similar to: callgroup and pickupgroup

Hi, Talking about messenger,,, it's still necesary to do HKEY_CURRENT_USER\Software\Microsoft\MessengerService\Corp2PC_Phone equals to '1' ??? But it's still sending the '+' digit, so it's necesary to stripMSD? Thanks a lot cmayor ___________________________________________________ Yahoo! Messenger - Nueva versi?n GRATIS Super Webcam, voz, caritas animadas, y m?s...

Hi everybody, I'm trying to use msn with * and for that, I'm reading all information on the mailing list. You used to recommend the page http://mcleod.pbx.nq.net/msn/, but I always get an error while opening. Has it changed? Is there another one? Thanks cmayor ___________________________________________________ Yahoo! Messenger - Nueva versi?n GRATIS Super Webcam, voz, caritas animadas, y

Hi, A question about transfert. How can I make transfert with the the person who call. X call Z and X transfert Z to Y. I only succeed to do X call Z and Z transfert to Y. If someone have a solution it will be very good =) regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, We are using asterisk 0.4.0 on debian sid with Cisco 7960 and ATA186 phones. All sip entries have: callgroup=1 pickupgroup=1 However I am unable to remotely pickup a ringing phone using *8#. I get fast busy tone. Is there some flag to add in extensions.conf ? Thanks in advance,

what is the format for an h323 entry in the dialplan? can I use chan_h323 without compiling anything else or should I compile oh323? basically what's the best way :) cheers Dave

Hi since "callerid=" in sip.conf doesn't set the Caller ID, I suppose it must be there for some other reason. Is this a not-yet-working feature for future releases of Asterisk? If not, what does it actually do? thanks regards bk

Hi everybody and sorry for posting this again to the list. I don't want you guys to think that I'm DEMANDING FOR SUPPORT (we all have just had several discusions about that) but my experience tell me that if a posted question is easy enough, it is answered immediatly, or it will never be answerd!!!(people forget it...) Mine are two very simple questions about modules.conf. It's only a

I have just compiled & installed the latest oh323, on a fresh asterisk installation however using a previously working oh323.conf file. When I try to dial an outbound oh323 call I get the following error : -- Going to extension s|1 because of immediate=yes -- Executing Wait("Zap/1-1", "1") in new stack -- Accepting call from '21382890' to 's'

Hi! When I try to make a call I have these messages: dial 06302@gatekeeper -- Executing Dial("OSS/dsp", "OH323/06302|20|tT") in new stack *CLI> WrapH323Connection::WrapH323Connection: WrapH323Connection created. -- Called 06302 WARNING[1272234688]: File chan_oss.c, Line 624 (oss_read): Error reading from sound device (If you're running 'artsd' then

Hello. I have Wine20030709 in Mandrake9.1 without Windows. I have installed MSOffice97, WinRar, AcrobatReader5, PowerTranslator,.... even my "Encilopedia Micronet". But I cannot see videotapes. If I copy and paste mplayer.exe, it doesn't work (clear). If I install MediaPlayer6, the installer says: all OK (extracting, copying, configuring,...). But he/she has not made

Hello... I'm having problems with H323/G729 setup. Below is the output of h.323 debug when making a call. I use a SIP phone connected to an * box in the same LAN. The * connects to a h323/g729 PSTN terminator through internet. Calls rings and are answered in the other side, but I get no sound at all nor the other side does (complete silence in both sides). I thought this would just happen

Hi everybody, I have three questions to ask us: a) R incorporates a function for the Non-central T distribution which unfortunately and, as you know, is not available in Splus 4.5. In http://www.stats.ox.ac.uk/pub/Swin I found the Don MacQueen?s noncent.zip but when I run it in Splus 4.5 the following error message appears: "Error in .Fortran ("vectnc",: "VECTNC" is not a

I'm trying to set up asterisk so any phone connected to channel 1-16 of my Adit600 channel bank can pick up a call coming in on channel 24. I do not wish to ring any of the 16 channels on an incoming call -- this is strictly so I can pick up the line if I see it ringing and wish to answer at work. I have channel 24 in call group 3, and channels 1-16 in pickup groups 1 and 3. However

Hi! Is it possible to merge two ext3/ext2 partitions into ONE ext3/ext2 partition? -- Ralf Hildebrandt (Im Auftrag des Referat V a) Ralf.Hildebrandt at charite.de Charite - Universit?tsmedizin Berlin Tel. +49 (0)30-450 570-155 Gemeinsame Einrichtung von FU- und HU-Berlin Fax. +49 (0)30-450 570-916 IT-Zentrum Standort Campus Mitte AIM. ralfpostfix

I have just started to play with callgroups and pickupgroups.. I updates my * from CVS this morning (about 15 mins ago).. I have placed callgroup=1 and pickupgroup=1 into each of my 3 phone configurations in sip.conf.. I place a call from phoneA to phoneB, then I go to phoneC and dial *8# , the call does not get picked up by phoneC and continues to ring on phoneB.. Have I not configured

This is what happens when i run the ./tools/wineinstall WINE Installer v0.74 Running configure... configure: creating cache config.cache checking build system type... i686-pc-linux-gnuoldld checking host system type... i686-pc-linux-gnuoldld checking whether make sets $(MAKE)... yes checking for gcc... no checking for cc... no checking for cc... no checking for cl... no configure: error: no

I have a test SIP account set up with WorldCom and I have been trying to have Asterisk register to the WorldCom server with no luck. It appears that the SIP headers are different coming from Asterisk. I have included a packet capture from a successful login with a Windows Messenger client for reference. I have also copied in the SIP packet I captured with sip debug turned on. In my sip.conf file,

I have problems using the call pickup under Asterisk 1.8.4.2. I have another Asterisk with 1.6 - and it is working fine with the same settings. I have setup the same callgroup and pickupgroup for all extensions in sip.conf - just to make things simple for testing. The sequence *8 seems to be completely ignored by Asterisk - the client shows "Call answered" when dialing *8 while the

HI,ALL, I have multiple PSTN lines registered as multiple SIP channels (e.g. SIP/line1, SIP/line2, SIP/line3, etc...), on the multiple gateways( I uses the SIPURA3000). I wants to arrange them into an ordered hunting group for outbound calling. I used http://www.voip-info.org/wiki/view/hunt-dial+macro for reference. My configure files like blew.

Hi, I am trying to install passenger 3.0 for apache2 on Ubuntu 10.10. It says that I need to install libopenssl-ruby. But when I did ''sudo apt-get install libopenssl-ruby'', I got the following error. Reading package lists... Done Building dependency tree Reading state information... Done Note, selecting ''libruby'' instead of