I want a provider that uses SIP, I live outside ..
nice group,Having a Problem with CDRs. If i change the context with Goto() Asterisk write the new exten in dst cdr field. How can i keep the old entry? Any ideas makes me very happy.Thanks for helping..
Can anybody please tell me any ENUM test DID from e164.arpa tree, which I
can use to test some features?
http://www.linuxsystems.it/2012/03/new-t-38-gateway-patch-against-asterisk-1-8-11-0/ I made a new patch from irroots branch and I ported it to 1.8.11. Unfortunately latest one is still against 1.8.8 and porting from subversion is quite time consumi..
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..
Ive some problem setting up Asterisk 1.8: the configuration of the server is 8 core Intel(R) Xeon(R) CPU E5310 @ 1.60GHz 4GB of ram CentOS release 6.2 (Final) asterisk-220.127.116.11 Im using sipp to do some tests, the command I use is sipp -sf /root/uac_pcap_G729…
A client of ours get lots of problem with there voice quality when the do a lot SIP calls. In a application I log the rtpqos audio jitter an lost packets.(see Below) Does anybody know what the numbers mean? If I look at a sample of the channel variabl..
we are working on a new codec for asterisk.Its early stages and the main goal is just to be able to hear it using normal client end points.So what I want is to be able to take a couple of normal G711 extensions, but have asterisk internally force a transc..
The Asterisk Development Team has announced the release of Asterisk 10.3.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 10.3.0 resolves several issues reported by the commun..