search for: iske

Hi, I am very happy. My problems are solved without one little thing: (Iske <- matrix(c(1, 1, 1, 2, 2, 2, 1, 1, 1, 5, 1, 2, 2, 2, 1, 1, 1, 4, 4, 4, 4, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 5, 4, 4, 4, 4, 4, 4, 2, 2, 2, 2, 4, 5, 5, 5, 5, 5, 5, 5, 5, 5, 1, 2), ncol = 5)) #My Matrix Iske<- Iske+33 #I want see the letters (Iske.char<-apply(Iske, 1, function(x) rawToChar(...

Hello, I'd appreciate your comments on the following problem I'm having, please forgive me if this is something obvious, I've been scratching my head on this for a while: I have Asterisk+Kamailio setup where I'm currently testing inbound calls from outside. I have both webrtc and sip clients, where webrtc peers are defined according to sip.js instructions (

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Hi All, I am trying to get Asterisk up and running on my new Mandrake 9.1 install. I've installed Linux in the "standard" mandrake security mode, and "su" to do my attempts at install. I managed to obtain the source from CVS, and have been able to compile Zaptel. I then ran insmod zaptel, and also make config. I think I have compiled and loaded Zaptel successfully as

>There is no real answer to your question. >just use one you're most familiar with. Second that, using FC2 for me, and it's the Devil You Know, right? I have a good handle on the RedHat Way which the Debian guys say is stupid, but it's not, it's just different. All of the little RedHat-y things you get used to and when you use a different distro, it's like, WTF?? I like

Me too, with 3X Snom 360's in CAP positions. I just gave up in the end and cron'd a SIP REBOOT from my * box at strategic times. If your phone supports the REBOOT directive I'd just reboot your phone every day, 4 AM or what have you. -----Original Message----- From: Bartosz Jozwiak [mailto:bartek@cq-link.sr] Sent: Monday, April 24, 2006 12:35 PM To: Asterisk Users Mailing List -

Hello all, Some time before Christmas, a project was started. The task was to enhance an existing LiveCD solution with User-mode Linux. A number of requirements was stated, where the greatest challenge was to fit 10 virtual routers within an old no-name PC with only 256MB Ram. And the solution must still be run from a LiveCD. Thanks to Xen, the task was possible to complete. -But, what does it

On 05/12/14 16:46, Olli Heiskanen wrote: > INVITE that Asterisk (at port 5070) receives: > PU.BL.IC.IP:5060 > PU.BL.IC.IP:5070: SIP, length: 1046 > INVITE sip:660 at testers.com > <mailto:sip%3A660 at testers.com>;transport=UDP SIP/2.0 > Record-Route: <sip:PU.BL.IC.IP;lr=on;ftag=41030177> > Via: SIP/2.0/UDP >

I've tried the precompiled version of gastman, but it does'n work properly under windows, so I would like to try to compile it under windows, so maybe it begins to work. I've tried to compile gastman with mingw, but the db3.1 libraries request the cygwin include files and libraries. Then I've copied the cygwin's include files and libraries to mingw's directory, but the make

I download app_valetparking.c from http://www.loligo.com/asterisk/misc/apps/app_valetparking.c and followed the directions on http://www.loligo.com/asterisk/misc/apps/app_valetparking.README I am using asterisk-1.0.0 any suggestions [root@localhost asterisk]# astxs -install apps/app_valetparking.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat ions -g -Iinclude

Very friggen cool, that you very much for the information it looks like it will do the job nicely. What did you use in your extensions list to activate the relay? James > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jay Milk > Sent: Sunday, 20 February 2005 6:24 PM > To: 'Asterisk

I'm trying to connect my Asterisk server via sip using my vonage soft phone account. Has any anyone successfully got to work? I get error from asterisk saying: == Parsing '/etc/asterisk/sip.conf': == Parsing '/etc/asterisk/sip.conf': Found Aug 24 11:01:11 WARNING[1125329600]: acl.c:146 ast_get_ip: Unable to lookup '216.115.25.199:5061' when trying to register with

Hello, I hate to ask here, but.. Does anyone know of an IAX/SIP DID provider in Vancouver, British Columbia? I'm looking for a voicepulse isk service, one DID with standard calling features and some sort of long distance package. I've looked around on voip-info.org's list of VoIP providers but so far I haven't found one that offers a 604/778 number. Thanks Matthew

Is it possible to modify CDR variables before insert into MySQL or CVS? how? thanks all; Alex -------------- next part -------------- A non-text attachment was scrubbed... Name: alexm.vcf Type: text/x-vcard Size: 334 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060116/5a095cdc/alexm.vcf

Can anyone tell me the features of the Cisco 7905 with SIP? I mean things like number of lines, speakerphone, transfer buttons, etc. I've seen the Cisco material, but all it told me was how nifty it is and how wonderful the XML interface will be ;) Thanks, --Ernest

Hi All, I successfully installed a 32bit W2003 domu using the following config file (as the CD image wasn''t persistent between reboots, otherwise I would have used virt-install ;-) ) : isk = [ ''file:/images/xvm/w2003/w2003-disk.raw,hdc,w'',''file:/images/isos/en_win_srv_2003_r2_enterprise_cd1. iso,hda:cdrom,r'' ] memory = 512 name = "w2003"

The chan_sccp page at http://www.voip-info.org/tiki-index.php?page=chan_sccp2 has been updated. See the bottom of the page. Thanks. Comments welcome. -- respectfully, Joseph =============== ---------------------= ********** =

Hello Users, We have Thirdlane Multi tenant PBX system in production. Asterisk version is 1.6.2.15. Attendant transfer is working, but blind transfer is not working with Grandstream (gxp-2000) phone. We have read from google that it is a bug in Asterisk 1.6.2.15. We saw the below links: <http://www.odesk.com/leaving_odesk.php?ref=http%253A%252F%252Fwww.freepbx.o

Hi people, what is the current status of send/receive fax on asterisk extensions, i dont want to receive the fax and send an email or viceversa, i want to connect a standard fax machine to a Linksys' ATA (FXS RJ11 port) . Webdoc?, pointers? Thanks

Hello all, I noticed that Cisco claims that you can do station to station intercom with the 7940/7960 phones, and the Cisco Call Manager. Does anyone have an example SIP header that shows this in action? Or is it something else that triggers the intercom? I would like to add this in to *. Thanks, Mike