similar to: disposition "answered" after authenticate??????????

Hello, I wanted to add a failover trunk to my asterisk configuration. I got 2 gateways for my calls.. one is a pri other is voip trunk. I want to keep my trunk for failover. I am using ast 1.6 with asterisk-gui. But when i add a failover trunk for test purposes asterisk-gui adds the following line to my extensions.conf. where superonline is my voip provider and span_1 is failover trunk. exten =

Hi everyone! I'm quite a newbie at this Asterisk stuff so please bear with me. We've recently decided to start training in Asterisk via AsteriskNow! Asterisk version is 1.4.18.1 through AsteriskNow! 1.02 The box we have is paired with a Digium TE110P and we've managed to get it to the point where incoming calls via a single DID (from NTT Japan) can be received and answered

I am trying to help a guy out with his Atcom IP04. He has set it up like this. He has a handful of IP phones all connecting via SIP. He has two phone lines connected to the FXO ports one from telecom, another from vodaphone. He has set up the dialplan so that one of the trunks fails over to the other trunk. Everything seems to be working OK except for outgoing calls. He can call from

Hi, I'm having problem with call parking. When I park call, either via transfer to xten or park digit sequence from features.conf, I hear the parking lot number read to me and the user gets transferred. However, MOH stops for the caller the moment user is transferred. The user can be retrieved by dialing the parked extension and voice resumes. If the parked user hangs up, the channel state

--- span_1 = DAHDI/g11 1,1,dial(${span_1}/${EXTEN:0}) --- The configuration was rsync'd from a working pair of asterisk servers in another office. The only difference was the version 1.4.22 for the original servers that were operating as expected, 1.4.24 and 1.4.24.1 for the new servers. Included in both working and non working servers is the following configuration:

I need some extensions logic assistance, I'm trying to dial out one of multiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only allow 1 call per trunk) and roll over to a second or third depending on that busy status Here's what I've got for a macro thusfar, but it's not working(fails if the 1st trunk is busy) extensions.conf: [globals] trunk_1 => SIP/trunk1

Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D = 31999) -- Remote UNIX connection Verbosity is at least 8 -- Executing [00425298582 at numberplan-custom-1:1] Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20", "SIP/trunk_3/0425298582")

Hi, there. I have asterisknow beta 5 with the following data: Ip 192.168.0.60 mask 255.255.255.0 gw 192.168.0.1 the router (a linksys) has port forwarded the port udp 5060 and from 16384 to 16482 udp-tcp from the internet to the asterisk machine. the only protocol allowed is g729. Which work fine for the ip phones I already have setup in the LAN. My problem is trying to register to a voip

Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D = 31999) -- Remote UNIX connection Verbosity is at least 8 -- Executing [00425298582 at numberplan-custom-1:1] Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20", "SIP/trunk_3/0425298582")

I have a SIP phone calling via a SIP trunk another asterisk system, that then sends the call out a ZAP channel. When I press any of the features defined in features.conf, The end user on the ZAP side hears the DTMF tones, and none of the features work. My DTMFmode on the SIP users definition is rfc2833 Asterisk console doesn't register that a feature is being recognized, any ideas? Below

Hi everyone, having a issue with asterisk and my new Voip providers service. Iv set up many asterisk systems before but never seen this and have tried to fix this with no luck.. I have used this exact same sort of setup for 5 other providers and never had this issue, If i replace the trunk login details with my works voip account and set it to IAX then it works perfect, Just not the new

hi all I have a PRI line which i have connected to my asterisk server. I am able to make calls to mobile no through my asterisk server, while i am not able to make calls to land line nos. This is strange. Where do u think the? problem is , is it from the service provider or? mis configuration of my asterisk. I am from India and using airtel pri lines. Below i am pasting you my configuration file

Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D = 31999) -- Remote UNIX connection Verbosity is at least 8 -- Executing [00425298582 at numberplan-custom-1:1] Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20", "SIP/trunk_3/0425298582")

Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D = 31999) -- Remote UNIX connection Verbosity is at least 8 -- Executing [00425298582 at numberplan-custom-1:1] Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20", "SIP/trunk_3/0425298582")

Hi, I am attempting to setup an asterisk server, current specs: CentOS release 5 (Final) Asterisk 1.4.11 Asterisk-gui checked out from SVN last week I started with a fairly basic setup involving one VOIP provider who provided one dial in number, and a couple of handsets. Config files are below. It was pretty much totally built by Asterisk-gui, except for the fact I had to add

A call comes in from the PSTN, Asterisk answers it, it goes to the directory, and then to the extension the caller designates and the user at that extension answers. The user at the extension then wants to transfer the call to another extension; on the Cisco 7940 they push the ?more? soft key, then the ?Transfer? soft key, then enter the extension number they want to transfer to, and hit the

I'm trying to set on-demand call recording. Here's a snippet of the pertinent dialplan. The purpose of this is to allow one user in particular to be able to receive an email recording of the call everytime he dials *91 + number. The problem is that the email is not going out or being generated when I use the ${CALLFILENAME} variable. When I use the actual file name of the gsm recording,

I need some help getting a system running for one of my company's plants. I am running AsteriskNow 1.7 with Asterisk 1.6.2.10 and FreePBX 2.8.0.2. My D-Channel keeps bouncing. The telecom tech told me he thought that I might be using the wrong sync source, and I think I might have been, but I changed DAHDI system.conf to "span=1,1,0,ESF,B8ZS" (from

--- span_1 = DAHDI/g1 1,1,dial(${span_1}/${EXTEN:0}) --- I can only presume some form of precedence overrides the group configuration in the recent asterisk installs and not for the servers set up earlier. On 26/5/09 4:01 PM, "Kal Feher" <kalman.feher at melbourneit.com.au> wrote: > Ok I've solved the problem. I do not think it was as switchtype issue after > all as

Folks, I have a Merlin Legend R7 V10.0 with a 2 100D cards. I have 1 card in slot 4 going to CenturyTel, and the card in slot 10 going to a flip cable to a TE110P card in a Asterisk 1.6.x box. I have routing setup on the Merlin to send a block of numbers to the Asterisk. Currently the PSTN can dial the Asterisk Extensions. The Asterisk can dial Merlin Extensions. The Merlin can Dial Asterisk