similar to: Re: newbie question-asterisk username/password

Greetings, I am new to all this VoIP stuff and have been having a bit of a hard time getting my soft phone working as a SIP client thru Asterisk. I apologize to start off with such a simple question and hope it's ok to post this and see what others have done. THE GOOD NEWS: I have successfully setup Asterisk 1.07 on an OSX machine. The build is running and working successfully. I am able

Greetings, I am new to all this VoIP stuff and have been having a bit of a hard time getting my soft phone working as a SIP client thru Asterisk. I apologize to start off with such a simple question and hope it's ok to post this and see what others have done. THE GOOD NEWS: I have successfully setup Asterisk 1.07 on an OSX machine. The build is running and working successfully. I am able

Does anyone know of any good step by step tutorials on getting sip set up? I have asterisk installed but can't seem to figure out how to get an account set up and connect from my xTen phone so I can try the demo. The tutorials I read online seem to go into voicepulse stuff and all and I don't have an account there so am a bit lost. Thanks! -------------- next part -------------- An HTML

Hi, which OSX softphone do you use that supports IAX2 protocol with Asterisk? thanks Mike

I'm an asterisk newbie, so initially tried to install it on a Mandrake 10.1 (see config below) and with a bit of messing about using sample config, have been able to make the test call to device 1000, and also through to the IAX test number at Digium. However, operation is extremely flaky - I cannot reliably startup and use the system on a regular basis. I have several problems listed below

Hello there, I'm a new Asterisk user and I'm having difficulties to connect to and from my Asterisk server. Can anybody give me a hand? Here's some background information: * I'm running RedHat Linux Enterprise 4.0 * When iptables is stopped, my server can register with IAX service providers and receive registrations from IAX softphones. However, it does not succeed in

Excuse the ASCII diagramme - you will need a fixed width font to understand it. ------ ------- ------- ----- | A | ==> | NAT | === === | NAT | <== | B | ------ ------- | | ------- ----- ------------------- | The Internet | -------------------

vpbx*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status 621/621 192.168.250.76 D N 5060 OK (65 ms) 626/626 192.168.250.109 D N 5060 OK (180 ms) 616/Ronald Softphone (Unspecified) D N 0 UNKNOWN 615/Ronald office 192.168.250.103 D N 5060 OK (41

Hello, Please forgive me if this has been addressed previously. I have been searching the archives and have not come across what I thought was a solution. My * server is behind a DSL router using a NAT IP address of 10.0.0.9. A colleague running XP and X-Lite can register with * from his home, specifying my public IP as the SIP proxy in X-Lite (however this is only true if I have the NAT flag

Hi, folks, Is it possible to connect a IAX softphone to a SIP softphone via Asterisk? IAX client -- Asterisk -- SIP client I tried that, and I was able to dial and talk to my IAX client from the SIP client. But not the other way around, I couldn't dial the SIP client from the IAX client. The SIP client was not ringing. Asterisk showed some WARNING messages: Aug 21 20:24:38

Anyway, in my P.S. yesterday (the main post was on Codec problems), I described a situation where any IAX softphone was registering successfully, and then having zero sounds heard on either side of the call. Here is an "iax2 debug" output from a DIAX call to a local * server, dialing the extension that goes directly to the "demo" application. AsteriskHouse*CLI> iax2

hi guys, i am getting today my dev kit with fxo and fxs boards. i intend to do the following : 1) be able to route an incoming call from the pstn fxo port to an ip (answering with netmeeting or anyother sip softphone) 2) be able to call from netmeeting to my pstn fxo port to place calls. questions : how can i do this ? what are the commands for this simple setup ? how can i place calls

Hi, We have set up our Asterisk server, our extension.conf and sip.conf according to http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=4 It's quite basic, and extension.conf is set up properly. The difficulty we are now encountering is in sip.conf, in trying to get any softphone to register at our own Asterisk server. We have searched the mailing list, and find bits and

i read that asterisk supports iax,sip and h323 protocols.... i've used sip & iax softphones ... now i've a hardphone... an IP phone (Netphone) that supports h323 ..... i've compiled pwlib ,oh323 and asterisk -oh323 successfully ... but i m unable to place calls to/by my phone... i m confused whether to use h323.conf or oh323.conf and how ? i think it's different from iax.conf

I have an Ubuntu system and went into Synaptic and checked asterisk for installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc and got the following output with several errors and notices. Do I need to do more or are these ok? I expected to have some conf files in /etc/asterisk but there is nothing there. Thanks! Created by Mark Spencer <markster@digium.com>

hi, I use IAX softphone with asterisk and I notice that a call between two IAX softphones end after 1 min. Then I can't hear anything but the call still in progress. I have this log in asterisk IAX debug: Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00016ms SCall: 21589 DCall: 00001 [192.168.1.22:4569] Tx-Frame Retry[000] --

Hi, Try as a first step a tcpdump capture to verify if the softphone is actually sending the register message to the server. For me it seems like the softphone is not able to reach the server ! Best Regards, On Fri, Apr 24, 2015 at 10:55 AM, Helvio Junior <helvio.listas at gmail.com> wrote: > Hi Akhilesh, > > SIP protocol use port 5060 (default) and many other ports to stablish

Ok, I have trixbox working how I want. How do I now (cheaply as possibly) get a phone number so people can call it from any number? I am just doing a prototype so just want it done cheaply so I can demo it to my supervisors. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

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