similar to: One extension, multiple endpoints

Ok, I have a 7960 that's plugged into my laptop. my home network is wireless so I don't have a switch anywhere to plug the phone into directly. I'm running siproxd on my OS X laptop and I can make outbound calls from the 7960 fine (I guess I don't have the phone configured to register inbound calls via SIP), but the phone isn't registering to the asterisk box via siproxd

Hi...I've got two Grandstream phones attached to my Asterisk on the same subnet. The phones have fixed IP addresses. Asterisk is generated an error for one of them only, even though both appear to be registered correctly. The current state of the sip.conf is included below. Anyone know what is going on here? Both appear to be working fine between each other and between themselves in and

Hello all, This afternoon I had a BRI line installed by Telstra (our telco in Australia). I'm using an Eicon Diva 2.02 PCI ISDN card with the isdn4linux driver. Incoming and outgoing calls with Asterisk work fine (and with no echo - my main reason for getting ISDN). However, I can't seem to get outgoing DTMF working (incoming works fine). I made a call from my desk phone (Cisco 7940G)

Hello all, first post, need help. I'm running a complex asterisk 1.8 install with five machines. I inherited it and don't fully understand it, nor the deep mysteries of asterisk either. I would appreciate any insight you might have. I scoured the 'net and the Digium wiki and my Google-Fu has failed me. I've been asked to somehow prohibit transfers to extension 3232. It has to be

I would like to clarify the email I sent yesterday. There are two ethernet segments in two different cities that I would like to operate as one logical network. Both physical lans have a switch/hub, a gateway with one external IP address that NATs traffic and can port forward tinc ports to the internal debian stable machine (where tinc is run), various client computers ('c' in the

Hi, Is there any way to pick up a remote phone/extension wich is ringing for a long time but no one is availabe to answer that call? OR the scenario is like, suppose my extension is 1000, and I am working at a colleagues desk. At this time my phone is ringing in my desk and I want to pick up the extension I am presently at and get the call. I have another question about the remote login. Is

I'll start by saying I'm a trixbox user, and a new one at that, so hopefully you can respond to me on those terms. I have a user who works from home 1 day a week. On that day I'd like for him to be able to connect with a softphone and be reachable by just dialing his extension as we normally would. I could set him up a new extension, then he could forward his phone there on

Greetings List. we're facing a strange case with my system where in the middle of the call .. after like 7 minutes (not necessarily ) the callee is unable to hear the caller however the caller is able to hear the called party. the scenario is the following. 1- 15 computers running Windows XP SP3 joining a Windows Domain Controller with DHCP , DNS, ISA Internet Acceleration Server. 2- Internet

I have Asterisk version 1.0-RC1 running on Debian Woody. I have 1 analog phone working, 2 inbound lines working, X-Lite is working. The problem that I am having is with Cisco 7960 with SIP version 7.2 software. I can make outbound calls and they work fine, I even get a notice that I have voice mail on the phone and it seems to register properly but I can seem to dial to the phone.

I have an asterisk server up and running, using Firefly in IAX mode works great, even with Firefly behind a NAT (as expected, since IAX works really well with NAT). Now I'm trying to get X-Lite and/or Firefly to work in SIP mode from behind the NAT, and I can't seem to get there. At this point, the phone will successfully register with Asterisk, and the Asterisk qualify messages get

Hello, I want to record the message of both parties during a dial. My extensions.conf at the line where dial is looks like this: exten => s,803,Dial(SIP/arjankroon2,30,rR) My Sip.conf look like this: [arjankroon2] ;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! ;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed type=friend

I have a need to have the two sip phones register with the same extension (at least I think I have the need :) A client wants an incoming call to ring at the receptionists desk and also at their desk. If the receptionist is in it will be answered there and put on hold followed by a "Joe, you have a call on line 1". Is there a way to do this w/ asterisk? I've played with two

Hi, I am using the asterisk-oh323 wrapper and I am looking to allow registration of h323 endpoints and allow Asterisk to act as a gateway. The idea is simple: H323 endpoints would register with Asterisk. They each would have their own internal extension (like SIP). If a H323 endpoint dials an outbound extension, then the h323 call gets routed to a H323 Gatekeeper which then terminates

Hi folks - I've recently returned to asterisk after an eighteen month break. I've two sip providers - gradwell in the UK (inbound and outbound) and talklite in the US (outbound only). I've managed to get outbound dialing working but am not receiving any calls from gradwell. I've included my sip.conf and extensions.conf as well as the output from tethereal. When a call is placed

*This message was transferred with a trial version of CommuniGate(tm) Pro* Will asterisk actually convert between two different codecs????? ie, a SIP endpoint running GSM and another running G.711? Wouldn't that add quite some latency? I was always under the impression Asterisk did not recompress and was smart enough to negotiate the right codec at each end and just pass through the RTP

On Thu, 6 Aug 2015, Murthy Gandikota wrote: [trimming cruft nobody cares about anymore] > I use the same password for INBOUND and it works fine! Something amiss > with Asterisk OUTBOUND? because I used the same password with X-Lite and > X-Pro Vonage soft phones with successful calls. Would comparing an INVITE from X-Lite or X-Pro with the INVITE from Asterisk yield any clues? --

We have a system where calls are coming in from telcos via an opensips server and then being redirected out to a regular sip destination. There is no NAT, DTMF features, call recording, or codec translation being performed so I would expect asterisk to issue a reinvite after the call is answered and switch the audio however it is not happening. Here is the sip peer information for the call

Hi Muhammad From: "Muhammad Talha" <talha@worldcall.net.pk> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Wednesday, December 22, 2004 2:07 PM Subject: Re: [Asterisk-Users] Zaphfc/BRI Configuration help > Thanks for lan for your reply can you share your extention.conf > setting . > >

asterisk-users-request@lists.digium.com is believed to have said: >Hey Everyone, > >I downloaded and installed the X-Lite softphone the other day (the lite >version) and cannot seem to get it to work well. > >Don't get me wrong, it registers with my asterisk server and everything >seems to work well, except the call quality really is horrible. > >I thought it may

So I've got a bit of a head scratcher. Wanted to get some insight. I've got a PBX running 12.3.0 We're a ULAW shop from end to end. But I've been playing with G722 just for fun. I've got a Yealink T46G on my desk, And my colleague, A Polycom IP650 (Same office). Basically, Whenever I make an outbound call to a destination to something not G722 ready, I get no