similar to: No audio on ISDN PRI calls

We are trying to configure a Sangoma A101 card to allow both incoming and outgoing calls on a UK (BT) ISDN30e line with only 24 channels enabled. At present incoming calls work fine. We can't call out -- we get a BUSY/CONGESTED error. Do we need another context in our zapata.conf? In other words, do we need to reserve, say, channels 17-24 for outgoing calls? I also wonder if the signalling

Hello, I have to debian lenny servers serving SAMBA shared folders to XP SP3 clients. One server is running samba 3.2.3, the other 3.0.24. On one of the shares there is a folder which contains email message files with names like: 1164373321.H21047P2656.mail.domain.com:1Gnagg-0000gi-0p On the 3.2.3 samba server: >From XP I can browse the folder containing that file, but if I try to copy the

Hello, I'm trying to enable CONSOLE logging in Asterisk 1.4.14 on a 2.6.18 Debian Lenny server. It is enabled in logger.conf, however no log file is created in /var/log/asterisk/ Also I was wondering if I manage to get it working and use logrotate daemon as well, is logrotate going to restart the asterisk process each time in order to rotate the log file? Thank you.

Hello, I have an Asterisk running with a Sangoma A200 card with Hardware Echo cancelling connected to the UK PSTN. If a PSTN call comes in, voice both ways is OK, however if an outgoing call over the PSTN is made I can hear the other party OK but they can not, they can barely understand what I am saying, my voice is unclear fading and skipping. Internal SIP and IAX2 calls are OK,

Im trying to install and configure sangoma ... every thing is OK but when type the command "wanrouter start" the following error apears: wan Driver not found. Thanks for any help

Hello, We are having issues with a NEW Sangoma A108D: -- Executing [691918892 at pbx1:1] Dial("SIP/xtravoip200-009d24b0", "DAHDI/g0/691918892|30|m") in new stack [Oct 1 10:04:43] WARNING[14171]: channel.c:3170 ast_request: No translator path exists for channel type DAHDI (native 76) to 256 [Oct 1 10:04:43] WARNING[14171]: app_dial.c:1237 dial_exec_full: Unable to create

I upgraded from Tormenta2 to Sangoma A101. I followed the instructions, and installation was successful. zttool, ztcfg, all show card is installed properly. I copied the parameters from my old working zaptel.conf, zapata.conf and zapata-auto.conf. Verified on Sangoma website that these files are correct. Also configured wanpipe1.conf. But doing all this didn't start the PRI channels. It says

Hi, I have A101DE Sangoma Card( http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a101.html ) lspci shows as 03:04.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card [root at asterisk ~]# lspci -vvv -s 03:04.0 03:04.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card Subsystem: Unknown device

I have installed Asterisk on a CentOS4 box and then installed Asterisk from CVS. I installed a Sangoma A101 and connected it to a Adtran 600 using a T1 Crossover cable. The 600 has 12 x FXS, 12 x FXO interfaces. I ran through the wanpipe install instructions and configured it, now I can run [root@altpbx asterisk]# wanrouter hwprobe ------------------------------- | Wanpipe Hardware Probe Info

We will be connecting our Asterisk server to ISDN 30 and intend using the Sangoma A101 card. The install location is in London (UK). Sangoma card at Voipon http://www.voipon.co.uk/sangoma-a101-pri-isdn-card-p-132.html?gclid=CI32vJz22I0CFQXklAodIgjHaA I would be grateful to hear if this is the right choice of card. Usage reports would be helpful. Regards Rory -- Rory Campbell-Lange

Hi All, I'm about to setup an Asterisk install to take over an old legacy PBX system. At present, the legacy system has modules in it which provides 4 * data ISDN links to the video conferencing unit (Tandberg 3000 MXP) on site, these use the ISDN30 (uk) that the normal voice calls go over. Is it possible to emulate this in asterisk? I've seen zapras but I'm not sure if that's

Hello All, I have a setup of ABE on rPath linux,Sangoma A101D, and a T-1 line (Not PRI) which is all happily coexisting and all lights are green. The T-1 comes in from the world into a "Shark Box" which splits the T into 384K data and 6 channels voice. The data side is working great. The voice side, not so great. It was originally broken out to 6 pots line and Verizon came back

Hi all, I've done this a few times with other PBX's but swyx has stumped me! I'm having some trouble getting Asterisk connected to a Swyx system using a sangoma A104dx... currently the setup is: BT <-> Swyx The above setup works fine... what i'm trying to achieve is BT & SIP Trunks <-> Asterisk <-> Swyx I have connected to our BT (2 x ISDN30 UK) with

Has anyone seen something like this before. Randomly, on longish calls, the local side of the call audio goes dead. Meaning remote caller can hear us but we cannot hear the remote person? Linux voip 2.6.18-128.1.6.el5 #1 SMP Wed Apr 1 09:10:25 EDT 2009 x86_64 x86_64 x86_64 GNU/Linux Asterisk 1.4.24.1, Copyright (C) 1999 - 2008 Digium, Inc. and others. WANPIPE Release: 3.4.1 Wanpipe Config:

Hi guys, Currently I just purchased the TE205P for the installation of my ISDN30 setup. I compiled, setup the card correctly I think. When I do the ztcfg -vvv it shows all 31 channels clear, with 30 bchannels using for voice ranging from 1-15, 17-31, and 16 is the dchannel. However, when I do a cat /proc/zaptel/1 I see only 24 channels configured, and then, when I do a cat /proc/zaptel/2 I see

Hello, I'm considering an Asterisk box for up to 10-15 concurrent calls. Incoming PSTN/ISDN/IAX2, outgoing PSTN/ISDN/IAX2. Could someone roughly suggest the minimal hardware requirements for this kind of setup? Trying to come up with the cheapest solution. Thank you. Veselin K

Hello, I'm using Asterisk with an ISDN30e PRI line (only 16 channels active). Every now and then I get a CONGESTION error even-though there are only 1 or 2 channels in use out of the 16 at that time. When this happens, the user just needs to re-dial and the call goes through OK. On a SNOM phone when the problem occurs, a "Service Unavailable 907" error is shown. [2008-10-14

Hi, Are there any hardware for ISDN30 ? if yes any problem with this ? is i out-of-box like ISDN2 but with 30 linies ? Do I need more than the cable from my teleprowider and a PCI-card ? /HHA _________________________________________________________________ Find high-speed ‘net deals — comparison-shop your local providers here. https://broadband.msn.com

Hello, I'm running an Asterisk 1.4.14 on a linux machine. Serving SIP Snom users. I've noticed that each time Asterisk is restarted, for the first 5-10 minutes, the SIP users can dial but cannot be dialed until each phone re-registers itself against the server. So only after the "Saved useragent...for peer 111" line appears on the Asterisk console, then the 111 user can be

Quoth Jon Morgan <jon.morgan at motors.co.uk> > >We have a 2 port Digium TE220P card, one span is configured to connect to our ISDN30 provider (British Telecom), the other span connects to our internal PBX. Here's the zaptel.conf snip: > >span=1,1,0,ccs,hdb3,crc4 >bchan=1-15 >dchan=16 >bchan=17-31 > >span=2,0,0,ccs,hdb3,crc4 >bchan=32-46 >dchan=47