similar to: Installing Xen on a Toshiba A200

Hi, I've just built my first asterisk server. Current information: OS Version: Linux asterisk.visinet.com.au 2.6.18-8.1.8.el5 #1 SMP Tue Jul 10 06:50:22 EDT 2007 i686 i686 i386 GNU/Linux Asterisk Build: Asterisk 1.4.11 Asterisk GUI-version Revision: 1479 $ Server Date & TimeZone: Thu Sep 6 02:37:11 EST 2007 I've used the Asterisk GUI for setup with two IP

Update on this: I found that by changing insecure = very to insecure = invite, adding the second trunk no longer stopped calls working. I've read the documentation on this switch and still don't see how it applies/is meant to get used. Anyway, with this change in place, the following may help: asterisk*CLI> sip show registry Host Username

Hi, I am attempting to setup an asterisk server, current specs: CentOS release 5 (Final) Asterisk 1.4.11 Asterisk-gui checked out from SVN last week I started with a fairly basic setup involving one VOIP provider who provided one dial in number, and a couple of handsets. Config files are below. It was pretty much totally built by Asterisk-gui, except for the fact I had to add

Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D = 31999) -- Remote UNIX connection Verbosity is at least 8 -- Executing [00425298582 at numberplan-custom-1:1] Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20", "SIP/trunk_3/0425298582")

Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D = 31999) -- Remote UNIX connection Verbosity is at least 8 -- Executing [00425298582 at numberplan-custom-1:1] Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20", "SIP/trunk_3/0425298582")

Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D = 31999) -- Remote UNIX connection Verbosity is at least 8 -- Executing [00425298582 at numberplan-custom-1:1] Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20", "SIP/trunk_3/0425298582")

Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D = 31999) -- Remote UNIX connection Verbosity is at least 8 -- Executing [00425298582 at numberplan-custom-1:1] Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20", "SIP/trunk_3/0425298582")

As far as I can tell, after discussing this matter with other asterisk users in my area, my telco _does_ provide disconnect supervision.. It seems that the problem is actually related to the Sangoma A200 card I'm using, as two other people both using this same card have expressed the same problem.. Are there any other users on this list using the Sangoma A200 FXO port card, and experiencing

Hi Guys, Looking for a powerful box that is compact, can take two hard drives for Raid-1 (no SSD, too expensive), have at least two Gig ports or two 10/100mbps ports. Fit two PCIe or one PCIe card plus it's daughter card which needs as much room as a PCIe and doesn't need the actual slot. That is for Sangoma A200 + Daughter card. I would be really glad if I can get some sort of a KVM

Hello, i am trying to install sangoma a200d to my centOS server but i am receivig this error message: ZT_CHANCONFIG falied on channel 1: invalid argument (22) Please help me on hopw to solve this issue! tnx! kenjie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060309/fbeeb280/attachment.htm

I've been using Asterisk 1.2.6 with a 4 port FXO Sangoma A200 card for the past month without many problems (other than the fact that the Sangoma card doesn't disconnect hung up calls immediately, which I posted about in another thread, and has still not been fixed), however, I had a call from one of our clients today and they complained that our phone system kept ringing through when they

I use Trixbox 2.0 with a Sangoma A200 I also have echo so bought the HPEC and yes it works brilliantly. The problem I have is I used to use Trixbox 1.? with this sam hardware and had a few inbound CLID issues on my UK BT lines, Sangoma support suggested changing the RXGAIN in zapata.conf and it worked. Now using Trixbox 2.0 and have upgraged to latest software Asterisk 1.2.16 Zaptel 1.2.15

Evening all. I'm looking at building an Asterisk system for one of the projects of the Charity where I'm the SysAdmin. The project has two analogue phone lines - BT Featureline Compact, we're in the UK - that I'd like Asterisk to handle. My current quandary is which FXO interface to use. I've been looking at the Digium TDM400P and Sangoma A200, which are similar prices for

Hi, Does some one experience the Sangoma A20X-ec series card that cant detect the hangup tone? I got * server 1.2.5 and running on Centos 4.3, I hock up to two PSTN lines and each time some one call in and my phone delay 1-2 sec (this is Asterisk delay nothing to do with Sangoma) and it rings on my phone, however, end on the day I got not less that 10 empty messages. I found out that Sangoma FXO

Hi I was wondering if any one has used these cards, I am looking at this as a replacement for the tdm410, I have some issues with installing the tdm410 in a small case because of the power plug being at the end of the board. I am in australia seems like we have a different setup for out fxs voltage, any one in oz using this card ? Thanks Alex -------------- next part -------------- A non-text

Hi. I have 3 lines connected to a SangomaA200. All was working normally, but 3 days ago we experiment this issue. The outside callers can call us normally; but when we try to do an external call, just receive the message "all circuits are busy". Only happen with outgoing call. Incomming calls work fine. if the conf was not modified, what can i check to search the reason of that

Dear Folks, Anyone knows if Sangoma supports or going to provide support for battery removal detection on FXO lines?? As Tzafrir said earlier DAHDI supports it, which is a very nice feature but what about Sangoma? Regards. -- M. Shokuie Nia. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Has anyone ever experienced crosstalk with sangoma analogue cards on 64bit? We have exhausted every test to try and replicate this and find a solution with Sangoma tech support, but we can not fix it. We are about to try the card and four *seperate* UK BT lines in a 32bit system. The current system is a 4gb, dual core cpu with pbx in a flash 1.4, Zaptel and Asterisk 1.4.21-2 Currently we

Hi I have been having a rough time getting a Sangoma A200/Remora FXO/ FXS Analog AFT card set up properly. The main issue is that the card has four ports and as far as I can tell Asterisk is only seeing two. On the two that it recognizes the "Green" FXS ports are not green, they just are not lit. The "RED" FXO ports are indeed red, but from what I have read your not

Hi all, I have setup a Dell PowerEdge 2550 with a Sangoma A200 card with 2xFSO and 1XFS modules. The PowerEdge specs are 1 x P3 1133MHz, 512MB RAM. Sangoma A200 has 3 analogue PSTN lines connected. This server is based in Office 1, with 5 users all with a Linksys SPA942 VoIP Handset. There is another Office (Office 2) connected to here using VPN. There are two users in Office 2 with the