Digium IP Phones

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Hello, Im looking to buy a digium phone D70 unit just for testing on lab; to
really understand the phone and features. I cant find any website with opinions; any here? Are they really valuable
to the price? (D70 quite expensive) Does the SDK for building apps is usable? Can you build powerfull apps?
Examples? Many thanks

Asterisk Users 3.3 years ago 4 Answers

Why do I get call twice in one go?

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I understand why do I get call twice to my mobile when I execute the
following AMI command sets: ACTION: Originate
Channel: Local/800@test
Timeout: 60000
Priority: 1 and my dialplan look like this: [test]
exten => 800,1,DIAL(SIP/447xxxxxx@voip);
exten => 800,n,Hangup()
How to prevent getting called twice in one go when I execute this AMI
command? Thanks...

Asterisk Users 3.3 years ago 0 Answers

what to use for a B410P

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Hi, I'm experiencing difficulties to get a B410P running with Asterisk
10.3.1 and DAHDI 2.6.1.
Am I supposed to use DAHDI for this card and ISDN BRI for my country
(Belgium)? thx, BC

Asterisk Users 3.3 years ago 7 Answers

British Telecom ISDN BRI line issues

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Should I understand that no Asterisk user has issues with ISDN "system
access" configuration from UK? or maybe no one is using Asterisk In UK :) ? On Tue, May 8, 2012 at 12:46 PM, khalid touati wrote: > Hi All,
> I am posting this thread with the hope that someone in UK (or elsewhere)
> had a similar issue:
> Our issue is simple, we cannnot use our ISDN line, when watching asterisk
> console it gives a bunch of ISDN errors where the following is probably the
> most relevant:

Asterisk Users 3.3 years ago 19 Answers

No compatible codecs, not accepting this offer! - after upgrading to 1.8.11

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Hi, I've upgraded my asterisk 1.4 to the version 1.8.11. After making some
adjustments to the configuration files to port it to the new version, calls
between registered phones in asterisk, work fine, but inbound calls coming
from the SIP trunk I have with a telco to asterisk, don't work anymore. I
don't know why!... This is the SDP portion that comes in the INVITE messages of calls through
that trunk (let's say, whose endpoint has the IP x.x.x.x, purposely
omitted). Nothing seems to be wrong with that to me:
v=0
o=CSM…

Asterisk Users 3.3 years ago 4 Answers