Re: [asterisk-users] OT: OpenSIPS vs Kamailio -- which do you use and why?

2011-03-04 Thread Amit Nepal
is changed to kamailio I guess. It had a fork, but now they have merged together. Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net On 3/4/2011 11:49 AM, Steve Edwards wrote: I'm starting a new project similar

Re: [asterisk-users] Asterisk to asterisk t.38

2011-01-21 Thread Amit Nepal
. Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net On 1/20/2011 4:11 PM, Bryant Zimmerman wrote: Amit Make sure that the trunk you have between the two servers has the t.38 enabled on it. Do you have any NAT

[asterisk-users] Asterisk to asterisk t.38

2011-01-20 Thread Amit Nepal
I have a setup of asterisk 1.6 in one box and asteirsk 1.4 in another. I can send recieve faxes from both boxes fine to and from pstn. But the faxing between 1.6 and 1.4 extensions does fail. Any ideas please ? -- Thank You Amit Nepal

Re: [asterisk-users] Asterisk to asterisk t.38

2011-01-20 Thread Amit Nepal
1.4 and ast 1.6. ATA (T.38 capable) AST 1.6 AUDIO CODEAST 1.4ATA (t.38 Capable) Thank You Amit Nepal On 1/20/2011 1:56 PM, David Backeberg wrote: On Thu, Jan 20, 2011 at 3:14 PM, Amit Nepalami...@phoenixinternet.net wrote: I have a setup of asterisk 1.6 in one box and asteirsk 1.4 in another

Re: [asterisk-users] Asterisk to asterisk t.38

2011-01-20 Thread Amit Nepal
Yes Tom, I am sending via the PSTN gateway which is audio code in my case. Thank You Amit Nepal On 1/20/2011 3:07 PM, Tom Rymes wrote: On 01/20/2011 4:26 PM, Amit Nepal wrote: I have an Audio code gateway between two asterisk servers. The audio code has PRI connected for PSTN. I can