similar to: "Cheapest" way to determine channels in a group from outside asterisk?

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

Hello list, we are developing a new application that uses the Manager API in order to find a set of channels where variables are set in a predefined way. To do this, we currently send a Status command to obtain all available channels and then query them all, one by one, fot the status of a certain dialplan variable. As you can imagine, this gets rapidly pretty tedious as the number of

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

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

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

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

> Asterisk Project Security Advisory - ASA-2007-013 > > +----------------------------------------------------------------------------------+ > | Product | Asterisk | > |----------------------+-----------------------------------------------------------| > | Summary | IAX2

> Asterisk Project Security Advisory - ASA-2007-013 > > +----------------------------------------------------------------------------------+ > | Product | Asterisk | > |----------------------+-----------------------------------------------------------| > | Summary | IAX2

Dear Tim; What is folks? Where I can find it about VPN solution? Regards Bilal > Hi, > > Greetings to All, > > Im looking for some help on configuring VPN on the Asterisk PBX that I > have hosted in US. Im currently in Middle East and as everyone knows > some countries here has taboo to VOIP. Im not able to get phy phones > registered to my PBX as they are blocking SIP

Hi All, Thank's for your hint Tim Panton I could connect my asterisk machine to my oracle machine. I used unixODBC-2.2.11.tar.gz, oracle-instantclient-basic-10.2.0.3-1.i386.rpm, oracle-instantclient-sqlplus-10.2.0.3-1.i386.rpm and the drive from www.oracle.com (odbc-oracle-3.1.0-linux-x86-glibc.tar) to configure my asterisk machine. I can connect to my oracle machine with isql and in

I've hit an odd issue in a test 1.8 deployment, playback() stalls mid file. The call stays up, but asterisk stops sending packets. It doesn't always happen - but on demo-congrats it happens about half the time. It only happens in IAX calls. Anyone else experienced it ? (I filed an issue just in case it isn't just me) T. Tim Panton - Web/VoIP consultant and implementor

Hi People, I had success to do my asterisk to record CDR in a databese MYSQL... Now, I need to do it to record CDR in Oracle... Does Anybody knows how to do this?? Every hints are welcome.... Thank`s all Everton Goularth Uberlandia - MG - Brazil _______________________________________________________ Yahoo! Mail - Sempre a melhor op??o para voc?! Experimente j? e veja as

<!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

This is a bit off topic, because I 'think' it isn't an Asterisk problem. However I'm not sure and anyhow I'm hoping someone may recognize the symptom. We moved offices a month ago. Our trusty SNOM190s (all between 3 and 5 years old) were packed up for the move, then unpacked a couple of weeks later. On unpacking them and connecting them to the new network, several of

Hello, I have been reading up on the capabilities of the Asterisk-Java library. I believe that this library can act as an interface between a Java GUI(custom softphone) and the Asterisk server. Seems like the Live API would be easiest to use to make the connection to the Asterisk server and to set-up calls. One thing I am not sure about is how to transmit the audio streams between users'

Can someone give me a little detail as to what this error message means and why it may be occuring? I keep seeing tons of these roll by on the CLI on one of our systems. Thanks! Apr 9 11:05:40 WARNING[19263]: chan_iax2.c:7538 socket_read: Received mini fra me before first full voice frame

Hi there, I'm looking for a compact fanless solution preferrably wall mountable and not too exotic. It needs to be commercial grade. I don't really consider most of the Via ITX solutions I have seen commercial grade but perhaps someone can convince me otherwise. This solution is about the best I have found. Maybe a bit on the exotic side but I like the fact it is wall mountable AND

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

Hello List, We are deveoping apication/system based on PHP5, Postgre,Ajax,ect.. It should be compleate sistem for realystate agensy and road worers(agents) and it will be distributed system. We made very good inplementation based on asterisk and OSP for distributed offices and it will be part of system (integrated). We would like to implement some options in system: Agent have contact list with

How does one process a return code in Asterisk? Example... exten => s,n,Playback(/tmp/podcast/${CALLERIDNUM}) exten => s,n,System(rm "/tmp/podcast/${CALLERIDNUM}.gsm") If the caller hangs up on the playback command the file remove System statement after it never gets executed. The playback command returns a -1 in this case and logs a warning. The only thing mentioned in the