similar to: Crackly Prompts but Voice OK

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

I have 2 servers, configured identically. Each has a TE405P and 2 PRIs. One server was experiencing crackly audio on one circuit, accompanied by HDLC bad FCS messages. The telco recabled and moved me to another port on the DMS-100. The audio is better, but there are still bad FCS problems on the span. I have moved the PRI in question to the other server, and the problem does indeed move with the

I've freshly installed Asterisk on a Fedora C2 machine. Dual P4's, 2GB RAM. 15KRPM Drive. Using the default configs and added one Soft Sip phone. While listening to the demo the quality isnt very good. It's kind of crackly and skips a bit. Should the sound be better or is that just what you get using IP phones/Asterisk? (I ran the X-Lite phone). Nihal -----BEGIN PGP PUBLIC KEY

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 have had this issues for ages but have ignored it, but my handsets get faint background crackle on the FXS port. When I connect the handset directly into my PSTN, it goes away complete, so I am generating this from somewhere in my asterisk box. I get it immediately when the phone goes off-hook and it stays through the dialing and call progress. I can't think of any other devices

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

We are running asterisk with a single POTS line for local calls and a voip line for long distance. Whenever we receive a call on the POTS line it is more than likely, but not always, going to have significant distracting echo. In addition to that there is occasional heavy crackle or static. I have tried to follow the guidelines at :

Hi, I am about to purchase an Eurotech PRI E1 GSM & CDMA Gateway to operate with my Asterisk's TE412P interface. Anyone here has any experience of having this combination? Any success or failure stories would be greatly appreciated. Thanks in advance. Ash

Yo all, I have a problem with a Dutch (KPN) PSTN-line, connected to an X100P. The call wil sound OK at first, but after 10-20 minutes, the audio will start to crackle. Soon after that, this crackle turns into a continuous noise and the parties won't be able to hear eachother anymore. It also sometimes happens that the party on the TDM400P hears a very loud, short-delay echo of themselves,

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:

Anybody have suggestions on having a 56K dialpool and VOIP connections with an Asterisk box over the same set of PRIs? We've done the PM3 with PRIs for just dialup, but are looking for a way to integrate our Asterisk box and move our voice calls onto the same PRIs. Ian -- Ian White Victoria Free-Net Association email: iwhite@victoria.tc.ca http://victoria.tc.ca/

Hey guys, I'm looking for some realworld specs on somebodys machine that will work with the Digium 4-port T1/PRI card and that will support 96 simultaneous calls. Dell is soon to release the PowerEdge 1850: 2U, Dual 3.6Ghz Xenon, 1Gb DDR2 RAM, Dual 36GB Ultra320 SCSI RAID, Hot swap Powersupply, one 64bit 133Mhz PCI and one 64bit 100Mhz PCI for about $3,000. Tack on a 4 port Digium card and

Hello, I've found a little documentation on voip-info and on the asterisk- users list, although I was hoping for an example of a tried-and-true failover setup between PRI and SIP. We are an outgoing call center that uses asterisk 1.4 connected to 2 PRIs from the local telephone company in one group (g1) and a SIP trunk from bandwidth.com. The PRIs are the primary outgoing service,

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

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

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

Has anyone figured out a way to instantaneously swing over PRIs bearing calls in progress to another media gateway without dropping them? Obviously, this would require a DACS of some sort. But I am thinking that it is possible to swing T1s over in a DACS without actually causing the endpoint to reframe as long as the other endpoint is kept in sync. So, it'd be nice, for example, to bring

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

With multiple inbound PRIs (and hunting across them) coming to multiple [asterisk] servers it is important to be able to do administration, i.e. control which PRIs in the same hunt group take (and which don't take) calls from telco at any given period of time. Our pre-asterisk platform uses SERVICE commands for this purpose to put B-channels into 'out-of-service'/'maintenance'