similar to: IAX2 Trunk two Asterisk boxes.

> 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

What types of ISDN BRI cards work well in Europe (Guadeloupe, Martinique and France) ? For example: AVM C2 or AVM C4 or eicon Diva server 4 BRI? Any others? Which driver is appropriate? Ray Burkholder ray@oneunified.net http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. -------------- next part

Anyone know a sip-client that will work on a Handheld PC running WINCE for HPC. I can find some for PocketPC, but the wont work on my HPC ?? /HHA _________________________________________________________________ Scope out the new MSN Plus Internet Software — optimizes dial-up to the max! http://join.msn.com/?pgmarket=en-us&page=byoa/plus&ST=1

Hello r helpers! Below is the whole coding for my programme. Before proceed more further, let me explain for you. First of all, I need to compute trimmed mean. Till that step is ok. Then I need to compute ssdw which is sum of square deviation. If I do equal trimming at both tail of distribution that I chose, I will use the first ssd formulae which is "a". But if I am doing unequal

I'm trying to link up Cisco MGCP-enabled router (residential gateway) with Asterisk, and it looks like some sort of protocol mismatch, could it be MGCP 0.1 vs 1.0? Look at this (x.x.x.99 is the router, x.x.x.98 is Asterisk): MGCP read: NTFY 2 aaln/0@voip-gw1 MGCP 0.1 X: 0 O: hd from 192.168.154.99:2427MGCP read: NTFY 2 aaln/0@voip-gw1 MGCP 0.1 X: 0 O: hd from 192.168.154.99:2427Verb:

i'm looking at potential asterisk service providers and came across oneunifed.net i googled for opinions and feedback, but haven't come across anything yet. is anyone using them or does anyone have feedback on their asterisk support and expertise? tia, george

Hi, Does anyone know of Canadian VoIP termination providers? I have Canadian customers and would like to provide Canadian dial in and dial out (canadian callerid). Thanks!

I have a few 7960 Skinny phones. I've edited the skinny.conf file, but I'm a little unsure as to how get the phone to figure out which ip address it should register with when it boots. How do I do that? I already have a tftp server for my SIP based phones. Do I need a tftp server for skinny configs at all? And if so, can it be the same tftp server as the SIP ones use (I'm not sure

Hi there I am pretty close on giving up on Asterisk :-/ I am (still) trying to make a call from a H323 phone to an Asterisk provider using AIX. But H323 does not route the number to AIX. All it is transmitting is an "s". *CLI> -- Executing Dial("OH323/R27865", "IAX2/demo:demo@gw1.musimi.dk/s") in new stack -- Called demo:demo@gw1.musimi.dk/s Jul 2

I assume there are several people on this list that have Cisco Call Manager implementations under their belt.... We are beginning a call manager implementation and the first question I asked Cisco was, should we use SIP or Skinny. Cisco is pushing me towards Skinny, saying that I will lose some functionality with SIP. They also say that most of their customers implement skinny. I see two

We needed the client browser to be open all the time for dynamic data to load without the page refreshing. After looking at all of our options we decided on programming it ourselves using flash rather than java. We have a flash frontend thats tied to our backend mysql DB. We use it for loading web site traffic data, email opens, click-throughs, bouncebacks, stats, etc. It could also be used with

Hello list, hope someone can help me - I'm going crazy using the FXO port a SPA3012. I would like the SPA 3012 to act as a simple FXO port to an Asterisk, that is, once it detects a call, it should simply send it over to the local Asterisk server. No intelligent routing, PIN, anything else.... I configured it like this: PSTN-To-VoIP Gateway Setup PSTN-To-VoIP Gateway Enable: yes PSTN

That was the issue, thanks. I now am able to get the caller ringing on an outbound call, but an external phone number (E164) I am dialing does not ring. On Sun, Mar 15, 2015 at 12:19 PM, George Joseph <george.joseph at fairview5.com > wrote: > > > On Sun, Mar 15, 2015 at 8:32 AM, Sonny Rajagopalan < > sonny.rajagopalan at gmail.com> wrote: > >> I have setup my

Hi All Will cisco 7910 ip phone compatible with Asterisk? I know that 7960 are fine. David Kwok -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 1878 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040112/e8023f35/smime.bin

Hi Who concern, I setup TINC VPN follow these. 192.168.1.x / 24 (Client groups) | 192.168.1.1 (eth1) (GW1) 202.44.34.206 (eth0) || Internet || 202.44.45.14 (eth0) (GW2) 192.168.2.1 (eth1)

I have several Cisco SIP gateways sending calls to Asterisk. Because the gateways don't have user-agents, they don't authenticate with Asterisk. And because they don't authenticate, they use the default context in the sip.conf file. Is there a way to either: A) identify the inbound gateway with a variable, in channel info, or the manager interface? If there was a ${SIPDOMAIN} for

Hi, I don't know enough C to know for certain if this is a programmer or compiler error: In a Objective-C source file I have: . static const char sessionEntriesKVO = ' '; . Later I use that variable as a ID by taking it's address like this: [feedManager addObserver:self forKeyPath:@"sessionEntriesCount" options:0 context:&sessionEntriesKVO]; and later . if

I have setup my Asterisk 13.1.0 server with PJSIP on AWS/EC2. My basic configuration works, and I am connected to a SIP trunk using SIP.US, and have set up my inbound calling which works correctly (when I call my PBX DID, the call does come into my PBX network). The issue is that I am not able to make outbound calls, because the call fails with the error:

Does anyone have a working way of having a Cisco 7960 reload its config remotely. I have tried some of the scripts that I have found on the web, but to no avail. Thanks for the help. B. J. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040116/aa4eda3c/attachment.htm

Hi, I've searched the archive but can't seem to find the answer to my problem. i have two gateways running with asterisk , my question is : is there any possibility to do failover with gateways with asterisk ? i mean that if one gateway is down , asterisk switch automatically to other gateway . i have succefully used failover with limit number off calls (if gw1 have max calls ,asterisk