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 Post subject: web-meetme and a2billing integration - done
PostPosted: Wed Jan 30, 2008 4:05 am 
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Joined: Wed Jan 23, 2008 10:21 am
Posts: 15
Hi everyone,

I have successfully added support for a2billing in web-meetme - although I need to do more testing but it seems to work fine so far. Outbound calls placed from a conference (e.g. by conference presenter inviting/calling out to a phone number) are billed to the conference owner's A2billing account.

If anyone else needs this please post here.

When I have time I will post the changes made (as there were a few) to get this working.

~ Rodney


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 Post subject:
PostPosted: Wed Jan 30, 2008 4:37 am 
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Joined: Sun Jan 20, 2008 12:00 am
Posts: 71
I would like instruction on hwo to do this. Please post.

thank you


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 Post subject:
PostPosted: Wed Jan 30, 2008 4:27 pm 
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Joined: Fri Oct 27, 2006 6:17 pm
Posts: 161
i'want to see this too :D


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 Post subject:
PostPosted: Wed Jan 30, 2008 4:48 pm 
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Joined: Tue Jun 06, 2006 12:14 pm
Posts: 685
Location: florida
Even a demo of it would be nice to see as well.


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 Post subject:
PostPosted: Wed Jan 30, 2008 7:29 pm 
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Joined: Wed Jan 23, 2008 10:21 am
Posts: 15
ok, there's clearly interest in this :-) I'll put together a post. Things are a little busy today but should get to it by the weekend.


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 Post subject: web-meetme A2B update...
PostPosted: Sun Feb 03, 2008 1:06 am 
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Joined: Wed Jan 23, 2008 10:21 am
Posts: 15
I've run into a problem with the meetme integration, which I am hoping someone can help with.

When the outbound call is placed from web-meetme via the "Invite" button (which calls call_operator.php) a call is initiated with one leg connected to the meetme room, and the other leg placed outbound to the remote party, which get's placed through A2Billing. The call is connected fine, but unlike a normal A2 call, A2B completes immediately the call is bridged to meetme. In A2B the call is placed with params something similar to:

SIP/<number>@<provider>|60|HRgrL(<a>:<b>:<c>)

phpagi exec_dial returns {DIALSTATUS} on completion of the call.

When the call gets bridged to the meetme conference room, exec_dial returns answered (time 0) even though the call is still in progress (in a normal call it would not return until one of the parties hangs up). A2Billing then completes processing recording a call of 0 seconds duration, even though in fact it is still in progress.

I will continue playing around with this to see if there is a simple solution. But for now wanted to ask if anyone has any bright ideas of things I could try?

~ Rodney

Dial cmd documentation: http://phpagi.sourceforge.net/phpagi2/docs/


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 Post subject:
PostPosted: Sun Feb 03, 2008 7:24 am 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
Hi

Are you running into the same issues with a double dial - which are outlined here: -

http://forum.asterisk2billing.org/viewtopic.php?t=2127

I've done an implementation the other way round, where people phone into a conference number, and are charged to the owner of the conference, but never dialing out as you have done.

Joe


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 Post subject:
PostPosted: Tue Feb 05, 2008 8:26 am 
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Joined: Wed Jan 23, 2008 10:21 am
Posts: 15
I think I've solved this one. Will test some more and if it's a solution will post more tomorrow...

~ R


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 Post subject:
PostPosted: Thu Feb 14, 2008 9:43 pm 
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Joined: Fri Oct 27, 2006 6:17 pm
Posts: 161
:hmm:


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 Post subject:
PostPosted: Tue Feb 26, 2008 3:21 am 
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Joined: Tue Feb 26, 2008 3:09 am
Posts: 1
Location: Orlando, FL
I have had a similar issue. I've tracked it down the the "masquerade" done by the Meetme App. In Vicidial, they have a workaround, but it's ... COMPLEX involving screens and reassigning the context/exten/pri of the running script into the "masqueraded" channel at the moment it moves. But there's more. :shock: They also use the callerid name for permanent tracking and run their script more than once (so it can detach and come back later).

I am presently trying to combine the two (Vicidial is a Predictive AutoDialer and is EXCELLENT, when combined with A2Billing ... well, you get the idea).

But for my purposes, I can't use their solution, it's already in use, unless I piggyback on their script and re-execute the A2B script afterwards. I'd prefer to find another solution (run a script to track the meetme until IT dies, then return to finish the billing ...).

Any ideas?

Thanks in advance!

Bill

code (from the moment the call is answered until the charges are calculated, and NO the call hasn't been terminated, as you can see from the timestamp this all happens within the same second, the call is still Active):

Feb 24 17:44:09 VERBOSE[24580] logger.c: > Channel Local/73001@default-d02e,1 was answered.
Feb 24 17:44:09 DEBUG[24580] manager.c: Manager received command 'Logoff'
Feb 24 17:44:09 VERBOSE[24580] logger.c: == Manager 'sendcron' logged off from 127.0.0.1
Feb 24 17:44:09 DEBUG[15193] devicestate.c: Changing state for Local/73001@default - state 2 (In use)
Feb 24 17:44:09 DEBUG[15193] devicestate.c: Changing state for Local/73001@default - state 2 (In use)
Feb 24 17:44:09 DEBUG[24646] app_queue.c: Device 'SIP/192.168.10.62' changed to state '2' (In use) but we don't care because they're not a member of any queue.
Feb 24 17:44:09 DEBUG[24647] pbx.c: Launching 'MeetMe'
Feb 24 17:44:09 DEBUG[24648] app_queue.c: Device 'Local/73001@default' changed to state '2' (In use) but we don't care because they're not a member of any queue.
Feb 24 17:44:09 DEBUG[24649] app_queue.c: Device 'Local/73001@default' changed to state '2' (In use) but we don't care because they're not a member of any queue.
Feb 24 17:44:09 VERBOSE[24647] logger.c: -- Executing MeetMe("Local/73001@default-d02e,1", "8600051") in new stack
Feb 24 17:44:09 VERBOSE[24647] logger.c: == Parsing '/etc/asterisk/meetme.conf': Feb 24 17:44:09 DEBUG[24647] config.c: Parsing /etc/asterisk/meetme.conf
Feb 24 17:44:09 VERBOSE[24647] logger.c: == Parsing '/etc/asterisk/meetme.conf': Found
Feb 24 17:44:09 VERBOSE[24647] logger.c: == Parsing '/etc/asterisk/meetme_additional.conf': Feb 24 17:44:09 DEBUG[24647] config.c: Parsing /etc/asterisk/meetme_additional.conf
Feb 24 17:44:09 VERBOSE[24647] logger.c: == Parsing '/etc/asterisk/meetme_additional.conf': Found
Feb 24 17:44:09 VERBOSE[24647] logger.c: == Parsing '/etc/asterisk/meetme_custom.conf': Feb 24 17:44:09 DEBUG[24647] config.c: Parsing /etc/asterisk/meetme_custom.conf
Feb 24 17:44:09 VERBOSE[24647] logger.c: == Parsing '/etc/asterisk/meetme_custom.conf': Found
Feb 24 17:44:09 DEBUG[24647] chan_zap.c: Using channel -2
Feb 24 17:44:09 DEBUG[15193] devicestate.c: Changing state for Zap/pseudo - state 2 (In use)
Feb 24 17:44:09 DEBUG[24650] app_queue.c: Device 'Zap/pseudo' changed to state '2' (In use) but we don't care because they're not a member of any queue.
Feb 24 17:44:09 VERBOSE[24647] logger.c: -- Created MeetMe conference 1023 for conference '8600051'
Feb 24 17:44:09 DEBUG[24647] channel.c: Set channel Local/73001@default-d02e,1 to write format gsm
Feb 24 17:44:09 DEBUG[24582] rtp.c: Ooh, format changed from unknown to ulaw
Feb 24 17:44:09 DEBUG[24647] channel.c: Scheduling timer at 160 sample intervals
Feb 24 17:44:09 VERBOSE[24647] logger.c: -- Playing 'conf-onlyperson' (language 'en')
Feb 24 17:44:09 DEBUG[24582] channel.c: Planning to masquerade channel SIP/192.168.10.62-081d1e18 into the structure of Local/73001@default-d02e,1
Feb 24 17:44:09 DEBUG[24582] channel.c: Done planning to masquerade channel SIP/192.168.10.62-081d1e18 into the structure of Local/73001@default-d02e,1
Feb 24 17:44:09 DEBUG[24582] chan_local.c: Not posting to queue since already masked on 'Local/73001@default-d02e,2'
Feb 24 17:44:09 DEBUG[24647] channel.c: Actually Masquerading SIP/192.168.10.62-081d1e18(6) into the structure of Local/73001@default-d02e,1(6)
Feb 24 17:44:09 DEBUG[24647] channel.c: Got clone lock for masquerade on 'SIP/192.168.10.62-081d1e18' at 0x813b084
Feb 24 17:44:09 DEBUG[24582] channel.c: Didn't get a frame from channel: Local/73001@default-d02e,2
Feb 24 17:44:09 DEBUG[24582] channel.c: Bridge stops bridging channels Local/73001@default-d02e,2 and SIP/192.168.10.62-081d1e18<MASQ>
Feb 24 17:44:09 DEBUG[24647] channel.c: Set channel SIP/192.168.10.62-081d1e18 to write format gsm
Feb 24 17:44:09 DEBUG[24647] channel.c: Set channel SIP/192.168.10.62-081d1e18 to read format slin
Feb 24 17:44:09 DEBUG[24647] channel.c: Putting channel SIP/192.168.10.62-081d1e18 in 2/64 formats
Feb 24 17:44:09 DEBUG[24647] channel.c: Released clone lock on 'Local/73001@default-d02e,1<ZOMBIE>'
Feb 24 17:44:09 DEBUG[24582] channel.c: Hanging up zombie 'Local/73001@default-d02e,1<ZOMBIE>'
Feb 24 17:44:09 DEBUG[15193] devicestate.c: Changing state for Local/73001@default - state 2 (In use)
Feb 24 17:44:09 DEBUG[24582] app_dial.c: Exiting with DIALSTATUS=ANSWER.
Feb 24 17:44:09 DEBUG[24651] app_queue.c: Device 'Local/73001@default' changed to state '2' (In use) but we don't care because they're not a member of any queue.
Feb 24 17:44:09 DEBUG[24647] channel.c: Done Masquerading SIP/192.168.10.62-081d1e18 (6)
Feb 24 17:44:09 VERBOSE[24582] logger.c: a2billing.php: file:Class.RateEngine.php - line:1154 - -> dialstatus : ANSWER, answered time is 0

At this point, the A2B rating engine does its thing and bills the call, since as far as it is concerned the call has terminated. But it hasn't ... the Meetme app has just killed the Zombie channel (after transferring the sound into another channel in the conference), and with it dies the "Dial" command that was executed by a2billing.php.


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 Post subject:
PostPosted: Tue Feb 26, 2008 9:04 am 
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Joined: Wed Jan 23, 2008 10:21 am
Posts: 15
quick update - it's been busy the last couple of weeks, and I am still testing my solution in real world conferences. I need to add inbound charging for DID (for users dialing in on a toll free number).

When I get everything working properly I'll post details of how I am going about doing this.

~ Rodney


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