similar to: DTMF not completely muted

Hello everyone. I have a timing slips errors and I can't understand what source of the problem is. My installation has 2 digium cards: TE420 and TE220 cards in one server. There are 3 spans (E1) to PSTN and 3 spans to internal PBS stations - normal installation for transit communication. Span configuration is: span=1,1,0,ccs,hdb3 #TE420 - first port. To PSTN. span=2,0,0,ccs,hdb3 #TE420 -

So historically I've done one of two things on systems where I've needed to use MeetMe * used a real Digium card, and I've only ever used a TE400 or a TE420 for that purpose, and I know they have the timing chip * used dahdi_dummy, which works well with light load, but I had it running on a very overloaded server and had audio quality issues. I may have had quality issues even with a

Hi, folks. I have a problem using Asterisk 1.2. I create conferences using app_meetme and Zap channels, and for every participant I run the script defined by AGI_BACKGROUND_SCRIPT to be able to listen and react to DTMF tones. As the docs tell me, when using the AGI background script one loses the ability to control the meetme conference via the command line so for muting conference participants I

Hi, One of our customers has an issue with the callee not being able to hear them. It seems to happen very frequently on one number in particular where there are about 3 IVR menus to dial through before getting to a live person. However, this does not happen on every call. Running tcpdump on the RTP packets, I can see that RTP is setting sent, but the values in the packet are all very close to

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

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, I'm having problems with a TE420P card, in which I cannot make calls using spans 2 through 4. After a couple of days of working correctly, spans 2, 3 and 4 start failing (can not make calls). The system is configured to work with SS7. After the ACM message goes out, immediately a REL message is returned. I searched for error messages in /var/log/messages and could not find a clue

Hello! I am trying to configure my digium TE220 dual-span pci express card with Dahdi. I seemed to have managed to set up the card with the Dahdi kernel, as demonstrated by executing dahdi_scan: [1] active=yes alarms=RED description=T2XXP (PCI) Card 0 Span 1 name=TE2/0/1 manufacturer=Digium devicetype=Wildcard TE220 (4th Gen) location=Board ID Switch 0 basechan=1 totchans=31 irq=16

This may not be the right place to ask, but I have an annoying issue with a Sipura/SPA1000-2.0.10(e) ATA device connected to an Asterisk box. (The system is remote to me, so I've only been able to observe this by dialling into a VoIP phone on-site, then run commands on the box remotely!) First of all it's all working fine connected to an Asterisk box and the user can make/take calls

I am having a problem trying to do inband DTMF passthru via asterisk. My setup: PSTN gateway MAXTNT v11.0 SIP (T1 PRI/NT2) Asterisk HEAD or v1.0 makes no difference (I am using HEAD mostly) 12/10/04 and 01/17/05 (no difference) CAC ABII-T100P to/from analog lines to/from asterisk BTW, I have used a ABI and it works just like the ABII with asterisk. What I am seeing is: I make a call from a

We are attempting to configure SIP trunking between asterisk 1.2.22 and a Mitel 3300 CX box. The Mitel machine will gateway to the PSTN for us. I found this earlier post about doing this from July: http://lists.digium.com/pipermail/asterisk-users/2007-July/191630.html Unfortunately the promised configs never came ;(. We're having the exact reverse problem: we can register with the Mitel

Hi I'm having a very odd phenomenon happening on our production server (1.4.17 and using realtime). Sometimes a call will disconnect 30 seconds after the SIP phone hits the mute button but it doesn't happen all the time. I've done a sip debug while watching this happen and that doesn't show anything other than a BYE message being sent out of the blue. The rtptimeout and

My Setup looks like this: Mitel 200 SX (1st T1) -------- Bell South (2nd T1) | | | Digium TE110P Asterisk MITEL CONFIGURATION Primary Timing Source: 1st T1 Card Secondary Timing Source: 2nd T1 Card ASTERISK CONFIGURATION span=1,1,0,d4,ami (Look to the Span for timing) We are getting a lot of Frame and Slip errors.... Time Slip Frame 7:00 736 950 8:00 690 1200 9:00 437

hallo everyone, fyi ... working SIP Trunk configuration between Asterisk and Mitel 3300 ICP attached. let's refine further, please test and share your feedback, regards, Joseph Okoegwale Abuja, Nigeria -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070709/4f7f15e1/attachment.htm --------------

I have an Asterisk box, 1.4.30 with a PRI. A Mitel 3300 is connected to the Asterisk box via SIP trunking. When a user calls from the Mitel through the Asterisk box the user can speak but can not hear the far end. But - when I route the Mitel user to echo() it works, send and receive. The Mitel user also can record and playback greetings. One thing I have noticed is that when the Mitel user

I've just done some rudimentary testing to see how Vorbis degrades at absurdly low bitrates without downsampling. In summary, don't hope for anything decent below -q 0 for now. I tried oggenc -b <bitrate> -M <bitrate> for the below and a few in between: 24k - spectral energy "floor" captured decently, but many pure-tone blips (think old computer movie sound effects)

Does anyone have any experience getting a Mitel SX-200 EL/ML PBX system to work with *? I'm not getting inbound or outbound calls to work. (Inbound caller gets line busy tone.) SETTINGS FROM MITEL: I built a Crossover cable and connected it like this: PSTN--T1--ASTRISK--T1--OLD MITEL -Crossover Cable Pin-out: 1 - 4 2 - 5

Hi: I want to have conference call with asterisknow and need 2 ports E1.Which Digium card is better?TE212 or TE220.I haven't problem with motherboard. Regards. --------------------------------- Get the Yahoo! toolbar and be alerted to new email wherever you're surfing. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a problem which I need to resolve. We are trying to put an asterisk between a Mitel PBX and the world. We are adding Voip service via Asterisk. Here is are config files for the settings but our problem is the following. We are able to send calls to the Mitel pbx and it's the T1 connections is green saying it's ok. The support department from Mitel said that they use e&M and

Hi All, Our experience with * to date has been a bit limited. It's a 4xCisco 7960 network, linking our head office with a faraday caged datacenter. As a way of putting voicecomms into a sealed room, it was quick and easy to deploy, and works very well. As typically happens, we've now thought about extending the use of asterisk - and a new opportunity has cropped up. In three months