similar to: TE220B

I've seen this consistently on three systems, with three different cards, and multiple versions of DAHDI. At first I thought the issue only occurred on newer, Nehalem-based, systems, but I reproduced it on a Core 2 Duo box as well. I've tested with dahdi-linux 2.2.0.2, dadhi- linux-complete 2.0.0+2.0.0, 2.1.0.2+2.1.0.2, and 2.2.0.2+2.2.0. The card is a Digium TE220B which uses the

Hi list, After a lot of testing + troubleshooting, I guess I'm observing what I am now calling a regression with zaptel 1.4.10 (is it?) As such I call for peer feedback, before either asking Digium install support or filing a bug. Thanks in advance! System: HP Proliant DL380 G5 with 2x PCI-X + 1x PCIe riser card OS: Centos 5 Kernel: 2.6.18-53.1.14.el5 (also tested under

hello everyone, I have a trixbox server with an E1 card(not digium).It connects to an AVAYA pbx use E1. It works fine.But when i change the E1 card to digium TE220B,there is a problem. When sip extension A(on trixbox) call PSTN extension B(on avaya),A must wait longtime before B start to ring.From the log I find there are two times call. I don't know why the first request be rejected

When I reload the asterisk I get the following warnings: Oct 13 08:29:17 WARNING[10170]: chan_zap.c:10874 setup_zap: Ignoring switchtype Oct 13 08:29:17 WARNING[10170]: chan_zap.c:10874 setup_zap: Ignoring signalling Oct 13 08:29:17 WARNING[10170]: chan_zap.c:10874 setup_zap: Ignoring rxwink Everything works fine as far as I know, I can dial and complete calls. So why am I getting this

I have a pair of snom 360s at a customer and they were giving me Low Memory errors. The distributor suggested updating the firmware. I did that, to the one just below 4.0 (which wasn't released yet). One of the phones is still giving the Low Memory error every 3-4 days. The other one had a broken display that was just RMA'd, so it' hasn't been up long enough to know if the

Today I had channel 1 on my TDM400 go to sleep, verified by pluggin in known good telco lines in various combinations on channel 1 through 4 - problem is channel 1, not anything external. So after seeing lots of stuff on the list re: TDM400's I power cycled, removed board and let linux say nothing's home, replaced board, told linux to ignore it etc. No OS/asterisk/etc changes made on

We have 2 T1's coming from our phone switch to a digium TE220B. We have managed to get CPN and the extension outpulsed from the switch, but call setups are really slow. Our T1's are set up as E&M Wink, and they send us the last 5 digits dialed followed by the 10 digit calling party number (we couldn't get the switch to be happy with *CPN*+5* to use featd). We are using asterisk

Just noticed this today: Digium TE220B Dual Span T1/E1 PCI Express Card with Octasic DSP-based Echo Cancellation Module<http://www.voipsupply.com/product_info.php?products_id=3352> It's about time Digium got on the ball and made PCI-e cards. What are people's experiences with this card? Anyone know if there are plans for a PCI-e analog card for FXO use? -------------- next part

I am thinking of going with a Dell PowerEdge 1950 ||| for a new CentOS/Asterisk set up. It will have dual 2.33GHz processors, 16GB memory, two 500GB hard drives (presumably mirrored). I also plan to get a Digium TE220B to go with it. (a non-dell server is not an option, but I am wondering if there is a better one to consider) The system will be a voice mail repository for 4-6,000 students.

I'm running Asterisk 1.2.26.1 svn rev 79171 on Trixbox 2.2. libpri 1.2.7 and zaptel 1.2.22.1. The hardware is a HP dl360 single cpu with a TE220B. The system load is below 0.10. I moved the server into production, with one PRI, on Friday. On that day we handled a couple thousand calls and I only saw one HDLC abort message. On Saturday half the calls and two abort messages an hour

Hello folks. I've got a strange issue. When I modprobe TE220 I do not see mesages like Launching card: 0 <..> Setting up global serial parameters. You can see how I loaded and unloaded the card for several times - http://asteriskpbx.ru/pastebin/11 lspci can detect the card: 03:08.0 Communication controller: Digium, Inc. Device 0220 (rev 02) dahdi_hardware also: astpbx ~ # dahdi_hardware

Hi I am not sure why this is happening or whether it has anything to do with my iaxmodem setup. When receiving a fax via iaxmodem, I got an error message saying *chan_iax2.c:7542 socket_process: Call rejected by 127.0.0.1: Busy*

Hi, I'm trying to figure out the cause of a soft lockup I experienced: Mar 29 09:38:24 pstn1 kernel: BUG: soft lockup - CPU#0 stuck for 10s! [asterisk:32029] Mar 29 09:38:24 pstn1 kernel: Pid: 32029, comm: asterisk Mar 29 09:38:24 pstn1 kernel: EIP: 0060:[<c046e7fe>] CPU: 0 Mar 29 09:38:24 pstn1 kernel: EIP is at kfree+0x68/0x6c Mar 29 09:38:24 pstn1 kernel: EFLAGS:

Hello list, I have about 20 Digium TE-405Ps out in the field, and I started having trouble with one just recently. The card had worked fine for a month with 4 PRIs in NFAS configuration, and then all of a sudden I started getting a disappearing D channel. A restart of asterisk / ztcfg /module unload-load did not fix the problem, but a reboot [not power off, just a restart] would bring it

Hi All, I have a WD RE4-GP which dropped an Adaptec 51645 RAID controller. I ran a smartctl short test on the drive and it failed with a read error. So I ran the Western Digital's own diagnostic software (DLGDIAG), both the short and extended test on the drive and it passed with no errors. So I ran the smartctl short test again and again it failed. I then ran smartctl long test and that

Hi, Is there a way I can stop logging this specific messages: Dec 18 13:24:52 ERROR[6913] chan_sip.c: Call to user 'remi' rejected due to usage limit of 1 Without having to completely stop logging all error messages in my log files. Thanks, Remi

Hi, I'm getting crazy about ztdummy. I have to replicate a PBX where ztdummy is working fine but for some reason I cannot. The two machines have the same kernel, motherboard, the same gcc version and the same zaptel 1.4.8. On the second machine zaptel compiles without errors and ztdummy.ko is generated but when I modprobe it I get the following error in messages: rtc: lost some interrupts

When dtmfmode is set to inband for SIP, and i originate a call from sip out to the PSTN, I can hear the DTMF digit twice in the audio stream. Once very briefly and once for normal duration. Our Theory: While Asterisk is parsing the DTMF, for a fraction of a second, while the end user generated DTMF is being detected, the DTMF is passed inband. Once the DTMF is detected Asterisk silences it

Hi, I have a Asterisk that connects to the PSTN via a PRI. After Asterisk sends the setup message it immediately sends a 183 Session Progress. Is there a way I can change it so that it sends a 100 Trying instead? Because I am having some issues with a equipment where it does not play a busy tone as a result of sending a 183 Session Progress then the 486 Busy. Thanks Remi

Hi list, I'm having trouble with DTFM tones detection. Usually, some tones are being received duplicated in Asterisk, some not. As you can imagine, that's a very big problem involving IVR menu options, Meetme conferences protected with passwords, and so on. We are currently using DAHDi 2.2.0.2, module wct4xxp, which is managing a Digium TE220B card, with a hardware echo canceller