similar to: Faxing via a sip extension with a digium e1 card

1. Search the archives 2. Search again 3. Now search the internet 4. The fact is that faxing over VoIP without T.38 doesnt really work. Because it works 60% of the time it doesn't mean that it works, until it works 99% of the time. Since Asterisk does NOT support T.38 it doens't support faxing over VoIP other solutions are available, search the wiki. On 9/26/05, Mark Armstrong

Why is what he is doing different than having the fax machine on a Sipura ATA? Just because both those ports are on the pci card that doesn't make them not Voice in between....if I'm wrong....eh...oh well.... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of C F Sent: Wednesday, September 28, 2005 9:45

Hello, it is now 2am presently where i am and i have been at work since 8am yesterday. I am not allowed to go home unless i get this system up and running, else i shouldnt expect a job when i come back :s... now that you have some background, I am having no luck installing these two cards - i have already confirmed they are on their own IRQ etc, and if i run genzaptelconf, they are coming up

Hello, I have several customers describing something like annoying "silence suppression". So I did some tests and I can confirm. After disabling echo cancellation in zapata.conf the "silence suppression" effect is gone, but there is a little echo of course. I do not have this problem on other boxes where I am using oslec as an echo canceller. All my calls are SIP to ZAP (E1).

Hullo everyone, Iam new to Asterix but am trying to get R2 signalling working with a Digium E100P E1/PRA Card. My Telecom doesn't have an implementation of ISDN PRI yet, so it's R2 that is supported. Could anyone have a patch to fix this for me. Am told it has something to do with Assembly Language programming. Please help me fix this. Regards, Nicholas

Hi, I've been asked to see whether it is possible to do call logging for call center environments at a lower budget than the usual $1000 per channel. Afaik, with PRI this is possible through a high-impendance Y connection, but I wonder whether this would work with the Zapata cards. Anyone ever tried this? Regards, Cees -- XP SP2 can cause cancer in rats

>> Hi! >> I have an Asterisk Box with one E1. This is connected >> with PSTN. My problem is that periodically the >> Asterisk console shows the following message. >> >> -- B-channel 0/1 succesfully restarted on span 1 >> -- B-channel 0/2 succesfully restarted on span 1 >> [..etc...] >> >> I don't know if this behavior is correct.

Hi, I tried so many but can't find the right setting for my problem. What do i have to configure so that the complete number including extension is displayed at the called party. At the moment the called party only sees the number 7837-0 not the 7837-134. Everything works fine. Incoming and Outgoing calls. Is there someone who configured a german pri with that digium card? I really

Hi it's possible to use a Digium TE110P Single T1 / E1 PCI Interface for supply a E1 link to a old PABX ? Thanks

hi! anyone please recommend/guide me of purchasing a resonably high performance server system regarding processor(s) & motherboard (+ other compulsary peripherals i.e. VGA, Soundcard). Mentioning up-to-date vendor+model will be more helping I've to use Digium TE411p Quad E1 card signalling on the E1 is SS7 no. of simultaneouse calls is 100+ regards -------------- next part

Hi, I am looking to buy 2 used 1 or 4 ports E1 Cards. If you have one, would you please contact me? Thanks, Pete -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081217/48afd68d/attachment.htm

Hola! I've a Digium E1 card, zaptel recognizes it as (loading the module with debug=1) Setting up DMA (write/read = 321b2000/321b2200) Controller version: 43 Framer: DS21554, Revision: 5 (E1) Found a Wildcard: Digium Wildcard E100P E1/PRA Using CCS/HDB3 coding/framing with CRC4 120 Ohms Calling startup (flags is 4099) Started DMA Got interrupt: 0x0005 /etc/zaptel.conf:

Hi, one short question: Is it possible for the zaptel driver to deal with multiple phone numbers on one single E1 PRI line? I could make my carrier route +49 xxx aaaaa-zzz and +49 xxx bbbbb-zzz and others down one single PRI trunk to our asterisk box terminating in a Digium TE410P. Does the driver handle this and can I put calls coming in all on the same physical interface put into

Hi! I have an Asterisk Box with one E1. This is connected with PSTN. My problem is that periodically the Asterisk console shows the following message. -- B-channel 0/1 succesfully restarted on span 1 -- B-channel 0/2 succesfully restarted on span 1 -- B-channel 0/3 succesfully restarted on span 1 -- B-channel 0/4 succesfully restarted on span 1 -- B-channel 0/5 succesfully restarted on span 1 --

Hi, For the project I've used the Eicon DIVA card. It has 8 BRI ports, and for about 25% of the time there are 7 or 8 in use. So we want to replace it with an E1 card. Only issue is, replace it with what? The idea we have been playing with was to get a Digium E1 card (we already have bought lot of Quad E1 cards :-) and then just put it back to back against Asterisk server. And instead of

Hi all I am beginning in asterisk and am making tests with an ata-186. For the time being the tests are going well, however have a doubt. I am thinking about using a canal e1 with plate digium. Assuming that the company of telecommunications supplies e1 with 30 canals and numeration to me 4000-0001 4000-0029. she is possible to configure asterisk in way that somebody of is dials 4000-0025, to

Hi, I bought a Digium Quad E1 card model TE406P. Till now, I can't make it work... I mean, I have red alarm when I configure one E1. The provider is in France (France T?l?com) and I use the following zaptel config : span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 I'm using Linux 2.6.15 and when I run ztcfg -vvvv, it seems that all channels are configured... So can someone

> The documentation mentions that the Digium channels can be split into some > voice channels and the remainder of the channels used for routing IP > traffic. > > Does any one have this in use in conjunction with Asterisk? Does it work > well? Would you recommend it for a production server? > > Obviously, if this works, this makes for a cost effective platform where

Hi.... Apologies for the length of this, but any help would be greatly appreciated ;-)... I have installed and configured a Digium E100P on my Asterisk PBX and I have connected this to an 8 port E1 VoIP gateway on a Cisco 6509. There is also an external E1 link from our Telco plugged into one of the other ports on this gateway and this gateway is in turn registered with a Cisco CallManager