similar to: Issues with sound quality and HDLC

Hi all, I was looking to build a SIP-to-PRI gateway using Asterisk (as in my other post), but there is also an alternative of using a Cisco router with something like an NM-HDV module with a T1 VIC module and DSP channel banks. The question is, would it be more reliable to offload all dahdi/zaptel/libpri type stuff to a dedicated gateway device (Asterisk or Cisco) and have the Asterisk PBX only

Hi all, I'd like to setup a really lean Asterisk installation that essentially has a full ISDN PRI (AT&T, T1, 23 B-chans, 1 D-chan, BZ8S, 5ESS, National dialplan) on a Digium TE207P adapter that all it does is convert the ISDN channels to SIP/IAX channels. Then I would add this Asterisk 'gateway' as a provider on one (or many) Asterisk systems on the back. With such a config I

New never used bought by mistake ( 5V pci slot ) and stored... pls make offer, thanks and regards, Jean-louis Curty The TE207P is a bundling of our leading TE205P product and our new VPMOCT064 Octasic DSP-based echo cancellation module. The TE207P is Digium's and the industry's first two-port digital card featuring hardware-based echo cancellation. The new module enables users to

I need to place a SIP FXO gateway in Central America. I've been looking at Quintum products, but the prices are about $150/FXO port. I have a Dell SC400 on the shelf, and I'm considering just installing Asterisk and two TDM04B cards and shipping it down. Does anyone have multiple TDM cards in the SC400? FXO ports on a TDM card are about $75/FXO port. With the Dell, there will be

If you are having trouble reading this email, read the online version<http://now.eloqua.com/es.asp?s=491&e=78675&elq=55426a8b6c714f5bb6f2bf4b5d37bf55> . <http://app.en25.com/e/er.aspx?s=491&lid=215&elq=55426a8b6c714f5bb6f2bf4b5d37bf55> Dear Lito, *The information in this email is given to you in advance to make you aware of an impending product release

As you may have heard, Digium announced this morning that it's acquired Switchvox, a well known provider of Asterisk-based phone systems. Since several people have already asked me about the deal, I figured I'd let you all know my feelings on the matter. First of all, let me say that I personally think this is a great thing for all the parties involved. Obviously this gives Digium a

Hello, I need to have an extension on a SwitchVox server dial out to one on an Asterisk (FreePBX actually) box which will host a voice directory. The Asterisk server will then need to dial one of the SwitchVox extensions if it gets a voice match. Anyone has done that, and could let me know how? So far it looks like IAX peering (what SW calls "SwitchVox peering") could work? Thanks in

Does anyone know what version of Asterisk Switchvox uses, and if it is modified in any way? FWIW, I am dealing with a provider that claims compatibility with Switchvox but not Asterisk for their SIP trunking service.

I just inherited a client that is using a Switchvox system. I normally install a CentOS based system with freePBX and some custom endpoint management stuff for Polycom phones. This Switchvox is making me feel a bit stifled. I am having nightmares of another recent encounter with Trixbox Pro. Can I really not ssh into this box? If I could is there anything useful that I might change

I have a customer that wants to get switchvox, since I have never used it, I would like to hear some feedback from active users of switchvox. In specific: 1. Does it use realtime or conf files 2. Is it possible to change it manually? 3. Is SSH access to login to console/shell available? 4. Are you or your customers happy with the user interface? TIA

I'm impressed with the call log reporting that Switchvox offers. There's a screenshot here: http://www.switchvox.com/sv?cmd=screenshots&pic=13 Can normal Asterisk do this? I installed AsteriskNow on an old computer, but I couldn't find anything like that in the GUI. I found the CSV file in /var/log/asterisk/cdr-csv, but that's not much help unless there's some other

I need help on my zaptel.conf and Zapata.conf for a TE207P I'd like Span 1 to receive a PRI from the phone company(US PRI). I'd like Span 2 to interface with a Nortel Phone system as a PRI(acting as the phone company) Essentially my asterisk box is a man in the middle intercepting calls from the PRI passing certain DID to the Nortel, also intercepting calls from the Nortel passing them

Hi Related to this message: http://lists.digium.com/pipermail/asterisk-users/2005-February/090205.html After having a lot of trouble with abort and breakdown of the PRI connection I finally figured out what was wrong: The kernel was compiled for generic ide chipsets which doesn't support DMA. When our daily backup created some load for the disks the interrupts for the digium card

I am interested in IMAP Voicemail storage for some of my customers. Does anyone know of any vendors of asterisk appliances (physical PBXs) that provide this as a "standard feature" (or an optional standard feature)? Ultimately, I'd like to be able to have a single point of accountability for the system as a whole. I would like an intuitive & powerful configuration GUI (such as

Dear All, we have installed a TE410P card on a Dell Poweredge 1750 running a slackware 10 with 2.4.26 Kernel. Then we have made two loops on the card and we have configurated all the 120 channels. Our goals was to perform some stess tests even if in this scenario we used the same box as generator and target. The stress test comprised to generate up to 60 calls at the same time by placing a file

I just upgraded my asterisk install from january 2005 CVS HEAD to current CVS HEAD : * zaptel * libpri * asterisk My asterisk have one TE405P with one span (the clock source) pluged into a telco PRI E1, a second span is a PRI E1 to another PBX and the third one is a T1 to a rhino channel bank. Everything went smoothly except that the two PRI span restart every ten seconds after some HDLC

my target is to produce a linux router with T100p to Fract T1 24chan to ethernet and TDM 4channel FXO. I have not been able to get nethdlc to work, just get zaptel networking not configured if I set bchan and dchan the card comes to life, but I don't believe this is correct for what I want to accomplish. according to markster@ this is the way they do it. using (multiple distro versions

I have a pair of sites tied together with a T1 line running zaptel PPP on either end. The system works, but I keep getting these messages in the kernel logs, and users are reporting connection problems (TCP timeouts, and hangs) especially under high usage. --snip-- HDLC Receiver overrun on channel WCT1/0/1 (master=WCT1/0/1) HDLC Receiver overrun on channel WCT1/0/1 (master=WCT1/0/1) HDLC

Hello, I contacted Krzysztof Halasa, the maintainer of the current sethdlc code. This is the response he gave me. Of course, I have this working now. Still some minor configuration to do, but It does work, and the connections to the internet are _much_ faster than when I went through the adtran... I would highly recommend to anyone with a fractionated T1, with data and voice channels to

Hi, After upgrading to DAHDI 2.4.0 from Zaptel, I noticed a lot of HDLC errors on my console: [Nov 27 01:15:09] NOTICE[2743] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 [Nov 27 01:15:09] NOTICE[2743] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 These errors prevented calls from being made and received on my PRI spans. This seems