similar to: Activate/Deactivate Hardware echo cancellation on TE406/TE411 when briging

Hi below are my configs: pstn(e1)--->asterisk (span1)----->legacy pbx(connected via span2)-----> legacy pbx analog extensions. my dial plan is like callers dial into asterisk(span1) , hear an IVR option and they are connected to the agents via the legacy pbx (which is in sync with asterisk on span2)....This works perfectly fine until about 200 calls or so...After that time when asterisk

Hi! I have the chance to play with a couple of E400P cards, each installed in a IBM e330 XSeries servers (2 x 1GHz P-III CPU 2 Gb RAM, 36Gb SCSI HDD with RH8.0 2.4.18-smp kernel), and I'm trying to test/benchmark this e330/E400P combo generating calls thru /var/spool/asterisk/outgoing One e400P if doing the carrier work making calls and the other just receives the calls: Server#1

Has anyone on the list tried one of these new cards with built-in echo cancellation? This electronic message transmission, including attachments, is for the exclusive use of the individuals to which this e-mail is addressed and is to be reviewed and used exclusively for authorized company purposes. This transmission may contain proprietary, confidential or privileged information. If you are

I've googled / voip-info'd / searched until my eyes are blurry, but couldn't see the info I was looking for. I've turned here for help! Asterisk CVS head (9/7/04) Fedora Core 2 (updated to 2.6.6 kernel) DE405P (jumpers set to E1) I want to put asterisk in the middle of our current pbx (Meridian Option11) Currently the meridian has a 2MB pri EuroISDN card linked via a rj-45 into

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Ok Guys, I've done a tonne of hunting around on this problem, but can't find much help. I'm running: asterisk 1.4.19.1 libpri 1.4.3 and zaptel 1.4.9.2 which I believe has been modified by RedFone to add the ztd-ethmf module. My interface is a RedFone foneBridge2 4 Span; and I'm connecting to a BT E1 PRI / ISDN30e with 15 lines on span 1, and a legacy Panasonic PBX on span 4. Upon

I just wanted to post here and let everyone know that the TE406P (quadspan T1/E1 with hardware echo can) kicks some serious ass. We've been running a PRI now for over a year with Asterisk (every single call in and out is through two Asterisk boxes, including faxes) and while the software based echo cancellation is more than adequate, we'd get the occassional "edgy" echo and

Dear All, I was wondering will there be any problems or changes that I will need to do to compile the current Asterisk(1.2.7)/Zaptel(1.2.5)/Libpri(1.2.2) source from www.asterisk.org into a 64-bit binaries? I am currently using the following hardware for my new server. CPU: Pentium D 930 3.0 GHz Mobo: Intel D945PSN Motherboard RAM: 512MB 533MHz DDR-2 Drive: SATA II Seagate 160GB Card: TE406

Hello, Is it possible, for testing, to connect an cat5 straight patch cord between 2 ports of a Digium B410P card and use these 2 ports as a normal dahdi trunk ? I've tried this: One port is set as NT, the other as TE. I would expect timing to come for system hardware so I choose in /etc/dahdi/system.conf : span=1,0,0,ccs,ami span=2,0,0,ccs,ami Results: - both ports lights are green -

asterisk-users Any help will be appreciated. This card did not connect with E1 line how to loop E400P card to test ? now I loop the card. span 1 ---span2 RJ45 pins 1--4 2--5 but show : When calling ,showing error: app_dial.c:764 dial_exec: Unable to create channel of type 'Zap' Asterisk Ready. *CLI> -- Registered SIP '2002' at 192.168.139.59 port 3289 expires 120

Hi, No one seems to have any issue with the following posting. Can any one suggest how to install/configure channel drivers to work. Basically I am trying to send the SIP calls to GNUGK but Asterisk reports the error "No channel driver found". >>> I was trying to compile the oh323 channel driver but unable to compile the openh323_1_13_5 (which is the only required version as

Hello, I am in Argentina configuring an E1 in R2 and have some inconveniences. When the call takes the salient line, I receive an error of signaling. In * I have loaded the following: - libr2 installed - The [ar] in zonedata.c and indications.conf is configured - zaptel.conf o: span=1,0,0,ccs,hdb3,crc4 span=2,0,0,cas,hdb3 #span1 bchan=1-15,17-31 dchan=16 #span2 cas=32-62:1001

Hello, I have a busy problem with Asterisk when I try to transfer a call from PRI directly to IVR. This problem appear sometime after 2 hours or 2 minutes. The log file contain : Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) When this problem appear I must restart Asterisk to solve it. Another thing, I don't know why the alarm is set to NOP on SPAN

I have a two port TE205P Digium card. I have set everything up to create a native zap bridge between the two spans. Everything works perfectly except one thing. Our telco has a "password" that has to be entered as soon as a long distance call is made. So if I dial a long distance call from my meridian system, asterisk bridges the call between two channels, my telco picks up and gives

Hello! If I have more than a hundred analog telephones (analog lines) that need to be connected to Asterisk PBX, what kind of hardware do I need, and where can I buy it? Thanks in advance!

Firstly, the problem... Ever since I installed and setup asterisk, I have had various problems, initially it was echo caused by the ISDN (isdn4linux) card I was using. So, I upgraded to the X101P from digium. I still had echo, so I figured it was also caused by the ATA186 (cisco) I was using. So, I upgraded again to the TDM40B quad FXS card. This solved pretty much all my problems, except

Y'all will know me on this circuit for the past year or so, and you'll know I've done some pretty intense testing with various aspects of Asterisk and Zaptel drivers. The Kris Boutilier's modifiecations to the MARK2 echo canceller are A#1. I have always had a little residual echo on my home zap lines and the odd echo on my PRI lines (not connected through the TE406). It was

Hi there, Im getting this notice in CLI, but the call quality is okey, Im using digium TE406 and asterisk 1.2.4. here are the CLI actual logs: -- Executing SetAccount("Local/50015308467418@default-ca2e,2", "XXXXXX") in new stack -- Executing AGI("Local/50015308467418@default-ca2e,2", "call_log.agi|50015308467418") in new stack -- Launched AGI

I found a bug in the latest T38 passthrough patches, the effect is that a non-SIP call after being put on hold is then lost, no resume is possible. The fix is to be applied in the chan_sip.c file: } else { /* No bridged peer with T38 enabled*/ transmit_response_with_sdp(p, "200 OK", req, 1); } - } + } else transmit_response_with_sdp(p, "200

Nothing from me is posting to the list either. Julian