Support A2Billing :

provided by Star2Billing S.L.

Support A2Billing :
It is currently Thu Mar 28, 2024 6:01 pm
Predictive Dialer


All times are UTC




Post new topic Reply to topic  [ 3 posts ] 
Author Message
 Post subject: No sound when calling from a DID
PostPosted: Mon Nov 09, 2009 4:07 am 
Offline

Joined: Mon Nov 09, 2009 3:51 am
Posts: 6
Hi, a2billing friends!

This is the scenario:

I call in Canada to a local number 1647... (I bought this number from DIDWW.com).
It is a number which is routed to my Asterisk (TRIXBOX 2.8.0.1) server.
Asterisk answers the call, recognizes the DID.
The inbound route takes it to an IVR, where I select an option.
That option takes me to a2billing.
I give the PIN, I dial the number, and it rings,... someone answers.
I, in Canada, originating the call, hear the person who answers.
That person doesn't hear me.

I know it can be some NAT issue, but just wanted to know if there's some sort of limitation in a2billing regarding the source of the call.
The usual origin of the call is a PSTN trunk, so I thought maybe it cannot work with other origins.

If not, I may need to sweat a bit more dealing with NATs and routers :).

Thanks indeed!


Top
 Profile  
 
 Post subject: Re: No sound when calling from a DID
PostPosted: Wed Nov 11, 2009 7:18 pm 
Offline

Joined: Thu May 29, 2008 9:07 pm
Posts: 72
You need to start from the least complicated setup, and work your way up.
Try to call in to the did in freepbx like you were, and have it send it to a sip phone in freepbx that is setup. do you got 2 way audio like that?


Top
 Profile  
 
 Post subject: Re: No sound when calling from a DID
PostPosted: Fri Nov 27, 2009 9:15 am 
Offline

Joined: Mon Nov 09, 2009 3:51 am
Posts: 6
Hi! Thanks for your concern. I found a solution but forgot to write it here.

When calls failed, the Asterisk CLI told me it was trying to make some "sip bridge". When calls went right no "sip bridge" were tried.

Making some google on "sip bridge" a solution appeared: in the Trunks associated with the DID, add these:
canreinvite=no
nat=yes

It is working now. Thank God... and Google! he he


Top
 Profile  
 
Display posts from previous:  Sort by  
Post new topic Reply to topic  [ 3 posts ] 
VoIP Billing solution


All times are UTC


Who is online

Users browsing this forum: No registered users and 3 guests


You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot post attachments in this forum

Search for:
Jump to:  
cron
Powered by phpBB © 2000, 2002, 2005, 2007 phpBB Group