similar to: te110p and interrupts

I was browsing the applications developed in zaptel and came across zttest. After I run it, I get the following: Opened pseudo zap interface, measuring accuracy... 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 100.000000% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.975586% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%

I have some DID numbers that come into Asterisk via PRI, then connect to a Panasonic DBS PBX via PRI. Outbound calls from the PBX go out via PRI to Asterisk, then out to the Telco's PRI. The sync LEDs on my PBX show that it is synced to Asterisk via the PRI. I have users complaining about random hangups. What is the best way toi approach finding the issue? ref: zttest looks good. ./zttest

Hello, I have some ugly numbers given by zttest for ztdummy on an AMD64 box running linux-2.6.15 compiled for Athlon64. linux-2.6.15, zaptel/branches/1.2 r900, jiffies ./zttest Opened pseudo zap interface, measuring accuracy... 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586%

Another request make me test my t1 card, which has no quality problems, but all that I get is: [root@SIP2-MI zaptel-1.2.1]# ./zttest Opened pseudo zap interface, measuring accuracy... 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793%

I am so fed up with Digium cards. My company first owned a TE410P, I installed it in a Dell server and "enjoyed" its instability (we bought it months before Digium warned about the incompatibility issues). Then we switched to a TE411P for the hardware echo cancellation. Now we want to receive fax (< 20/day) on it and guess what ? Since April 2006 (again a few months after we bought

Has anyone had any compatibility issues with a TE110P card installed on a Dell Poweredge 1950? I noted the following error on the LCD display of the Dell Poweredge 1950: E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0. The Dell hardware owners manual states that it means the system BIOS has reported a PCI parity error on a component that resides in PCI configuration space at bus 0,

Hi Francois, Thank you for your interest. I tried the card alone so I don't think is an IRQ problem (anyway is there a way to be sure?) To be sincere, in all systems I saw (even working ones) there is an IRQ balance failed where Linux boots. The system is new, I tried different processors (P4 and Celeron) with SMP kernel and without. The card has the echo module, could not be

Helo, I have problem with Digium TE110P connected to CISCO 3640 (port on NM-HDV-2E1-60) wth PRI E1 link. I use CISCO now for testing but when I tried with real PBX problem was exactly same. I have this messages in Asterisk conole and log sometimes: NOTICE[1115] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Usually 2-5 such messages in series, can be repeated after 10

Hi, I have a TDM400P running with only one FXO port running on a VIA EPIA MS10000 (1Ghz Via Eden). When I run zttest from Zaptel 1.2.1 it hang and when I interrupt it with Ctrl-C that is the result: ?anyone have some idea about why isn't working? Some additional info: # /usr/src/zaptel/zttest -v Opened pseudo zap interface, measuring accuracy... --- Results after 0 passes --- Best:

> They instantly got us to look at the output of zttest and we found that this was (in their words) 'extremely low', with 'best' and > 'worst' readings of 99.975586% and 99.963379% respectively. Might want to give PCI latency setting a try, it helped for me. My ZTTEST would drop occasionally to 99.95% until I set: setpci -v -s 01:01.0 latency_timer=ff

I just needed to test a dialup modem connection (don't ask) and I had a modem connected to a TDM card (FXS port) which then dialled out via a E1 PRI on a TE4xxp card. See my log below: atdt0198xxxxxx CONNECT 36000 V42bis ** Dial IP ** Username: xxxx Password: Entering PPP Session. IP address is xxx.xxx.xxx.xxx MTU is 1500. OK ATH0 OK ATDT0198xxxxxx CONNECT 31200 V42bis ** Dial

Hello, I had noticed strange crackling sound on my phone calls going through my zaptel device (TDM400P), so i decided to check on possible timer issue, and found lots of issues on forums concerning the sensibility of zaptel with IRQs, and tried about everything: moving PCI slots, noapic and acpi=off boot options, play with different kernel options: iosched/preemption/timer/..., play with

Our asterisk system gains access to the PSTN through a voip provider. We have no digium or other telephony hardware in our system. Do the zttest results still matter to us? Our results were as follows: --- Results after 1007 passes --- Best: 100.000000 -- Worst: 99.780273 -- Average: 99.975763

I am trying to get to the bottom of audio clicks, pops, dropouts with my Asterisk server. These occur even when the system is under minimal load (e.g. 1 Zap device in a queue being played music on hold) and occurs with both Zap and Sip devices so isn't network related. The audio problems occur at the same time on all channels and seems to be when Asterisk "gets busy" and uses

We are having intermittent problems where the person we call reports static when we place an outgoing PSTN call. Only the person called hears static, to us the conversation sounds fine. Never happens on inbound calls. It doesn't matter what channel you call from (IAX, SIP, or Zap). We have a Sangoma A108D with hardware echo cancellation with 2 PRIs to Level3 and 2 PRIs to a Nortel Option

hi all, What is the best Asterisk-compliant for Dell machine is recommended? I will have roughly 400 users in a production office. thanks!!

Hi, I'm trying to connect to Telewest/Virgin Media with a TE110P using asterisk 1.4.13/zaptel 1.4.6. No matter what I try, my span always appears as PRI span 1/0: Provisioned, Down, Active My zapata.conf is currently ----------------------------------- [channels] echocancel=yes echocancelwhenbridged=no echotraining=yes switchtype=euroisdn contect=from-pri signalling=pri_cpe group=1 channel

I'm running asterisk/zaptel 1.4.5. If I load the ztdummy module, the dialplan hangs when it tries to play audio (i.e. Playback) -- and I just hear static on the line. I'm running this on a debian system. I actually have it working on a different debian system but have yet to discover the important difference between the two installations. Any advice on where to look? Thanks John

Hey All, I've been working on trying to get asterisk to play nice under Xen and I've run into a bit of a road block. I'm not using any hardware stuff only ztdummy. First I had issues getting ztdummy to work but that was solved by recompiling the xenU kernel to have CONFIG_CRC_CCITT=y which it was missing. Now ztdummy is seen fine in the virtual machine though when I run

Hello, 1-2 sec pauses happen while * plays (streams) messages/prompts. We get reports about that from users and experience it ourselves randomly. Cannot reproduce it for debugging though, so need to figure out some other ways to fix it. 1. It's not silence recorded within or pauses between audio files 2. It's not load related - can happen with no load at all 3. We use decent boxes - dual