similar to: Re: 32 e1's with asterisk

I would look at using serveral machine splitting up the load using one 4 port card in each. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > janvb@caselaboratories.com > Sent: Tuesday, January 10, 2006 6:03 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion >

Hi, I found E400P quad PRI card quite cheap (749USD): http://www.govarion.com/product_info.php?cPath=1&products_id=2&osCsid=68cdd6e3d08754 in comparison to te410p (approx 1500 USD ) http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TE410P Now newer generation with HW echo canceling emerged (te411p). I'm not sure in what things those two cards

can someone overthere help? the server specs are as follows HP DL380G4 Dual Intel Xeon 3.2GHz processor with 4GB RAM, running fedora core 3 asterisk-1.2.5 ss7-0.8.3d. using sip as advised to receive calls from another gateway in US. using g729 in transcoding way. however, I noticed the call hit the 51 active calls which is 102channels, I run "top" to check the system resources usage

hi I want to compress and trunk 2E1 capacity over a satellite SCPC link using asterisk. I am nw to asterisk and I need suggestions on how to implement this. Best regards Arinze --------------------------------- ALL-NEW Yahoo! Messenger - all new features - even more fun! -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am planning to go in for a E1 line and whould like to know what questions i need to ask my service provider so i can connect that E1 to my asterisk box using the digium E1 card. what I mean is will my service provider give me info like LBO, framing , coding etc which i need to configure the span tag in the zaptel.conf and what about B and D channels am I allowed to setup whichever channel I

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

Hello All, I have got my TE110P working at the hardware level, turned out to be a dodgy cable causing the Yellow errors in zttool. However, now I am getting yellow errors in asterisk, but zttool shows nothing out of the ordinary. here is my current config, and some asterisk console errors, any suggestions most welcome. zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31

Can anyone tell me what the capacity is of 2 E1's in minutes. Ie how many minutes can 2 E1's take. Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070413/de59fcf5/attachment.htm

Hi all- This question is for those familiar with EuroISDN setup. I have a customer in Europe where I'm going to install an asterisk based system with 4 E1's. The customer will configure them all in one large hunt group. My question is about the E1 channel configuration. I know that usually single E1 spans are set up with 30 channels (1-15 and 17-31) as the voice (bearer) channels and

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

Greetings! I'm about to embark on a journey to implement an Asterisk solution for providing VoIP telephony to a large number of locations for a client. The idea is to have a central server(s) accepting IAX2 registrations from remote Asterisk boxes and then routing calls via an upstream ITSP using SIP trunking. As you can imagine, I have a few questions.. Is there a limit or recommended amount

I'm kind of swamped today so I'll answer what I can get away with until tonight: --- Joshua Haberman <joshua@haberman.com> wrote: > The most interesting questions to me are ones you didn't address: > > 1. Will Ogg FLAC become the default manifestation of the FLAC codec? > If not, why not? What does Ogg not offer that makes it worth having > two different file

Hi, we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls. Can this be done with Asterisk? Has it been done before? I really would like an input on this. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I've been debuging the call disconnection problem in our architecture: PSTN---E1---OldPBX---E1---Asterisk This is our problem: -SIP user agent "A" calls a pstn phone "B". -"B" hangs up the call. -SIP user agent "A" starts listenning busytones... But the call still on. (and being payed). - Call only ends when it is correctly hanged up in the

El Tue, 9 Jun 2020 09:46:32 -0600 George Joseph <gjoseph at digium.com> escribió: Hi George > > > > > > If transcoding is enabled Would it be possible to do the same but handle a > > 488 > > back from Bob and failover to another INVITE with Bob's allow list to > > handle > > transcoding? That way we would always try no-transcoding before

hi, we are currently planning are large site which will migrate from an old siemens hicom pbx to asterisk. it will be a slow migration, the asterisk server will be inserted between the telco E1 and the hicom. new phones will be sip ones. the customer has several fax machines and analog phones (some of them have to be explosion-proof). around 50 analog ports in total are needed. as we are in

Hi, I just got hold on an Orion E1 30 port GSM Gateway, and I am having problems trying to get the E1 link to come up. I am using Asteisk 1.2.12 with a Sangoma A101 card. I am quite familiar with E1's, both the Digium and Samgoma types, as I have successfully hooked up to many PBX's and such, but I just cant seem to get this one to work. None of the 30 channels 'come up'.

Hi All, I am facing problem with E1 line. I have installed Asterisk (1.4.20.1) on a system with Digium TE420 card (Zaptel- 1.4.10) But every now and then I face problem of down E1's. The log show lot of entries like "pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2" This happens on a regular basis and the E1 becomes up after some time. My

Miguel, How many simultaneous calls (incoming and outgoing) you did ? Kind regards, Miguel Date: Sun, 3 Oct 2004 11:21:04 -0500 From: Miguel Cavazos <miguel@cavazos.com.mx> Subject: [Asterisk-Users] Working E1 MFC/R2 M?xico !!! To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID:

Asterisk Project Security Advisory - AST-2019-005 Product Asterisk Summary Remote Crash Vulnerability in audio transcoding Nature of Advisory Denial of Service Susceptibility Remote Unauthenticated Sessions Severity Minor