[asterisk-users] Removing `chan_dahdi.conf`

Hi guys, I'm currently playing around with optimizing an Asterisk install (trying to remove as many possible configuration files as possible) for testing and debugging purposes. I've been able to remove most of the files and maintain an error-less Asterisk full file, with a single exception: I

Re: [asterisk-users] realtime sip peers : musiconhold class

I took this from the wiki, but it's not working : [r...@asterisk testing]# sox test.wav -r 8000 -c1 test.alaw resample -ql sox sox: effect `resample' is deprecated; see sox(1) for an alternative sox formats: no handler for file extension `alaw' Jonas. On 08/14/2010 04:30 PM, Motiejus Jakštys

Re: [asterisk-users] realtime sip peers : musiconhold class

And even when I think the format is correct : [r...@asterisk testing]# sox test.wav -r 8000 -c1 test.wav resample -ql sox sox: effect `resample' is deprecated; see sox(1) for an alternative sox wav: Premature EOF on .wav input file [r...@asterisk testing]# file test.wav test.wav: RIFF

Re: [asterisk-users] Removing `chan_dahdi.conf`

On Sat, Aug 14, 2010 at 11:46:49PM -0700, Randall Degges wrote: Hi guys, I'm currently playing around with optimizing an Asterisk install (trying to remove as many possible configuration files as possible) for testing and debugging purposes. I've been able to remove most of the files and

[asterisk-users] Timing on Asterisk

Hi All I am occasionally hearing a slight pop or skip in an audio message playback on one ALL SIP installation. There are other Audio problems with the installation too (underwater Audio, 1 second 1 way audio delay (takes 1 second for Audio spoken by the customer to reach the agent, and Robotic

[asterisk-users] 603 error

Hi, I have an interesting problem that the dial out via sip always generates 603 error The following is the sip debug Your help is appreciated. CK == Using SIP RTP CoS mark 5 -- Executing [998560...@dlpn_dp1:1] Dial(SIP/6100-005b, SIP/13398560...@hkbn2b) in new stack == Using SIP

[asterisk-users] Use of Storage Area Network with Asterisk

Are there any best practices for using a SAN with Asterisk? In the past we've kept config files local, but voicemail on a SAN. Aree there any issues with latency putting voice prompts, configs, etc. on a SAN? Anyone have some best practices to share? MD --

Re: [asterisk-users] Use of Storage Area Network with Asterisk

On Sun, Aug 15, 2010 at 9:09 AM, Michelle Dupuis mdup...@ocg.ca wrote: Are there any best practices for using a SAN with Asterisk? In the past we've kept config files local, but voicemail on a SAN. Aree there any issues with latency putting voice prompts, configs, etc. on a SAN? Anyone

[asterisk-users] Realtime Context

Hi, I'd like to be able to create contexts in real-time when I add new clients to my asterisk box. Currently, I have to create a blank context in extensions.conf and add:- switch = Realtime/@ Is there any way to avoid the step of creating the blank context and simply include all the entries

[asterisk-users] Fwd: 603 error

Hi, I have an interesting problem that the dial out via sip always generates 603 error The following is the sip debug Your help is appreciated. CK == Using SIP RTP CoS mark 5 -- Executing [998560...@dlpn_dp1:1] Dial(SIP/6100-005b, SIP/13398560...@hkbn2b) in new stack == Using SIP