similar to: VPM450: Not Present

Hi all, We need to get better echo cancellation on an Asterisk gateway. Currently it has two TE410P (1st gen) cards. So would it be possible to just buy two VPM450M cards ? Or do we need to buy two new TE412P cards ? In that case a Sangoma A108d card would be nice as well ? What configuration gives the best audio quality ? Thanks, Leon de Rooij leon at scarlet-internet.nl

Another day, another apparant unexplained hardware incompatibility. I have a TE412P and a TDM400B living quite happily in a whitebox using an Intel motherboard: http://www.intel.com/design/servers/boards/se7230nh1-e/index.htm I tried to move to an IBM x3650 system. It uses a slightly newer chipset, but apparantly it's in the same family. The SE-7230 board has been EOL'd and the

Hi, I was able to implement T122p one port PRI and was able to call out, but I am planning to use TE412p (includes echo cancellation) 4 port digital card (PRI), I wanted to know can asterisk support 3 four port PRI cards (12 PRI connections) with proper hardware like dual core quadcore processor and 8gb RAM in one server? Also I was planning to implement using 64 bit architecture with Asterisk:

Can someone point me in the right direction to find documentation on best practices when setting up a new Asterisk server? I'm using RHES4 and Dell 1750 with TE412P. My current problems are frequent crashes and choppy audio so I think I can easily tweak these out of the picture. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello list, I'm currently looking into building a new Asterisk server, due to some codec problems i've got to transcode most of my channels between Alaw -- G729. Is there any indication on how many channels you would be able to transcode on a certain platform? I'm looking into dual Xeon or dual Opteron configurations, which of these platforms would perform better? And how much power

Can I run two instances of asterisk sharing a single te412p ? I want to be able to have several asterisk servers (for testing various scenarios) running on one server. I was wondering if these asterisk processes could share a zaptel/dahdi card nicely. Julian

Just to share some experience with everyone about what happened today to our Asterisk 1.4 box with Digium TE412P card. We had an unscheduled power outage which shut down the Asterisk box. When the power went up, Asterisk came back up okay but the ports on the card were all red. Zttool show red alarm and cat /proc/zaptel/1 show red alarm today. Both incoming and outgoing cannot be made. When a

More than 60% of our outbound calls are now to mobiles, so the time has come to whack in a gsm channel bank. Does anyone have any preference of bank ? Do you use a PRI or VOIP connection from the bank to asterisk ? Real-world experiences are sooooo much better than marketing blurb ;) We currently have a TE412P with a free socket, so we have a choice either way. I am looking for up to 30

Dear Users, I am fairly new to Digium and Asterisk. I wanted to know that if I use the Digium product THREE Digium Wildcard TE412Ps? (Quad E1 Card) how many calls can I handle simultaneously. I want to use the cards with the following Configurations: Intel? Xeon? 3.00GHz/800MHz, 2M Processor 1GB PC2-3200 DDR2 SDRAM (2x512MB + 2X1GB) memory Integrated Dual Channel Ultra320 SCSI Adapter NC7781

Hi all, I have 1 Card Digium TE412P and 2PRI E1. I have more problems with drops lines. The asterisk log is this: May 16 10:52:26 NOTICE[4465]: chan_zap.c:8194 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 1 May 16 10:52:26 WARNING[4465]: chan_zap.c:2287 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! May 16 10:52:26 NOTICE[4465]:

I am testing analog phone and fax machine plugged into Rhino Channel Bank which is connected to TE412P card. This site is in China. I am running RHEL 5, Asterisk 1.4.21.2, Zaptel 1.4.11 and libpri 1.4.4 I ran into a problem which is analog phone can hear dial tone and can make outgoing calls. Another phone (ether internal or external) can call the analog phone ***but the phone does not

I'm scoping out HA for a relatively simple Office/Call Center PBX. Current setup uses a TE412P with 4 PRI our telco with SIP hard/soft phones for users. Some outbound also goes to a SIP provider. Active/Active looks to be too much hassle for an installation this size, so we're looking at adding an extra * in an active/passive configuration with Linux-HA in between them. Does anyone

Hi all I have a server that has two TE412P (T1/E1+DSP) cards installed. One of them configured as an E1 PRI connected to PSTN and another one configured as a T1 E&M connected to Avaya PBX. Each card only uses two ports, so there are 2 E1 lines and 2 T1 lines connecting to this server. The purpose of this server is as a TDM trunk gateway that gets call from E1/T1 and then forward to an IP-PBX

I am using Asterisk 1.2 with a TE410P connected to E1 PRI trunks. The application relies on a DTMF digit string sent by the phone after the call has connected. This DTMF is detected by Asterisk under the control of WAIT FOR DIGIT commands send from an AGI processor over a FastAGI connection. Usually the DTMF is detected without error, but on a significant minority of calls, Asterisk is missing

We're looking at using 4 or 8 port T1 cards with echo cancellation and are evaluating brands to go with. We know that Sangoma has excellent solutions especially when it comes to echo. But we still have to hear about actual performance of a Digium card using the same Octasic DSP echo canceller. Would appreciate hearing something on this. --------------------------------- Sponsored Link

I have been reading with great interest the posts on trouble shooting echo cancellation with *. Is it just coincidence that all of this discussion has been with analog lines. Are PRI's susceptible to echo problem like POTS lines. Thanks for clearing this up. Alan

I'm hoping people can suggest some ideas for debugging a problem that I'm having with DTMF. Unlike most of the DTMF problems reported here, it has nothing to do with Asterisk interpreting DTMF. My problem is with the synthesis of DTMF tones on outbound calls on a PRI connected to a TE412P card. I'm running * 1.4.10.1 with Zaptel 1.4.4. It is important to note that these problems

Has anyone successfully run * 1.4 with the following configuration (or something very similar)? HP DL380 G5 (3Ghz Xeon) CentOS 4.5 (kernel 2.6.9-55) Asterisk 1.4.5 (or 1.4.4) Zaptel 1.4.3 (or 1.4.2.1) TE412P TDM400B (2x FXO and 2x FXS modules) I've had this rig running * 1.2.18 with Zaptel 1.2.17.1 for several months without any issues. Upon trying to upgrade to * 1.4.4 and Zaptel 1.4.2.1 a

Hi Team Can someone advice me on how i can lower the load average on my asterisk server? dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.10.1 asterisk-1.4.25.1 2 X TE412P Digium cards on ISDN PRI Im using the system as an IVR without any transcoding or bridging ************************************** top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75 Tasks: 149

Greetings list, I've been asked to provide a system for 200 extensions, most of which will be existing analogue POTS handsets, not IP handsets. I've not really had any experience with large channel banks in the past (since most of our deployments are strictly IP-only to the desk), so I'm at a loss as to which ones are worth looking at. If anyone's had experience using channel