similar to: audio delay over time on Zap to SIP

Since i'm sure there are others out there in the same position as me, being disappointed that the original T400P and E400P cards are no longer available from Digium, I thought I would pass on a place I found to get them. I needed the older card, since I didn't want to have to get a motherboard with 3.3v pci slots on it.

I recently purchased a Sipura 2000 and connected a phone to it which is connected to my asterisk box via sip. Calls to the Sipura 2000 work fine from another sip device connected through *, from either an fxo or fxs (via adtran channel bank connected to a T400P card) port. However, when a call comes in from the phone company over a T1 with em_w trunks, the phone on the Sipura will ring but I

Hi- I have an upcoming order for a bunch of asterisk boxes, and I'm considering using an "assembled" package for the server, instead of building them from components as I usually do. Does anyone have experience with the Dell PowerEdge 750 server, or any other 1U rackmount server for use with asterisk? Thanks in advance Scott Stingel Scott M. Stingel President, Emerging Voice

Hi all, Does somebody know how I have to setup my TE405P ? Is it correct my configuration below ? Otherwise, can somebody help me ? Thanks, Angel. zaptel.conf span=1,1,0,ccs,hdb3 span=2,0,1,ccs,hdb3 span=3,0,1,ccs,hdb3 span=4,0,1,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31

Hi- I have posted a photo of the TE410P Digium card on my site, so that those wishing to purchase a compatible motherboard can see physically what the PCI slot requirement is: http://www.evtmedia.com/TE410P.htm I believe the required slot is a 64-bit, 3.3 Volt PCI, most commonly found on Xeon-based motherboards. A 64-bit slot is longer than the TE410P board requires, but the PCI connector

Hello, I have had several instances over the last month of Asterisk freezing, sometimes after 12 hours, sometimes after 8 days. The common elements are that: - all Zap channels lock[hangups don't register and no new calls in or out] - no new in/outbound calls can be made on Zap or SIP channels - people who are still connected to calls can continue to talk - in the CLI interface, you can

Max TE410P card on an Asterisk Hello, Does anybody know the max number of TE410P/TE405P card we can put in an asterisk box? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040504/f220d3ae/attachment.htm

Do they fall under FCC certification if they're built to the same specifications as the ones from Digium? If I build my own T100Ps from the schematics and board layouts that are available, are they legal to plug into the PSTN?

Hello, I have a big old Fax machine that will only pick up a ring at 24volts and 20mA(minimum) ring[according to the technical specs manual]. None of my SIP -> Analog phone adapters supply this: Cisco - 50 volts SIPURA - 70 volts Handytone - who knows, but it doesn't work can anyone tell me if there are any analog adapters or channel banks that can send a 24V 20mA ring signal? Also,

Is there anything to choose, in performance, between a TE405P and a TE410P? I understand the difference between the PCI bus voltages, and certainly don't intend to try Andrew's hacksaw operation :-). But if I choose the card first, and a compatible mobo second, does it make any difference which? Cheers Tony -- Tony Mountifield Work: tony@softins.co.uk - http://www.softins.co.uk Play:

Hi all, I've seen a couple of posts recently from people who are doing something with Asterisk and a T100P and a Nortel PBX. However it's not clear exactly what they are doing. Does anyone know if it's possible to use a T100P or T400P as a DSX-1 interface to connect to a T1/PRI CO module in a Nortel PBX? We have a Nortel Norstar Modular ICS PBX and I'd like to plug its CO

Hi- I'm almost embarrassed to ask the following simple question, following John's excellent and rigorous bandwidth analysis (see earlier thread): I have a straightforward Asterisk application, IVR-only (no connections between channels). It will simply decode DTMF's and speak prompts (probably A-Law encoded) on a number of E1 circuits simultaneously. Realistically, how many

I just had the same exact problem this morning. The only thing I've done in the last couple of days is update update zaptel. I rolled back my zaptel to 2/11/04 from 3/8/04. And kept my libpri from 3/8/04. I never had this error before updated. I had other issues, but not this one. -sb -----Original Message----- From: asterisk-users-admin@lists.digium.com

Anyone care to elaborate on the differences between the T400P and the TE405P? -Darren -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050908/e817f8bb/attachment.htm

Hi I plan to setup an asterisk box to function as a SIP gateway forwarding lots of calls to/from a backend of several other asterisk boxes, each with a TE410 card for PSTN connectivity. It will only gateway the calls into the PSTN gateways. No transcoding is planned - only plain ALAW. How many concurrent calls would you think this can handle? I'm asked to plan a system that can handle

Why PCI 3.3V for E1/T1 card!? I can not use it because I can not find any motherboard with PCI 3.3 . Any sugestions!? Cristian VASILIU AccessNET International S.A. Software Programmer mail to :<cvasiliu@accessnet.ro> www:<http://cvasiliu.home.ro>

Hi I want to connect a Windows machine over TAPI with the Asterisk PBX. So is it possible to connect the Windows machine directly to Asterisk (Zaptel card)? Thanks Harry Baron

Hi, I have a project coming up which will need to interface Asterisk to E1 trunks in the UK. I have a couple of questions which I hope someone can answer, or give me some pointers: 1. If I want two E1 trunks, is there anything to choose, performance-wise, between using two ports on a single TE405P, and using two E100P cards? 2. How can I test the E1 operation in the lab, which doesn't

Hi, I am an asterisk newbie and looking around for information . I wish someone could take their valuable time off to answer my query in detail. I wish to set up an IVR system that can allow user authentication and therefter accept 2-3 inputs from users ..generate a key and transmit the same in voice back to the user . The system will intially have small load but if the whole package in future

Hi all- Over the past week or two, I've been trying out asterisk under Fedora 1 Linux (RedHat). In my setup (single and dual Xeon motherboards), I have so far had a very good experience in terms of performance. In doing E1 load testing, I've found that Fedora handles heavy load much better than RedHat9, probably because of its better use of the multi-threading capabilities of the Xeon.