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Hey everybody, I have a situation where we have 2 Asterisk (CVS as of 08/25/2005) connected via IAX. On the corporate side, we have 1 TE110P connecting to a Definity G3R and it's connecting to a TN464F card, giving a 23 channel connection. I have echocancel=yes, echotraining=yes and echocancelwhenbridged=yes. One the remote office side, they a Adit 600 channel bank for 10 outside

Can anyone help me here? ----- Forwarded message from Larry Rosenman <ler at lerctr.org> ----- Date: Sat, 9 Nov 2013 08:46:26 -0600 From: Larry Rosenman <ler at lerctr.org> To: freebsd-stable at freebsd.org Subject: 10-BETA3: Bad negotiation on AHD controller User-Agent: Mutt/1.5.22 (2013-10-16) Ever since I put 10 on this box (source upgrade from 8), I've been getting slow

Is it not for a card with 4 FXO? I spent several hours the other day trying to figure out what I had done wrong and I ahd forgotten to connect the power cable. I setup several of these before and couldn't figure out why this one didn't work. It appears that's all it waqs. Without the power connecter the card will probe, and even appear to be working but when the lines ring (coming

Hello, I'm a bit new to SIP, and I've set up a SIP line with Asterisk and my wholesale provider. That worked, fine. I ahd to open up the ports on my router, forward them to the correct box, again fine. Now, if I get one of my customers to connect his SIP phone to my Asterisk box, and HE'S behind a NAT firewall, does he have to go through the same process, or is it just the Asterisk

Thanks Moises. I was kind of hoping that, at least if I hosted my Asterisk server somewhere where there was no NAT for the * box that the SIP phones wouldn't create any issues. How do you people with Hosted PBX handle the deployment of SIP phones behind NAT firewalls? Is it just elbow grease and configuring every single phone for the customer, or is there a way? Mike you can redirect

In conjunction with my last post on Tellme I want to write another suggestion for an application I had. I don't know if you guys have come across Google Gas http://www.ahding.com/cheapgas But basically it is an application that this guy has developed using the Google API to search an online database on gas prices in your area. One of my strong beliefs about how Asterisk is going

Hello, a local prison contacted us regarding some calling card solution. they need 4 E1s to serve 120 rooms in that prison. we are planning on using 4 servers to serve the calls and one for the database servers' specifications are: 2.8 Dual Core Proccessors 2 GB Ram 160 Sata Drive each server will be provided with 1 E1 card Questions are: 1- will those servers be able to handle that ammount

That was my thought too. Even if it *does* work without it, there may be a reason internally you can't see why it is required. (Ex. Putting extra stress on a component causing it to fail in 6 months) -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tom Rymes Sent: Wednesday, August 10, 2005 3:49 PM To:

I've started to implement checks for package versions on dependencies in install.packages(). However, this is revealing a number of problems/misconceptions. (A) We do not check versions when loading namespaces, ahd the namespace registry does not contain version information. So that for example (rtracklayer) Depends: R (>= 2.7.0), Biobase, methods, RCurl Imports: XML (>=

Hello, As i check the call logs, some of my clients seem to make successful calls but, in logfiles, Source field seems empty..Still I can see who is the source from Channel tab as SIP/XXXX, and the called number and the time etc but.. nothing on Source and the Called ID tab. Just some clients has this problem. But as i check nothing special in their settings. What might cause this problem. Using

Hi, I use realtime and I found that changing accountcode needed to restart asterisk to activate that code and shown in CDR. Does it has a way to update accountcode without restart asterisk? ango

Hi All. Can anyone tell what is sip tunneling ahd how its works want a make a sip tunnel between to Asterisk Server.... Please share your Kind Toughts -- Best Regards Shazi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090820/091e1ed5/attachment-0001.htm

Dear List, i have been thinking of building a calling cards solution based on Asterisk and a2billing.. i have a few questions regarding this solution and was hoping you may have the answers and could be generous enough to offer them. the servers i'm thinking of are with the following Specs: Processor: Intel X3210 Ram: 8Gb HDD: 2x500 GB Sata Internet Link: 100mbps Dedicated was thinking of

samba samba ? how are you? After a serious accident, I am ready to go back to work. At monsanto they let the solutions dry out in these silly racks. I don' work ther any more. Then I was working at the Danforth plant sience center. tThe whole building was down toone autoclave so I was there early But I dropped a 4l erlenmeyer and slipped and fell on the broken glass. They had to take

Hi, all! Look here http://www.artlebedev.ru/tools/technogrette/etc/admin/ (the bottom example) Can I do somethinhg like this using prototype and script.aculo.us? I think we need to: 1. get X and Y position of the Draggable 2 set revert = true or false by calculating positiong of the nearest Droppable 3 If Draggable is accepted by Droppable - set snap:[xd,yd], where xd and yd - Droppable

Mark, Thanks a lot for the feedback. It's reassuring to say the least Mike Message: 18 Date: Sat, 21 Jan 2006 15:36:18 -0500 From: Mark Phillips <g7ltt@g7ltt.com> Subject: Re: [Asterisk-Users] SIP and NAT - best practices? To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <43D29B42.3060705@g7ltt.com> Content-Type:

Is it possible to make use of queues for incoming calls but to have agents that do not need to log in ? If I create a queue and make certain SIP-users member of the queue, do these SIP-users always need to log in to the queue to be able to receive calls that are in the queue ?? Can't a member be just available when the phone is registered to the Asterisk-server ? In stead of also having to

Hi I have On func_odbc [EXEC] readhandle=ressqlserver writehandle=ressqlserver readsql=${ARG1} writesql=${ARG1} I'm trying an update on dialplan: exten=> 141,3,Set(dummy=${ODBC_EXEC(UPDATE Tabla set campo = ${EXTEN})}) On Cli: WARNING[3579]: func_odbc.c:353 acf_odbc_read: Error -1 in FETCH [UPDATE Tabla set campo = 4356] Any idea why is this?? The query

Hello list, Currently I have upgraded again to Linux-Mandrake 7.0. I don't think I'll upgrade anytime soon. Nonetheless, I have made tremendous inroads with Samba. With RedHat 6.0, I could not get the Linux box to show in Windows Networking Neighborhood; now I can visually see it and it prompts for a password. Cool, I said no! It's just a password issue and I have it fixed in no

I am looking for Video Conference Software (Open Source) , But but not for free Trial.. please give reference about it. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090427/cc9690ee/attachment.htm