similar to: Unable to get IP of eth0

I am having a problem trying to get a particular softphone working on my setup. The machine it runs on has more than one interface. When the softphone registers, it registers fine, and asterisk is given the correct IP for registration. Whenever RTP is set-up however, the client gives the wrong IP to connect to and I get the inevitable problem with one-way media. Is there any way of forcing

Hi all, I'd like to connect a softphone at home (nat, dynamic-ip) to a sip-phone in the office via asterisk 1.4.21 (nat, fixed-ip). SIP works well, the phone is ringing, but when I pickup the call, there's no audio on both sides. I debugged the rtp-traffic at home. As long as the phone is ringing, everything is fine. But after the pickup, asterisk sends a SIP/SDP package with its

Hi I've a softphone account to a AVAYA H.323 system, basically, it has a numeric ID (which is the extension number) and a numeric password. Instead of using the default AVAYA softphone (H.323), can I make asterisk as a H.323 client and login to the AVAYA system via any one of its h323 modules? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hey all, I've setup my asterisk install on a CentOS5 server, I've got a few IAX2 and SIP softphone clients connected on the same subnet and at least 1 external IAX2 softphone. However I'm having some difficulty getting the Polycom hardphone to function correctly. Watching the logs and debug trace it: - Registers correctly - Is able to make calls to other peers However it is not able

Is it possible to have asterisk override the connection information embedded in a SIP 200 packet with the registration information? I have multihomed machines with softphones and they register just fine and sip works fine, but the RTP packets get sent to the ip from the SIP connection information and the softphones are sending the wrong ip. I can't find an option in the softphone to change ip

How do I select a radio button when both id and name are identical?: <input type="radio" name="BILFPB.bilPremieUppgifterFI.under24" tabindex="13" value="J24" checked="checked" id="forare">Ja <input type="radio" name="BILFPB.bilPremieUppgifterFI.under24" tabindex="14" value="N24"

Hello, ------------- -------- --- -------- |Sip Softphone|-------|Internet|--------|F.W|-----|Asterisk| ------------- -------- --- -------- IP addresses: a.b.c.d q.w.e.r The SIP softphone(x-lite) is configured to register with the asterisk server through port 9090 (Domain q.w.e.r:9090).Firewall(F.W) is setup as the

I've noticed the following situation: In two softphones, I've configured the next codec order for each one softphone 1: 1 - PCMA 2 - GSM softphone 2: 1 - GSM 2 - PCMA and in Asterisk, the order is: disallow=all allow=gsm allow=alaw If I call from softphone 1 to softphone 2, I presume that Asterisk should do transcoding (canreinvite is set to no):

I have a software-only PBX set up. I can register various softphones and they will call each other -- but I've never succeeded in getting any voice routed from any of the softphones. Only the console will transmit audio. I am writing to ask if I have missed some obvious step in configuring the system. Conditions: (1) Softphones running on the same machine as the PBX: Only Kphone seems

Hi list, I am faced by a situation where I am trying to make a softphone and a Siemens C450IP talk to each other. Both are hooked up directly to the same asterisk, in the same IP net. - a softphone runs on 192.168.14.3 - the C450IP is at 192.168.14.30 - asterisk runs on the machine known as 192.168.14.1 I am running Asterisk 1.4.11, backported to Debian Etch by Xorcom. If I set

Hi, I'm having an issue with all my calls going out my SIP provider. I'm using a softphone registering to a local Asterisk PBX (I'm using Jitsi by the way - it's great and actively growing). I register as extension 4053 to asterisk server at 10.215.147.115 (alias IP - real IP addr. is 10.215.147.111) and dial a phone number that is routed via an Internet SIP provider. The call

Hi, everybody: I have looked at the Polycom entries on www.voip-info.org, and they're outdated and convoluted and full of errors. All I want to do is get my Polycom 501 to register with a working Asterisk server. I want to do the configuration locally on the phone through the console. (The server works with an Xten X-lite softphone.) Has anyone done this? What do I need to do? Thanks,

All, I have a simple question and a complicated reason for asking: Is it possible to change asterisk's source port for outbound IAX2 connections? I've tried using "sourceaddress" to no avail. I can set it to: proper.ip.of.box:4569 or 0.0.0.0:4569 and it works as expected. But if I try to set it to: proper.ip.of.box:5000 or 0.0.0.0:5000 it fails around line 8536 in

Hello! I'm searching for a way to call ATA (IAX or SIP) that is not registered with any server or proxy. Is it possible to make such a call from a softphone to an ATA just with IP? Something like (sip:// or iax://)1111@210.12.34.45 (where 210.12.34.45 is ATA's public ip)? Regards, CuPoTKa.

Thirdlane Connect can be used as a softphone. It works in modern browsers (no installation is required), on Mac, Windows and Linux desktops, and on mobile phones. Besides basic softphone functionality, it provides instant messaging, group chat (channels), voice and video conferencing, and screen sharing. It integrates with a variety of applications and CRMs such as Salesforce, Zoho, Zendesk,

Hello! I'm new to asterisk and because I try to configure the package for my needs the last days without success, I'd like to ask a basical qestion. I need asterisk to work together with the German VoIP provider sipgate (http://www.sipgate.de). Asterisk should act as a softphone, I want to recive and make calls only with the software under linux, no softphone should be used. Is this

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

Hi, I have my asterisk server inside a NAT. When i connect a softphone SIP inside my net, all go well. Ok. But when I try connect a SIP softphone client outside my NAT, I get: NOTICE[6087]: chan_sip.c:7691 handle_request: Registration from 'Alex <sip:phone2@192.168.1.21>' failed for '83.41.119.25' Can someone help me with this? PD: Sorry for my english

Hi Folks, I'm trying to register softphone(X-lite) but I'm not able to register softphone whenever I'm trying to register softphone I got below error [image: Inline image 1] Is there any document/guide line where I will get process to register softphone in asterisk(Which is installed in EC2 Cloud). Regards Akhilesh -------------- next part -------------- An HTML attachment was