similar to: dahdi timing

Hello, Anyone out there knows the steps to get a Sangoma UT50 or UT51 VoiceTimer USB stick working with an OpenVZ instance of Asterisk? I have Dahdi + UT50 driver installed on mother node running fine but not sure what to do in OpenVZ which has Asterisk installed. Do I have to install Dahdi on OpenVZ? Do I link the Dahdi from mother node to OpenVZ? Sangoma wiki is not clear. Thanks,

Asterisk 11.1 We have a situation where one of our incomings POTS lines will not answer. There are 2 lines configured by the Telco as a rollover group (rings the line that is not busy) and they feed into a Digium AEX410 on the server. The most recent time this happened, I did a /etc/init.d/dahdi status and saw this: ### Span 4: WCTDM/1 "Wildcard AEX410" *53 FXO FXSKS

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

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

Hi, No matter I try, I can not register to Voipwise with Trixbox. It is always in "unregistered" state in sip registry. Here is my last sip trunk configuration: PEER DETAILS: allow=g729 bindport=5060 disallow=alldtmfmode=rfc2833 fromdomain=sip.voipwise.com fromuser=username host=sip.voipwise.com insecure=very maxexpirey=120 pickupgroup=1 port=5060 secret=pass type=peer

Aloha, I'm not sure why I'm getting this error, but I can't seem to get chan_dahdi to load. SIP & IAX2 are working fine. Debian 4 w/ 2.6.28 kernel. Asterisk 1.6.1.5, dahdi-linux 2.2.0.2, dahdi-tools-2.2.0 CLI> module load chan_dahdi.so Unable to load module chan_dahdi.so Command 'module load chan_dahdi.so' failed. [Sep 1 10:57:51] WARNING[31696]: pbx.c:4550

Poking around the zaptel SVN earlier today i see support was added for an AEX410 card recently... I'm going to go out on a limb and assume this is the PCI-Express version of the TDM410? Any hints on a general availability date? -- Matt

I'm having an issue with some Snom 300s on a server running Asterisk version 13.9.1, Dahdi 2.11.1 w/OSLEC and pjsip 2.5.1. There is _*NO NAT*_ involved. Phones and server are plugged into the same network switch, all on the same IP range. The server is running a Wildcard AEX410 analog card with 2 FXO modules receiving incoming analog lines. Occasionally, in the middle of a call, the

Hi Asterisk Users, We've been experiencing some tough time regarding a new Asterisk installation connected to the PSTN via an ISDN PRI with a Digium TE121 with the optional VPMADT032 echo cancellation module. For now, I'll focus on something very specific which is summarized on this email's subject. However, here are some general facts for the context: - System pbxfri went

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

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

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

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

I'd like to have my Asterisk system pick up a certain extension on an existing Panasonic PBX when it rings. (It's connected to some proprietary Panasonic doorphones that I haven't replaced yet.) I connected that extension to an FXO port on a Digium AEX410 card, and set that channel to have the context "doorphone". The problem is that the extension is never executed. With

The Asterisk Development Team announces the release of DAHDI-Linux 2.4.1.1. DAHDI-Linux 2.4.1.1 and DAHDI-Linux-Complete 2.4.1.1+2.4.1 are available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete 2.4.1.1 is a maintenance release to fix a regression in DAHDI-Linux 2.4.1 from DAHDI-Linux 2.4.0

The Asterisk Development Team announces the release of DAHDI-Linux 2.4.1.1. DAHDI-Linux 2.4.1.1 and DAHDI-Linux-Complete 2.4.1.1+2.4.1 are available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete 2.4.1.1 is a maintenance release to fix a regression in DAHDI-Linux 2.4.1 from DAHDI-Linux 2.4.0

I have two servers each with one TDM card from Digium and three POTS lines going into each server (each POTS line is one individual number). and I want to have all incoming calls into server B be directed to the IVR on server A and to be able to fail over to Server B in case Server A has an issue (it can be a manual failover and I actually prefer it to be a manual process). I've

Hi, Continuing with the saga of Digium vs MTNL Mumbai, looking for suggestions on handling incoming Caller-ID issues. The card manages to grab a couple of (random) digits of the incoming CID, but they're more or less useless. Is there any way to fix this? Asterisk 1.8.13, Dahdi 2.5.0.1 on Debian Testing (Wheezy), MTNL Mumbai. Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express)

>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

The Asterisk Development Team has announced the release of Asterisk 1.6.2.17. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.17 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this