vicidial.org

by **nitinsrivastava** » Tue Jun 27, 2006 7:28 am

hi,
i have seen that mysql is running but asterisk isnot running. now how to make it run.

by **mflorell** » Tue Jun 27, 2006 7:37 am

Are you sure you read the SCRATCH_INSTALL:

### start up asterisk
/home/cron/start_asterisk_boot.pl

If it does not start then you should start it manually and post the errors:
asterisk -vvvvvvvvvvvvvvvvvvvvvgc

by **nitinsrivastava** » Tue Jun 27, 2006 7:41 am

hi,
i have started manually but there is error

Jun 27 18:09:47 WARNING[27910]: chan_zap.c:923 zt_open: Unable to specify channel 1: Device or resource busy
Jun 27 18:09:47 ERROR[27910]: chan_zap.c:6878 mkintf: Unable to open channel 1: Device or resource busy
here = 0, tmp->channel = 1, channel = 1
Jun 27 18:09:47 ERROR[27910]: chan_zap.c:10314 setup_zap: Unable to register channel '1-2'
Jun 27 18:09:47 WARNING[27910]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1
Jun 27 18:09:47 WARNING[27910]: loader.c:554 load_modules: Loading module chan_zap.so failed!
Ouch ... error while writing audio data: : Broken pipe
Junk at the beginning 49443303
Warning, flexibel rate not heavily tested!

by **mflorell** » Tue Jun 27, 2006 9:05 am

Looks like there is a problem with your zaptel driver. What zaptel card are you using?

If none, what are you using for your zaptel timer?

What kernel are you running on?

by **nitinsrivastava** » Tue Jun 27, 2006 9:31 am

Hi,
I am not using any zaptel device, i don't know where is to set zaptel timmer, please tell me details where and what to change.
regarding your question for kernel
Linux version 2.6.9-34.EL (buildcentos@build-i386) (gcc version 3.4.5 20051201 (Red Hat 3.4.5-2))

by **mflorell** » Tue Jun 27, 2006 9:59 am

Then you must be using ztdummy. What are you using for your ztdummy timer? the USB driver or the kernel timer?

by **Nitin** » Tue Jun 27, 2006 10:25 am

I am not sure. How do I find out.

Thanks

by **mflorell** » Tue Jun 27, 2006 10:47 am

I am not an expert on ztdummy(we recommend getting at least an x100p card for timing), but here is the voip-info.org page on the subject:
http://www.voip-info.org/wiki-Asterisk+timer+ztdummy

by **enjay** » Tue Jun 27, 2006 11:19 am

Just compile Zaptel and use the ztdummy kernel module

after compiled (make clean, make, make install) run modprobe ztdummy

pretty sure thats all that needs to be done.

-enjay

by **mmerani** » Tue Jun 27, 2006 2:40 pm

btw.. for those venturing into the wild beyond a fair warning...

I had contacted everyone on the page mflorell recommended, only one contacted me back, we hired this firm, they installed a complex environment for us and when they could not make it work.. they abandoned us after thousands of dollars of expense. We are limping along with the vicidialer and trying to get our sysadmin up to speed so we are back to doing it the hard way.

If there are any REAL consultants monitoring these boards.. please contact us.. we need an experienced person to make this environment work.

by **mflorell** » Tue Jun 27, 2006 4:28 pm

Please let me know which company you worked with off-forum.

As for the companies on the list, I have talked with them recently and they are all very busy. I am guessing they they may be picking their favorite or most profitable clients to take work from and dropping the rest.

It is difficult to find quality consultants for Asterisk right now since we are all very much in demand and, if they are like me, they have a few enterprise-level clients that demand our attention and pay very well for that attention at a moment's notice. This means that the smaller clients do get pushed aside from time to time.

I currently am only taking a few new small clients that will work with my schedule. I do consulting for $200/hour if you are interested, and since I wrote almost all of VICIDIAL I can usually get things running rather well for most of my clients.

If there are any consultants out there that would like to be added to the consultants list please contact me off-forum.

by **nitinsrivastava** » Wed Jun 28, 2006 7:06 am

Hello,
I have reconfigured my PBX and VICIDIALER. I think it is running 90% correctly as when webclient logins with his UID/pwd,Phoneno/pwd there is one incoming ring to my SIP softphone atteched with pbx. But when i try to make call manual then it shows waiting for ring for 50 sec and then call get disconnected showing server time out. But when click on Call Agent Again link then it rings my softphone.
I am using this link http://MYIP/agc/vicidial.php I think some thing more required to make calls. I am not using any TDM400P card, i am using H323 codec means VOIP to directly transfer calls from my PBX.
Please help me soon.
With regards,
Nitin Srivastava

by **mflorell** » Wed Jun 28, 2006 7:10 am

VICIDIAL does not work with H323.

VICIDIAL only works with Zap, IAX or SIP channels.

by **nitinsrivastava** » Wed Jun 28, 2006 7:13 am

hi,
My PBX is working on SIP channel, But i think that there is little problem somwhere in the Config. files because it is waiting for ring on the webform.

by **mflorell** » Wed Jun 28, 2006 7:23 am

Please post the Asterisk CLI output of the call being placed as well as the output from this command line command:
'screen -r'

by **nitinsrivastava** » Wed Jun 28, 2006 7:30 am

[root@asterisk asterisk]# screen -r
There are several suitable screens on:
3938.ASTsend (Detached)
3952.ASTupdate (Detached)
3943.ASTVDauto (Detached)
3949.ASTVDremote (Detached)
3929.ASTlisten (Detached)
Type "screen [-d] -r [pid.]tty.host" to resume one of them.

by **Nitin** » Thu Jun 29, 2006 12:21 am

Below is the Asterisk CLI output that I am getting:

[root@asterisk asterisk]# screen -r
There are several suitable screens on:
3938.ASTsend (Detached)
3952.ASTupdate (Detached)
3943.ASTVDauto (Detached)
3949.ASTVDremote (Detached)
3929.ASTlisten (Detached)
Type "screen [-d] -r [pid.]tty.host" to resume one of them.

I am unable to dial out using the dialer.

by **mflorell** » Thu Jun 29, 2006 3:39 am

That is not Asterisk CLI output.

I need to see the Asterisk CLI output from the "asterisk -vvvvvvvvvvvvvvvvvvvvgc" command while you are trying to dial.

by **nitinsrivastava** » Thu Jun 29, 2006 5:15 am

Hello,

asterisk *CLI output
-- Accepting AUTHENTICATED call from 59.144.165.12:
> requested format = gsm,
> requested prefs = (),
> actual format = ulaw,
> host prefs = (ulaw|alaw|gsm),
> priority = mine
-- Executing Macro("IAX2/6666-2", "dialout-trunk|2|19497330418||") in new stack
-- Executing GotoIf("IAX2/6666-2", "1?3:2") in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("IAX2/6666-2", "user-callerid") in new stack
-- Executing Set("IAX2/6666-2", "AMPUSER=6666") in new stack
-- Executing Set("IAX2/6666-2", "EMERGENCYCID=") in new stack
-- Executing Set("IAX2/6666-2", "AMPUSERCIDNAME=6666") in new stack
-- Executing GotoIf("IAX2/6666-2", "0?6") in new stack
-- Executing Set("IAX2/6666-2", "CALLERID(all)="6666" <6666>") in new stack
-- Executing NoOp("IAX2/6666-2", "Using CallerID "6666" <6666>") in new stack
-- Executing Macro("IAX2/6666-2", "record-enable|6666|OUT") in new stack
-- Executing GotoIf("IAX2/6666-2", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("IAX2/6666-2", "recordingcheck|20060629-154135|1151575895.60") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060629-154135|1151575895.60: Outbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("IAX2/6666-2", "No recording needed") in new stack
-- Executing Macro("IAX2/6666-2", "outbound-callerid|2") in new stack
-- Executing Set("IAX2/6666-2", "USEROUTCID=") in new stack
-- Executing GotoIf("IAX2/6666-2", "0?4") in new stack
-- Executing Set("IAX2/6666-2", "CALLERID(all)=2132912210") in new stack
-- Executing GotoIf("IAX2/6666-2", "1?6") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing GotoIf("IAX2/6666-2", "1?8") in new stack
-- Goto (macro-outbound-callerid,s,8)
-- Executing NoOp("IAX2/6666-2", "CallerID set to "" <2132912210>") in new stack
-- Executing Set("IAX2/6666-2", "GROUP()=OUT_2") in new stack
-- Executing GotoIf("IAX2/6666-2", "0?108") in new stack
-- Executing Set("IAX2/6666-2", "DIAL_NUMBER=19497330418") in new stack
-- Executing Set("IAX2/6666-2", "DIAL_TRUNK=2") in new stack
-- Executing AGI("IAX2/6666-2", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("IAX2/6666-2", "OUTNUM=70999719497330418") in new stack
-- Executing Set("IAX2/6666-2", "custom=SIP/outbounding") in new stack
-- Executing GotoIf("IAX2/6666-2", "0?16") in new stack
-- Executing Dial("IAX2/6666-2", "SIP/outbounding/70999719497330418|120|W") in new stack
-- Called outbounding/70999719497330418
-- SIP/outbounding-1692 is ringing
-- SIP/outbounding-1692 is making progress passing it to IAX2/6666-2
-- SIP/outbounding-1692 answered IAX2/6666-2
This out put we get while dialing through softphone.

by **nitinsrivastava** » Thu Jun 29, 2006 5:28 am

Hi,
My VICIDIALER OUTPUT IS GIVEN DOWN
Asterisk*CLI>
Jun 29 15:56:55 NOTICE[20126]: chan_local.c:494 local_alloc: No such extension/context 8600051@default creating local channel
Jun 29 15:56:55 NOTICE[20126]: channel.c:2443 __ast_request_and_dial: Unable to request channel Local/8600051@default/n

by **mflorell** » Thu Jun 29, 2006 6:59 am

Are you using freepbx?

It's going to take you some work to get that working with all of this callerID altering that you are going to have to delete from your dialplans.

The problem seems to be that you don't have the 8600051-8600100 extens in your default context. Can you confirm that they are in there?

by **nitinsrivastava** » Thu Jun 29, 2006 7:44 am

Hi,
I have all that numbers in my dial plan,means extensions.conf file of /etc/asterisk but i have that error now too and output on this url
http://MYIP/agc/vicidial.php
STATUS: Calling: 7147423863 UID: M0629181026000000020 Waiting for Ring... 25 seconds

Asterisk*CLI>
Jun 29 18:10:29 NOTICE[25693]: chan_local.c:494 local_alloc: No such extension/context 8600051@default creating local channel
Jun 29 18:10:29 NOTICE[25693]: channel.c:2443 __ast_request_and_dial: Unable to request channel Local/8600051@default/n

by **mflorell** » Thu Jun 29, 2006 8:05 am

Do you have those entries in your meetme.conf file?

by **nitinsrivastava** » Thu Jun 29, 2006 8:09 am

hi,
yes i have those settings in my meetme.conf file too. If u want to see it in GUI through web browser then i can give u my URL, to access it from ur side.

by **mflorell** » Thu Jun 29, 2006 8:46 am

Your issue is with the dialplan. That messge means that Asterisk is not finding the 8600051 exten in the [default] context.

I ask you again, are you using freepbx?

by **nitinsrivastava** » Thu Jun 29, 2006 8:54 am

Hi
yes i am using freepbx.

by **nitinsrivastava** » Thu Jun 29, 2006 9:05 am

Hi,
i have gone through my extensions.conf file, here is the section of default.

[default]
include => ext-local
exten => s,1,Playback(vm-goodbye)
exten => s,2,Macro(hangupcall)

what to be added here please specify in detail.

by **mflorell** » Thu Jun 29, 2006 9:07 am

freepbx creates a mess of dialplan rules that cause all sorts of problems for VICIDIAL. The biggest one is the callerID issue where freepbx overrides the callerID set by VICIDIAL unless you setup several special rules in freepbx so that it will not mess with callerID on VICIDIAL calls. The issue you are having with it is unrelated to the callerID issue and I do not know how to fix that either within freepbx.

We are working on testing freepbx with VICIDIAL but we still haven't gotten it to work. When we do, we will post our results on this forum. Until then I suggest you remove freepbx and use basic conf files.

by **nitinsrivastava** » Thu Jun 29, 2006 9:13 am

Hi,
then let me know how to make vicidialer work, without FREEPBX (Asteriak) Version 2.0.1 . what changes are required to make it run without any PBX or some other PBX. If it works with some other pbx then let me know that which PBX i must install to make it run.
My Asterisk version is 1.2.7.1
There must be some solution to make VICIDAILER active at this stage too. please reply.
with regards,
Nitin Srivastava

by **mflorell** » Thu Jun 29, 2006 9:30 am

All I can suggest is to follow the SCRATCH_INSTALL document. If you follow it your system will work.

It would probably be much easier and faster for you to hire a consultant to install VICIDIAL for you on your system.

by **nitinsrivastava** » Thu Jun 29, 2006 9:52 am

i have told what errors and outputs are coming there. I want to know that will it work by doing some entries in some config files or it will not work at all. Since i am folliowing ur scratch_insatll file from begining, i have done all the changes u have suggested me, but it really not started. But why there is one call to my SIP softphone when Agent logins to web form using user id/ pwd, phone_no/pwd. It is amazing that call is coming from web form to my Softphone but when trying to make out calls by manual dialing or auto dialing then it is waiting for ring and finally ring time outs. Where is the basic problem here, is it in PBX Configuartion or VICIDAILER configuration.
reoply soon.

by **mflorell** » Thu Jun 29, 2006 10:00 am

If you have followed the SCRATCH_INSTALL I would like to know where on that document it says to install freepbx?

Your problem is with your dialplan(extensions.conf). You need to delete it and all other freepbx files that you installed and start again from PHASE 5 in the SCRATCH_INSTALL document.

by **nitinsrivastava** » Thu Jun 29, 2006 10:04 am

hi,
i was using FreePBX before the installation of the vicidialer. I have just followed all the steps u have said from begining of installation of VICIDIALER.but i have never removed my free pbx at all. According to you what should i do. I have earlier told you that i am not a linux guy, so tell me what and how to do all changes required to make successfull run of this VICIDIALER.

by **mflorell** » Thu Jun 29, 2006 10:44 am

You need to delete all extensions files that are installed and start again from PHASE 5 in the SCRATCH_INSTALL document.

by **nitinsrivastava** » Thu Jun 29, 2006 10:46 am

means i must delete extensions.conf file and Sip.conf, iax2.conf files only.
reply soon

by **mflorell** » Thu Jun 29, 2006 12:39 pm

Yes, you might want to keep copies just in case you need to refer to them if you have already customized some things.

by **marvin** » Thu Jun 29, 2006 12:52 pm

ic thats why.... i guess ur using aah or trixbox
go to macro-dialout-trunk look for

exten => s,15,Dial(${OUT_${ARG1}}/${OUTNUM}) ; Regular Trunk Dial

and change it to

exten => s,15,Dial(${OUT_${ARG1}}/${OUTNUM},55,to) ; Regular Trunk Dial

and add those extensions in extension_custom.conf

by **nitinsrivastava** » Thu Jun 29, 2006 1:50 pm

Hi,
Sorry for twice posts, here is the output after the reconfiguration. please see it.
Asterisk*CLI>
== Starting SIP/7777-213f at default,8600051,1 failed so falling back to exten 's'
== Starting SIP/7777-213f at default,s,1 still failed so falling back to context 'default'
-- Sent into invalid extension 's' in context 'default' on SIP/7777-213f
-- Executing Playback("SIP/7777-213f", "invalid") in new stack
-- Playing 'invalid' (language 'en')
-- Invalid extension '7' in context 'default' on SIP/7777-213f
== CDR updated on SIP/7777-213f
-- Executing Playback("SIP/7777-213f", "invalid") in new stack
-- Playing 'invalid' (language 'en')
-- Invalid extension '7' in context 'default' on SIP/7777-213f
== CDR updated on SIP/7777-213f
-- Executing Playback("SIP/7777-213f", "invalid") in new stack
-- Playing 'invalid' (language 'en')

by **nitinsrivastava** » Fri Jun 30, 2006 11:25 am

Hi,
I have reinstalled all again. Now when i use to run asterisk -r then it does not connects to it
when i use to start Asterisk manually by command asterisk -vvvvvvvvvvvvgc
then i get this error
Jun 30 21:51:49 WARNING[7261]: manager.c:1742 init_manager: Unable to bind socket: Address already in use

where is problem, reply me soon

by **enterux** » Fri Jun 30, 2006 11:39 am

Hey Nitin,

I read through your entire post and figured that you didnt install astGUIClient / Vicidial from scratch.

And now that the entire thing has become lot more complicated

If you still care, do write to me, I provide professional consulting on setting up Vicidial and also we are official Digium Hardware Reseller and can help you with the stuff.

Thanks & Regards,
Mitul Limbani,
Enterux Solutions,
www.enterux.com
(P.S.: Send me a PM if you really need some help badly, and are willing to pay for the same)

vicidial.org

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Mflorell says hire consultants.. well we did..

Unable to Dial Out

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