I'm At A Total loss Just To Dial A Call! UK

[dailafing]

Well, I'm trying to set up a new Dialler from a newly downloaded ViciBox Preloaded Express Install, and cant seem to get it to place calls...

This is my Asterisk readout during the time I place the call...

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[Sep 24 15:16:53]     -- Created MeetMe conference 1023 for conference '8600051'
[Sep 24 15:16:53]     -- <SIP/102-00000015> Playing 'conf-onlyperson.gsm' (language 'en')
[Sep 24 15:17:02]  Reloading SIP
[Sep 24 15:17:02]   == Parsing '/etc/asterisk/sip.conf': [Sep 24 15:17:02]   == Found
[Sep 24 15:17:02]   == Parsing '/etc/asterisk/sip-vicidial.conf': [Sep 24 15:17:02]   == Found
[Sep 24 15:17:02]   == Parsing '/etc/asterisk/users.conf': [Sep 24 15:17:02]   == Found
[Sep 24 15:17:02]   == Using SIP CoS mark 4
[Sep 24 15:17:02]   == Parsing '/etc/asterisk/sip_notify.conf': [Sep 24 15:17:02]   == Found
[Sep 24 15:17:56]     -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00000023;2", "8600051,F") in new stack
[Sep 24 15:17:56]        > Channel Local/8600051@default-00000023;1 was answered.
[Sep 24 15:17:56]     -- Executing [4415547****3@default:1] AGI("Local/8600051@default-00000023;1", "agi://127.0.0.1:4577/call_log") in new stack
[Sep 24 15:17:56]     -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=1001))
[Sep 24 15:17:56]     -- <Local/8600051@default-00000023;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Sep 24 15:17:56]     -- Executing [4415547****3@default:2] Dial("Local/8600051@default-00000023;1", "SIP/1234:PASSWORD@sip.provider.net/445547****3,,tToR") in new stack ; This is how is appears in the CLI!!!
[Sep 24 15:17:56] NOTICE[2066]: chan_sip.c:27535 sip_request_call: Conflicting extension values given. Using '1234' and not '445547****3'
[Sep 24 15:17:56]   == Using SIP RTP CoS mark 5
[Sep 24 15:17:56] ERROR[2066]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("sip.provider.net", "(null)", ...): Name or service not known
[Sep 24 15:17:56] WARNING[2066]: chan_sip.c:5711 create_addr: No such host: sip.provider.net
[Sep 24 15:17:56] WARNING[2066]: app_dial.c:2345 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[Sep 24 15:17:56]   == Everyone is busy/congested at this time (1:0/0/1)
[Sep 24 15:17:56]     -- Executing [4415547****3@default:3] Hangup("Local/8600051@default-00000023;1", "") in new stack
[Sep 24 15:17:56]   == Spawn extension (default, 4415547****3, 3) exited non-zero on 'Local/8600051@default-00000023;1'
[Sep 24 15:17:56]     -- Executing [h@default:1] AGI("Local/8600051@default-00000023;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Sep 24 15:17:56]     -- <Local/8600051@default-00000023;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL---------- completed, returning 0
[Sep 24 15:17:56]   == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-00000023;2'
[Sep 24 15:17:56]     -- Executing [h@default:1] AGI("Local/8600051@default-00000023;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Sep 24 15:17:56]     -- <Local/8600051@default-00000023;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Sep 24 15:17:57]   == Manager 'sendcron' logged off from 127.0.0.1


My Dial plan is:

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exten => _X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _0.,2,Dial(${SIPTRUNK}/44${EXTEN:1},,tTor)
exten => _440.,2,Dial(${SIPTRUNK}/44${EXTEN:3},,tTor)
exten => _44.,2,Dial(${SIPTRUNK}/44${EXTEN:3},,tToR)
exten => _44440.,2,Dial(${SIPTRUNK}/44${EXTEN:5},,tTor)
exten => _X.,3,Hangup

And from the Asterisk, you can see my numbers are formatted correctly to the extension.

I've been on the phone with OrbTalk, my carrier, and they were kind enough to go through my asterisk with me, but were unable to find the fault. They were certain that they were not receiving calls from the dialler, and said it was an issue with the configuration of the dialler.

Thanks for any advice

Version Info:

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OpenSuSE v.13.1 64-bit
Kernel v.3.11.10
Asterisk v.1.8.28.2-vici
DAHDI v.2.9.1.1
LibPRI v.1.4.14
Amfletec VoiceSync v.1.3.7
OpenR2 v.1.3.3 for MFC/R2 support
ViciDial SVN Trunk v.2.10-443a build 140617-2017 revision 2130

[omarrodriguezt]

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[Sep 24 15:17:56] ERROR[2066]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("sip.provider.net", "(null)", ...): Name or service not known
[Sep 24 15:17:56] WARNING[2066]: chan_sip.c:5711 create_addr: No such host: sip.provider.net
[Sep 24 15:17:56] WARNING[2066]: app_dial.c:2345 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[Sep 24 15:17:56]   == Everyone is busy/congested at this time (1:0/0/1)



I've been on the phone with OrbTalk, my carrier, and they were kind enough to go through my asterisk with me, but were unable to find the fault. They were certain that they were not receiving calls from the dialler, and said it was an issue with the configuration of the dialler.

It looks like your carrier is not set up correctly. Share your carrier details (don't forget to hide sensitive information)
Also, ask your carrier for a sample asterisk trunk settings.

[dailafing]

It looks like your carrier is not set up correctly. Share your carrier details (don't forget to hide sensitive information)
Also, ask your carrier for a sample asterisk trunk settings.

Hi there, Here is my Carrier...

Reg:
register => <USERNAME>:<PASS>@193.104.103.6:5060/<USERNAME>

Acc entry:
[orbtalk]
host=sip.orbtalk.co.uk
username=<USERNAME>
secret=<PASS>
type=peer
nat=yes
dtmfmode=RFC2833
allow=alaw&ulaw
port=5060
fromuser=<USERNAME>
fromdomain=sip.orbtalk.co.uk
context=from-trunk

Globals Str:
ORBTALK=SIP/orbtalk

Dial Plan:
exten => _X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _44.,2,Dial(${orbtalk}/44${EXTEN:2},,tToR)
exten => _X.,3,Hangup

Hope to here from you soon, and thank for this...

[dailafing]

I've been trying to fix it myself in the mean time...
I now I have a different error in the Asterisk CLI....

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[Sep 25 15:20:07]     -- Call accepted by 192.168.1.66 (format gsm)
[Sep 25 15:20:07]     -- Format for call is gsm
[Sep 25 15:20:09]        > Channel IAX2/103-885 was answered.
[Sep 25 15:20:09]     -- Executing [8600052@default:1] MeetMe("IAX2/103-885", "8600052,F") in new stack
[Sep 25 15:20:09]   == Parsing '/etc/asterisk/meetme.conf': [Sep 25 15:20:09]   == Found
[Sep 25 15:20:09]   == Parsing '/etc/asterisk/meetme-vicidial.conf': [Sep 25 15:20:09]   ==  Found
[Sep 25 15:20:09]     -- Created MeetMe conference 1023 for conference '8600052'
[Sep 25 15:20:09]     -- <IAX2/103-885> Playing 'conf-onlyperson.gsm' (language 'en')
[Sep 25 15:20:10]   == Manager 'sendcron' logged off from 127.0.0.1
[Sep 25 15:20:16]   == Manager 'sendcron' logged on from 127.0.0.1
[Sep 25 15:20:16]     -- Executing [8600052@default:1] MeetMe("Local/8600052@default-00000001;2", "8600052,F") in new stack
[Sep 25 15:20:16]        > Channel Local/8600052@default-00000001;1 was answered.
[Sep 25 15:20:16]     -- Executing [441269****06@default:1] AGI("Local/8600052@default-00000001;1", "agi://127.0.0.1:4577/call_log") in new stack
[Sep 25 15:20:16]     -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=1001))
[Sep 25 15:20:16]     -- <Local/8600052@default-00000001;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Sep 25 15:20:16]     -- Auto fallthrough, channel 'Local/8600052@default-00000001;1' status is 'UNKNOWN'
[Sep 25 15:20:16]     -- Executing [h@default:1] AGI("Local/8600052@default-00000001;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Sep 25 15:20:16]     -- <Local/8600052@default-00000001;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Sep 25 15:20:16]   == Spawn extension (default, 8600052, 1) exited non-zero on 'Local/8600052@default-00000001;2'
[Sep 25 15:20:16]     -- Executing [h@default:1] AGI("Local/8600052@default-00000001;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Sep 25 15:20:16]     -- <Local/8600052@default-00000001;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Sep 25 15:20:17]   == Manager 'sendcron' logged off from 127.0.0.1
[Sep 25 15:21:02]   == Manager 'sendcron' logged on from 127.0.0.1


And this was only changed because my dial plan is now:

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exten => _X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _44,2,Dial(${orbtalk}/44${EXTEN:2},,tToR)
exten => _X.,3,Hangup

I removed a '.' after my _44...

So does anyone know what

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 Auto fallthrough, channel 'Local/8600052@default-00000001;1' status is 'UNKNOWN'


...means?

Google and searching here talks about DID and ingroups... But i've not messed with any inbound calls.... just trying to get my first ring out on a new dialler

Thanks

[dailafing]

Ok, so another update....

I'm still trying to get m head around the DialPlan, im just using other peoples to be honest, and dont really understand the syntax....

So I figured I should change the X's to 44....

My current Dialplan...

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exten => _44,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _44,2,Dial(${orbtalk}/44${EXTEN:2},,tToR)
exten => _44,3,Hangup

And now the agent hears "I am sorry, that is not a valid extension", rather than the previous, connect, then disconnect sounds...
And the Asterisk looks different also...!

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[Sep 25 15:42:06]   == Manager 'sendcron' logged on from 127.0.0.1
[Sep 25 15:42:06]   == Manager 'sendcron' logged off from 127.0.0.1
[Sep 25 15:42:19]   == Manager 'sendcron' logged on from 127.0.0.1
[Sep 25 15:42:19]     -- Call accepted by 192.168.1.66 (format gsm)
[Sep 25 15:42:19]     -- Format for call is gsm
[Sep 25 15:42:21]        > Channel IAX2/103-9242 was answered.
[Sep 25 15:42:21]     -- Executing [8600052@default:1] MeetMe("IAX2/103-9242", "8600052,F") in new stack
[Sep 25 15:42:21]   == Parsing '/etc/asterisk/meetme.conf': [Sep 25 15:42:21]   == Found
[Sep 25 15:42:21]   == Parsing '/etc/asterisk/meetme-vicidial.conf': [Sep 25 15:42:21]   == Found
[Sep 25 15:42:21]     -- Created MeetMe conference 1023 for conference '8600052'
[Sep 25 15:42:21]     -- <IAX2/103-9242> Playing 'conf-onlyperson.gsm' (language 'en')
[Sep 25 15:42:22]   == Manager 'sendcron' logged off from 127.0.0.1
[Sep 25 15:42:25]   == Manager 'sendcron' logged on from 127.0.0.1
[Sep 25 15:42:25]     -- Executing [8600052@default:1] MeetMe("Local/8600052@default-00000002;2", "8600052,F") in new stack
[Sep 25 15:42:25]        > Channel Local/8600052@default-00000002;1 was answered.
[Sep 25 15:42:25]   == Starting Local/8600052@default-00000002;1 at default,441269****06,1 failed so falling back to exten 's'
[Sep 25 15:42:25]   == Starting Local/8600052@default-00000002;1 at default,s,1 still failed so falling back to context 'default'
[Sep 25 15:42:25]     -- Sent into invalid extension 's' in context 'default' on Local/8600052@default-00000002;1
[Sep 25 15:42:25]     -- Executing [i@default:1] Playback("Local/8600052@default-00000002;1", "invalid") in new stack
[Sep 25 15:42:25]     -- <Local/8600052@default-00000002;1> Playing 'invalid.gsm' (language 'en')
[Sep 25 15:42:26]   == Manager 'sendcron' logged off from 127.0.0.1
[Sep 25 15:42:29]     -- Executing [i@default:2] Hangup("Local/8600052@default-00000002;1", "") in new stack
[Sep 25 15:42:29]   == Spawn extension (default, i, 2) exited non-zero on 'Local/8600052@default-00000002;1'
[Sep 25 15:42:29]     -- Executing [h@default:1] AGI("Local/8600052@default-00000002;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Sep 25 15:42:29]     -- <Local/8600052@default-00000002;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
[Sep 25 15:42:29] WARNING[11920]: app_meetme.c:3956 conf_run: Unable to write frame to channel Local/8600052@default-00000002;2
[Sep 25 15:42:29]   == Spawn extension (default, 8600052, 1) exited non-zero on 'Local/8600052@default-00000002;2'
[Sep 25 15:42:29]     -- Executing [h@default:1] AGI("Local/8600052@default-00000002;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Sep 25 15:42:29]     -- <Local/8600052@default-00000002;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
Dialler2*CLI>


I hope this makes sence to someone, thanks.

[omarrodriguezt]

In the asterisk CLI run this command: sip show peers like orbtalk

Also, ask the carrier whaT is the format they are expecting for calls, like 44 +phone or 044 +phone, etc

[dailafing]

In the asterisk CLI run this command: sip show peers like orbtalk

Also, ask the carrier whaT is the format they are expecting for calls, like 44 +phone or 044 +phone, etc

Quick update.
I already phoned them, and they remoted in to have a look about. I covered number syntax with them as that's what I thought the issue was. Besides, I already tried different configurations of number presentation.
But they say, although it's "registered" with them, they are not receiving any placed calls from my dialler, from their side of the trunk.

I'll try the command suggested shortly, however they did similar commands when they were in control, and found that it was indeed registering. They may have read more than that from the readouts, but they were unable to highlight to me any direction to what could be wrong.

At one point he said my phone may be set up incorrectly. But since when I log in, my zoiper rings just fine, I can hear the "only person.gsm", and I can hear the connected/hangup sounds playing as the failed attempts are being made.
So, the phone is set up right, right? Or are there are other phone settings that could interfere with how calls are placed?

Did you see my dial plan?
I've this feeling it may be the syntax of the dial plan?
I know nothing about how it works, or what is even does?
It's just a copy and paste...
So any pointers on that would help.

Thanks

[williamconley]

/etc/asterisk/sip.conf

check the value of "externip"

also be sure your dialer can ping their server.

verify the "hostname" and that the dialplan (of the Admin->Carrier) coincides with the Dial Prefix of the campaign when combined with the Dial Code and Phone Number being dialed.

Most importantly: Post a CLI output example which will tell us what's going wrong. From the moment the call starts until it ends, but just THIS CALL, not 3000 lines of unrelated code. LOL

And remember that you have altered your dialplan since the earlier entry, so previous posts are not helpful for what may be wrong NOW.

[dailafing]

/etc/asterisk/sip.conf

check the value of "externip"

What should this be, or are you saying I should post it here.?

also be sure your dialer can ping their server.

It can.

verify the "hostname" and that the dialplan (of the Admin->Carrier) coincides with the Dial Prefix of the campaign when combined with the Dial Code and Phone Number being dialed.

I've seen several uses of the term host name across vicidial.
The name I gave the dialler, during yast lan first run? I called it Dialler2. And then another use is as my carrier name?
And yes, my dial prefix is set to 44, just as my dial plan.

Most importantly: Post a CLI output example which will tell us what's going wrong. From the moment the call starts until it ends, but just THIS CALL, not 3000 lines of unrelated code. LOL

The CLI above is the latest one on my current configuration. Did you see the error?

And remember that you have altered your dialplan since the earlier entry, so previous posts are not helpful for what may be wrong NOW.

I kept them there since other people who have downloaded the same disk as myself, and followed the book to the letter, may also come in to the same chain of events as have I, and would there for benefit from reading the progress of this issue. Additionally, as this thread progresses, there will be different errors along the way, and those additionally people also going through similar issues, may be at different stages of the issue depending on their experience, and depending on what stage they are at in the error before turning to the Internet for help, well they are then more likely to find this post if I am more verbose in my error progression.
It also leto the wonder full contributors of the fourm, like your kind self, diagnose my issue if they can see my previous attempts.

Thank not a rant, just the only way an Aspie can explain my train of thought.
Your help is much appreciated

[dailafing]

Ok quick update...

I managed to get my working setting from my old Dialler, and the DialPlan looks like this:

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

And my Acc Entry Looks like this:

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[orbtalk_wmsx]
username=<MYUSERNAME>
secret=<MYPASSWORD>
type=friend
nat=no
qualify=yes
insecure=port,invite
dtmfmode=RFC2833
disallow=all
allow=ulaw
allow=alaw
port=5060
context=trunkinbound
host=sipgw3.orbtalk.co.uk
fromip=sipgw3.orbtalk.co.uk
fromdomain=sipgw3.orbtalk.co.uk

But I'm still getting this error which MUST be an error in the ViciBox system!!!

Code: Select all

[Sep 30 10:46:17]   == Manager 'sendcron' logged off from 127.0.0.1
[Sep 30 10:46:23]     -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00000005;2", "8600051,F") in new stack
[Sep 30 10:46:23]        > Channel Local/8600051@default-00000005;1 was answered.
[Sep 30 10:46:23]     -- Executing [12*******6@default:1] AGI("Local/8600051@default-00000005;1", "agi://127.0.0.1:4577/call_log") in new stack
[Sep 30 10:46:23]     -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=1001))
[Sep 30 10:46:23]     -- <Local/8600051@default-00000005;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Sep 30 10:46:23]     -- Executing [12*******6@default:2] Dial("Local/8600051@default-00000005;1", "SIP/1234:PASSWORD@sip.provider.net/12*******6,,tTor") in new stack
[Sep 30 10:46:23] NOTICE[6349]: chan_sip.c:27535 sip_request_call: Conflicting extension values given. Using '1234' and not '12*******6'
[Sep 30 10:46:23]   == Using SIP RTP CoS mark 5
[Sep 30 10:46:23] ERROR[6349]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("sip.provider.net", "(null)", ...): Name or service not known
[Sep 30 10:46:23] WARNING[6349]: chan_sip.c:5711 create_addr: No such host: sip.provider.net
[Sep 30 10:46:23] WARNING[6349]: app_dial.c:2345 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[Sep 30 10:46:23]   == Everyone is busy/congested at this time (1:0/0/1)
[Sep 30 10:46:23]     -- Executing [1269268906@default:3] Hangup("Local/8600051@default-00000005;1", "") in new stack
[Sep 30 10:46:23]   == Spawn extension (default, 1269268906, 3) exited non-zero on 'Local/8600051@default-00000005;1'
[Sep 30 10:46:23]     -- Executing [h@default:1] AGI("Local/8600051@default-00000005;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Sep 30 10:46:23]     -- <Local/8600051@default-00000005;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL---------- completed, returning 0
[Sep 30 10:46:23]   == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-00000005;2'
[Sep 30 10:46:23]     -- Executing [h@default:1] AGI("Local/8600051@default-00000005;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Sep 30 10:46:23]     -- <Local/8600051@default-00000005;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Sep 30 10:46:24]   == Manager 'sendcron' logged off from 127.0.0.1




Any Ideas what these errors mean?

Code: Select all

 NOTICE[6349]: chan_sip.c:27535 sip_request_call: Conflicting extension values given. Using '1234' and not '12*******6'

[omarrodriguezt]

In the CLI run

sip show peers

and confirm that your carrier peer is OK
also contact the carrier and ask them if they can see the call in their system

[dailafing]

It's my dial plan...
Working now...
Also will post a copy of my working dialplan shortly...

But from memory, the biggest change I think is that the campaign dial extention was x and so was the dial plan. Now they are both 9....