Live Monitoring

[adymeblack]

Just to get this out of the way:

Cluster: DB & 5 dialers
VERSION: 2.8-409a
BUILD: 130809-1410
Asterisk: 1.4.21.2
ISO Install: 5.0.4 - 64-Bit
2x- Intel Xeon Quad-Core CPU
16GB RAM
[Please let me know if i'm forgetting something]

This cluster is setup to handle outbound dialing only. Some of our supervisors need to live monitor the calls for quality. Whenever we try to to that, Vicidial shows success in calling the extension but the phone never rings. If calling a desk phone, after about 5 minutes a voice mail of the call appears sounds just as it should, but again the phone never rings.

While troubleshooting and watching what was actually happening i noticed this:

Code: Select all

[Feb 13 14:06:48] WARNING[16667]: pbx.c:1475 pbx_exec: The application delimiter is now the comma, not the pipe.  Did you forget to convert your dialplan?  (Dial(SIP/3020|60|))
  == Using SIP RTP CoS mark 5
[Feb 13 14:06:48] ERROR[16667]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("3020|60|", "(null)", ...): Name or service not known
[Feb 13 14:06:48] WARNING[16667]: chan_sip.c:5866 create_addr: No such host: 3020|60|
[Feb 13 14:06:48] WARNING[16667]: app_dial.c:2345 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [3020@default:2] Goto("Local/010*000*000*047*08600052@default-000009c5;1", "default,850266666666663020,1") in new stack
    -- Goto (default,850266666666663020,1)
    -- Executing [850266666666663020@default:1] Wait("Local/010*000*000*047*08600052@default-000009c5;1", "1") in new stack
    -- Executing [h@default:1] AGI("Local/010*000*000*047*08600052@default-000009c5;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----1-----0") in new stack
    -- <Local/010*000*000*047*08600052@default-000009c5;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----1-----0 completed, returning 0
  == Spawn extension (default, 010*000*000*047*08600052, 1) exited non-zero on 'Local/010*000*000*047*08600052@default-000009c5;2'
  == Manager 'sendcron' logged off from 127.0.0.1
    -- Executing [850266666666663020@default:2] VoiceMail("IAX2/vicidial4s-10924", "3020,u") in new stack
    -- <IAX2/vicidial4s-10924> Playing 'vm-theperson.gsm' (language 'en')

The part i'm most interested in is this:

Code: Select all

WARNING[16667]: pbx.c:1475 pbx_exec: The application delimiter is now the comma, not the pipe.  Did you forget to convert your dialplan?  (Dial(SIP/3020|60|))

It's seems pretty self explanatory, however the only place i can find a dial plan that looks like that is in a vicidial auto-generated conf file.

The phones i tested with are registering just fine and able to make outbound calls.
I have an older cluster [version 4 i think] and the live monitoring works just fine.

Not sure if it helps, but currently we are able to live monitor only if we use the session ID.

Any ideas where i need to change the dial plan to make this work?

Thank You in advance.

[williamconley]

A couple challanges:

there is no "5.0.4" iso installer. Perhaps you should paste the full name of the .iso image you downloaded.

Your version of asterisk would be quite different if you had installed with any of the "5.X.X" installers. Asterisk 1.4 is not available in that series.

You did not specify the actual method of monitoring the calls. Button for Button. (Certainly more than "whenever we try to do that" is in order, LOL).

If there was some sort of database port and/or upgrade, beware that the later version of Asterisk will not work properly with the prior version of the asterisk configuration files. You must use the new ones (which I believe have replaced all the | with , properly ... among other things!). And be sure your running Asterisk on each dialer matches the asterisk version listed in "admin->servers" for that server. had a lot of problems with that, too. It is actually quite normal to "perl install.pl" on a dialer to get those new configuration files installed, there is a question during the install that will grab the new ones for you IF you have answered the prior version question correctly. And all the "vicidial generated files" will be regenerated anyway (never hurts to request them all to be rebuilt in admin->servers either). If you've customized them .. and don't want to give up the customizations ... you should seriously look into figuring out how to put the custom entries into the Vicidial Interface instead (so they land in the vicidial generated versions of the file instead of the hard-coded and maintained static files). Or pay someone to modify the interface to duplicate your handiwork. It's pretty much always possible.

Happy Hunting!

[adymeblack]

My apologies, the ISO used was ViciBox5.x86_64-5.0.3 on a fresh server so no upgrading was involved.

We were trying to live monitor via X-Lite or Zoiper, as well as desk phones. I did manage to get the live monitoring figured out, for some reason Vicidial was showing it had asterisk 1.4 installed, but it actually had 1.8 installed. Told Vici that the asterisk version was 1.8 in the webgui, re built conf files, rebooted [just to be sure] live monitoring works with no problems. Fixed a conferencing issue i also had.

However now i'm having a lot of other issues, such as agents pulling multiple calls as well as randomly being logged out of vicidial. I've also noticed it seem to take quite a bit longer for it to come up sometimes.

I'm considering running the install script again, but i'm worried that it will take out records of previous calls, recordings and all that fun stuff.

[adymeblack]

Well i spoke too soon. Turns out i am only able to live monitor if using an external extension. If i try to use an in-house extension, the GUI shows a successful call, but the CLI comes back saying that the extension is invalid. I know for a fact the extension exists, my phone will register with it, and it is able to make calls which confuses me..... Makes me think something in the dialplan maybe?

Code: Select all

 -- Executing [010*000*000*047*08600056@default:1] Dial("Local/010*000*000*047*08600056@default-00001138;2", "IAX2/vicidial5s:test@10.0.0.47:4569/08600056,55,oT") in new stack
    -- Called IAX2/vicidial5s:test@10.0.0.47:4569/08600056
    -- Call accepted by 10.0.0.47 (format ulaw)
    -- Format for call is ulaw
    -- IAX2/vicidial4s-15135 answered Local/010*000*000*047*08600056@default-00001138;2
       > Channel Local/010*000*000*047*08600056@default-00001138;1 was answered. 
== Starting Local/010*000*000*047*08600056@default-00001138;1 at default,9000,1 failed so falling back to exten 's'
  == Starting Local/010*000*000*047*08600056@default-00001138;1 at default,s,1 still failed so falling back to context 'default'
    -- Sent into invalid extension 's' in context 'default' on Local/010*000*000*047*08600056@default-00001138;1
    -- Executing [i@default:1] Playback("Local/010*000*000*047*08600056@default-00001138;1", "invalid") in new stack


[geoff3dmg]

Are you using phone aliasing? As monitoring from the realtime screen doesn't understand phone aliases (I presume this is a bug).

[adymeblack]

No, we don't use phone aliases. We just enter the direct extension that points to their computer or desk phone.

[perci100]

I usually create a new phone and user for every QA or admin and make the ID out of the range of my normal phones just so I dont get confused. When you click on "choose report display options" on the real time screen make sure that the correct phone extension is in the text box and that matches your softphone. It sounds like maybe you have multiple users with the same phone extension or it doesn't match the user in your softphone.

Hope that helps.

[williamconley]

but the CLI comes back saying that the extension is invalid. I know for a fact the extension exists, my phone will register with it, and it is able to make calls which confuses me..... Makes me think something in the dialplan maybe?

Don't see your problem directly, but I can help you with a conceptual issue that may allow you to resolve the issue yourself.

Based on your quote above, you may think that you register an "extension". But you do not. SIP accounts are not extensions. Your sip account may be numeric, but that is your choice (and a bad idea, too, BTW, but that is a different topic, LOL). consider that your phone could also be "xxgqigj9000" which would be in the extension field in the Vicidial Phones list. Since one cannot "dial" xxgpigj9000, it's safe to say that xxgpigj9000 is not an extension field. It would be better if it were "Account Name" instead of "Extension" as it causes confusion.

The "extension" is actually called "Dial Plan Number" in the Vicidial phones interface.

And don't forget that the context (phone context and exten context) way down below can make a difference in whether it can be dialed or not.

You can also use "dialplan show XXXXX@default" to determine whether or not it is possible to dial an extension (in the default context, which is where pretty much everything Vicidial happens)

Happy Hunting!

[adymeblack]

I figured out the issue.

For some reason Vici was reporting it only had Asterisk 1.4 installed, informed the system to use the 1.8 version installed and rebuilt the conf files and it is working.

[luckypig]

I got the same error as above.
I installed (DB+Web) on Server A. Asterisk on Server B has two phone "6009 and 6008"
when 6009 call to 6008, I get error:

== Using SIP RTP CoS mark 5
-- Executing [6008@default:1] Dial("SIP/6009-00000011", "SIP/6008|60|") in new stack
[Jun 18 14:28:15] WARNING[11846]: pbx.c:1475 pbx_exec: The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? (Dial(SIP/6008|60|))
== Using SIP RTP CoS mark 5
[Jun 18 14:28:24] ERROR[11846]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("6008|60|", "(null)", ...): Name or service not known
[Jun 18 14:28:24] WARNING[11846]: chan_sip.c:5869 create_addr: No such host: 6008|60|
[Jun 18 14:28:24] WARNING[11846]: app_dial.c:2345 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [6008@default:2] Goto("SIP/6009-00000011", "default,850266666666666008,1") in new stack
-- Goto (default,850266666666666008,1)
-- Executing [850266666666666008@default:1] Wait("SIP/6009-00000011", "1") in new stack
-- Executing [850266666666666008@default:2] VoiceMail("SIP/6009-00000011", "6008,u") in new stack
-- <SIP/6009-00000011> Playing 'vm-theperson.gsm' (language 'en')
...........


Can you help me way to solve?

Thank in advance