similar to: Possible timing issue?

Carlos and Stefan (and other who have helped): I DON'T HAVE the res_timing_timerfd.so file. Can I build it? Recompiling Asterisk is unrealistic in my position but I wonder if I can build the one module. Here's what I do have: apbx:~ $ locate *res_timing_timerfd* /usr/src/asterisk-1.8.23.1/res/.res_timing_timerfd.makeopts /usr/src/asterisk-1.8.23.1/res/.res_timing_timerfd.moduleinfo

hi: my current system is 1.6.2. I have dahdi hardware card. I must disable res_timing_timerfd module or sometimes phone calls would become silent suddenly. I don't know the situation in 1.8. I heard that timing is still a problem in 1.8. should I keep using "res_timing_dahdi" or I can use "res_timing_timerfd" to get some benefit if I upgrade to 1.8? thank a lot for

Hi, We heavily use meetme/SLA functionality in Asterisk, and continuously run into issues with dahdi timing. The two errors we get are: ERROR[6518] res_timing_dahdi.c: Failed to configure DAHDI timing fd for 0 sample timer ticks WARNING[22024] app_meetme.c: Unable to write frame to channel Right now, dahdi in our setup uses the software timer (with res_timing_dahdi.so which gives much better

On 1/14/19 4:02 PM, Duncan Turnbull wrote: > > > Sent from my iPad > > On 15/01/2019, at 10:34 AM, Thomas Peters <TPeters at mcts.org > <mailto:TPeters at mcts.org>> wrote: > >> Duncan: >> >> You may have it right—I took one phone and set the ring time to 60 >> seconds. I now get about 4 rings on that one. >> >> I wonder how I

>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

I sent this last night but it never showed up in the thread list so I'm trying again. Please pardon me if it duplicates. So I've been banging my head against the rack on this one and am now turning to the group for help. I'm in the process of bringing five Asterisk servers (all originally built from source code by myself) from various versions (1.6.2.x,11.6-cert13, and 13.1-cert2) up

Hello, Can't get chan_gtalk.so module to load, neither res_jabber.so: Asterisk*CLI> module load chan_gtalk.so Unable to load module chan_gtalk.so Command 'module load chan_gtalk.so ' failed. [Dec 1 16:10:05] WARNING[2931]: loader.c:387 load_dynamic_module: Error loading module 'chan_gtalk.so': libiksemel.so.3: cannot open shared object file: No such file or directory [Dec

Hi all Sometimes (about every three months) some of my Asterisk 1.8 boxes will start running this message thousands of times in the CLI: [Feb 12 14:18:23] ERROR[28129]: res_timing_timerfd.c:180 timerfd_timer_ack: Call to timerfd_gettime() error: Invalid argument [Feb 12 14:18:23] ERROR[28129]: res_timing_timerfd.c:180 timerfd_timer_ack: Call to timerfd_gettime() error: Invalid argument [Feb 12

Can anyone recommend some kind of virtual t.38 fax software? I'd like to test/debug some of the t.38 stuff, but it'd be much easier if I had a software client that could just generate the faxes from a workstation, rather than having to sit with the fax machine + t.38 ata to source faxes from. There doesn't seem to be much out there, and the stuff that's out there is kind of

starting 161.1-beta3: chan_iax2.c:10925 build_user: Unable to support trunking on user 'iax-out' without DAHDI timing But I have these "timing" modules: ls /usr/lib/asterisk/modules/res_tim* /usr/lib/asterisk/modules/res_timing_dahdi.so /usr/lib/asterisk/modules/res_timing_pthread.so Do I need to do some magic to get these loaded? modules.conf is set to auto. Is this what

We are running load capacity tests using Amazon AWS configurations. For the tests, we are basically scaling up calls to a second Asterisk box. First box that is calling the second box plays music on hold for 60 seconds, then hangs up the call. My initial thought was jitter problems, but that doesn't seem to be the case. I believe I found the cause while looking at the asterisk logs. I am

Folks, I've read some sources claiming that Asterisk does not need DAHDI for timing in 1.6.1. Is this true? Searching the web, all I can find is pages celebrating the fact but no actual documentation on which version it was introduced in and how one would go about configuring an external time source. I'm having a devil of a job trying to compile DAHDI on a hosted Xen VM and thought I

1. Subject. 2. asterisk16-1.6.2.6-1_centos5.src.rpm have not asterisk.logrotate in SOURCES 3. for "--without dahdi" diff SPECS/asterisk16-my.spec SPECS/asterisk16.spec 750a750 > %{_libdir}/asterisk/modules/res_timing_dahdi.so 879d878 < %{_libdir}/asterisk/modules/res_timing_dahdi.so

We are running Asterisk 1.8.5.0 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 was using the DAHDI timing module. We needed the DAHDI timing module loaded so that paging would work. However, at that time we upgraded to 1.8.5.0 and

Hi, How would I go about troubleshooting this: [Dec 24 07:15:11] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc32ed at 192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:12] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaaec88 at 192.168.1.95 for seqno 101

Does anyone have a suggestion on how to handle this? For example, if I have a list of numbers that I want to go out a certain sip channel and another that I want to go out the dahdi device, is there a way to do this? None of the numbers will fit into a pattern, so just plain pattern matching won't do. The most straightforward way would be to just define explicit patterns. Obviously that

Hi, I''ve read ADSL-Bandwidth-Management-HOWTO http://www.tldp.org/HOWTO/ADSL-Bandwidth-Management-HOWTO/implementation.htm l#AEN166 and I''ve a doubt from script: [ ... ] # DNS name resolution (small packets) iptables -t mangle -A MYSHAPER-OUT -p udp -j MARK --set-mark 21 [ ... ] That is a bug ? I think that " DNS name resolution (small packets) " is better

Hello List! My Asterisk stopped making SIP-calls today, I could call from external, and saw Call coming in over PRI, but calling the SIP/Device wont work. I saw 5 open channels - all chan_spy. Only a restart helped. In the messages-file i found from yesterday: [Mar 4 17:28:01] NOTICE[25769] app_chanspy.c: Attaching SIP/209-0000170fto SIP/210-0000170e [Mar 4 17:29:38] NOTICE[25790]

Because some users have requested transfer beep confirmations I've switched our phones over to using the asterisk transfer feature instead of the built in transfer functions of the phones. While testing it was working fine, but I changed something in features.conf and suddenly any time I hit transfer (*2), I can only enter one digit before asterisk immediately tries to dial that extension.

The Asterisk Development Team announces the release of Asterisk 1.8.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.7.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! Please note that a significant numbers of changes and fixes have