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Hi All, After reading the sparse info and attempting to get this running, I'm unsuccessful and could use some guidance. I already have a MRTG server up and running serving hundreds of router interface graphs. I would like to add SIP/IAX channel graphs for all our asterisk servers. I'm running asterisk 1.2 and MRTG 2.4.17. I tried the script from http://karlsbakk.net/asterisk/ but get

Hello, Junghanns and BeroNet offer 8 BRI ports cards. Do you know if Digium's B410P has an inner TDM bus so that an 8 BRI ports subsystem (2 PCI slots used, but 1 one set of interrupts) could be made out of 2 B410P ? I know you could (theorically) do this with Junghanns and BeroNet cards. Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Is it possible to get the XML config off of a Linksys SPA-941 or 942 phone? I've tried http://[ip address]/admin/spacfg.xml however that file doesn't appear to exist. Thanks.

Today I had a problem where the internet connection is unstable so calls are getting dropped all over the place. The one thing I do not understand is that at least 30 phones on the internal network went to "No Service". Since they are on the same network segment and on the same subnet I do not see why the Internet connection sould affect them. The asterisk server is behing NAT and we

I know this is slightly off topic, but I was wondering if anyone can help with a problem getting my PAP2's to connect to Asterisk. I use a provisioning file, and I recently re-wrote the files for each PAP2. I had a small typo and the PAPs logged it as a corrupt file. I corrected the file, however, Line 1 on both of the PAP2's now wont register. Line 2 works fine though. I've done the

Hello list, using asterisk 1.4.30 and trying GROUP() and GROUP_COUNT() for the first time... Having some troubles. This the dialplan (using a sub) : exten => s,n,Set(_custID=${custID}) exten => s,n,GROUP(${custID}) exten => s,n,NoOp(grouppcount = GROUP_COUNT(${custID})) exten => s,n,GoToIf($[ ${GROUP_COUNT(${custID})} > 2 ]?maxreached) The CLI shows : [Jun 5 16:06:26] --

...us line (ie, SPA942, SPA962..) worked differently. When the second call was coming in, your soft keys would not change. You could still XFER the current extablished call before you answered the next. How can we make the new phones behave in that way?? Thanks -- Technical Support http://www.neuroredes.com

If you've used Linksys phones against recent Asterisk 1.4.x you may have noticed that they may drop registration for a quick bit and then go back to being ok if your phone is behind NAT. If you turn Asterisk's sip debug information on, you'll probably find errors like these in your logs: NOTICE[30179] chan_sip.c: Correct auth, but based on stale nonce received from

Hi pals, Pardon me if this question sound basic please. This is the first installation where I have to use the analogue card and therefore a little lusty. I have googled a lot, but though there is a lot of information about the above file, none indicate where the file lives. I have a installed asterisk and zaptel software on a fresh installation of CentOS 5. This all from source and following

Given the following SRV records: _sip._udp.tollfree.sip-happens.com. 38400 IN SRV 10 0 5060 sometimes.sip-happens.com. _sip._udp.tollfree.sip-happens.com. 38400 IN SRV 20 0 5070 ares.sip-happens.com. Why is asterisk (1.4.17) not honouring the priority and not failing over to using other records when a connection fails? For a given call to tollfree.sip-happens.com ares.sip-happens.com was chosen

For the 3rd consecutive term, the US Senate has introduced the "Truth in caller ID Act of 2009". It was passed by the Senate (finally) in January, and has moved to the House for a vote. A lot of states have ambiguous or overly restrictive language on how caller ID may be manipulated. For instance, if you have a PBX, and a call comes in from the PSTN, which you then loop back out

Hello all What I'm looking for is some plain speaking advice on ISDN. Currently using 4 analog lines connecting via a four port TDM400P FXO card. We need to physically move our installations, and on requesting the analog lines be moved - our telco (BT) is suggesting we replace our analog lines with ISDN2. We would have 3 x ISDN2 connections, giving us six voice channels. They've

Alright I've been having an issue when trying to dial out locally when coming from SIP. This used to work no problem, now it doesn't. Now the local PRI to Bell Is working fine I have calls coming in and out of it constantly right now. BUT if I try and make a local call from SIP (from X-Lite or one of our Linksys SPA2102s) It fails every time with errors like these == Using SIP RTP

Hi, How can I create a web page allowing people to listen (with their own PC) a couple of .wav/a-law files stored on a Linux server ? Chances are users would access this web page from Internet Explorer but if I could make it available to other browsers, that would be better. I googled a bit and couldn't find a tag such as media://myaudiofile.wav that would fulfill this spec. As much as

Is there any way to find-out the peak number of calls that an asterisk system has had? Not the total number of calls, but the maximum number of simultaneous calls. I know I can porobably go through the CDR logs and look for calls which have overlapped in time, but I'm wondering if there's some counter somewhere I could access... (I'm looking for evidence for an ISDN client who

Hi to all, I actually have an asterisk server configured to write CDR mysql records in the same machine (localhost), but I want to write this records to another machine also in mysql at the same time, It is possible? It means that I want save the records in both machines. Thanks in advance. Ricardo Melendez -------------- next part -------------- An HTML attachment was scrubbed... URL: