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 Post subject:
PostPosted: Thu Dec 21, 2006 2:15 am 
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Joined: Thu May 04, 2006 6:14 am
Posts: 76
Location: Manta - Ecuador
Quote:
Question off topic -> can you call SIP friend from PSTN???


Yes, with DID. You can get one free from www.ipkall.com and route to your sip/iax friend.


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 Post subject:
PostPosted: Thu Dec 21, 2006 2:51 am 
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Joined: Mon May 29, 2006 7:07 pm
Posts: 287
Location: Denver
No my question is call you call sip friend via 9 from regular phones?
I can can SIP to SIP, SIP -> PSTN, but when i dial my access numbers and enter 9xxxx ( SIP friend ) is not recognized as SIP call so it cannot connect.


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 Post subject:
PostPosted: Wed Jan 24, 2007 1:51 pm 
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Joined: Wed Jun 21, 2006 11:14 am
Posts: 75
Soultion what you have suggested is great and it works the only problem is we have to edit voicemail.conf file and add the Cardnumber
+ password and email into it to make it work Thankyou for the great work i will get the script which can add this automatically when the sip or Iax friends are created and also sip account generation and IAX genration automatically when user signup and when we generate pin with Sip and IAX automactic script generation and asterisk reload.

regards
surender


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 Post subject:
PostPosted: Wed Jan 24, 2007 6:24 pm 
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Joined: Thu May 04, 2006 6:14 am
Posts: 76
Location: Manta - Ecuador
Thank you, I'll be waiting for your voicemail script :)


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 Post subject:
PostPosted: Mon Apr 30, 2007 8:04 am 
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Joined: Tue Nov 21, 2006 2:45 pm
Posts: 25
Does anyone know if this feature is going to be included in the next release (as per KrzyKat's suggestion) I guess that raises aother point is there any definnitive list of features/fixes for the next release and ahy news on the timescale?

Best Regards to all


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 Post subject:
PostPosted: Mon Apr 30, 2007 5:20 pm 
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Joined: Mon May 29, 2006 7:07 pm
Posts: 287
Location: Denver
I haven't seen any news from Areski, who knows maybe there is no next release...


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 Post subject:
PostPosted: Mon Apr 30, 2007 5:31 pm 
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Joined: Thu Oct 19, 2006 9:56 am
Posts: 300
Location: Athens, Greece
With open source you can rest assure that there will be one! ;)

I guess Areski is working hard at the moment..


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 Post subject:
PostPosted: Tue May 01, 2007 4:08 pm 
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Joined: Tue Jun 06, 2006 12:14 pm
Posts: 685
Location: florida
Areski and his team (mostly Areski I think) ARE working very hard on a new release. And I think it is getting close to release. I know for a FACT that it is one day closer to release today than it was yesterday. LOL 8)


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 Post subject:
PostPosted: Tue May 01, 2007 5:49 pm 
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Joined: Mon May 29, 2006 7:07 pm
Posts: 287
Location: Denver
Hi, i know he is working hard - i wish he can post some of the new features... it's hard to wait....


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 Post subject: Re: Alternate Forwarding method ?
PostPosted: Sat May 26, 2007 10:18 am 
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Joined: Mon May 14, 2007 5:41 am
Posts: 12
Hi, all, please help:
I tried to add this for
BUSY , it works, Channelunavailable also works
but not for NoAnswer
-- Called 1105503076
-- SIP/1105503076-091fce98 is ringing
-- Nobody picked up in 30000 ms
-- AGI Script a2billing.php completed, returning 0
Thank you very much


jono_fowler wrote:
I decided to edit the Class.A2billing.php and call it from there. Figured that way i could put some database stuff in for whether or not the user wants VoiceMail active or not and what action to take accordingly. (around line 886). The same prinicple could be used in the BUSY status.

Find the following:

} elseif (($dialstatus == "CHANUNAVAIL") || ($dialstatus == "CONGESTION")) {
$answeredtime=0;
// FOR FOLLOWME IF THERE IS MORE WE PASS TO THE NEXT ONE OTHERWISE WE NEED TO LOG THE CALL MADE
if (count($listdestination)>$callcount) continue;
} else{

Change To :

} elseif (($dialstatus == "CHANUNAVAIL") || ($dialstatus == "CONGESTION")) {
$answeredtime=0;
// FOR FOLLOWME IF THERE IS MORE WE PASS TO THE NEXT ONE OTHERWISE WE NEED TO LOG THE CALL MADE
if (count($listdestination)>$callcount) continue;

// The following section will send the caller to VoiceMail with the unavailable priority.
$did_number = "u".$agi->request['agi_extension'];
$this -> write_log("[STATUS] CHANNEL UNAVAILABLE - DIVERT TO VOICEMAIL ($did_number)");
$agi-> exec(VoiceMail,$did_number);

} else {


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 Post subject:
PostPosted: Sun Mar 29, 2009 11:23 am 
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Joined: Sat Jul 07, 2007 10:00 pm
Posts: 57
exactly same problem here, can't get voicemail to work for noanswer :(


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 Post subject:
PostPosted: Mon Mar 30, 2009 8:16 am 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
There is a changelog and features list in the SVN trunk.

The best way everyone can help speed up the release of 1.4 is to get it installed and with plenty of eyes on it, we can get it to commercial strength very quickly.

Joe


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 Post subject: Re: DID forwarding to voicemail if not answered by SIP_FRIEND.
PostPosted: Wed Jan 19, 2011 4:22 pm 
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Joined: Wed Nov 24, 2010 1:02 pm
Posts: 3
Hi All,

I see this thread is probably defunct, but I'm having a Voicemail issue and I can't get on top of it.

I've only started using Asterisk and A2Billing in the last few months and I'm still quite wet behind the ears, so if you are replying(and this would be much appriciated) please do so with a good bit of detail.

Here's the back ground...
In our last A2Billing server, The Customer Extensions were setup though FreePBX and the Account codes were passed to A2Billing for billing, and the likes of Voicemail etc was handled by FreePBX.

It was an older version of the A2Billing software, and after a number of bugs were identified and a failed attempt to update the software to get it properly working with FreePBX, I contracted [email protected] to install everything onto a clean CentOS Server.

Part of the deal, was that Training Videos would be supplied and I would use these to help me roll out a Commercial VoIP Service on our Network.

The Training Video described how to create the User Accounts and Extensions through A2Billing, and not through FreePBX as I had done in the past, so this is what we did.

After fixing various teething problems, we deployed the server and started rolling out basic services.

Shortly after the initial deployment, after connecting about 100 customers, I wanted to enable Voicemail for some or all users, but I couldn’t identify how to do this with A2Billing.

I contacted A2Billing, where I was told that this is an optional extra (Would have been nice of them if they told me in the first place!!), though I had the option to have the extensions defined in FreePBX and let it handle Voicemail also.

But the problem at this stage was that, we had 100 working customers defined in A2Billing, and another 100 or so waiting to get connected and a process that was complicated enough without introducing a whole other system(FreePBX) to our team that would have the Extensions and Voicemail configured in it.

So we took our only practical option, we paid up and got [email protected] to install the bolt on to enable Voicemail; which we were told would work in exactly the same way as the FreePBX voicemail system did.

The Point of this paragraph is that we couldn’t roll back to using FreePBX for extensions.


So Voicemail was enabled.
There were some initial problems after this were configured, which it would have been nice of A2Billing to let us know about before hand, but these have since been dealt with.

And then we have our current issue. (My thanks to those who have stuck with this post so far! :shock: )

The Problem...
When someone from the outside world dials the DDI of one of my customers and my customer does not answer, it gets forwarded to voicemail.

But the only ‘User Recorded’ greeting that I can get the system to play, is what the VoiceMail system refers to as a ‘Temporary Greeting’ which apparently overrides any other greetings.

If I delete the ‘Temporary Greeting’ and record any of the other greetings, ‘Unavailable’, ‘Busy’ or ‘My Name’, none of these greetings will be played to me.

Regardless of what greeting I setup, ‘Temporary’ or otherwise, the 'prepaid-callfollowme' message is always played before hand.

So in the example below, you can see the 'prepaid-callfollowme' message followed by my ‘Temporary Greeting’, followed by a Beep and my message recording...

-- AGI Script Executing Application: (DIAL) Options: (sip/10015|20|HiL(3600000:61000:30000))
-- Limit Data for this call:
> timelimit = 3600000
> play_warning = 61000
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 30000
> start_sound =
> warning_sound = timeleft
> end_sound =
-- Called 10015
-- SIP/10015-00000f3e is ringing
-- AGI Script a2billing.php completed, returning 0
-- Nobody picked up in 20000 ms
-- Playing 'prepaid-callfollowme' (escape_digits=#) (sample_offset 0)
-- AGI Script Executing Application: (VoiceMail) Options: (6216519519|s)
-- <SIP/DigiWebOfficeTrunk-00000f3d> Playing '/var/spool/asterisk/voicemail/default/6216519519/temp' (language 'en')
-- <SIP/DigiWebOfficeTrunk-00000f3d> Playing 'beep' (language 'en')
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/6216519519/tmp/CDinjh format: wav49, 0x52664c8
-- x=1, open writing: /var/spool/asterisk/voicemail/default/6216519519/tmp/CDinjh format: gsm, 0x52f1608
-- x=2, open writing: /var/spool/asterisk/voicemail/default/6216519519/tmp/CDinjh format: wav, 0x535bdb8
-- User hung up
-- AGI Script a2billing.php completed, returning 0

As a temporary workaround, I have renamed the ‘/var/lib/asterisk/sounds/en/prepaid-callfollowme.gsm’ to ‘prepaid-callfollowme.bak’, and recorded a so called ‘Temporary Greeting’ as my main greeting.

So now, a call not answered will be forwarded to VoiceMail where the ‘prepaid-callfollowme’ is skipped and the Temporary Message is played, then a Beep and a Message is recorded.

Functionally, this is exactly what I wanted, but it’s confusing for customers to tell them to ignore ‘Unavailable’, ‘Busy’ and ‘Name’ greetings and record a ‘Temporary Greeting’ which won’t be temporary.

If I delete the ‘Temporary Greeting’, and record an ‘Unavailable Greeting’...
When I dial the DDI and it gets passed to voicemail, this is what happens...

-- Nobody picked up in 25000 ms
-- AGI Script Executing Application: (VoiceMail) Options: (6216519519|s)
-- <SIP/DigiWebOfficeTrunk-0000fe85> Playing 'beep' (language 'en')
-- Recording the message

Again, no effort is made to play the ‘Unavailable Greeting’.

[email protected] referred me to this link http://www.voip-info.org/wiki/view/Aste ... +VoiceMail where it states that the ‘s’ flag skips any Greeting, and we can see from the above example that the ‘s’ flag is indeed specified.

[email protected] said that they “cannot use the ‘u’ option, because if the customer fails to record a message, then the customer's accountcode would be played to anyone who rang, and that is a big security risk.”

So now I’m completely stuck.

In my reply to [email protected], I say the following...
“To get around the Security Risk, why not play a generic Unavailable Greeting by Default, unless the customer records their own.

Or perhaps we could write a script that places a Generic Unavailable Greeting in every mailbox.”

This way, the greeting would always exist and if the customer fails to record a message, then the customer's account code would not be played to anyone who rang, but instead would play the default greeting.

I also said, “If only one flag can be selected, and it’s a global setting, then the other Voicemail options of ‘Busy Greeting’ and ‘Name Greeting’ should be removed.”

We’re trying to be a professional TeleComs provider and to have a situation where we can’t offer a properly working Voicemail solution seems off the wall.

I don’t think it’s acceptable that all our customers should be told that due to a bug in the Voicemail system, that the only Greeting that works is the ‘Temporary Greeting’.

That’s going to be very confusing for alot of people and reflects very badly on our fledgling Phone Service.
If anyone has any suggestions of how to get a proper voicemail service working with A2Billing SIP Extensions, I’d love to hear from you.

All the best,
MadMax


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 Post subject: Re: DID forwarding to voicemail if not answered by SIP_FRIEND.
PostPosted: Sun Dec 09, 2012 2:57 pm 
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Joined: Sun Dec 09, 2012 2:53 pm
Posts: 2
Hi Madman,

I also have the same issue like yours. Were you able to resolve the issue or did alternate tweaks?


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 Post subject: Re: DID forwarding to voicemail if not answered by SIP_FRIEND.
PostPosted: Thu Dec 27, 2012 12:16 pm 
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Joined: Sun Dec 09, 2012 2:53 pm
Posts: 2
I am really struggling to get this VM setup right. Anyone who can help to resolve this?


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