similar to: IAX2 protocol

DOes anyone have any good suggestions as to good SIP or IAX clients for linux? I have set up and am currently testing asterisk in a controlled environment. I have gnophone running on one of my boxes but the gnophone site has been down. So I can't seem to fing the IAX and Ix-devel rpms or the gsm and gsm-devel rpms. So that prevents me from setting up gnophone on another box. I am

Hey all...I'm trying to use gnophone to connect to my asterisk box behind my firewall..I thought I could just setup a tunnel with something like ssh host.com -L5036:asteriskserver:5036 and just change my gnophone to connect to localhost:5036 but I never see anything happen on the asterisk server. I'm even trying this on the same network just in case there is something funky with NAT.

Sorry, this is a re-post. I sent the message from the wrong e-mail address. I was checking up on the cvs changes. Hi. I just tried setting up the send url function of Queue app, but it did not seem to work with gnophone. I must have missed something. It never tries to go to the page. I have this in extensions.conf: exten => 1,3,Queue(support,t,http://www.yahoo.com) It shows in the

Is it possible for two PDA's to communicate like telephones via SIP channels on a PC running Asterisk? If that is possible, does there exist any applications that can be installed on a Zaurus 5600, which is a PDA with an Xscale processor running on a Linux OS, that can essentially turn it into a softphone? Thanks in advance for any input, Daniel

Anyone else having problems connecting to http://downloads.digium.com/pub/ ?? Jonn

Dear Asterisk-Users: Question: ======== How do I get asterisk to pass DNID/RDNIS information between asterisk machines using iax2, in a Dial(IAX2...) command ? Setup: ===== I have two asterisk boxes, MASTER and SLAVE. MASTER is running 1.2.0 and SLAVE is running 1.2.1. The main box handles incoming calls on a multiple lines (both via hardware connection to our internal PBX and calls

Hi, I'm working on a call center application where callers input some information and get transferred to an attendant, or waits in a queue until one is available. The operator is using a PC-based system that needs to have access to the information previously input by the caller. I was thinking about making * write some control info somewhere and then make the application get it through

Would like to deploy * in a small help desk environment (five to ten people) using call queues and some sort of CTI interface to pop Remedy screen data in front of the help desk person receiving the call. Data to be popped would be based on CallerID. Anyone doing something similar? Anyone interfacing to an external Remedy system? Any reference sites that I could read/learn more of the

Andres sounds as if this is Andres's own development. He mentioned IAX, not IAX2. My guess is that he might have used one of the IAX GPL Libraries and source trees, based on iaxClient and not libiax2. It is possible that Andres is not aware of the GPL terms that he has to adhere to, if he wants to commercialize this product. The Source code for IAX Phone is available from Steven Sokol's

On 20/04/15 17:45, john wrote: > > > > Is this wheezy ? if so, it might be an idea to use backports, this > will get you 4.1.17 which is still in development, 3.6 is now EOL > OK, I understand a bit better where your problems lie. I would still use backports, supported code is (hopefully) better code :-) > > > I'd be willing to do that if it got me support

I have been working on a cross platform library for connecting to asterisk along with Steve Kann. I recently was testing the library against the recent code for asterisk from CVS and got the following error when using the command line client (similar to miniphone). Text Based Telephony Client. TeleClient> call 192.168.2.40 TeleClient> Unknown event: 15 dump Dumping call! TeleClient>

I have looked through the archives, and can only find old references to this problem that appear to be no longer relevant, so I thought I'd ask again. I am having a problem with periodic breaks in audio over an IAX trunk. The interruption only happens in one direction, and (I think) only with clients built on the open source libiax. Codec is irrelevant, and jitterbuffer on/off seems to

I am having a really hard time getting gnophone working with asterisk. Gnophone tries to register with my server but there is no response. I can direct incoming calls to gnophone but if gnophone answers them, asterisk does not recognize it. Here is my configuration: iax.conf [jambo] type=user host=dynamic defaultip=136.159.99.100 permit=136.159.99.100 username=jambo secret=fubar

Hi all, I installed all needed RPMs by GnoPhone to be installed without problems but when attempting to install GnoPhone itself I get this message: # rpm -Uvh gnophone-0.2.4-1.i386.rpm error: Failed dependencies: mozilla >= 0.9.2 is needed by gnophone-0.2.4-1 libgtkembedmoz.so is needed by gnophone-0.2.4-1 libgtksuperwin.so is needed by gnophone-0.2.4-1 I'm using

Hello, from what I've understood Iax2 should support aes128 encryption. I've found this old info: http://www.voip-info.org/wiki/view/IAX+encryption and this (unanswered?) post http://lists.digium.com/pipermail/asterisk-security/2005-August/000060.h tml Is this the libiax used currently on asterisk http://ftp.digium.com/pub/libiax/ ? I would like to understand if someone is using this in

On Sun, Aug 17, 2003 at 03:44:21AM -0500, Gnophone Support wrote: > Hello, and thank you for registering at gnophone.com. Your login > information is listed below: > > Username: miernik > Password: ******* > IAX Phone Number: 17002916107 > > Please login as soon as possible to > http://x.linux-support.net/directory/ to complete the

How do I get Gnophone to register to my Asterisk server? I have set up iax. conf as follows: [tim] type=friend ;username=tstornes host=dynamic ;defaultip=207.194.60.56 secret=1111 context=from-iax callerid => "Tim" <5000> auth=plaintext qualify=10 permit=0.0.0.0/0.0.0.0 and extensions.conf includes a section in the context from-iax: exten => 5000,1,Dial(IAX/tim/s|100|r)

Hi All, I know, IAX is NAT friendly, but... I have a problem running gnophone from a box behind NAT firewall. I can register gnophone with * through NAT, but when I try to make a call it instantly disconnects. CLI iax show peers command tells me that peer is unreachable. However this peer is registred. Gnophone also tells me that it is registred. It seems that registration handshake has

hello everybody well...while trying to make gnophone to gnophone call using IAXtel's PBX server......i am not being able to establish a connetion.... possibly whats happening is, we are not being able to transmit our message packets properly and result is its not being able to establish connection n is resending frmaes again and gain here is message we are getting Tx-Frame Retry[000]

hello all i'm interested in setting up a small pbx using asterisk and the primary goal is keeping the cost down. the general layout of the net is as folows: * 4 phone lines (2x isdn+ 2x analog) [or 2x isdn + 1x analog, as one might be put aside for a traditional phone/fax with no fancy stuff] * a server box * several client hosts (all linux with x) currently, the isdn lines are unused (to