search for: westhawk

...another. I've tried a fresh asterisk install, but that didn't help either. So I am forced to conclude that something went 'bad' in those (old) phones while they were switched off. Has anyone got any clues for me? Thanks! Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 2419 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090509/20ae0b8c/attachment.bin

Hi, I''m struggling to get the validates_associated to work as I think it should be. I''m using: JRuby 1.1 rails-2.0.2 activerecord-2.0.2 activerecord-jdbc-adapter-0.8.2 My tables in MySQL: CREATE TABLE area_codes ( id INT UNSIGNED auto_increment primary key ... ); CREATE TABLE markets ( id INT UNSIGNED auto_increment primary key, ... ); CREATE TABLE

...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 www.westhawk.co.uk

Hello, Let me explain a scenario There are different Asterisk Servers at different Remote locations. Recording in different formats for FIVE seconds reveals that Format : Size wav : 84 KB gsm : 8.3 KB sln : 84 KB It can be recorded in any format. This is size for five seconds only. We need to transfer these files from different remote servers to a centralized server. We need to play these

...k with a Digium Quad E1 card. Am I right in thinking that I'll need a special 'crossover-E1' RJ45 cable? If so, any clues where I might buy one in the UK? The Digium card sellers don't seem to stock such a thing. Thanks. Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk

...is claiming these are the pre-requisites ? How can I best track down what it _thinks_ is missing ? (This is on asterisk 1.8 svn trunk - but I don't think that is important, I think it is a package number issue) Thanks in advance, Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk

Hi, I am using a 64-bit RHEL 5 machine. I built Asterisk latest 1.6 branch. The system has ODBC and Postgres installed. psql, isql and odbc work fine. Asterisk "make menuselect" for some reason does not see the installed packages and refuses to build res_odbc and other packages. How do I force it to do that? Is there a way to modify the output file from menuselect and make it

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

...------------------------------------------------------| > | Reported By | Tim Panton, Mexuar, <tim@mexuar.com> | > | | | > | | Birgit Arkesteijn, Westhawk, <birgit@westhawk.co.uk> | > |----------------------+-----------------------------------------------------------| > | Posted On | May 4, 2007 | > |----------------------+-------------------------------------------...

...------------------------------------------------------| > | Reported By | Tim Panton, Mexuar, <tim@mexuar.com> | > | | | > | | Birgit Arkesteijn, Westhawk, <birgit@westhawk.co.uk> | > |----------------------+-----------------------------------------------------------| > | Posted On | May 4, 2007 | > |----------------------+-------------------------------------------...

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

...an 3 Is this likely to be a 1) config error 2) cable issue (I made them) 3) hardware problem with the Digium card 4) software (lib pri) Any clues? Anyone seen this recently (google shows it in 2005 but not since as far as I can see) T. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk

Anyone connected up to it yet? http://www.skypeforsip.com/ It would seem to make Digiums chan_skype rather pointness, or am I missing something? Or is this Digiums chan_skype in a hosted box somewhere? Gordon

hi, i'm searching solution for playing media(moh,prompts,voicemail,recordings - wav format) from adobe flash player (web browser) flash cannot play wav directly (imho) i must convert files to any other format on-the-fly - i cannot use mp3 because of royalties - next option is swf (with ffmpeg), supported free audio codecs (http://en.wikipedia.org/wiki/Flash_Video#Format_details) *

...pen - 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 > www.westhawk.co.uk > > Are both Asterisk's 1.8? I had unhappy results doing IAX between 1.4 and > 1.6 (1.8 is built on 1.6???) The far end is our voip supplier's asterisk - no Idea what version. But it also happens when talking to our Java IAX stack which isn't asterisk at all, so it isn...

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

...ifferent port. If they are blocking it by content inspection (unlikely but possible I suppose) you could try turning on encryption. Personally I'd try doing both the above before going down the VPN route. However I know folks have VPNs working very well. Tim. Tim Panton www.mexuar.net www.westhawk.co.uk/ ____________________________________________________________________________________ Pinpoint customers who are looking for what you sell. http://searchmarketing.yahoo.com/

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

I think I've forgotten something obvious.... I've got 2 incoming calls, I want to bridge them - how can I do this ? (assume I somehow know which calls should be paired up...) I could dump them both in a meetme - but that seems wasteful as i _know_ there will only ever be 2 parties. (And I need DTMF to flow through). I may want to record the bridged call, but that isn't vital.