similar to: h323.conf new try

Hello everybody, First time installer and I need the lists advice. My plan is to use asterisk PBX with some hardware to terminate my calls coming from several operational gnugk gatekeepers. Do have RH9 and downloaded the latest asterisk from CVS. Compiled according instructions and is running fine. Could hardly find any info on h323 implementation untill the REAME in the channels directory.

Hi list, I am new to this list and played around with some h323 issues that I can't get solved. Anybody who can point out what I am doing wrong? This is in short the scenario: I have 20 endpoints connected to openh323 gnugk and forwarded the calls to asterisk box. When a call comes in asterisk will pick up the call answer it with a message and user can select an extension to dial out.

Hello Everybody, I am trying to configure Asterisk to listen into a database which is created in PostgreSQL. Whenever asterisk starts up, it is unable to connect to the pg database and gives the following error: [cdr_pgsql.so] => (PostgreSQL CDR Backend) == Parsing '/etc/asterisk/cdr_pgsql.conf': Found Jun 24 21:20:53 DEBUG[1074494336]: cdr_pgsql.c:284 my_load_module: cdr_pgsql:

I am running a small network consisting of a few Win98 machines, 2 NT (SP6) machines and a Samba 2.0.6 PDC. I am using domain logons, and everything is working fine. However, when I try to list the shares on a Win98 box fron an NT machine, having logged on with a domain-authenticated account, the Win98 boxes return an "Access Denied" error. If I do the same with a local account on the

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

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

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)

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

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 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

Hello, I have had numerous issues with PJSIP outbound calling in Asterisk 13.1.0 and SIP.US SIP trunk. My Asterisk server is on EC2 and I have opened up the appropriate ports. The SIP clients can be anywhere on the Internet, including behind NATs. I am able to get to my Asterisk server's internal extensions via the DID (and appropriate dialplans) but I am not able to make outbound calls to

I have a working Asterisk 13.1.0 running, and I am trying to configure a SIP trunk for outbound and inbound calling, and a DID for the Asterisk server, which is used for incoming calls from PSTN. I configured my SIP.US trunks (showing one gateway, gw1, here for brevity, have two: gw1 & gw2, which are both configured on my end): [sonnyGW1] type=registration transport=transport-udp

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

Hello there. I am setting up a router using openwrt. Part of the project is load balancing among 2 broadband lines. I made use of the line: ip route add default scope global \ nexthop {GW1} dev {IF1} weight 1 \ nexthop {GW2} dev {IF2} weight 1 somewhere on the configs. On the test phase, automatically switching (routing) to line GW2 when line GW1 is suddenly disconnected takes a long time.

Hello, I have a problem with the following setup, I hope you can help me. I have two internet gateways, one for LAN1 and the second for LAN2. +--------------+ GW1 more eth0| |eth4(SNAT) GW2 ---...routers...-----+ router +----------------- | | +---+------+---+ eth1|

Hello, I''m having a very strange problem concerning traceroute and routing and didn''t know if lartc or netfilter would be the correct choice for asking. (so sorry if my question is misplaced) I have the following setup: public ip -- gw1 -- 172.16.0.1 --- 172.16.0.2/and public ip''s --- gw2 --- switch --users (public and private ip addresses; ip-user-pub) from the

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

On Sun, Mar 15, 2015 at 8:32 AM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote: > 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

On Sun, Mar 15, 2015 at 10:34 AM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote: > 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. > Any error messages? If you set 'core set verbose 3' and try it, does the Dial get executed? > > On Sun, Mar