similar to: UK Gateway

We have an Asterisk server that speaks IAX2 to Magrathea to get to the PSTN. Our local phones are a mix of Cisco 7940s and Grandstream BT100s all configured for SIP with silence-suppression disabled. Everything is configured to use a-law encoding. The version is: sip*CLI> show version Asterisk CVS-05/06/04-18:45:57 built by root@sip on a i686 running Linux Incoming callers are complaining of

[I'm sorry to trouble the list with this, but this is the only way I know to contact the person concerned] This message is for Stephen Karrington - it appears that you have over-agressive 'spam' filters and we can no longer email you. Please rectify this if we are to have meaningful conversation! The original message was received from Linus Surguy

I am looking to install a web interface for Asterisk to transfer calls and look who's on the phone. If anybody has a working web interface please let me know. I installed the www.asternic.com (operator) But when I bring up my web browser it says transferring data and does not bring a browser. -----Original Message----- From: asterisk-users-admin@lists.digium.com

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)

[BEGIN] 2018/8/24 16:57:41 [root@localhost puduct]# guestmount -v --rw -a /home/kvmsystem/VSOS_2G.qcow2 -m /dev/sda1 /mnt/ids/ libguestfs: trace: set_verbose true libguestfs: trace: set_verbose = 0 libguestfs: create: flags = 0, handle = 0x7f243deb9920, program = guestmount libguestfs: trace: set_verbose true libguestfs: trace: set_verbose = 0 libguestfs: trace: set_recovery_proc false libguestfs:

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

The freenum.org beta continues to roll forward. If you have an Asterisk or SER SIP gateway/proxy, please see if you can make some sense of the examples below and install them in your system. Your users will hopefully be able to dial toll free numbers in various nations just like they dial regular numbers in those same country codes. I'd like to get some additional people trying to make

DID's from Allstream (AT&T) are $2 Cdn/month but I think they have a rule that it has to terminate on their network somewhere... -----Original Message----- From: Linus Surguy [mailto:linus@magrathea-telecom.co.uk] Sent: Monday, June 14, 2004 6:53 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Canadian DID Can anyone point me in the direction of a wholesaler of

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 use Jack for getting callee sound. Dial with option M(): JACK_HOOK(manipulate,i(rec_737219:input),o(rec_737219:output),c(rec_737219))=on This works fine, but I need to connect the sound channel to Jack *before* the actual answer. As you can see in the AMI log, between "Ringing" to JACK_HOOK there is a 6 second break. I don't want that. I need a way to launch Dialplan function

Hello, I have configured a load balancing router using Julian''s patches and as described in "nano.txt" for two ISP links as shown below. ISP 1 ISP 2 . . | | | | | | | WAN

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:

Hello, i have had a terribe day today, why? because i have been so 4 or 5 hours trying to do work this and i have not finished it. I feel very pleased if anything can help me. I have two isp connections: eth0: dhcp ip1: 82.198.*.* gw1: 82.198.*.* and a router adsl in ip 192.168.0.253: eth1: static ip2: 192.168.0.254 gw1: 192.168.0.253 the same machine has 3 more networks cards but i am

HI, I have been pulling my hair out trying to get a Cisco MC3810 to interface my Asterisk box with a T1. I am able to make outgoing calls but incoing calls never reach my Asterisk box. The cisco give a fast busy when I try to call one of the DID's. When playing around with the dial-peers I can get the cisco to pick up the call, but then it forwards the call back to the ANI that is dialing.

Hi, I''ve setup a gateway using multiple default gateways and netfilter MASQUERADE to load balance traffic between two DSL interfaces and one dedicated link, and when I try to download something big, or when I''m using MSN (both in clients under this gateway), sometimes, or most times, after a while the connection timeouts. The connection doesn''t seem to change its

I am relatively new to iproute2. Any information on the topic below would save me considerable time. I appreciate any help, thanks. We want to plug two lines into a webserver to increase the bandwidth available to the hosted sites. Rather than use round-robin DNS we would prefer to use iproute2 to use equal cost routing across the two lines. However, I am still not sure how well that will

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

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 :) My employer is looking to move a call centre to a new office, and has been increasingly frustrated with their legacy PBX (call-logging licensing and hardware upgrade costs). So I've stepped forth as the Open Source Pedant and suggested Asterisk so we can do all our own CallerID / call logging / analyses, and make use of IP Phones / teleworking, etc. The problem begins in that I only

Anyone, willing to take a lead on this one, since Tom is taking a rest: " I am hosting all servers by myself. I have five static IP addreses with a DSL line. My DSL router from the ISP provider is configured as bridge, so no traffic is filtered. I checked the logs and getting: Jan 5 23:05:12 gw1 kernel: Shorewall:all2all:REJECT:IN= OUT=eth0 SRC=66.58.99.86 DST=216.35.73.164 LEN=68