similar to: Multiple IAX clients behind a firewall

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

Is anyone aware of an milliwat/echo test number for telstra or similar? I want to fiddle a bit with gains but can't seem to get a hold of a miliwatt test number Im aware of a number they distribute for line quality testing for modems for BigPond, but I can't track it down .... -- Adrian Carter Technical Manager Leading Edge Internet Web http://www.lei.net.au http://support.lei.net.au

I am playing around with IAX encryption and have had good success. I read somewhere, that trunked packets are not encrypted. Does anybody know if this means the trunk packets themselves are not encrypted but the voice frames in them are encrypted or does this mean that if you are using trunking then encryption of the voice frames will not occur. I have used Wireshark to sniff the packets and it

I couldn't find any specific reference to this but maybe Im missing something completly... anyways, when trying to mix a few wav files together post-recording (the -in/-out files) using a pretty vanila soxmix line, I get the error: Done Mixing OUT115-20060215-150749-1139976460.7898-out.WAV..... Mixing OUT115-20060215-155022-1139979011.8787..... /usr/bin/soxmix: invalid wav gsm frame size: 1

I don't follow. Are you saying there's a limit of 10 contexts within extensions.conf? If, on the other hand, your saying that each context can only have 10 #include statements, I tried that too. See below. I put 12 #include's across two contexts, and still got the same error, I not sure that goto's are much of a solution, but I'll have to proof it. And that's my

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

Hi all, All of a sudden all my IAX DIDs have gone down. I couldn't find any reason other than that the ISP is blocking port 4569. DIDs register fine from my home server, but not from office server, which is not behind any NAT. SIP registers fine. I am trying to change IAX port but it apparantly IAX works only on 4569. Changing it in iax.conf doesn't do anything. Changing it is

Ooops did Google just become a carrier :) http://googleblog.blogspot.com/2007/07/all-aboard.html I hear stocks crumbling worldwide as I type. Cheers, Dean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070703/92218fc6/attachment.htm

Hi, I have an IVR that sounds just fine and dandy over ZAP. However, when I dial in through an 800 number from a provider that I connect to via IAX I get this 'blip' in the sound file. At first I thought it was just packet loss, but it happens at the exact same spot every single time. There are several parts to the IVR menu and several places where this blip happens every single time

Can someone tell me if passwords are sent in plain text when using IAX? I have been told already that SIP automatically encrypts the password? Anyone know of some good Asterisk security links, docs, articles? Thanks!

Does anyone know how many simultaneous calls can a WRTG54GS handle? Assuming SIP phones are connected locally using G711.u codec and the WRTG54GS connects to a remote Asterisk server using IAX2 trunking using GSM codec. Thanks, Daniel

HEllo list ! When user A calls user B via Asterisk (Users A and B are registered on the same Asterisk server ) and an ISDN PRI, user B phone always shows Asterisk server telephone number. How to hide it and how to forward user A number ? We tried usecallerid, callerid, hidecallerid, restrictcid, usecallingpres in zapata.conf but we always see Asterisk server telephone number ! Thanks

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

Hi, Just a quick question. Is there a way when making an IAX call to transmit some additional call-data, perhaps in a variable? I could overload callerid-name, but that is nasty and ugly :) Thanks for any suggestions. Regards, Steve

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

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:

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

Hi All, I'm having problem with IAX, I'm trying to connect to speex.co.il from asterisk using: register => username:password@speex.dyndns.org and I cant get it to work. Maybe someone who already got this to work will help... When dialing my speex extension I see the next output from consol: IAX2 Debugging Enabled *CLI> Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno:

Some more info: Just tried this on a server without using any NAT and no port forwarding, no masquerading, and I still have the same problem. So there goes that idea. I do not know what this VNAK error means. By the way, I am using the latest version (1.2.6) of asterisk, have also tried other versions with the same problem [1.0.9 (Ubuntu Breezy) and 1.0.7 (Debian Sarge) and 1.2.1 (Ubuntu

Hi, someone know a good webphone, possibily a free one Thx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060627/0e83bc29/attachment.htm