similar to: Linksys SPA3012 inbound FXO problems

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:

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)

Hello List, Im just dipping my feet into the asterisk world, and im having major fxo problems Im running Asterisk (from svn) + libpri (from svn) + asterisk-addons (from svn) + asterisk gui (svn 1.4 branch) + zaptel (svn 1.4) on a Debian Etch box, with 1gb ram, running all of the services for my home server (web / db / music server etc), and i would like to run my PSTN line from Kingston Comms,

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

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

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:

I configured my * with the instructions found here http://www.voip-info.org/wiki/view/Asterisk+Google+Talk to work with gtalk. The Phone rings and connects - but no audio! I am using a self-compiled asterisk 1.4.2 There is a lot of output on the CLI but I can't make sense of it. Perhaps somebody can help? Michael Output from the CLI: JABBER: gtalk_account OUTGOING: <iq

This post is simply documenting a spouse-friendly way of using the spa-3000 as both a fxs and fxo port for basic soho environments in the US, allowing asterisk to participate as needed/wanted. All home phones are connected _only_ to the spa-3000 fxs port. The incoming home pstn line is connected _only_ to the spa-3000 fxo port. Defined Line 1 (fxs) to register with asterisk via sip (extn

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

Hello, I have a strange situation with an SPA3102 FXO/FXS device. I'm in situation that when i receive a call from PBX line I must forward the calls to 2 VoIP numbers. Right now i have the following settings: (S0<:1010 at GW1>). I want to forward at 1020 too. I tested (S0<:1010|1020 at GW1>) and doesn't work. Did you have any other ideea? Thank you.

Ok I upgraded tonight a server from CVS in Late NOV to one just downloaded tonight. It all runs up OK and I can contact it from my ATA 186 using g729a codec and that all works fine. What I am having trouble with is connecting through IAX ATP.org.au in AUS to my server. The connection comes through OK I can see all the tracking info in the console OK but I get 0 audio in either direction.

I have the following in extentions.conf: exten => 2111,1,Dial(SIP/2111 at gw1.langley) exten => 2111,2,Voicemail(u2111) exten => 2111,3,Hangup exten => 2111,100,Voicemail(b2111) exten => 2111,101,Hangup I have the following in sip.conf: ; Cisco 1750 [gw1.langley] type=friend host=172.16.17.1 context=default canreinvite=no Like the ATA, lots of stuff doesn't work on the 1750

Hi list, I have a very strange problem with my network. I have 2 internet connections: A - 1 Gbit, B - 100Mbps. Network layout: A, B | | [Brd1] / \ [L1] [L2] \ / [ GW1] ................... Clients ..................... Brd1 runs bgpd, and balances the traffic through L1 and L2. L1 and L2 do traffic shaping. GW1 does some packet

Yes, I think the dial does get executed (sonny calling outbound 202-555-1212): core set verbose 3 Console verbose was OFF and is now 3. -- Executing [912025551212 at from-internal:1] Log("PJSIP/sonny-00000031", "NOTICE, Dialing out from "" <sonny> to 12025551212 through fromgw") in new stack [Mar 15 19:27:06] NOTICE[16648][C-00000022]: Ext. 912025551212:1 @

I am still haveing problems (echo) with my X100P but I'm thinking it has more to do with the server it is in which is not a negotiable item at this time. My question then is to the use of SPA3000's as a replacement from the FXO standpoint. 1. Can you setup the FXO port to recognize distinctinve ring and call a different context like you can do with Zap channels? Being able to call a

I'm curious what asterisk solutions there are out there for inbound call centers with multiple DIDs. I'm looking for solutions for a setup where single system may have 1000 DIDs going to it, one for each account. Each account may not get that many calls. Solutions that will all reporting on calls coming into different accounts, call routing for queues based on distribution groups. Like

Aki Tuomi wrote in <1907575568.4364.1597984769802 at appsuite-dev-gw1.open-xchange.com>: |> On 21/08/2020 02:17 Steffen Nurpmeso <steffen at sdaoden.eu> wrote: ... |> Wietse Venema wrote in |> <4BXSTk189nzJrP3 at spike.porcupine.org>: |> ... |>|Steffen Nurpmeso: |> ... |>|> until SASL says it is done?!. How could EXTERNAL ever work

Hello, So installed Fedora15. No problem, wish it had regular menus but the steamline effect isn't totally terrible. Heard that Guild Wars1 is playable in linux and so installed wine. It seems that you aren't supposed to install as root so I removed it as root. I then try to install anything via yum as non-root and I get the error of needing to be root. So, I read that you can bypass this

Hello all. I am not 100% familiar with the Linux advanced routing capbalieties yet, so I thought , after reading source code, documentation and more that I might be better off asking the experts on this list. I am not using this ina production environment, but at home, as a test case. First of all, here is the network topology. Internal LAN <--> Switch <--> [NAT/FIREWALL/INTERNAL LAN