vicidial.org

guys plz help me

I installed vnow1.2 when i receive the call in x-lite I can here the sound like
you are the only preson the in this conference
But when i dial the next number and click on Dial leads I get the error sound saying " I am sorry that is not a valid extension"

plz help me

Hi softlin! Give us more info so we could help you.

Actually I forgot to give extension no so that was the problem
thx for esponsing

plzzz help me with this as i am also facing same problem .......... i am dialing to uk and my voip provide coad is 00 so wht should....... i had configure in dialing coad =44 and vendor coad =00 and i am facing same problem plzzzz help me with this

i have encountered this one before. it was just when i changed the format of the numbers i loaded in the leads solved it.

make sure that your numbers should match on what is in your extension configuration. also on the campaign settings, the dial prefix.

hope this one might help you.

i had cheq many times but facing same problem can anyone help me with this my dial prifix is 9 in managar

you must find from your provider exactly what your "dialed number" should be (do they expect 011445551112222?) then look at your CLI> in asterisk and see if that is what you are dialing.

if you have "9" in the campaign and "44" in the "dial code" of the lead and a 10 digit number, then you need to check and see what actually comes out (this is also affected by exten:1, which will strip off the "9" because :1 means strip one digit off the front). so 9445551112222 becomes 445551112222, but what if they want 00445551112222? then you need to change your "dial code" on each lead to "0044" instead of "44" (or temporarily change your "9" to "900" until you can load the leads properly).

compare your asterisk CLI "dial" command with what the provider wants to receive in the dial command.

sir i had change my voip provider so i can make call using 44

so wht should i update in leads dial coad=44 or dial code =944


or should i have to add any vendor code ?

if you want to have an answer to that question, my first instinct is to say 44, but really you have not given what us what your provider wants. it could be in a format like this:
44NXXNXXXXXX

The N's and X's represent the actual phone number to be dialed. Any real numbers would be the "phone code" (in this case 44).

They may use many other formats, but you should share with us the exact request. They may even say "Dial(XXXXXXXXXXXXXX,tTor)" or something else entirely. But if all they said was "dial 44" you may want to look for a more professional / helpful carrier.

the new voip provider use 44 only i will cheq that and let u know .....

LOL. They can dial "UK" only, but they can't dial "44" only. It's like having a rug salesman (he only has one rug and can't sell it, he'd be out of business). They have to dial numbers IN the UK, and accept a dial plan accordingly, such as "0044XXXXXXXXXX" or "44XXXXXXXXX" or something. If they say "10 digit dialing prefixed with 44" that's fine. But something ...

If you use something like _44.,2,Dial, that will match anything with a 44 beginning and you don't have to worry about matching specifics. Also, check with your provider the exact format they want the number in, e.g. if they want only 44+NUMBER and you are sending 0044+NUMBER it won't work until you strip the leading 00 from the dial string.

i had confirm with voip provider they support 44 as uk code and then number
eg 441934398492

i try this but still same problem ???

show the CLI output from the attempt (you will see a "dial" command ... and then either it will reject the call or accept the call ...). You can also (if the call is rejected or accepted) use "sip debug" to see the sip packet headers which contain useful debug information (generally a 3 digit code followed by its english translation, like 200 OK or 404 not found ...)

if you don't get a "dial" command on your CLI, then you are not dialing a number that matches your setup and your system won't dial until you match your own setup (dial prefix + dial code + number must match a pattern available in extensions.conf or an included file in the proper context)

Hi All,

Please I need some clarifications. Does it mean that vicidial only uses SIP for channel dialing? I have an FXO card to use for my service but from all indications, most of the conversations are on SIP.

My FXO card installed but whenever I dial into it, it gives me I am sorry this is not a valid extension.

These are my configurations:

1. I am running vicidialnow version 1.2

2. I am running it on a 4 port FXO card

3. /etc/asterisk/zapata.conf
[channels]
context=pstn-incoming
signalling=fxs_ks
channel =>1,2,3,4

4. /etc/zaptel.conf
Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1"
fxsks=1
fxsks=2
fxsks=3
fxsks=4
loadzone = us
defaultzone = us

5 /etc/asterisk/extension.conf
[pstn-incoming]
exten => _106.,1,Answer()
exten => _106.,2,Dial(newsip:test@192.168.19.2:5060,20,tr)

Please note, the signal to the FXO card is from my PBX

6. This is my asterisk -r screen

Connected to Asterisk 1.2.30.2 currently running on vici (pid = 2794)
Verbosity is at least 5
-- Remote UNIX connection
-- Starting simple switch on 'Zap/4-1'
Jul 23 15:58:18 NOTICE[3252]: chan_zap.c:6248 ss_thread: Got event 18 (Ring Begin)...
Jul 23 15:58:19 NOTICE[3252]: chan_zap.c:6248 ss_thread: Got event 2 (Ring/Answered)...
Jul 23 15:58:23 NOTICE[3252]: chan_zap.c:6248 ss_thread: Got event 18 (Ring Begin)...
== Starting Zap/4-1 at pstn_incoming,s,1 failed so falling back to exten 's'
== Starting Zap/4-1 at pstn_incoming,s,1 still failed so falling back to context 'default'
-- Sent into invalid extension 's' in context 'default' on Zap/4-1
-- Executing Playback("Zap/4-1", "invalid") in new stack
-- Playing 'invalid' (language 'en')

The FXO call went it but could not be routed to the appropriate extensions. Does it matter that all channels must be based on either SIP or IAX?

7. I have configured by DID with number 106 as I am connecting from my PBX.

I will appreciate your immediate response.

Thanks

oshonubi:

Sorry noone got back to you on this one, but I think you jumped into a non-related thread because it had the same error message you were getting.

In answer to one of your questions, no it is not required that SIP or IAX be used at all.

The rest of your questions are hard to understand. So we will begin by giving some advice on how best to use this forum to get quick answers to get your system working:

Read the "stickies" at the top of the forum. This is a requirement before posting (and often you will find your answer there). This will help you to get the best experience from this forum. You need to post your setup (hardware, software, vicidial version, and any OTHER software installed in this server). It's a good idea to put this information in your "signature" so you don't have to remember it each time (two lines should do it nicely).

Also read the Manager's Manual (required reading and very often resolves a lot of your issues) which is available on EFLO.net. It has free and paid versions, I recommend the Paid version, it will save you its purchase price very quickly. You will also find an installation guide for VicidialNOW which is very useful on the VicidialNOW site (if you are using VicidialNOW instead of Vicibox or scratch install of Vicidial).

If it is of any help, remember that with a "Card" you must be able to use the card with Asterisk for Vicidial to be able to make use of that card as a "trunk". So getting the card configured properly is a function outside Vicidial (you may be able to find much more help on getting the card to work properly from the seller or from an Asterisk forum). However, 4 port cards are usually very easily configured (but a cheap one that doesn't really work can waste a lot of your time).

Handle your issues one at a time as you go through the installation in the Manager's Manual and/or VicidialNOW guide and seriously do it one line at a time and STOP and get help if anything doesn't work properly. Do not wait until you have 5 problems and 10 questions, one step at a time from beginning to end and it will work.

Hi William,

Thanks for the response. I believe I followed the thread very well. Meanwhile, I was able to resolve the issue by understanding that the vicidial manual would help in zap configuration. I fell back on the asterisk skill. Be as it may the challenge has been resolved.

one of the possible solutions is to change in the campaign, the Prefix Dial 9 to X (for 91NXXNXXXXXX value would be 9, for no dial prefix use X), this worked wen i perform the migration to the new version.