Comments for Asterisk FAQs

Comment on asterisk 1.4.39 and dahdi 2.6 by RFadam

RFadam — Mon, 23 Apr 2012 17:30:37 +0000

hey i installed first asterisk and then DAHDI. bt nw i came to know that for meetme() application i need DAHDI as im a beginner to asterisk. can anybody kindly suggest me how to configure DAHDI completely with asterisk? i have install DAHDI, then configure it with asterisk but when i open the menuselect the required modules are showing XXX. i do not want to reinstall the asterisk again as i do not want to lose my progress in it…:(

Comment on SIP jitter and packlost channel variables by Audio faqs | Billpearson

Audio faqs | Billpearson — Fri, 30 Mar 2012 23:08:58 +0000

[...] SIP jitter and packlost channel variables – Asterisk FAQs [...]

Comment on About Me by PHP/MySQL developer needed

PHP/MySQL developer needed — Tue, 07 Feb 2012 22:55:09 +0000

[...] About Me [...]

Comment on Exceptionally long voice queue length by Sys_eng

Sys_eng — Mon, 23 Jan 2012 23:04:18 +0000

It is also occurring with 1.8.8.1 and will cause lockup, loss of registration. Only recourse is to reboot.

Comment on Speech recognition in Asterisk using Google Voice API by Brian Knep

Brian Knep — Wed, 04 Jan 2012 16:21:59 +0000

Currently I have seen that most trusted speech system with asterisk is Lumenvox but as it is not free so hoping Google speech engine is a nice way of doing speech recognition. I have also tried this but I hear only silence when agi is launched, and looks like it is stuck. Nothing happens if I press # or any button, until I hangup and agi script exis. I am working to resolve this and will update accordingly. Any one, any idea about this?

Comment on Get the total amount of lines/channels for a SIP-trunk? by krdian

krdian — Fri, 14 Oct 2011 11:56:24 +0000

Hi,

You can use GROUP() to group channels as You need.

Cheers,

krdian

Comment on (no subject) by krdian

krdian — Wed, 07 Sep 2011 12:01:20 +0000

Have You tried to add g oprion to Dial command ? for example:

exten => _X.,7,Dial(DAHDI/7/09501032209,10,S(60)g)

regards

Konrad Rozycki

Comment on Make ConfBridge hang up on last participant by ZemaFus Asterisk Master

ZemaFus Asterisk Master — Mon, 29 Aug 2011 04:22:41 +0000

Well thats an easy one budd, just insert a hangup line as the next priority line of the extension, that will free the lines immediately upon all internal lines disconnecting from the call. BTW could you please post your bridge configuration, sounds like you have a pretty handy makeup that I’d like to see… thanks….

Comment on trunk not working if I register a phone at the same IP as the trunk peer’s IP by mbithi

mbithi — Mon, 22 Aug 2011 07:44:38 +0000

Why should u do that?

Comment on Cisco SIP 8.5 and 9.0 Issues by anjo2

anjo2 — Sun, 21 Aug 2011 14:56:19 +0000

In line(s) configs, the proxy should be:
USECALLMANAGER

You need that in 9.x firmwares

Anyone can uplooad cmterm-7942_7962-sip.9-2-1.zip ?

Comment on Extension Not found in Context? by Stivaro

Stivaro — Fri, 19 Aug 2011 00:43:32 +0000

Are you running Asterisk 1.8? I’m running into the same issue. Let me know if you find out.

Comment on outbound calls via google voice not answered by toll free numbers with ivrs by BobL

BobL — Wed, 17 Aug 2011 13:31:38 +0000

I experience the same with my companies conference call numbers through google voice.

Comment on DAHDI-Linux-complete On CENTOS Kernel 2.6.32 by DAHDI-linux-complete on CENTOS kernel 2.6.32 | Linux Blog

DAHDI-linux-complete on CENTOS kernel 2.6.32 | Linux Blog — Tue, 16 Aug 2011 21:00:26 +0000

[...] source, I can see the latest DAHDI-linux-complete version is 2.5.0+2.5.0. Original post: DAHDI-linux-complete on CENTOS kernel 2.6.32 This entry was posted in Uncategorized and tagged dahdi, dahi, latest, subversion, the-latest, [...]

Comment on One way audio when using originate… by Graham

Graham — Sun, 14 Aug 2011 17:35:15 +0000

I know why this is happening. I had similar issues with an Avaya kit using SIP a while back. Took me a lot of investigation work to find my. Take a look at my One Way Audio article to see the fix. Let me know if this fixes the issue for you.

Comment on IP-phone configration by Pete

Pete — Fri, 12 Aug 2011 12:53:28 +0000

On Cisco phones, under VOICE —> EXT 1 (or 2 or 3)
Look at your Dial Plan it should show something like this by default:
(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)

Change the first entry (*xx to *xxxxx) that way the phone will know to wait for 5 char after * has been pressed.

Let me know if that helped