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Hello Haki I fixed this problem following the instructions in /usr/src/zaptel-1.0.9/README.udev. Regards Cec?lia -----Mensagem original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]Em nome de Erdem HAKI Enviada em: sexta-feira, 15 de julho de 2005 05:11...

Does any of you know where I can find an explanation of lapack errors codes? I get error code 17 when using optim(). Is there a way to handle errors in R such that depending on the type of error I can decide what to do next? Thanks, Haky

...ls from 2 softphones to anotherone, second caller listens " the person have extension xxxx is on the phone ...." . So we couldn't make two or more calls at the same time for a SoftPhone. What should we do to make several calls at the same time? Thanks for your interest. Erdem HAKI - erdemh@tesas.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050620/6649b28d/attachment.htm

...users (so i want to use asterisk as a signaling proxy and bypass RTP sessions)? I used "canreinvite=yes" but it didn't work. Description from asterisk conf. File; (canreinvite=yes ; allow RTP voice traffic to bypass Asterisk) Thanks Erdem HAKI - erdemh@tesas.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050624/4633b33c/attachment.htm

Hello, I'm trying Meet Me Feature. I read wiki , searched google and i configured my extension.conf and meetme.conf. But I receive "this is not a valid conference number, please try again" message, so what could be the problem? Thanks for your interest. Erdem HAKI -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050715/b2916d2a/attachment.htm

...ser in 1th Asterisk can call the user in 2nd Asterisk. First asterisk server ip : 192.168.3.101 Second asterisk server ip : 192.168.3.102 can someone give me some ideas about how to write this configuration in asterisk config files and which conf file should i use? Thanks, Erdem HAKI -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051005/d05f1f40/attachment.htm

...sterisk 902123645789-------------x not registered to Asterisk *So first number want to make a call second one (desired situation) 902121234567----------------------------> Asterisk --> Trunk Thanks for your interest. Erdem HAKI - erdemh@tesas.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050622/0d5ccab6/attachment.htm

Hi, Is there anybody who has a working video conference config? I use Asterisk@HOME 1.0.7a , I couldn't use Video Conferencing feature of eyeBeam. Thanks Erdem HAKI -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050526/3527af70/attachment.htm

--- listas iPfone <listas@ipfone.com.br> wrote: > Hi all > > I need to know if the video support for h.263 is > active in version stable > 1.0.7 to use with eyeBeam in asterisk it works for me... [2222] type=friend secret=xxxx auth=md5 callerid="myCallerId" <2222> canreinvite=no host=dynamic disallow=all context=default allow=alaw allow=ulaw allow=speex

Hello, I'm setting up AST Post Paid application, is there anybody who set up astpp ? I followed the directions, i visited the astpp admin page in my web browser. But i couldn't setup the brands and routes etc. "Database unavailable -- please check configuration" appeared on the top of the page, so i went to "configure" section, I filled in the blanks according to my

so what should "astpp db" be exactly, where can i find its name? what should i write there? Thanks again.. > The Database field should contain the name of the astpp db, something > along the lines of "astpp" is what I would put in there. Here is a fixed > version of the script. It did not post properly to the wiki: >

Hi all of you. I am trying to configure voice mail in asterisk and i am facing problems. I have found following warning message in /var/log/asterisk/messages -------------- No application 'Voicemail' for extension (macro-mainmenu, s, 5) I have configured voice mail accordingly in extention.conf [headoffice] -- ------------ ------------- exten => _63,1,Macro(mainmenu)