I made a new patch from irroot's 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 consuming, hopefully my work will be useful to
Today I had no time to properly test it, so feedbacks are welcome.
Squeeze debian packages with t38 gateway will follow. Cheers,
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 timer ticks
According to the following from Asterisk wiki, DAHDI is required for MeetMe. "Some confusion has arisen regarding the fact that non-DAHDI timing interfaces are available now. One common misconception which has arisen is that since timing can be provided elsewhere, DAHDI is no longer required for using the MeetMe application. Unfortunately, this is not the case. In addition…
I've 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-18.104.22.168 I'm using sipp to do some tests, the command I use is "sipp -sf
/root/uac_pcap_G729.xml -s 17000 192.168.200.64 -l 1000"
the file "uac_pcap_G729.xml" just establishes a call, wait 120 seconds
and hangup. Finally the problem is: I cannot manage more than 80 concurrent calls.
but I know that is a limit too low, the same machine in the production
Hi, 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 variables, I see the following number.
local_lostpackets = 7706
local_jitter = 2
local_maxjitter = 11
local_minjitter = 0
remote_lostpackets = 0
remote_jitter = 0
remote_maxjitter = 70000
remote_minjitter = 14000
Hi, we are working on a new codec for asterisk. It's 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 transcode to my new
codec. In this way we can easily run up a bunch of listening tests with
a wide variety of users very easily Ok, so how do I force this transcode in the middle, ideally without
using two asterisk…
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/asterisk The release of Asterisk 10.3.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: * --- Fix potential buffer overrun and memory leak when executing "sip show peers" (Closes issue ASTERISK-19231. Reported by Thomas Arimont, Jamuel Starkey) * --- Fix ACK routing for non-2xx responses. (Closes issue ASTERISK-19389.) * --- Remove possible segfaults from res_odbc by adding locks around usage of odbc handle (Closes issue…