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Hi All, Could anyone tell me the real use of "internal_ timing=yes" option on asterisk.conf file? I am using asterisk 1.4.22. As per my understanding if we don't have any TDM card installed with appropriate driver, we use internal_timing = yes to get the timing from ztdummy /ztDahdi. Is there any advantage on enabling "internal_timing=yes" even if we are proving timing from TDM card? I would really appreciate your feedback. ThanksSam -------------- next part -------------- An HTML attachment was scrubbed...

...never sent back to the phone. Instead I just see the usual RTP flows. I've been trying to fix this for hours, does anyone have any ideas how to get this working correctly? Asterisk version is 13.25.0 The settings I think are relevant (I'm using chan_sip): (sip.conf) ignoresdpversion=yes internal_timing=yes progressinband=never silencesuppression=no prematuremedia=no (Per peer) progressinband=yes directrtpsetup=no dtmfmode=rfc2833 directmedia=no silencesuppression=no prematuremedia=no TIA Mark. -- Mark Farmer farmorg at gmail.com -------------- next part -------------- An HTML attachment w...

>Information on timing sources can be found here: >https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces >As noted on that page, ConfBridge can use any timing interface Asterisk >provides, and is not limited to the DAHDI timing interface. Generally, >timerfd is a good timing interface. >That aside, I would try to rule out external issues with the garbled audio

Hi, I am using Debian OS kernel 2.6.22-3-amd64 and zaptel driver 1.4 with ztdummy module for meetme application. I use meetme with SIP channels. I have such problem that when one connects to the conference voice is "cut". Each voice sequence is disturbed. Does any one have similar issue and could give me some advice?? my extension.conf for meetme: ;switch =>

...39; timing module for this test. It has been 1001 milliseconds, and we got 510 timer ticks *CLI> timing test 10 Attempting to test a timer with 10 ticks per second. Using the 'DAHDI' timing module for this test. It has been 1080 milliseconds, and we got 11 timer ticks in asterisk.conf, internal_timing = yes -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100815/d235beb6/attachment.htm

...for DMTF input or when "Wait()" is called. Also, there seems to be a small gap of 2 or 3 not sent packets when playing several files one after another. At the moment I'm using Asterisk 1.8 Certified (Cert 5). In asterisk.conf I've set: highpriority = yes languageprefix = yes internal_timing = yes defaultlanguage = de transmit_silence = yes transcode_via_sln = yes documentation_language = en_US "core show settings" says: PBX Core settings ----------------- Version: 1.8.15-cert5 Build Options: DONT_OPTIMIZE, LOADABLE_MODULES, BUILD...

...1. A dialplan uses the Milliwatt application with 'o' option 2. The internal_timing opion in asterisk.conf is off 3. The attacker sends a large audio packet. The number of samples in the audio packet determines the number of int...

Hi Gang, My production box with my DAHDI cards is a 1.4.26 build. I have 3 test machines that I do IAX communication with. Machine 1 is a real Dell POWEREDGE 1500 running CENTOS running 1.4.30. Machine 2 is a SUSE 11.1 VM running 1.4.30. Machine 3 is another SUSE 11.1 VM running 1.8.0. I can SIP into all 4 machines and life is great. When I try to IAX from the live machine to

Hi, I have set up asterisk to run as non root, and allow admin users to log in to the server as asterisk, which gives them privileges to edit configs in the asterisk home directory. As for connecting to the console with 'asterisk -r' - this by default does not work as asterisk is owned stored in /usr/sbin/asterisk I am reading that the best way to solve this is to use 'visudo' -

We have recently started making more frequent use of the meetme conference of our * system. We are using v1.0.8 with a 2.6.11 kernel on our system. We generally have 4 callers in it: two with the gsm codec and 2 with g729. Initially, the conference works fine and there is little latency. After about 15min., though, the latency is very noticable and by 25min it's unbearable. If we all leave

Hi, Having problems with getting either RxFax or FaxReceive to compile. Running Asterisk 1.4 on CentOS 5. Does anyone have the free/open source executables that you could send me? Thanks for your help! P. S.: TxFax and FaxSend would also be appreciated.

Hi, I've been getting regular CPU usage spikes(50%-80%), due to asterisk (according to top). I never noticed this on 1.4, and I have top running in the background pretty much all the time. In between those spikes Asterisk stays under 10% CPU usage (I have a transcoder card, which helps). It's very regular, never any missed spike, or any spike in between the regular spikes. I

...9;t like to have an expensive Astribank attached to the ALIX board just to provide timing. So my question is: is there any way to improve dahdi_dummy's performance, or maybe some other way to get this to work without the need for a physical DAHDI device? By the way, I have already tried to set internal_timing=yes on asterisk.conf, nothing changed. Here's some relevant info on the system: voyage:~# cat /proc/interrupts CPU0 0: 23 XT-PIC-XT timer 2: 0 XT-PIC-XT cascade 3: 1 XT-PIC-XT 4: 255 XT-PIC-XT...