similar to: Zap PRI load testing

I have 12 T1's going into 3 servers, 4 in each into "Digium, Inc. Wildcard TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02)" cards. Each "group" of T1's have the primary D on 24 and the secondary D on 96. The first server (ts20) and the last server (ts22) can playback "demo-congrats" fine. The "middle" server (ts21) cannot -- just dead air.

Guys I am having some serious issues with my call queue and Management is breathing down my neck pretty bad, and I am running out of ideas. I have a single queue for my tech support department. I originally was using the AgentCallbackLogin for them and it tested out great on our testing weekends, but it hasn't worked out since. It would only let one of them take calls at a time, no matter

Hello All, I have googled and wikied but must not be searching correctly. Assuming the TE110P has same ability as old T100P to use some voice and some data channels, lets say I have a TE110P set to accept voice on 10 channels and pass the other 14 channels as data. Under this scenerio i am "guessing" that * should still be able to accept VoIP calls on the data channels and still

Hello * users I've have a rather disturbing problem, which I don't know how to debug or how to solve, but first a brief description of the set up. One Asterisk server with a TE410P card installed (first line used on this only), and a number of Wellgate 3504A (4 port FXS devices with SIP firmware). There is no connection from the Asterisk server to the outside world or any other

Hello * users Sorry I forgot to send the mail in plain text the first time... I've have a rather disturbing problem, which I don't know how to debug or how to solve, but first a brief description of the set up. One Asterisk server with a TE410P card installed (first line used on this only), and a number of Wellgate 3504A (4 port FXS devices with SIP firmware). There is no connection

I am getting a TON of these errors on the console. I Googled and wikied and greped found the error in the source but cannot understand why it is happening. The system works fine, no dropped calls, no echo, it will even run for weeks with this error. But it just scrolls and scrolls on the console. Temporary fix was to turn off the console monitor! :-) Any ideas. Apr 16 10:40:12

hi all, i am trying to get my TE410P (see previous posts) working in Austria (telekom Austria - i am still waiting for an answer for my questions). my /etc/zaptel.conf looks like -------------------------------- span=1,1,0,ccs,hdb3,crc4,yellow span=2,2,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 loadzone=at defaultzone=at -------------------------------- after

doesnt work for me Dne 29.5.2016 v 17:48 Niklas Larsson napsal(a): > Hi, > > > On 2016-05-27 18:28, Marek ?ervenka wrote: >> after downgrade to 13.8.2 >> May 27 18:21:06 ast kernel: asterisk[16286]: segfault at 1010024 ip >> b49162cd sp bfac0940 error 4 in >> libmysqlclient.so.16.0.0[b48f1000+12e000] >> >> after downgrade to 13.7.2 >>

for today we only have experience with BRI applications together with asterisk. is the following scenario possible and stable enough for production? FYI : We want to build a unified messaging application integrated with SIP. We have an E1 connection in Belgium with 100 msn's We would think about having 2 servers : Server A : Asterisk PRI card (Digium TE410P) Server B : Fax

<P>hi Everyone,</P> <P>We are configuring an ISDN PRI E1 with an E100P card, when you load the drivers, and starts the asterisk, cards also starts fine, givin following output,</P> <P>*CLI> <BR>&nbsp; == D-Channel on span 1 up<BR>&nbsp;&nbsp;&nbsp; -- B-channel 1 successfully restarted on span 1<BR>&nbsp;&nbsp;&nbsp; --

after downgrade to 13.8.2 May 27 18:21:06 ast kernel: asterisk[16286]: segfault at 1010024 ip b49162cd sp bfac0940 error 4 in libmysqlclient.so.16.0.0[b48f1000+12e000] after downgrade to 13.7.2 asterisk is ok Dne 27.5.2016 v 18:09 Marek ?ervenka napsal(a): > btw info from my segfault > > Core was generated by `/usr/sbin/asterisk -f -vvvg -c'. > Program terminated with signal 11,

Hi, one short question: Is it possible for the zaptel driver to deal with multiple phone numbers on one single E1 PRI line? I could make my carrier route +49 xxx aaaaa-zzz and +49 xxx bbbbb-zzz and others down one single PRI trunk to our asterisk box terminating in a Digium TE410P. Does the driver handle this and can I put calls coming in all on the same physical interface put into

hi, I'm trying to configure my newly acquired TE410P card to work as 4 E1 spans. This is supposed to be a drop-in replacement to the earlier E100P card. However, on loading the zaptel module it gets configured as T1 spans basically doing a 'cat' on /proc/zaptel/1 thru 4, it shows 24 channels per span. After this ztcfg fails saying 'ZT_CHANCONFIG failed for channel 97'.

Hi all! I'm trying to replace our old Siemens PBX with Asterisk. I've installed a TE410P card on a Linux box (Fedora Core 2 - 2.6.x kernel). I had no problem with compiling and installing modules and according to dmesg/zttool everything is ok. However, even though I've configured extensions.conf to play a greeting message when dialed, I can't get it to work. When I type

Hello Everyone, I have a system that was working on Sunday 1 P.M. and then gives Congestion on Monday morning. Sometimes over night it probably stopped working. It's a PBXinaFLASH with Asterisk 1.4 and libPRI with a 23 channel PRI connected and 24th D-Channel. This is all I see in /var/log/asterisk/full: [2010-04-19 08:45:50] WARNING[29707] app_dial.c: Unable to create channel of type

Don & others, Thank you for your answer. The fog maybe lifting ;). The zaptel.conf file has the following in its comments: # # The timing parameter determines the selection of primary, secondary, and # so on sync sources. If this span should be considered a primary sync # source, then give it a value of "1". For a secondary, use "2", and so on. # To not use this as a

I have a Sangoma Card with two PRIs. They are both configured in Zaptel and Zapata; In Zapata I have them separated in Group 1 and 2 but if I make a call and specify Zap/g2 it doesn't go when calling Channels : HERE IS what I get: Accepting AUTHENTICATED call from x.x.x.x > requested format = speex, > requested prefs = (), > actual format = gsm,

Has anyone else installed the official 1.4.0 release? I have, and it installed very easily. However, I don't have any of my usual command line tools for monitoring and debugging zap channels and PRI lines: asterisk1*CLI> pri show span 1 No such command 'pri show' (type 'help' for help) asterisk1*CLI> Ditto with zap stuff: asterisk1*CLI> zap show

I have asterisk 1.4 using Queue application. I have this error I must restart Asterisk to correct it. Any Ideas?? log: [Jun 20 16:42:27] WARNING[29339] channel.c: Unexpected control subclass '17' [Jun 20 16:42:38] NOTICE[29337] app_queue.c: No one is answering queue 'myqueue' (13/12/0) [Jun 20 16:44:16] WARNING[30203] file.c: Failed to write frame [Jun 20 16:45:11] WARNING[8044]

Is it possible to detect SIT tones on an outbound call? Specifically this is for an outbound call generated via a .call file over a PRI. I get answer supervision when the SIT tone starts and Asterisk believes this is a successful call. I'm using Asterisk 1.4.6. Thank you! -------------- next part -------------- An HTML attachment was scrubbed... URL: