similar to: IAX problem through intermediate asterisk box

Was my question not understood? Hello, I would like to connect 2 asterisk boxes together, so this is my scenario: Asterisk Main: it is connected to many sip providers and its main purpose as a call termination forwarder. Asterisk B: it?s connected to E1, and its purpose to terminate calls. It will receive SIP messages from Asterisk_Main, but there will be no voice traffic

Hello List! I have 2 asterisk servers, The Admin(.20), and the Call Center(.21). The Admin server contains the 1XXX extension and the Call Center hosts the 2XXX extensions. I would like for our Admin folks to be able to call the Call Center folks (and vice versa). The call will go over the server fine, but when the Call Center server answer, the CLI returns: "NOTICE[4296]: chan_iax2.c:7398

asterisk-users-request@lists.digium.com is believed to have said: >Budgetone and MWI > >The message button can be programmed to dial an extension that checks >voicemail >exten => 160,1,Voicemailmain(${CALLERIDNUM}) > Thanks, this is what I was thinking about. Still, how do you get the BT to dial 160? In my Asterisk setting I have the same mailbox numbers reused for the

I am trying to send calls from (telx-NY17S) to (telx-nyc) via an IAX2 channel. However the call is being rejected on the (telx-nyc) server. See error below copied from telx-nyc CLI> Apr 22 13:56:57 NOTICE[147465]: chan_iax2.c:5390 socket_read: Rejected connect attempt from 192.168.0.251 I have icluded the following conf files 1. extensions.conf (telx-nyc) 2. iax.conf (telx-nyc) 3.

Anyone know if - it is possible to limit 1 agent per extension where the last agent to log in overrides any previous agents or - a Command/application to clear all agents logged in on extension Does this look like it would require a custom mod to do it? J __________________________________ Discover Yahoo! Get on-the-go sports scores, stock quotes, news and more. Check it out!

i have 2 asterisk servers one on CentOS and one on Fedora , i configured IAX trunking between the 2 servers so that i dial -say from a sip extension 2000 on fedora server to a sip extension 3000 on CentOS server the call seems to be established but hangup automatically after very short time and here is the iax2 set debug command result on centos server and also my iax.conf and extension.conf and

Hi Joseph, With iax servers dispersed across the internet, you could still use the below setup, it would work but it's not as secure as you would want it. I would then have a context for each server and use the IP address deny and permit statements. Also, you can have 1 server with a public IP and have the other servers behind a NAT register to the public server. There are really several

Hi all, I'd like to setup a really lean Asterisk installation that essentially has a full ISDN PRI (AT&T, T1, 23 B-chans, 1 D-chan, BZ8S, 5ESS, National dialplan) on a Digium TE207P adapter that all it does is convert the ISDN channels to SIP/IAX channels. Then I would add this Asterisk 'gateway' as a provider on one (or many) Asterisk systems on the back. With such a config I

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On 6/12/2015 7:09 PM, jd1008 wrote: >> > I want to thank you for posting your installation experience with > Centos 7. > My laptop is Latitude E6500 and I am quite certain it will experience > the same issue > because it is almost the same as your laptop. Difference might be in > cpu speed > and in RAM. My cpu is 2.81GHz dual core, and RAM is 8GB. the E6x00, E6x10,

Hi List; How can one Asterisk Box A to send a SIP call for another Asterisk Box B, and that call to be authorized based on the username and password, and not on the IP (as the IP address of the source is not known because it keep changing)? I think the trick in the Dial command, how to write it properly in a way that other Asterisk Box can recognize the sip username and password which are existed

Hi I've 'inherited' an existing Asterisk with a number of users, and some pstn connections through its Zaptel card. I've recently set up another Asterisk which has no direct pstn access and I've connected the 2 systems with IAX. The original system has an extension number range 1xxx and the new asterisk has the number range 3xxx. I can call both ways, and when required can

Due to the unfortunate nature of Wikis, the section on voip-info.org that deals with dual asterisk servers is full of pretty bad and outdataed examples. What I'm trying to do is distribute small asterisk boxes to remote offices that have SIP clients connected inside the network, and ship any outbound calls to a central asterisk server via IAX that is in turn connected to the PSTN and

Hai, I still dont understand the fuss about "domain admins" and no GID because im running this for 3 years now. So... Again what was the problem here, i dont remember it.. (sorry) In my opinion, the problem is not "domain admins", the problem is Administrator. And because if that you need an other "administrator user", that is a copy of Administrator its

Good day all How do i set a connection between 2 asterisk servers via PRI In Bri I would set one to NT and TE How shoud the zapata.conf and zaptel.conf look And how should the cable be? All I got on the web was to set one to "pri_net"...this cant be all? And the cable > pin1 <--> pin4> pin2 <--> pin5> pin3 <--> pin6> pin4 <--> pin1> pin5 <-->

hello list i have 1 server installed with asterisk centos and digium card i have installed the same configuration in another unit but in this unit there is no card installed i have created a sip trunk between the 2 servers like that in the server 1 with card sip.conf [asterisk1] type=freind host=ipadresseserver2 context=internal insecure=invite allow=all [2000] type=friend host=dynamic

Folks, I have a NEC 2400 pbx(non-voip) behind a Cisco 3725, connected via standard wic-t1 card. The NEC needs to call two different asterisk servers with 4 digits. I have two way calling working with the one * box, but the other is perplexing me. Here's the layout * <--> Cisco 2811(192.168.13.1) <--> 1.54 point to point <- Cisco 3725(192.168.8.1)<-> NEC 2400. The

We are running our * server as a virtual PBX for 6 companies. I am having all of the Allison prompts plus our own custom IVR prompts being re-recorded for each company, in a different voice (marketing thing) with a different personality (perky, corporate, earthy) . I'm curious if someone could point out a dirty trick to get the voice to play right, for internal and external callers,

Hi Is there anyway to add logic to dialplan pattern matching? I would like to match all toll free numbers with one pattern, so 1800, 1877, 1866, 1855, etc. I can't figure out how to do this in dialplan syntax. As a programmer, I want to say 18[00 or 77 or 66 or 55 etc]. Can't figure out if this is even possible with dialplan pattern matching (though I suspect it is somehow). Andrew

Hello All, I installed Asterisk 1.6.1.11 on Redhat 5.1. I use X-lite SoftPhone. The softphone can call the other one but I can' t hear any voice. My configuration files are below: [root at localhost asterisk]# cat sip.conf [general] canreinvite=yes [1001] username=1001 password=1001 type=friend context=phones host=dynamic [1002] callerid=1002 username=1002 password=1002 type=friend