similar to: Problems w. chan_capi + ztdummy

Dear Sirs, Anybody ever tried running SIP up against Voicepulse? On their http://connect.voicepulse.com they claim they support both SIP and IAX, but I can't seem to get SIP running. I have as mentioned before on this list - huge problems getting any timing devices running on some of my machines, so IAX is not really an option right now. If I try I get a "Service Unavailable"

Hi Everybody, Has anybody tried the above (or indeed any other 4XBRI cards) successfully with Asterisk. As far as I can see the above mentioned card is an active ISDN card but supported by it's own I4L driver. This leads to interesting questions particularly regarding echo cancellations (which usually doesn't work on the cheap passive cards with one exception as far as I can see).

Hi Everybody Any significant changes to CVS HEAD over the last couple of days. I've got two asterisk boxes - both on public IP but one is dynamic. The one on dynamic IP registers at the other one - that part is fine. Calls going from the one with dynamic to the static one goes fine. Call the other way results now in: Failed to authenticate user "1101"

Hi Everybody, Got a weird problem here I think. Got a setup with an asterisk (current from cvs as of a few hours ago) in a box with an el-cheapo ISDN BRI card connected to the PSTN network and two Snom phones internally (one Snom-100 and one Snom-200). Dialing between the snom phones or dialing out to PSTN from any of the snom phones works perfectly. But when I receive a call FROM the PSTN

Hi! I have some problems with the download of Asterisk-addons. I try to follow instructions that I found in http://www.voip-info.org/tiki-index.php?page=Asterisk%20cdr%20mysql , but nothing to do. This is my shell: [root@obelix root]# cd /usr/src [root@obelix src]# export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot [root@obelix src]# cvs login Logging in to

Dear Sirs, I've got a weird problem with IAX2 transfers. My setup consist of 3 Asterisk servers. One is located in Europe on a public IP and a local PSTN connection through ISDN. Two are located in South-east Asia - both on public, but dynamic IP. These two each have a bunch of SIP phones attached. All 3 systems are running latest CVS (ok - might be a day or two old) on Linux 2.6.6 with

Hi Chris, I know this is quite an old email, but I was browsing through the archive :) I am currently working on "embedding" asterisk in one of Allwell's STB's. The idea is more or less exactly like yours. The STB will be solid-state and contain OS, Asterisk, Basic configuration and voice files on a flash disk. It will boot up and get a network share via DHCP. This network

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>

Does there exist a text file with all the 'standard' Asterisk voice messages? I'm planning to get them recorded in dutch, but need to know the exact text of each prompt... Michiel

If anyone is interested ... just in case! :-)... I have tried to write , based on the cdr_mysql.so module, an Sybase module. To compile you can use something like that: export SYBPLATFORM=linux export SYBASE=/opt/sybase cc -I$SYBASE/include -c -o cdr_sybase.o cdr_sybase.c cc -shared -Xlinker -x -o cdr_sybase.so cdr_sybase.o -lsybdb -lm -L$SYBASE/lib (anyone could write the corect Makefile

Hi Everybody, I've been asked to upgrade an old NT4 server with a new server running Debian/Samba. I've got no problem with migrating the old server, but I do have one "unknown". The company mail run on an Exchange server that is most likely part of the NT4 network. Has anybody tried this setup? I did look in documentation and google and found precious little - which to me

<P>hi Everyone,</P> <P>We are configuring an ISDN PRI E1 with an E100P card, when you load the drivers, and starts the asterisk, cards also starts fine, givin following output,</P> <P>*CLI> <BR>&nbsp; == D-Channel on span 1 up<BR>&nbsp;&nbsp;&nbsp; -- B-channel 1 successfully restarted on span 1<BR>&nbsp;&nbsp;&nbsp; --

Does anybody have any experience with the ZyXEL Prestige 2000W? I am having problems with the line tear down when I call another extension. If nobody picks up at the other end when I hangup the 2000W, the other extension continues to ring. Is there any way to hangup a SIP call if there is no more traffic? Asterisk seems to think that there is still a connection open. This is pretty annoying

Hello, I've been playing around with two generic X100P analog cards to create a proof-of-concept system before we go ahead and hook up a PRI. I'm running into a reproducible problem with sound quality of bridged calls, and am hoping someone will be able to point me in the right direction. I have in my dial plan a _9. extension so outgoing calls can be made... the first thing is

I am completely new to * ( I know read the archives but this is a little different case) I am trying to setup a Sip system out side my security firewalls for home users. I currently run a Cisco avvid solution internally but it's highly firwalled. I am planning on building a pri out of my 3745 cisco router and pluging it in to a 3810 which is on the outside setup with sip and running a *

Sorry Martin to bother you again! I have an ISDN flux with 100 numbers. The local PSTN is sending now the DNIS/DID (so they said!!!) (I have set the immediate=no in zapata.conf) but I have the same problem as before : NOTE : the number is alphanumeric-DID alphanumeric (I will make tests with numeric mumber!). WARNING[9226]: File chan_zap.c, Line 5275 (pri_fixup): Call specified, but not

I have Asterisk running with a combination of SIP and H323 clients. I am using the OH323 module instead of the H323 one. When the SIP clients ring each other, they can hear a ringing noise in the ear peice to let them know that the other parties phone is ringing. However, when the H323 client rings a SIP client, there is no ringing sound at all, although as soon as the called party picks up the

i have been using the asterisk for some three weeks. Previously i was using the softphone iax-phone and now i have to shift to the sip phone xlite. The problem is that there's always unbearable noice in sip to sip calls. Is there any way to get rid of this???? Kindest MM Luqman -------------- next part -------------- An HTML attachment was scrubbed... URL:

1. D/120JCT-LS card with 12 ports. This ports are FXS ports? 2. It is true that "Dialogic drivers cost of $15 per channel" ? 3. Can I use this hardware with asterisk (for E1-ISDN using Wildcard E400P <http://www.digium.com/index.php?menu=wildcard_e400p>) ? 4. Anyone with experiance can tell me how they work and can provide a configuration example? (2 DSP Motorola procesors - I

And someone goes and redesigns the look and operation of the Trac. Quite noice, but a couple of comments: * The main toolbar ("Wiki", "Documentation", "Timeline") ends up under the puppet logo on the left hand side, which is a pest. * I can''t click on any link, text box, or button in the main part of the page. All of the navigation links work fine, and I