* You are viewing the archive for March, 2012

Free calls to the US question

I want a provider that uses SIP, I live outside of the US.

Keep dst CDR Record If Context Change

Hello 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 me.
Daniel

any enum test number of e164.arpa tree?

Can anybody please tell me any ENUM test DID from e164.arpa tree, which I
can use to test some features?

Thanks,
Ricardo.

T.38 gateway patch against Asterisk 1.8.11.0

http://www.linuxsystems.it/2012/03/new-t-38-gateway-patch-against-asterisk-1-8-11-0/

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
someone else.
Today I had no time to properly test it, so feedbacks are welcome.
Squeeze debian packages with t38 gateway will follow.

Cheers,
Niccolò

meetme with timerfd

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[6518] 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 to

providing timing, DAHDI also provides a conferencing engine which the MeetMe

application requires.

I’m curious if DAHDI require/use res_timing_dahdi for it to run/function properly. Can we use res_timing_timerfd (instead of res_timing_dahdi) along with DAHDI for MeetMe?

Thanks a lot,
Matt

concurrent channels limit

Hi,
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-1.8.10.1

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
environment, with the same configurations and asterisk 1.4 manage up to
400 concurrent calls.

How can I find my real upper call limit?

tnx