similar to: Config for Asterisk with Sipgate behind Linksys Router

I'm running a Linksys WRT54GL router from my CentOS-7 home server. Every now and then (maybe once every 2 days) the router's WiFi cuts out, and I've found no way to solve this except to disconnect the power from the router, wait 10 seconds and then re-connect. This always works. The router is running under dd-wrt. My question is - which makes it a tiny bit CentOS-related - does anyone

I''m not a subscribed user, so please cc me on any replies (fier0@bigfoot.com). I know this has been asked a few times, but i have not been able to find a direct answer. I was using shorewall with 2 nics, and it worked fine, except if that linux box went down then nobody could get out to the internet (and the wife would kick my ass). I''ve now started to use my linksys

Voxilla news story: http://voxilla.com/voxstory84-nested-order0-threshold0.html Two new products * A Sipura 2000 in a linksys box: Linksys PAP2 Phone Adapter * A combination NAT router with 2 FXS ports: Linksys RT31P2 Broadband Router Jim James H. Thompson jht@lava.net

I'm running a Linksys WRT54GL router from my CentOS-7 home server. Every now and then (maybe once every 2 days) the router's WiFi cuts out, and I've found no way to solve this except to disconnect the power from the router, wait 10 seconds and then re-connect. This always works. The router is running under dd-wrt. My question is - which makes it a tiny bit CentOS-related - does anyone

And the firewall is in between the phones and both servers or are you registering the phones on a local server and trunking to the other server through the firewall? In terms of nat and Cisco 7960s I've never had a issue registering two of them behind nat to a server on the other side, however, if you called one phone from the other, you'd end up with one way audio. -----Original

Hello, I've been looking at the DialPlans by some poeple using Asterisk with SipGate, but the new Asterisk@home 1.0 allows you to create Outbound routes etc, does using the web admin give the same effects? When I add a SIP Trunk with my Sipgate settings and use a pattern of "8|." to place all calls with a 8 prefix tot he sipgate account the softphones dial the number, the Asterisk

I setup a vonage account last year, and cancelled it last night when I put my asterisk box together and signed up for a Broadvoice account to use with it. Now I would like to use my Linksys router as an MTA, but realize it is still programmed with all of vonage's proprietary information and I do not know how to clear it. I understand that just pushing the reset button will not do it.

i apologize if this has been asked before, but things are too busy to preclude a full search of the list. i have both cable and dsl from the local providers here. due to wiring issues here, i''ve been forced to put the cable modem in one end of the house and the dsl modem in the other. the cable modem is firewalled off by a cisco pix 501 (192.168.2.12). the dsl modem is firewalled off

Recently the small business I work at replaced an aging linksys wrt style router with a SonicWall one. Ever since this change I have issues with this node falling back to TCP constantly. TCP for me ends up performing very sub-par, and most of the time unworkable. For some reason too the connection usually ends up being routed through another node. I created a new tinc instance using TAP

Ok, I've done some more research and I don't think I want an FXO box... What I'd like to do is use BroadVoice (with their BYOD plan) and then run Asterisk on my WRT54G router. I'd also like to use my regular home phones without having to use a special "SIP" phone... (eg. I like my Vtech normal cordless phones) What do I need to buy to get this working? It sounds like I

I''m giving up on an old model Cisco AP350 (it was installed on the internal LAN) and installing a Linksys BEFW1154 v4 "Wireless-B Broadband Router" for my home setup. The settings do not seem intuitive when integrated with Shorewall. It seems to me that someone had done a write-up for this Linksys box, but I can''t seem to find it in the documentation. Is

On 05/11/2016 03:05 AM, Timothy Murphy wrote: > does anyone with such a router know of a way > to wake the router up in such a case through the computer? Enable ssh? ssh root at dd-wrt reboot

Hi, have some problem with incoming calls from sipgate. This was working in 1.4 but in 1.6 I get a 401 Unauthorized :-(. Sipgate has mentioned that I have to change the type to friend, but it is already friend, so what's wrong? Kind regards, Michael Here is the sip.conf: [sipgate_out] type=friend nat=yes username=1234567 fromuser=1234567 fromdomain=sipgate.de secret=secret host=sipgate.de

Hello all. I'm having an odd problem getting * and sipgate to work together. From Sipgate support I have gotten this repsonse to my query: ===== Your Asterisk is registering incorrectly with our servers. It registers like this: sip:s@217.XXX.XXX.XXX:5076 The "s" should be your SIP ID. Anything else is rejected. I don't know where you can find this setting, but from our

Hey, I'w got a problem (bug maybe?). I have recently got my Asterisk to work perfect and I'm not trying to setup some dial routes and get the system working as I wan't it to. Yesterday I was installing Festival and also did a "aptitude upgrade" on my Debian Unstable installation. After that the problem started. After some serious testing yesterday night and today I have

Problem I have is this. outside firewall (extension 2003) can call me inside firewall (extension 2000) and all is fine. If I call from inside firewall (extension 2000) to outside firewall (extension 2003) I hear no ringing and person at other end can pick up and I hear for maybe a half second then I go to voicemail. If I add another extension on the outside then communication between outside

Hello All, I'm installing some DIY routers, composed of Intel nuc's ,Linksys usb-ethernet adapters and Centos7 On most machines I just plug in the usb device, install Centos7 and everything works. But on this one machine the network interface doesn't work. After installing ip addr shows: [root at clgmol ~]# ip addr 1: lo: <LOOPBACK,UP,LOWER_UP> mtu 65536 qdisc noqueue

Could anyone who has successfully configured Asterisk to use g729 to conference 10-20 people please post their configs. I purchased and successfully installed 2 g729 licenses and but when I dial into my conference number on the Asterisk box from a SPA-2000 set to allow all codecs, it always appears to connect using ULAW. My iax.conf file includes the following under the general section

Hello i am trying to setup sipgate gateway i can get incoming calls fine, but when i dial in and then try to dial out i get this in asterisk command line -- Called 01179248615 at sipgate [Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885 handle_response_invite: Failed to authenticate on INVITE to '"01179553708" <sip:SIP-ID at sipgate.co.uk>;tag=as30eb9dd1' --

Hi, I have a few accounts with sipgate.co.uk to get some different DiD numbers. However, when an incoming call comes in, it seems to pick the wrong peer from sip.conf which sends the call into the wrong context and it fails because there is no extension in that context to match the register. Using the config's below, if I dial the DiD on account 2222222, it works fine - picks peer 2222222