HELP! - Dial timed out, contact your systems administor

[dailafing]

Every time I manual dial, I can successfully communicate within the phone call, but its like vici doesn't know the call has been placed?

I receive this message about 10 seconds into the call in an "Agent alert!" pop up box



Can anyone help me please?

The asterisk output it...

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[Jan 15 17:46:20] NOTICE[2442]: cel_custom.c:95 load_config: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs.
[Jan 15 17:46:20]  Added CEL CSV mapping for 0 files.
[Jan 15 17:46:20] ERROR[2381]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("VicidialServerHostName.vici.local","(null)", ...): Name or service not known
[Jan 15 17:46:20] WARNING[2381]: acl.c:590 resolve_first: Unable to lookup 'VicidialServerHostName.vici.local'
[Jan 15 17:46:20]   == Using SIP CoS mark 4
[Jan 15 17:46:20]   == Parsing '/etc/asterisk/sip_notify.conf': [Jan 15 17:46:20]   == Found
[Jan 15 17:46:50]   == Manager 'sendcron' logged on from 127.0.0.1
[Jan 15 17:46:50]   == Using SIP RTP CoS mark 5
[Jan 15 17:46:52]        > Channel SIP/gs102-00000000 was answered.
[Jan 15 17:46:52]     -- Executing [8600051@default:1] MeetMe("SIP/gs102-00000000", "8600051,F") in new stack
[Jan 15 17:46:52]   == Parsing '/etc/asterisk/meetme.conf': [Jan 15 17:46:52]   == Found
[Jan 15 17:46:52]   == Parsing '/etc/asterisk/meetme-vicidial.conf': [Jan 15 17:46:52]   == Found
[Jan 15 17:46:52]     -- Created MeetMe conference 1023 for conference '8600051'
[Jan 15 17:46:52]     -- <SIP/gs102-00000000> Playing 'conf-onlyperson.gsm' (language 'en')

[Jan 15 17:47:19]     -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00000000;2", "8600051,F") in new stack
[Jan 15 17:47:19]        > Channel Local/8600051@default-00000000;1 was answered.
[Jan 15 17:47:19]     -- Executing [PHONENUMBERHERE@default:1] AGI("Local/8600051@default-00000000;1", "agi://127.0.0.1:4 577/call_log") in new stack
[Jan 15 17:47:19]     -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=1001))
[Jan 15 17:47:19]     -- <Local/8600051@default-00000000;1>AGI Script agi://127.0.0.1:4577/call_log completed, return ing 0
[Jan 15 17:47:19]     -- Executing [PHONENUMBERHERE@default:2] Dial("Local/8600051@default-00000000;1", "SIP/orbtalk_wmsx/PHONENUMBERHERE,,tToR") in new stack
[Jan 15 17:47:19]   == Using SIP RTP CoS mark 5
[Jan 15 17:47:19]     -- Called SIP/orbtalk_wmsx/PHONENUMBERHERE
[Jan 15 17:47:20]   == Manager 'sendcron' logged off from 127.0.0.1
[Jan 15 17:47:20]   == Manager 'sendcron' logged on from 127.0.0.1
[Jan 15 17:47:20]     -- Executing [58600051@default:1] MeetMe("Local/58600051@default-00000001;2", "8600051,Fmq") in new stack
[Jan 15 17:47:20]        > Channel Local/58600051@default-00000001;1 was answered.
[Jan 15 17:47:20]     -- Executing [8309@default:1] Answer("Local/58600051@default-00000001;1", "") in new stack
[Jan 15 17:47:20]     -- Executing [8309@default:2] Monitor("Local/58600051@default-00000001;1", "wav,PHONENUMBERHERE_20140115-224719") in new stack
[Jan 15 17:47:20]     -- Executing [8309@default:3] Wait("Local/58600051@default-00000001;1", "7200") in new stack
[Jan 15 17:47:20]     -- SIP/orbtalk_wmsx-00000001 is making progress passing it to Local/8600051@default-00000000;1
[Jan 15 17:47:21]   == Manager 'sendcron' logged off from 127.0.0.1
[Jan 15 17:47:24]     -- SIP/orbtalk_wmsx-00000001 answered Local/8600051@default-00000000;1
[Jan 15 17:47:37]   == Manager 'sendcron' logged on from 127.0.0.1
[Jan 15 17:47:37]   == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-00000000;2'
[Jan 15 17:47:37]     -- Executing [h@default:1] AGI("Local/8600051@default-00000000;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Jan 15 17:47:37]   == Manager 'sendcron' logged on from 127.0.0.1
[Jan 15 17:47:37]     -- Executing [h@default:1] AGI("Local/8600051@default-00000000;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----18-----13") in new stack
[Jan 15 17:47:37]   == Manager 'sendcron' logged on from 127.0.0.1
[Jan 15 17:47:37]   == Spawn extension (default, 58600051, 1) exited non-zero on 'Local/58600051@default-00000001;2'
[Jan 15 17:47:37]     -- Executing [h@default:1] AGI("Local/58600051@default-00000001;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Jan 15 17:47:37]     -- <Local/58600051@default-00000001;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Jan 15 17:47:37]   == Spawn extension (default, 8309, 3) exited non-zero on 'Local/58600051@default-00000001;1'
[Jan 15 17:47:37]     -- Executing [h@default:1] AGI("Local/58600051@default-00000001;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Jan 15 17:47:37]     -- <Local/8600051@default-00000000;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Jan 15 17:47:37]     -- <Local/58600051@default-00000001;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Jan 15 17:47:37]     -- <Local/8600051@default-00000000;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----18-----13 completed, returning 0



Using ViciBox5.i686-5.0.3.preload.iso - Asterisk 1.8.25 and VERSION: 2.8-421a BUILD: 140108-0752

[geoff3dmg]

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[Jan 15 17:46:20] ERROR[2381]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("VicidialServerHostName.vici.local","(null)", ...): Name or service not known
[Jan 15 17:46:20] WARNING[2381]: acl.c:590 resolve_first: Unable to lookup 'VicidialServerHostName.vici.local'

You have a dns error for a start.

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[Jan 15 17:47:24]     -- SIP/orbtalk_wmsx-00000001 answered Local/8600051@default-00000000;1

The call was successful. I am unsure if it's related to the DNS error. However I have used orbtalk in the past, I know for a fact you have to add some sip-silences in your routing extension as there is a delay sending the audio with this carrier. Also what does your carrier dialplan and peer definition look like, I know orbtalk was fussy generally to get working (SIP headers have to be just so).

[dailafing]

Ok, I remember setting up the 3 DNS severs with yast lan. 1) my router internal up address, 2) 8.8.8.8, 3) 4.2.2.2.

Orbtalk were not very helpful. They said they supported asterisk before I spent with them, then got a bit arsy when I asked for a dial plan! I had to change the r to an uppercase R to get a ring on my manual dials! And they originally tried to blame me for it....

This is my dial plan...
exten => _X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _X.,2,Dial(${SIPTRUNK}/${EXTEN},,tToR)
exten => _X.,3,Hangup

Reading through the rest if your message, I am not familiar with: sip-silences, a routing extention, peer definition, sip-headers.

Additionally, I have also been struggling with a rather troublesome delay in voice teaching the destination. Resulting in awkward over speak with prospect customers.

Your continued advice is greatly anticipated and greatly appreciated.

Thanks

[geoff3dmg]

Here are my settings for Orbtalk PAYG SIP Trunks

Carrier definition (note the fromuser and fromdomain settings):

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[orbtalk]
type=peer
insecure=invite
nat=yes
host=sipgw3.orbtalk.co.uk
fromdomain=sipgw3.orbtalk.co.uk
fromuser=<username>
username=<username>
secret=<password>
qualify=yes
canreinvite=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=trunkinbound

registration string:

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register=><username>:<password>@sipgw3.orbtalk.co.uk/<username>

dialplan, campaign prefix of 9 and locked to phonecode 44. I also force the CallerID, Orbtalk will reject the calls if you send CallerID that is not your number:

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exten => _944.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _944.,2,Set(CALLERID(num)=<username>)
exten => _944.,3,Dial(SIP/${EXTEN:1}@orbtalk,,tTor)
exten => _944.,4,Hangup

[dailafing]

Ok I will try the exact setup on my dialler, and let you know what happens.

By the way, is there a way to 'comment out' text in my dial plan?
So I can leave my current setup in place, and add yours as a trial.... Just saves me copying out to notepad.
Thas

[dailafing]

OK, I tried your settings but with my username & pw, however the same problem applies to the letter. The only difference being that when making a manual dial, I didn't hear the ring, but this was because the R in tTor was lowercase again....

I'm interested to know more about this CallerID thing, since they want to charge me a monthly fee for the CLI, but dure to low use, I love the freedom of the PAYGO service and don't want to commit to 24 a year lol

ok, back to the issue... Agent interface on manual dial does not recognise when the call is answered...

I made a call with ne carrier settings provided kindly by the above poster who's name does not appear to my in the full editor! sorry about that!

here is my CLI Asterisk output....

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[Jan 21 18:35:33]     -- Registered SIP 'gs102' at 192.168.1.7:57082
[Jan 21 18:35:33]        > Saved useragent "3CXPhone 6.0.26523.0" for peer gs102
[Jan 21 18:35:36]   == Using SIP RTP CoS mark 5
[Jan 21 18:35:38]        > Channel SIP/gs102-00000006 was answered.
[Jan 21 18:35:38]     -- Executing [8600051@default:1] MeetMe("SIP/gs102-00000006", "8600051,F") in new stack
[Jan 21 18:35:38]   == Parsing '/etc/asterisk/meetme.conf': [Jan 21 18:35:38]   == Found
[Jan 21 18:35:38]   == Parsing '/etc/asterisk/meetme-vicidial.conf': [Jan 21 18:35:38]   == Found
[Jan 21 18:35:38]     -- Created MeetMe conference 1023 for conference '8600051'
[Jan 21 18:35:38]     -- <SIP/gs102-00000006> Playing 'conf-onlyperson.gsm' (language 'en')
[Jan 21 18:35:53]     -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00000006;2", "8600051,F") in new stack
[Jan 21 18:35:53]        > Channel Local/8600051@default-00000006;1 was answered.
[Jan 21 18:35:53]     -- Executing [0126*******@default:1] AGI("Local/8600051@default-00000006;1", "agi://127.0.0.1:4577/call_log") in new stack
[Jan 21 18:35:53]     -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=1001))
[Jan 21 18:35:53]     -- <Local/8600051@default-00000006;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jan 21 18:35:53]     -- Executing [0126*******@default:2] Set("Local/8600051@default-00000006;1", "CALLERID(num)=OBTR*****") in new stack
[Jan 21 18:35:53]     -- Executing [0126*******@default:3] Dial("Local/8600051@default-00000006;1", "SIP/0126*******@orbtalk,,tTor") in new stack
[Jan 21 18:35:53]   == Using SIP RTP CoS mark 5
[Jan 21 18:35:53]     -- Called SIP/0126*******@orbtalk
[Jan 21 18:35:53]     -- Executing [58600051@default:1] MeetMe("Local/58600051@default-00000007;2", "8600051,Fmq") in new stack
[Jan 21 18:35:53]        > Channel Local/58600051@default-00000007;1 was answered.
[Jan 21 18:35:53]     -- Executing [8309@default:1] Answer("Local/58600051@default-00000007;1", "") in new stack
[Jan 21 18:35:53]     -- Executing [8309@default:2] Monitor("Local/58600051@default-00000007;1", "wav,0126*******_20140121-233552") in new stack
[Jan 21 18:35:53]     -- Executing [8309@default:3] Wait("Local/58600051@default-00000007;1", "7200") in new stack
[Jan 21 18:35:54]     -- SIP/orbtalk-00000007 is making progress passing it to Local/8600051@default-00000006;1
[Jan 21 18:35:58]     -- SIP/orbtalk-00000007 answered Local/8600051@default-00000006;1
[Jan 21 18:36:36]     -- Executing [h@default:1] AGI("Local/8600051@default-00000006;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----43-----38") in new stack
[Jan 21 18:36:36]     -- <Local/8600051@default-00000006;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----43-----38 completed, returning 0
[Jan 21 18:36:37]   == Spawn extension (default, 0126*******, 3) exited non-zero on 'Local/8600051@default-00000006;1'
[Jan 21 18:36:37] WARNING[5063]: app_meetme.c:3942 conf_run: Unable to write frame to channel Local/8600051@default-00000006;2
[Jan 21 18:36:37]   == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-00000006;2'
[Jan 21 18:36:37]     -- Executing [h@default:1] AGI("Local/8600051@default-00000006;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Jan 21 18:36:37]     -- <Local/8600051@default-00000006;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
VicidialServerHostName*CLI>


I'm hoping to get this nailed soon since im about to be expected to completely ditch our hosted dialler and manage this server im trying to bang out...!

im really looking forward to hearing the sound advice from someone with the answer please
I'm happy to answer any questions about my setup, and another recent post of mine in a topic I made shows the guide I followed to install with.

thanks

[dailafing]

Additionally, I have just discovered that if I dial on ratio or auto dial and then transfer the call, it hangs up on the customer when the agent leaves the conf!
But strangely, the agent 3cx is still left connected with the 3rd party??? even though the agent gui is 'ready' for the next call, and the dialler has started to churn numbers again!!!

Really strange...

Please help, im acutally panicing now, because I said I could do the dialler for them, and hot transfer in a major factor in their business...

thanks

[geoff3dmg]

The call worked in the above trace. But there was a meetme error. As I said, this is due to Orbtalk not passing the audio straight away. You must put some sip-silences to your routing extension to give it more time for the audio to come through. Check your campaign setting for the 'routing extension' number. By default it's '8368'. Then go open up /etc/asterisk/extensions.conf and scroll to about line 460. You should see something like this:

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; VICIDIAL_auto_dialer transfer script Load Balanced:
exten => 8368,1,Playback(sip-silence)
exten => 8368,n,AGI(agi://127.0.0.1:4577/call_log)
exten => 8368,n,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8368,n,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8368,n,Hangup()

You need to add a couple of sip-silences IIRC for Orbtalk, so you would alter the above to look like this:

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; VICIDIAL_auto_dialer transfer script Load Balanced:
exten => 8368,1,Playback(sip-silence)
exten => 8368,n,Playback(sip-silence)
exten => 8368,n,Playback(sip-silence)
exten => 8368,n,AGI(agi://127.0.0.1:4577/call_log)
exten => 8368,n,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8368,n,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8368,n,Hangup()

As for CallerID. You normally get an inbound number with Orbtalk when you go PAYG. If your outbound CallerID doesn't match then they reject the calls. However there is a form you can get off them and pay a one time fee to lift this restriction.