I am currently having a voice quality problem with one of our Asterisk servers.We have checked the network and we have found no problems that could cause the voice to sound cracked and with small interruptions.I am looking at the timing source for Aster..
I have an Asterisk 13 installation with an E1 card and I thought that DAHDI would be the default timing source for the system: pbxcore*CLI> module show like timing Module DescriptionUse CountStatus Support Level res_timing_dahdi.soDAHDI Timing Interf..
Im interested in using the testing a non-DAHDI timing source to have some assurance Im on a system thats not likely to give me grief over timing-related issues.Im familiar with dahdi_test and the guideline of needing 99.975% accuracy for reliable conferenc..
We are running Asterisk 220.127.116.11 with an uptime of 40 weeks. Just today our streaming music on hold stopped working. I remember when we had first installed 1.8 we had an issue where the streaming music on hold would not work because Music On Hold ..
If you are about to dive into the process of virtualizing some of your Asterisk infrastructure then the primary issue preventing you from moving might be the lack of proper timing as you need it for IAX2 trunking. These are your main options and th..
I know that Asterisk on virtual machine require a timing source. What would you suggest to use for timing? We will plan to use only SIP..
We use MeetMe with res_timing_dahdi as the timing source, and once a while we get the following error which then causes Asterisk to crash/restart (with safe Asterisk). ERROR res_timing_dahdi.c: Failed to configure DAHDI timing fd for 0 sample ti..
I have to make an asterisk gateway in front of several other asterisk. This gateway will essentialy be used for outbound call. This gateway will be connected to other asterisk by IAX trunk, outbound call will use SIP trunk (voip provider or patton isd..
The Asterisk Development Team is pleased to announce the release of Asterisk 18.104.22.168. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/The release of Asterisk 22.214.171.124 resolves several issues repor..
You mean say 0=Slave (Use PSTN clock) 1=Master(generate Internal clock) So best option is 0 for all span if you connected on PSTN right ? Date: Fri, 27 May 2011 17:27:43 -0300 From: email@example.com To: firstname.lastname@example.org Subject: ..