread
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Specss

Postby sasb » Wed Aug 25, 2010 8:26 am

Apologies -

Here we are:

Vicidial 2.4 Build: 0.5 | Asterisk 1.4.21.2 | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

Server uses g729 codec | it is an HP ML330 G6 - Intel Xeon 2.4Ghz Quad Core - 4GB Ram - 2 x 250GB SATA HDD - No Raid
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Postby williamconley » Wed Aug 25, 2010 8:33 am

show your "dial plan" for this carrier from the carrier entry
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dial plan

Postby sasb » Wed Aug 25, 2010 8:38 am

[toqi]
username=xxxxxxx
type=friend
secret=xxxxxxxxxxx
host=xxxxxxxxxxxxx
disallow=all
allow=g729
nat=yes
context=default
canreinvite=yes
dtmfmode=rfc2833

exten => _9.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9.,2,Dial(${TRUNK}/0${EXTEN:2},,tTor)
exten => _9.,3,Hangup
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asterisk cli

Postby sasb » Wed Aug 25, 2010 9:10 am

[Aug 25 15:08:05] -- Executing [907930895847@default:1] AGI("SIP/301-0000014 0", "agi://127.0.0.1:4577/call_log") in new stack
[Aug 25 15:08:05] -- AGI Script agi://127.0.0.1:4577/call_log completed, ret urning 0
[Aug 25 15:08:05] -- Executing [907930895847@default:2] Dial("SIP/301-000001 40", "IAX2/toqi/07930895847||tTor") in new stack
[Aug 25 15:08:05] -- Called toqi/07930895847
[Aug 25 15:08:05] -- Call accepted by xxx.xxx.xxx.xxx (format g729)
[Aug 25 15:08:05] -- Format for call is g729
[Aug 25 15:08:05] -- IAX2/toqi-6477 answered SIP/301-00000140
[Aug 25 15:08:06] == Parsing '/etc/asterisk/manager.conf': [Aug 25 15:08:06] F ound
[Aug 25 15:08:06] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 25 15:08:07] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 25 15:08:25] WARNING[3654]: chan_sip.c:2015 retrans_pkt: Maximum retries ex ceeded on transmission ZDdiZWU1NWQzN2VhMmQyOTllODIxMDIwZmRiZDBmZjc. for seqno 2 (Critical Response) -- See doc/sip-retransmit.txt.
[Aug 25 15:08:25] WARNING[3654]: chan_sip.c:2037 retrans_pkt: Hanging up call ZD diZWU1NWQzN2VhMmQyOTllODIxMDIwZmRiZDBmZjc. - no reply to our critical packet (se e doc/sip-retransmit.txt).
[Aug 25 15:08:25] -- Executing [h@default:1] DeadAGI("SIP/301-00000140", "ag i://127.0.0.1:4577/call_log--HVcauses-- ... ER-----20- ----20") in new stack
[Aug 25 15:08:25] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI -----NODEBUG-----16-----ANSWER-----20-----20 completed, returning 0
[Aug 25 15:08:25] -- Hungup 'IAX2/toqi-6477'
[Aug 25 15:08:25] == Spawn extension (default, 907930895847, 2) exited non-zer o on 'SIP/301-00000140'
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Postby williamconley » Wed Aug 25, 2010 9:51 am

1) Try SIP instead of IAX if the provider allows
2) Try another provider (in case THEY are disconnecting the calls)
3) A good “Test” is VOIPStreet. They will allow 100 minutes Free (certainly enough for a test) after you validate that you are “real” and not a Chinese “bot” by calling them to activate your service. Remember to sign up for “metered” service, not wholesale. Set up a test account (metered, no monthly fee or minimums) for YOU, and if the test goes well you will consider setting up a wholesale account LATER for the client. Their wholesale accounts cost less per minute but they DO have monthly minimums (generally at least $200/month).
4) Another good test is Vitelity, but they have a sign up fee.
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Vicidial / A2Billing

Postby sasb » Wed Aug 25, 2010 11:06 pm

Hey all

This was an issue with an A2Billing trunk registered in Vicidial - they were not working too well with each other - when tested with Vicidial registered direct to carrier it worked great.

Seemed like some delay/lag issue between A2Billing and Vicidial

Cheers for the help Bill,

Shahid
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Postby williamconley » Wed Aug 25, 2010 11:18 pm

Another vote for "Stay away from mixing TrixBox and Vicidial". :)

I wonder if elastix has the same issues? LOL
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Postby Trying » Thu Aug 26, 2010 8:48 am

Speaking of Elastix...........

http://www.elastix.org/en/component/kun ... html#58187

Towards the bottom is a link to a script which integrates the two if anyone wants to try.
Vicibox Redux 3.1.14 64 bit
Asterisk: 1.4.39.2-vici | VERSION: 2.8-433a | BUILD: 140411-1434
Sangoma A102D/E1
Servers: 1 x Database; 1 x Web; 6 x Telephony; 2 x Archive
Outbound and blended ratio 3:1 with full voice recording
No additional software
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Postby williamconley » Thu Aug 26, 2010 9:04 am

Wow. Vicidial as a FreePBX module in Elastix. They say it breaks some FreePBX functionality, but vici is fully functional, due to permissions issues. So figuring out how to get all functionality back into FreePBX is all that's left.

Actually that sounds mighty familiar.
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Postby mflorell » Thu Aug 26, 2010 10:39 am

We can continue the ViciDial <-> ElastixPBX interoperability discussion on this thread since it doesn't really belong on Jobs:

http://www.vicidial.org/VICIDIALforum/v ... hp?p=52380
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