Author Topic: Problems with create_amp_*.pl  (Read 9524 times)

pointman87

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Re: Problems with create_amp_*.pl
« Reply #15 on: January 26, 2011, 06:29:36 am »
Ok i thought you also used affinity telecom, a tip is trying the sipgate template or download mine from tracker, search affinity. I didn´t get the broadcom template to work either, same problem.

BR Daniel

willow3

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Re: Problems with create_amp_*.pl
« Reply #16 on: January 27, 2011, 08:17:14 pm »
@pointman87: I didn't get your script for affinity to work. It didn't successfully register to the sip server. I don't have much time to troubleshoot for the moment so I tried the sipgate template instead. I got the same result as you. Incoming calls work but not outgoing. I tried experimenting with the outbound dial patterns in the Free PBX settings without success. I also noted there are two sets of dial patterns; one under Basic->Outbound routes-sipgate->Dial patterns and another under Basic->Trunk SIP/Sipgate->Dial rules. They seem unrelated and I don't know the difference. Which one did you change, and what rules did you use?

regards

pointman87

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Re: Problems with create_amp_*.pl
« Reply #17 on: January 28, 2011, 12:32:11 pm »
Easiest way is to use x for rules. in my case in gävle, we have 026 + 6 numbers = xxxxxxxxx
Do you understand? You just use the number of x apposite to the numbers you want to be able to dial.

BR Daniel

willow3

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Re: Problems with create_amp_*.pl
« Reply #18 on: February 03, 2011, 12:20:11 am »
the syntax wasn't my problem. everything worked fine once i saw the "apply changes" button on top of the page  :)

I also managed to connect the linksys rtp300 to asterisk. When I get the time I will make a wiki page about it.

willow3

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Re: Problems with create_amp_*.pl
« Reply #19 on: February 04, 2011, 11:48:24 pm »
For those who have a linksys RTP300 collecting dust in their drawer, here is how to integrate it in LinuxMCE:

http://wiki.linuxmce.org/index.php/Linksys_rtp300

regards