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...th the phone if you relocate it from one desk to 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/2009050...

Hi We are about to publish a book, which contains figures made with R plots. An important detail that we did not take into account is that the book will not be printed in 4 colors (cmyk mode), but only 2 (black +"spotcolor"). The spotcolor we use is part of the big Pantone family. The problem is that both pdf() and postscript() offer either rgb or cmyk, but no spotcolors such as pantone. I'm afraid this constraint can't be solved at all, and we can't use R for creating these plots? I did not find any package that would extend the colormodel to includ...

...ment, 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 www.westhawk.co.uk

...'m looking to connect a BMC 450 to an asterisk 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

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 ! 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 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.

Just a quick question ---cut--- [Sep 17 15:52:14] WARNING[8232] app_dial.c: Unable to create channel of type 'IAX2' (cause 34 - Circuit/channel congestion) [Sep 17 15:52:14] WARNING[8232] chan_iax2.c: No more space [Sep 17 15:52:14] WARNING[8232] chan_iax2.c: Unable to create call [Sep 17 15:52:14] WARNING[8232] app_dial.c: Unable to create channel of type 'IAX2' (cause 34 -

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

...d run asterisk 1.4 beta2 on an intel mac? Did it work? I've got it building ok (once I installed Xcode, wget and bison) However Asterisk hangs on startup (halfway through loading the modules). I have not (yet) had time to debug it, but I wondered if anyone else had done this before me ? Tim Panton www.mexuar.com

...r/lib64/libltdl.so.3 - so I'm confused as it 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

On 8 Oct 2010, at 15:37, Danny Nicholas wrote: > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tim Panton > Sent: Friday, October 08, 2010 9:34 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Weird stalling of playback on IAX2 channels on > 1.8svn > > I've hit an odd issue in a test 1.8 deployment, > playback() stalls mid file. The...

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

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....

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

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'm setting up a small office PBX on asterisk, and I've got to the point where I have to decide on what to do about FAX. The current situation is: PRI (E1) connected to an E100 and sip hard phones on their own network. I'd like to add some very limited fax capability to the system. Basically we need to send a couple of faxes a month, and receive about the same number. My options

...---------------+-----------------------------------------------------------| > | Reported On | April 27, 2007 | > |----------------------+-----------------------------------------------------------| > | Reported By | Tim Panton, Mexuar, <tim@mexuar.com> | > | | | > | | Birgit Arkesteijn, Westhawk, <birgit@westhawk.co.uk> | > |----------------------+--------------------...

...---------------+-----------------------------------------------------------| > | Reported On | April 27, 2007 | > |----------------------+-----------------------------------------------------------| > | Reported By | Tim Panton, Mexuar, <tim@mexuar.com> | > | | | > | | Birgit Arkesteijn, Westhawk, <birgit@westhawk.co.uk> | > |----------------------+--------------------...