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Asterisk 1.2 or 1.4?
Posted:
Sat Apr 12, 2008 2:41 am
by ssin14
Hi
I just want to know which version of asterisk works fine with vicidial with no problems.
Thanks
Thanks
Posted:
Sat Apr 12, 2008 7:47 am
by kupak
only 1.2
Posted:
Sat Apr 12, 2008 3:07 pm
by yeshuawatso
Does 1.4 work at all, because I've been messing with this for a week and I can't seem to get the manual dial in astaguiclient to work for outside lines. Although I think it's an asterisk issue I'd like to be sure that 1.4 will work at all.
Posted:
Sat Apr 12, 2008 5:27 pm
by i_magic
It does work, but it is not recommended due to stability issues
Posted:
Sat Apr 12, 2008 8:23 pm
by mflorell
Asterisk 1.4.19 is tested for stability and compatibility and it appears to work properly with VICIDIAL.
We still do not recommend it because it has not been out for very long, but after it has been in production for a while longer we will start to recommend it.
Remember that Asterisk 1.4 has some changes to the dialplan that need to be made for VICIDIAL to work properly as compared to the defaults in 1.2.
Posted:
Sun Apr 13, 2008 1:21 am
by yeshuawatso
mflorell wrote:....Remember that Asterisk 1.4 has some changes to the dialplan that need to be made for VICIDIAL to work properly as compared to the defaults in 1.2.
Is there any documentation on this issue as it seems to be my problem. Whenever a call is issued, the phone rings but then goes into demo mode. I've removed the "import demo" line which only made the lines hangup when forwarding.
I've purposely only created two extensions and two conference rooms to test the Vicidial setup process using two servers to separate Vicidial and Asterisk. This may not be the most ideal setup but it's how I want to run things in the future but I'm open to change. I've used the setup from the scratch install but left out all of the conference rooms and extensions that are loaded into the DB and extensions.conf.
If I can get this working my plan is to create an easier install system and setup. Something less complicated that the non-asterisk and linux gurus could do. I also have plans to rewrite the user interface and integrate a Java SIP/IAX phone into the browser. This will make things more "plug and play" but still hold the capability to use Vicidial like it is now. Of course all of these changes would be released back to the project.
Posted:
Sun Apr 13, 2008 2:44 pm
by mflorell
Could you include some Asterisk CLI output from when you are having problems?
agiout logfiles would be helpful too.
Posted:
Sun Apr 13, 2008 11:58 pm
by yeshuawatso
This is what happens when I try to use the manual dial of the AstGUIclient to numbers outside the PBX. I have tried this with the extensions I created using the scratch install and the extensions that I had prior to installing Vicidial.
Note the extension listed above can dial out using the softphone, just not with AstGUIclient.
- Code: Select all
== Starting IAX2/3011-1 at default,914792XXXXXX,1 failed so falling back to exten 's'
== Starting IAX2/3011-1 at default,s,1 still failed so falling back to context 'default'
[Apr 13 23:20:27] WARNING[17997]: pbx.c:2470 __ast_pbx_run: Channel 'IAX2/3011-1' sent into invalid extension 's' in context 'default', but no invalid handler
-- Hungup 'IAX2/3011-1'
I have read in prior post that this is a dial plan issue, however I'm not sure how to resolve it considering that manually dialing using the softphone works.
Posted:
Mon Apr 14, 2008 12:23 am
by mflorell
Looks like you don't have "default,914792XXXXXX" in your extensions .conf
Do you have _91NXXNXXXXXX in your dialplan?
Posted:
Mon Apr 14, 2008 9:27 pm
by yeshuawatso
Well looks like I got things to work just fine. But I figured I might let some people know what to do for my scenario.
For those who are using the AsteriskNOW setup, follow the scratch install as needed, however do not recompile the kernel or re-install asterisk. Use the Asterisk GUI to setup your extensions that do not need to use the dialer (although, technically they could). Then follow the instructions to edit the config files. Use the built in File editor within the AsteriskGUI. Once everything has been copied over and the required programs and codecs loaded, restart asterisk and return to the file editor.
Within the extensions.conf file, copy all text from the numbering plan. This maybe labeled "numberplan-custom-1" where the number 1 can be incremental. Paste this information under the "default" segment of the file. This will allow your extensions to dial out and avoid the "default" problem I was facing earlier. I'm sure there is a more appropriate way to make this work, I however didn't have time to google any solutions.
This setup will at least get the ball moving with using the easy install of the AsteriskNOW setup. So far the only downside I've found, is that Vicidial can't determine if the call has ended or still ongoing. Although you can log it appropriately, the pop-up message about "Dial time out [or something similar], will get a little annoying.
For now, this is a quick and easy way to set things up very quickly but should not be used for production. The strain of installing this system this way will get your hands dirty and ready for the normal scratch install. Although you can install the AsteriskGUI later, you will at least understand how all of these files affect the areas of Asterisk.
If anyone see's an easier way to do this that someone may find beneficial, please post a response. My next plan is to completely redesign astGUIclient and Vicidial so they are more attractive and easier to setup. Vicidial lacks a lot of design elements that commercial software systems have. But for now, I have another scratch install to perform.
Thanks for everyone's help.
Posted:
Tue Apr 15, 2008 8:31 am
by mflorell
Thank you very much for posting. This will help those trying to use AsteriskNOW quite a bit.