search for: tesa

...phones 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

...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

...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

...e found in DNS Product: Portable OpenSSH Version: 8.7p1 Hardware: All OS: All Status: NEW Severity: normal Priority: P5 Component: ssh Assignee: unassigned-bugs at mindrot.org Reporter: lukastesar03 at gmail.com This bug was already reported back in 2022 in the openssh-unix-dev ML[1] with no response. Basically the OpenSSH client is not compliant with RFC4255 in the way it checks the SSHFP records. > "If the algorithm and fingerprint of the key received from the SSH server match...

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)

Hi list, I set up two asterisk servers , 1001 is the first asterisk server's sip user, and 2001 is the second asterisk server's sip user. Each of them work well, but I don't konw how to connect them. I want to let the user 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

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

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: >

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: