vicidial.org

Hello,
I am facing a problem with the calls when they go out of confrence in pridictive dialing. when i make a call from my softphone ie xlite using the extension 201 i made i can do the call but when it enters into the confrence ie 8600051 the call doesnt get connnected .I am using the viciserver.

I am posting my asterisk cli when making a call from sip phone .


Executing NoOp("SIP/201-081c01e8", "CALEERID===> 33686511214") in new stack
-- Executing Dial("SIP/201-081c01e8", "SIP/voiptalk/33686511214|30") in new stack
-- Called voiptalk/33686511214
== Spawn extension (default, 33686511214, 2) exited non-zero on 'SIP/201-081c01e8'
-- Executing DeadAGI("SIP/201-081c01e8", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0


when i am using the confrence the calls output is

Executing MeetMeAdmin("Local/55558600051@default-5ccb,2", "8600051|K") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
Sep 7 04:03:12 NOTICE[29583]: app_meetme.c:2210 admin_exec: Conference Number not found
-- Executing Hangup("Local/55558600051@default-5ccb,2", "") in new stack
== Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-5ccb,2'
-- Executing DeadAGI("Local/55558600051@default-5ccb,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
> Channel SIP/201-081c01e8 was answered.
-- Executing MeetMe("SIP/201-081c01e8", "8600051|F") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600051@default-ed28,2", "8600051|F") in new stack
> Channel Local/8600051@default-ed28,1 was answered.
-- Executing NoOp("Local/8600051@default-ed28,1", "CALEERID===> ") in new stack
-- Executing Dial("Local/8600051@default-ed28,1", "SIP/voiptalk/|30") in new stack
-- Called voiptalk/
-- SIP/voiptalk-08213d48 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup("Local/8600051@default-ed28,1", "") in new stack
== Spawn extension (default, 033686767523, 3) exited non-zero on 'Local/8600051@default-ed28,1'
-- Executing DeadAGI("Local/8600051@default-ed28,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----1-----CONGESTION----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-ed28,2'
-- Executing DeadAGI("Local/8600051@default-ed28,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0




my sip carrier conf are

[voiptalk]
type=friend
username=user
secret=pass
fromuser=user
host=ipaddservice_provider
context=arsh
insecure=very


for my dial entry is

exten => _X.,1,Noop(CALEERID===> ${CALLERID(dnid)})
exten => _X.,2,Dial(SIP/voiptalk/${CALLERID(dnid)},30)
exten => _X.,3,Hangup

I have not made any changes in the extensions.conf and sip.conf

Please help me as i m having a big problem.

That's a major problem alright. Try to create the appropriate dialplan for the number you're calling instead of doing a catch all statement.

exten => _X.,2,Dial(SIP/voiptalk/${CALLERID(dnid)},30)

That seriously messes up your Vicidial system.

arshsaini wrote:when i am using the confrence the calls output is

Executing MeetMeAdmin("Local/55558600051@default-5ccb,2", "8600051|K") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
Sep 7 04:03:12 NOTICE[29583]: app_meetme.c:2210 admin_exec: Conference Number not found
-- Executing Hangup("Local/55558600051@default-5ccb,2", "") in new stack
== Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-5ccb,2'
-- Executing DeadAGI("Local/55558600051@default-5ccb,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
> Channel SIP/201-081c01e8 was answered.
-- Executing MeetMe("SIP/201-081c01e8", "8600051|F") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600051@default-ed28,2", "8600051|F") in new stack
> Channel Local/8600051@default-ed28,1 was answered.
-- Executing NoOp("Local/8600051@default-ed28,1", "CALEERID===> ") in new stack
-- Executing Dial("Local/8600051@default-ed28,1", "SIP/voiptalk/|30") in new stack
-- Called voiptalk/
-- SIP/voiptalk-08213d48 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup("Local/8600051@default-ed28,1", "") in new stack
== Spawn extension (default, 033686767523, 3) exited non-zero on 'Local/8600051@default-ed28,1'
-- Executing DeadAGI("Local/8600051@default-ed28,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----1-----CONGESTION----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-ed28,2'
-- Executing DeadAGI("Local/8600051@default-ed28,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

What shall i do more now?

I am new to vicidial and i don't know how to make the dial plan.
I made the changes and this one worked out.
but for confrence it is not working.

Someone help .

I suggest you follow the VicidialNOW Getting Started Guide: http://carlo.taguinod.net/vicidialnow/v ... -guide.pdf . Afterwards, purchase the complete Manager's and Agent's Manuals here: http://www.vicidial.org/store.php .

I went to this and did all the same config using the starting guide but its not working.

Hello,

Please work on your dial plan very well. From all indication, your dial plan is not fitting into your operator's dial plan. I may be able to help if you let me have some information about your operator. Once you miss the dial plan from you operator, you may still have that challenge most especially when you are placing an outbound call where the dial plan is different from those of US, UK, Germany, Australia which are already hard coded at the bank end for it to work in those countries.

My advice, know the numbering plans (not dial plans) of the outbound calls you then use it to do your outbound dial plan. Please go to /etc/asterisk then vi extensions.conf, you will see a lot of examples though mainly for US. Then use it to write your own.

Wish you the best

I am using the Sip account and it is getting registered as i can make calls from the sip phone using vicidial and i ahve seen by cli that it gets registered .
As much as i see my default context has the problem . Can some one give me extensions.conf so that i can put in my vicidial and can get through it.

there exist extensions.conf in /etc/asterisk/extensions.conf. What matters most is how your SIP provider wants you to make a call. With that you can design your dial plan. Sending extensions.conf from the default does not matter as long as the dial plan will not work for you.

Can i have the sample dial plan for dialing the calls to France.

Hello,

I am not to sure I can help as neither am I based in France nor know the numbering plan for France. However, I can help in two other proxy ways.

1. Give you the dial plan for my country and you use it to build for France

2. Give me various operators numbering plan or generic numbering plan in France.

For my country, Nigeria, there two major numbering plans

A. 080 and 070 GSM/CDMA 11 digits Line

B. Seven digit POTS Numbering plans

For the 080 and 070, the numbering plan will be _XXXXXXXXXXX e.g 08023175974, 07029989498

For the seven digit POTS numbers, the numbering plans will be

i. _XXXXXXX for local calls e.g 5551166

ii _XXXXXXXX for trunk calls with single digit trunk code e.g 1-5551166

iii _XXXXXXXXX for trunk calls with double digit trunk code e.g 02-5551166

iv _XXXXXXXXXX for trunk calls with triple digit trunk code e.g 123-5551166

If you can give me various operators numbers in France or generic national numbering plan, I can help you come up with a numbering plan. In addition, you will also need the numbering plan of your operators as some use some prefix for off net calls.

Also go to the extensions.conf in /etc/asterisk to see some examples created there. It will go a long way to help.

Hope this helps a bit.

there is the extension is country code 33
just we need country code +33 6 88 27 27 42


these is france no

Hi

You can use this _XXXXXXXXX (9 Digits). However, you need to confirm that your operator does not have any dial prefix. In addition, please confirm if 6 is an Trunk (area) code, if yes, is there any area code or trunk code that is double digit, if yes then add an additional X, making the dial plan to be _XXXXXXXXXX (10).

Also note that dial plan can be more than one depending on who and how you plan to dial. Again, go to the /etc/asterisk/extensions.conf you will see various examples there.

If you need to create more than one dial plan, then, I advice you write it at the back end through the extensions.conf

By the way, what access protocol are you using SIP, IAX2 or ZAP?

For IAX2/SIP you can configure most of the stuff at the front end, but for ZAP you need to do some work at the back end, starting from configuring zaptel.conf, to zapata.conf

Wish you the best.

Can i know what shall i put up in my dial entry .
As i ahve tried this also its not working !!!!

Hi Please let me have your skype so that we can chat. I believe we can solve the problem based on this.

My skype id is arsh.saini