similar to: Asterisk not hanging up calls

I'm looking for a recipe for a 3 way call where one of the parties can (without using the flash button) dial-out and add a third participant to the call. I tried Googling but it seems I'm missing a key search term. The reason I wanted to avoid using the flash button is that some handsets don't have it (nokia E61 who's 2 way calling via sip is also broken) Something like: 1.

I am trying to setup an interactive menu where a caller hits the main menu and can then dial an extension. As far as I can tell the "Waitexten" app is failing to read 3 digits and just reading the first and then announcing that it is invalid since all extensions are 3 digits. How do I make Waitexten wait for 3 digits? I have setup the extension "100" for users to reach the

Hi all, i have asterisk 1.2.17 with sip tcp support and i am trying to connect asterisk with HiPath 4000 V.3.0 using SIP. I can see the registration from the HG3540. But when i try to place a call from Asterisk to HiPath, the call fails with SIP/2.0 603 Declined. The strange thing is that the first INVITE uses tcp and the response is a 100 TRYING, the next 7 INVITE are using udp and the respose

Hi Was able to communicate clearly with e60 and E61 with asterisk with new access point , even though the access point security setting was of ?opennetworks? , the previous one was of ?WEP? , I feel this was a major hurdle in communication , now I can clearly accept and make calls using Nokia E60 and E61 devices Next I will be trying to find out how to make this device work

Has anyone played with this phone? i cant seem to get it to work properly, i manged to get it registered and can make calls from it, but i havent been able to make it receive calls. Weird thing its that if you make a call from it and while you are on that call you dial its number does calls go thru in second line, but as soon as you terminate both calls it wont recieve any calls again. Heres

Hello Olivier Ive been testing the E61 phone for some days now, and we need to have an inhouse asterisk server, connected to our main asterisk server, to get it to work. That means, that you cant just walk down to your local airport, and use the IP part of the phone on their network. You have to have a non nat local server, to get it to run. Other than that, the phone can accept calls both

I'm experiencing a problem with a Sipura SPA-941 not available for incoming calls after Asterisk & Freepbx upgrade. I can dial out with the phone gto any other internal or external ext. It is registered with the Asterisk server. When I dial the Sipura directly from any other extension, it goes directly to vm. I have other Sip softphones that are working fine. A sip debug when calling the

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

Hello, I'm wondering why asterisk is not transferring the callerid to the sip device. Scenario as follows: sangoma <---> zaptel <---> asterisk <---> sip <---> SIP-Device zaptel is reporting the callerid, but in the sip packages the sip-address shows unknown as user part, as this sip debug package shows: Executing Dial("Zap/62-1",

Asterisk 1.4.29 BLF-SUBSCRIBE go to internal IP (ngrep output): U 2010/03/02 11:34:06.013515 212.78.xxx.xxx:2048 -> 62.134.xxx.xxx:5060 SUBSCRIBE sip:12 at 62.134.xxx.xxx SIP/2.0..Via: SIP/2.0/UDP 212.78.xxx.xxx:2048;branch=z9hG4bK-d28tfohos0vh;rport..From: <sip:K922002626 at 62.134.xxx.xxx>;tag=vyx8c0trgx..To: <sip:12 at 62.134.xxx.xxx>;tag=as13e7cb7c..Call-ID:

That's just the thing, and it sucks, because the VoIP implementation actually works very good. Jon _____ Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af list mail Sendt: 8. juni 2006 02:34 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: SV: [Asterisk-Users] I can hear only one way when I use

I am new user of WINE and need a little help. 1. From the best I can tell I have everything configured correctly. However I can't figure out how to get a Windows screen to show up. 2. How do I get a WINE icon over to my desktop so as start it with a mouse click(s) Yes as you can tell I am very new to linux systems as I just installed this system this afternoon, so please be genle with

I am the IT Development Specialist for a small community college and our CIO has asked me to explore an alternative to Microsoft Active Directory as we are separating from our parent university and funding is tight so we were looking into CentOS with 389 Directory Server. Any advise or suggestions would be very helpful. Jacob Tennant

Hey all, Has anyone had any issues with phones having multiple lines that are in different contexts? We've got a couple phones that we're testing intercom functionality for, and I'm noticing that for some strange reason, no matter what line we use, the phones tend to be completely in one context or another, not segregated like I would expect. Our contexts look like this: context

I have been running Asterisk for years on a machine with a public IP. Most recently, I have been running 1.2.17, from the day it came out, with no (noticeable) problems. Yesterday, I switched over to a new server that is on the same public subnet, one higher than the original server. I built 1.2.17 from source on that machine (as I did on the old server). My firewall on the new machine is

Hello Be aware that the Nokia E60, E61 and E70 does not support NAT. Just to be shure that you know that. A clever choice from Nokia, so that users has to have some local equipment from the telco. Jon -----Oprindelig meddelelse----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af John Joseph Sendt: 7. juni 2006 13:59 Til: Asterisk Users

Hi, I noticed recently when installing the GDD package for R under GNU/Linux that it required the gd library (http://libgd.org/) for generating graphics. The resolution of this was to simply install the library on my system, and then GDD successfully installed without any complaints. However, the variant of GNU/Linux that I use is Gentoo, so I filed a bug requesting that a USE flag be set for

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type"> </head> <body bgcolor="#ffffff" text="#000000"> I have an Asterisk 1.4.18 with a mix of cordless phones connected using Linksys SPA2102 ATAs and Cisco 7940G

Hi, Guus The use case the shared default gateway for multi-tenant, if that the case the node who own the default gateway will have problem to route with different tenant who has overlapped address scope? Is it true when no any other tools like the namespaces? (tenant1)\ (tenant2)——common node—— shared gw node—— Internet (tenant3)/ But if the each tenant have it’s dedicate default gateway, but