similar to: fallback password mechanism ?

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Hi All, This is more of a hardware question that an Asterisk question so I hope this is still the correct place for the post. I know with the Linksys phones you can create a hotline by using the dial string of (S0<:number>). I have been trying to do this with a PolyCom phone but I have not been very successful. Does anyone know how to create a hotline phone with a PolyCom?

hello.... I'm new to wine and tested and installed it, because I wanted to run the hotline server from windows under linux... upload, news works great but download from this maschine is not working! strange thing does anybody has the hotline server runing? do you have had similar problems with another network app? this are the errors.... imted from C:\windows\system\wininet.dll, setting

I will be out of the office starting 12/09/2005 and will not return until 16/09/2005. Dear Sir / Mdm, I'm currently on course and are not in office. During this period of time, I have minimal access to internet and email cccess. As such, I might not be able to reply to your queries promptly. I apologise for the inconvenience caused. In the meantime, for any technical assitance, please

How can I make it more readable? Name/username 601/601 123456789/123456789 voipbuster/abcd 601 = hotline 123456789 = Peter Pan only voipbuster/abcd is easy read/understandable! bye Ronald Wiplinger

I have two phones (111 and 112) on a LAN, and I have on a users site a phone 333. phone 111 uses sip.conf, while 112 uses real-time set-up. 111 can call 333 AND the audio is working 112 can call 333 but audio is just white noise. 333 can call 111 or 112 and audio is working. The phones are identically set-up (just user name = phone number and password are different) sip.conf (for 111 - all

Hi, I have the problem that WaitExten is not responding to key presses. Here are the sections from my extensions.conf: [globals] incoming_call=0 menu=0 announce=0 [internal] exten => 777,1,Goto(hotline,${EXTEN},1) [hotline] exten => _X.,1,Set(CALLERID(name)=Hotline) exten => _X.,n,Set(original_extension=${EXTEN}) exten => _X.,n,GotoIf($[${announce}=1]?4:10) exten =>

Dear R-users! I am using R 1.7.0, under Windows XP; I also have Internet Explorer 6.0.2600.0000, Norton AntiVirus 7.60.926. Our firewall seems to want to "protect" me from downloading precompiled packages for Windows. When I try to download packages, like http://cran.at.r-project.org/bin/windows/contrib/1.7/RODBC_1.0-3.zip for example, I am not allowed to and get a message like this:

https://bugzilla.mindrot.org/show_bug.cgi?id=2011 Damien Miller <djm at mindrot.org> changed: What |Removed |Added ---------------------------------------------------------------------------- Status|NEW |RESOLVED Resolution|--- |FIXED Blocks| |1986

https://bugzilla.mindrot.org/show_bug.cgi?id=2011 Damien Miller <djm at mindrot.org> changed: What |Removed |Added ---------------------------------------------------------------------------- Status|RESOLVED |CLOSED --- Comment #14 from Damien Miller <djm at mindrot.org> --- Set all RESOLVED bugs to CLOSED with

https://bugzilla.mindrot.org/show_bug.cgi?id=2011 --- Comment #8 from Petr Lautrbach <plautrba at redhat.com> --- Created attachment 2214 --> https://bugzilla.mindrot.org/attachment.cgi?id=2214&action=edit don't probe seccomp capability of running kernel in configure I'd like to add also possibility to build seccomp_filter sandbox on system with older kernel, E.g. Fedora

Hi, I have a server with 2 TE110P cards. 1 card is plugged to telco line, another card is plugged with a Hicom PBX. I want to send some call to VoIP phones and all other to my PBX. I don't known how to make my dialplan : ===========Extensions.conf========== [incoming_call] exten => 090200000,1,Goto(callcenter,100,1) exten => 022956353,1,Goto(callcenter,100,1) exten =>

Hi, I'm having trouble getting an answer to the following problem on -questions - I hope someone here has done something similar and can help. I'd like to compile support for FreeBSD OPIE into sshd. Presently I have to use PAM to achieve one-time password support. On a 4.x system I have in /etc/ssh/sshd_config ChallengeResponseAuthentication yes and in /etc/pam.conf sshd auth

https://bugzilla.mindrot.org/show_bug.cgi?id=2011 Bug #: 2011 Summary: sandbox selection needs some kind of fallback mechanism Classification: Unclassified Product: Portable OpenSSH Version: 6.0p1 Platform: All OS/Version: All Status: NEW Severity: normal Priority: P2

I am seeking to implement the following behavor: When a headset on phone1 is picked up, phone2 rings right away, without any need to dial numbers on phone1. Is this possible to implement? ScriptHead -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060120/0892441d/attachment.htm

Sorry for reposting but as I got no reply so far so I am just coming to try my luck again. I am doing a multi-lingual website which might have several layout pages (language dependent)available. I want to do a mechanism to fall back from one language to another default language if such layout file is missing. Any chance I can do an is_exist check on the layout? With my current code I really want

Is there any way to set up a backup WINS specification in dhcp/Win NT/Samba? Having my WINS server go down the last few days made this a little more desirable, especially since when I noticed the service was dead, I wasn't able to reboot it for a few hours due to usage. Thanks, Bill -- Bill Eldridge Radio Free Asia bill@rfa.org

Hi ! I'm just wondering if it is possible to control asterisk in a dynamic way, such as, for example, signal to asterisk to give info about callers that are queued ( like in a hotline, everybody is queued until somebody take their call), and explicitly signal asterisks to redirect a particular call to a computer through VoIP ( imagine a hotliner choosing a call from a list of waiting calls

I would like the ability for a sipura sip device to instantly connect to an asterisk server as soon as the sipura sip device goes offhook and before any digits are pressed. This way asterisk can provide the dialtone and the dialplan. This also allows me to play a greeting to the phone before giving them a dialtone. Is there any way to do this, like possibly having the sipura device dial a

Hi All, We have clients which dynamic IPs which we have setup with <host>.dyndns.org addresses. We have added these to the rsync.conf 'hosts allow=" but they are being denied access.. Is there anyway around this? Matt