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Telephony

Proposed Q&A site for information technology professionals that design, support and manage various telephone architectures, PBX systems and protocols such as: Asterisk, FreePBX, SIP, VOIP, TrixBox, ViciDial, NEC, Shoretel.
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more than a networking topic – B0nk3r Jul 21 '12 at 3:45

12 Example Questions

active oldest votes
up vote 10 down vote
Are there SIP clients available that can receive/send audio and commands over STDIO?
added by Brad Aug 11 '12 at 20:03
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up vote 10 down vote
Is it the role of SIP to control bandwidth usage and scale audio quality up and down accordingly? Or is this done at another layer?
added by Brad Aug 11 '12 at 20:05
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up vote 10 down vote
Asterisk not loading config file extensions.conf. Nothing shows up in the logs. How to debug the problem?
added by Petr Pudlák Aug 26 '12 at 15:10
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up vote 7 down vote
Asterisk 1.6 doest not listen to DTMF codes from our Intelbras Voiper's telephones. What can be wrong?
added by brandizzi Oct 1 '12 at 12:19
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up vote 5 down vote
I am looking at building an Asterix box that connects to 4 T1 lines. Should I get a quad T1 card or 4 T1 cards?
added by Chris Travers Oct 6 '12 at 4:00
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I'm undecided about this. It might be a bit too close to Not Constructive. – DaveRandom Apr 19 at 23:30
I want to explain the answer and why I think it is constructive. Basically most of the T1 cards that Asterisk support (or did last time I checked) issue one IRQ per sample frame timeslice. What this means is that 4 T1 cards will generate 32000 IRQ's a second with no guarantee that these will be unformly spread across the timeframe. A single quad T1 card will usually generate only 8000 IRQ's/sec. This is a fairly common mistake people make and the quad T1 card poses far fewer performance issues. This is why I think the question is good. – Chris Travers Apr 20 at 2:57
up vote 4 down vote
I have connected a Network Alchemy POTS unit to an Avaya IP Office call server and it appears to work. Are there any potential compatibility issues?
added by DaveRandom Sep 7 '12 at 23:19
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up vote 2 down vote
Is it possible to dial SIP phones directly by IP address/hostname without DID numbers?
added by Brad Aug 11 '12 at 20:03
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This is a yes/no question and will likely provoke a yes/no answer (based on my experiences on SO). Simply by adding the word "How" to the beginning, much better answers would be provided. – DaveRandom Sep 7 '12 at 23:14
up vote 2 down vote
How can the configuration for asterisk be converted for freeswitch?
added by txwikinger Mar 13 at 15:02
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up vote 1 down vote
How can I route my Inbound Routes based on DID AND CallerID in FreePBX?
added by Mbrevda Oct 1 '12 at 20:27
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up vote 0 down vote
How do I view the output from a POTS phone connected to a Splicecom Maximiser?
added by JMK Jan 9 at 15:04
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up vote 0 down vote
After upgrading my FreePBX modules, the CSS for the administration control panel appears broken. How can I fix this?
added by JohnB Mar 27 at 3:10
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up vote 0 down vote
I have 3 POTS lines. Lines A and B both roll over to the same line C. Is it possible to detect which line (A or B) was originally dialed/rolled to C?
added by JohnB Mar 27 at 3:20
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