similar to: Creating an SIP softphone

Displaying 20 results from an estimated 10000 matches similar to: "Creating an SIP softphone"

2007 Sep 20
10
IAX Java Softphone?
Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein
2007 Sep 17
1
Softphone RTP Session Start-up Delay
Hello, I have a small LAN network where I am running a Jain-Sip softphone on two user pc's. These softphones are connected through Asterisk(Trixbox). Although the phones do work in providing an audio conversation, there is a long delay(about 20 seconds) in the initial RTP session setup. I have tried a few values for the buffer length including setting it to zero. I assumed this would
2007 Aug 27
1
Can't create audio conversation between softphonesthrough Asterisk
Hi, In the early stages of deciding how to try and develop this environment, I looked at all the protocols that could be used. SIP was chosen just because it seemed to me that it was the most widely used protocol. I believe IAX is a new protocol with a little less documentation and examples. The good thing about this Jain-sip-phone is that it saves a lot of time since many of the important
2006 Nov 01
6
Java Web Phone
Hello list partners you know about a softphone made in java attachable in a web page? GNU! Thaks in advance!______________________________ Visita http://www.tutopia.com y comienza a navegar m?s r?pido en Internet. Tutopia es Internet para todos. -------------- next part -------------- An HTML attachment was scrubbed... URL:
2007 Sep 11
1
Chan_sip Entry
Hello, I am trying to get to Jain Sip softphones to call one another via an Asterisk server. When I call from phone 1 to phone 2 there is audio transmission both ways, but when I call from phone 2 to phone 1 I don't get audio transmission and reception both ways. When I look at the asterisk log file it has an entry which says: "Oooh, format changed to 2". Would anyone know why
2006 Nov 13
3
FW: Desktop integration
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta content="text/html;charset=UTF-8" http-equiv="Content-Type"> </head> <body bgcolor="#ffffff" text="#000066"> Hi Dean,<br> <br> I will check that site - thanks for the hint.<br> The biggest problem I see with
2007 Aug 23
6
Asterisk Message Logs
Hello, Is it possible to print the Asterisk message logs to a file, or is this already done? By message logs I mean the display that shows up on the asterisk server when a call is made from one user to another. I believe if the verbosity is high, it can show what parts of the extension.conf file that it uses when making the call. I am trying to use two Jain-sip-applet-phones, connected through
2006 Oct 23
8
Asterisk and dialer Running on Thin Clients
Hi everybody Im the IT Manager for a new call center and my bosses has assing to me a very dificult task i have to configure the call center using Hp 5520 thin clients, asterisk and some kind of dialer that allows outbound calls. I triyed using terminal services but it dind worked because the lack on the sound and the microphone do not work on the thin clients using terminal services, we tried
2007 Jun 12
2
Softphone behind NAT issues
We are trying to use a softphone from a location that is behind a firewall. We are using x-lite as the softphone. So far, we've been able to get the phone to register with the asterisk server, and it can receive voice from the asterisk server (IE, voicemail, etc). However, asterisk can't hear anything from the softphone. We have used 2 different machines to test this on. We are watching
2007 Feb 07
2
Softphone +Realtime
Here's an interesting issue we're facing... We would like users to be able to use softphones from home/work and to use their same extensions they do at work. The first step of getting the phones to log in as their same extensions as work is easy and works. However, on the database side, once the client closes, the sip table is cleared of the ip to the phone. This means that no calls are
2006 Apr 20
2
Cubix Softphone + Asterisk 1.2.6
I've tried Idefisk and Cubix Softphones, and they both work fine, except for two issues: 1. Idefisk seems to have a longer delay between the time I can hit tones, and 2. Cubix, while can send DTMF faster, never actually connects to an Asterisk-dialed call -- I can't hear the party who answers. #2 has been asked but unanswered here:
2009 Jan 09
1
Web Softphone
Hi all! Im looking for 1ezphone to use as a web softphone but I?cant access to 1ezphone.com. Anyone knows what happened with this site?. Thanks! Gustavo A. Gonz?lez Dto. de Infraestructura Despegar.com, Inc. ggonzalez at despegar.com -------------- next part -------------- An HTML attachment was scrubbed... URL:
2007 Aug 24
1
Can't create audio conversation between softphones through Asterisk
Hello, I have two user machines, each with a jain-sip-applet-phone installed on it. I use the following process to try to make a call: 1. Register each phone with the Asterisk server (working). 2. Add a contact in each phone which is the other user. (Get a "489 Bad Event" SIP error shown below in red) 202 at 192.168.1.252 has been added to your contacts. null send request:
2006 Apr 26
6
I am looking for a webphone on MY SITE
I am looking for a way of not to install a softphone, preferable as a link on a web site to a webphone on MY SITE !!! Has anybody an idea for that? AJAX? bye Ronald Wiplinger
2007 Apr 05
5
Open Source VoIP client (on a webpage)
I need to decide on the best way to add a voip SIP or IAX client to a website. I'm thinking that I'd like it to be inline, like an aplet, on the page. I've got some asterisk servers running to connect up to, so the real challenge is finding an easily integrated open source client. Any suggestions from those who know? Jason
2005 Mar 10
1
what is best free softphone.
I use xlite , but it isn't support video when it is free. who used better softphone ? Thank u. Best Regards Zhao Zigang ??? Alcatel Shanghai Bell Co., LTD *:388,NingQiao Rd.,Shanghai 201206 *:086-21-50554550-7762 *:Zigang.Zhao@alcatel-sbell.com.cn
2007 Aug 10
0
Sending live audio in Asterisk
Hello, I am trying to create a Java GUI that will interact with an Asterisk Server. This Java GUI will essentially be a custom made SIP softphone. I will most likely use the Asterisk-Java Live API to create the connection to the Asterisk server and to open a new call. Then, I plan to use the JAIN SIP API to initialize the session and the JMF to send the audio streams via RTP when the two users
2005 Jun 10
0
SoftPhone - Solaris
Hi, I am looking for a softphone (sip or iax) that works in Solaris/SPARC with sunray100 terminals. I found iaxcomm but it doesn't work. Also I am trying sip-communicator but I have several errors from JMF/RTP. Does anyone have a softphone working over this platform? which one? I don't care if it is a commercial product, I can buy it if works fine. thanks in advance. Sebas --
2007 Jan 16
2
IAX2 softphones can't (won't?) use PRI trunks....
I have call center PCs that switch between an IBEAM SIP softphone and a NEBU IAX softphone (for reasons that aren't germane here). The SIP softphones work fine, but the IAX softphones get a fast busy unless I give them an IAX trunk to use, instead of the PRI trunks that all the other phones are using. I am using Asterisk 1.2.3. svn rev 47264. I've appended a sample call trace. The
2007 Jan 16
1
Ring tone too loud on IAX channel
Hi, We are using MozIAX as a softphone with a USB headset and are making outbound calls using IAX with ulaw encoding to our voip provider. We're running asterisk 1.4 Users are complaining that the ring tone generated by asterisk is much louder than the voice call once connected. They are having to turn the volume down to avoid being deafened by the ring tone, but then have an unacceptably