similar to: VoIP Test Samples to test Asterisk

Hi Folks, I installed Asterisk@home on my PC. It went through the installation and all. But now i get a command line login window. Doesn't it has a KDE or some other type of OS GUI (i am not talking about Asterisk@home web GUI)? After i login, just the command line interface comes out. Any command to type here to get Linux OS GUI? Thanks, Kiran

Can anyone _recommend_ a downloadable OSS softphone that _works_ under Linux and is compatible with Asterisk. So far I have tried kphone and linphone and had problems with both, and I am still waiting to hear back from the X-Lite beta folks. -- Howard. LANNet Computing Associates; Your Linux people <http://www.lannetlinux.com> ------------------------------------------ "When you just

I'm trying to find a voip-suitable USB headset (I.E. headphones + microphone) which I can use with my laptop while I'm traveling and using Firefly or another softphone. I'm currently using a Logitech headset which works well (except the echo it generates toward the other caller when I turn up the gains too high), but it just doesn't carry well - in fact, I can't carry it in my

I am an evil M$ MCSE :), who of late is planning on opening his own business... Due to the large cost involved with setting up Win2K server, both Hardware and OS, we are considering a Linux server setup. We have SOME experience with Linux, but were after information specifically on the possible implementation of Win2K Group Policies (as well as 'Domain' Authentication_ for the client

Hi: Iam looking for a sip softphone that supports g729 codec Any one have an idea ? Reagrds; jonnyhashem --------------------------------- Don't get soaked. Take a quick peak at the forecast with theYahoo! Search weather shortcut. -------------- next part -------------- An HTML attachment was scrubbed... URL:

After many months of development, I'm pleased to announced Firefly - an IAX soft phone and network. The firefly softphone - free, runs under windows, allows connection to multiple networks, skinable interface, connection to firefly network, IAX2 protocol, (SIP in next release), codecs supported - iLBC, G.711 ulaw/alaw, GSM. - contact lists, selectable ringtones. download from here -

Is there a softphone for windows that supports IAX? I can't seem to find anything out there...maybe im looking in the wrong places... Jeromy Grimmett VoipEmpire.com jeromy@voipempire.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050523/e668136a/attachment.htm

Hello, I have a one-way audio problem. If any one can give me a clue on how to solve it, I'd highly appreciate. My configuration is: Both Asterisk server and a SIP phone run within a LAN. Asterisk: CVS-HEAD-06/27/04-11:42:23. SIP phone is X-Lite release 1103m build stamp 14262. The Linux box that running Asterisk server is RedHat 2.4.18-14. Asterisk server runs on IP: 192.168.1.102. X-Lite

Quick question from a newbie, I have asterisk configured to dial IAX extensions (which works). When dialing from one IAX extension (using Firefly) to another IAX extension (also using Firefly), the Firefly client rings on the receiving end and gives the option of accepting or denying the call. However, when I dial in to Asterix using a VoicePulse number and dial the same extension Firefly

Hi We've recently set up and are using with success 1.0.7 using a Junghanns quadbri card to BRI ISDN, and Firefly with IAX2 protocol as softphones Works very well, however we're getting cases where sometimes the call just drops. >From setting more verbose modes we get a log which is shown below. The problem seems to be the maxretries message which comes from chan_iax2. We are using

How come FireFly doesn't give me an Inband DTMF option? Only RFC2833 and Info. RFC2833 is the default, so I left it that way. I also unchecked all the codecs except g711ulaw to force that codecs usage. However, when I go to place a call, I get this: Nov 3 13:18:44 WARNING[53641241]: dsp.c:1468 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833 Nov 3 13:18:44

Using Firefly 1.9.5 (thirdparty) on Win2k Using Asterisk CVS HEAD 20041117 (also tried with 20040806 and 200410-something) IAX2, no NAT. Firefly->Asterisk audio works, but I can't hear anything from the other side. Using GSM codec, also tried ulaw. Any ideas? -A. relevant bits of iax.conf: [andrew-bt] type=peer host=dynamic trunk=no [andrew-bt] type=user context=fxs secret=12345

Hi List; I am using Firefly softphone Version 1.9.9 Build 4521 and I select IAX protocol and did the configuration in Network1 (and I checked the Active checkbox) as following: Server: 192.168.8.4 username: iax2user1 password: password In the Asterisk, I did the following configuration on the /etc/asterisk/iax.conf: [iax2user1] type=friend context=internal username=iax2user1 secret=password

Hi All; I tried Firefly softphone with IAX and it gave very poor quality. Any one advise a good strong softphone that can work with IAX fine? Regards Bilal ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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:

are there any open source sip softphone (Window OS version )?

Hi, I have the following situation: I've installed Asterisk at Machine 1 (M1 - IP: 192.168.1.145) and X-Lite (X_lite-Xten-Win32-1103m.exe from www.xten.com) at Machine 2 (M2 - IP: 192.168.1.100) and Machine 3 (M3 - IP: 192.168.1.200). I need to catch the SIP and MGCP messages that will appear when M2 calls to M3 and vice versa. The SIP messages are working (I don't have problems with the

Hello, I am looking for Ogg-vorbis streams with block sizes 4096 and 8192. Please let me how do generate these streams. This is to test our fixed-point implementation... Best regards Kiran Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential

Hi all, I have a problem with concatinating strings while taking as a path here the problem is i have to take path as FPATH<-"D:\\Kiran" and file name as Fname<-"FINDINGS.CSV" and while I am reading this table I have to take path with using these two strings because in "FPATH" there is many files like findings.csv, and path will be

Hi, I am running Redhat RHEL 4, authentification is via kerberos against and AD server, usernames are supplied via ldap service running on another redhat box - winbind has been seg faulting repeating when accessing samba - always the same error message... see logs below - can anyone tell me whats going on? Mar 14 16:12:45 firefly winbindd[14752]: [2008/03/14 16:12:45, 0]