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Joe, Thanks for the tips. However, what baffles me is why is custom-a2billing-2 not working even when it is the same dial plan as custom-a2billing. The problem is occurring when we are supposed to jump to the custom-a2billing-2 context. So, it is not the a2billing.php|2 (agi-conf2) issue...

The context looks exactly like that. I am still getting the same issue. If I change my DID target to A2B-Inbound (custom-a2billing context), the call is answered and "Please enter your PIN" prompt is heard. If I enter the card number it says press 9 to call SIP IAX friend. Upon pressing 9 it...

A2Billing: calling Card Access number does not proceed OK. so far everything worked fine. But now, I am not able to dial a DID number from my cell phone to get the calling card prompts. Here are the excerpts from the logs: -- Executing [s@macro-privacy-mgr:9]...

I am not getting the user prompts for prepaid-welcome etc. Are they synthesized on the fly? i.e. Do I need to have festival/cepstral on the same machine as asterisk/a2billing?

Thanks for the above suggestion for using the Jeff Howe method. In fact it is much better and cleaner than using the ATAs to prepend 9. The issue with ATAs prepending 9 was that a *97 would come in as 9*97 - unacceptable.

Only Certain extensions are sent through a2b I think you can set up 2 trunks. 1 set of trunks require 9 as a prefix. These are sent to a2b. The extension ATAs that need to be billed will have a dialing prefix of 9 set in their config so that any number they dial will be prefixed with 9.

Awsome tool The tool seems awesome. I do not quite figure out how you can ask the device to t*f*t*p it's config. I have the following tedious procedure: 1. Buy the phones/ATAs. 2. Connect them to my local network. 3. Access their GUI and change the URL where they download their firmware...

rtp.c: Unable to set TOS to 184 After doing all that, I see that outgoing RTP from my device are marked DSCP 0xbe. Incoming from asterisk is 0x00 Log has many such entries. [Apr 18 17:05:48] WARNING[1817] rtp.c: Unable to set TOS to 184 Problem seems to be that the asterisk process should be...

Hi, I am looking for a SOHO router to manage priority for SIP and RTP/RTCP traffic. The idea I have is that since the devices can set DSCP codes in their IP Packets, the router should be configured to treat those packets preferentially. This works seamlessly when we add more devices on the...

Hi, This might be a pure asterisk question. Is there a way to register SIP phones using SIP Alias and password? Currently, you'll have to register with the authentication id set to the numerical extension. It would be most helpful if the sip phones can register using SIP alias instead...

I have 3 sets of extensions for 3 different departments. Each dept. has some DIDs. I have mapped the incoming DIDs to their respective ring groups. The departments want outgoing calls on "their" set of trunks. How can I assign groups of outgoing trunks based on the originating extension? I...