similar to: outgoing call on ISDN PRI

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

Hi guys, Here is a bit more detailed information of my problem: If I connect Asterisk PBX to the Polish telco via E1, I don't get any red alarms or anything. The line seems to be fine and the inbound calls are also accepted by the Asterisk. However, whenever I try to make an outbound call, the call is either stuck (Asterisk just displays Called g1/482.... and then nothing), or I get the

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

My legacy PBX accepts CID number, but not name. My old PRI vendor never sent the name, so there was never an issue. I have wedged asterisk between the Legacy PBX and PSTN. PSTN - PRI - asterisk - PRI - Legacy. Any calls from asterisk (sip and iax extensions) which have callerID set, will not connect. The legacy PBX hangs up, but asterisk thinks that it is still ringing. I have added

Hello Group I have a Asterisk server running with 2 Digium T1 cards installed. 1 card connects to Telco via a PRI. The 2nd card is connected to a fax server via Digi DataFire RAS 24 PT1 Adapter (Digi0001). The idea is to have Asterisk route the calls based on DID or FAX tones. Everything is working great so far. The only problem is the Fax server does not see the DID. How can I tell if Asterisk

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

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

Hello, I am trying to install my E1 card to make a conection with an Ericsson MD-110 PBX. I installed dahdi drivers as: dahdi_hardware pci:0000:04:08.0 wcte12xp- d161:8000 Wildcard TE121 ran dahdi_genconf and it created all my e1 ports. On the other side i also configured the pbx to communicate with TE121. On ericsson side, i have no error messages. On asterisk side, no error messages. But

<!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

My outgoing FXO calls are answered but have no audio in either direction if I have callprogress=no in chan_dahdi.conf. If I change to callprogress=yes then the audio returns. My chan_dahdi.conf file is listed below. Can anyone point-out why callprogress=no isn't working? #cat /tmp/a [trunkgroups] [channels] language=en context=incoming toneduration=40 ;usedistinctiveringdetection=yes

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

I need to determine the number of active calls in a group from outside of Asterisk. Currently I poll the manager API and parse the channel status list but this is becoming too expensive on CPU. What are my options? What is considered "standard practice" ? Update a DB field? Poll the manager api? Use an asterisk -rv 'some command' call?