similar to: asterisk compatible, hot swappable PRI card

Displaying 20 results from an estimated 6000 matches similar to: "asterisk compatible, hot swappable PRI card"

2005 May 20
3
Help with follow me
I hope someone can help me with this. This is what I want to happen. Someone dials in and goes to my extension. First, the phone on my desk rings If there is not an answer, I would like to have the dialplan call my cell phone. If I answer my cell phone, speak the incomming number to me. I press one of the buttons on my cell phone to accept the call. If I don't answer, or I don't
2005 Jan 12
2
Trouble building appradius
I am having trouble building appradius from http://appradius.minitelecom.org/ I configure, make, make install cpprad-1.0, but when I configure, then make appradius I get :- obelix:/usr/src/appradius/appradius1.0 # make make[1]: Entering directory `/usr/src/appradius/appradius1.0/lib' make[1]: Nothing to be done for `all'. make[1]: Leaving directory
2005 Jun 09
5
Voicemail and MS Exchange
> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > George Pajari > Sent: Thursday, June 09, 2005 10:19 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Voicemail and MS Exchange Synchronization > > > We have a customer considering
2005 Mar 08
4
force SIP authentication
Hello, is it possible with Asterisk to force SIP authentication? Right now, it seesm that just any SIP client can at least connect to my PBX, which I don't want. I want users to authenticate with username and password and otherwise deny them access. Thanks Florian
2005 Feb 20
2
How many line appearance can Snom 200 handle?
Snom 200 has be set up with extended key pad. The product literature also mention multiple sip registration. How many registration can it handle? It does not seem to appear in the user manual. David Kwok
2005 Feb 25
1
SIP Errors
Can someone explain what this error is? -- Got SIP response 500 "Server Internal Error - Invalid CSEQ number" back from 209.xxx.xxx.xxx How do I fix this? .o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office
2005 Feb 25
1
Asterisk and 723,729
has any one implemented asterisk with 723 and 729 codecs, what is the cheapest way. is there a limitation in the open 723 implementation ?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050225/d5daf369/attachment.htm
2005 Mar 21
2
Compiling with gcc -shared on OS X
Hey all again, I have successfully compiled and am running Asterisk (stable release) on OS X (10.3). However, any make directive that uses the "-shared" option in gcc results in an error. Apple states that -shared is not supported under OS X. Is there a workaround or do I have just have to live life without those modules (zaptel, libpri, format_mp3, probably others)? Thanks, Zach
2005 May 20
1
How can you keep agents logged in across a restart?
The persistentmembers=yes is suppose to keep agents in a queue over a restart. It might do this, but it doesn't do much good as even if they all remain in the queue, they are all logged out on a restart. Is there any way to keep the agents that are logged in, logged in across a restart? Thanks, Jon.
2005 Jun 10
1
Re: Voicemail and MS Exchange Synchronizatio n
> -----Original Message----- > From: Iassen Hristov [mailto:ih.ng@databrokers.net] Dumb, hacky idea...but just so crazy it might work: Have Asterisk include a read receipt request when sending the voice mail message. Write a script, triggered from a sendmail alias or .forward file, that will parse the incoming receipts and handle the message deletion. Bonus points: When someone listens
2005 Sep 09
2
"Registered SIP '202' ... expires 1800". Why does it expire
Hi, When a SIP client registers on Asterisk server, why it expires after certain amount of time?
2005 May 31
3
Opinions of Sphinx?
I'm planning a system of 120 SIP or PRI channels using speech recognition (fixed grammar of 500 words) menus. I could use a Cisco router and VoiceXML, but would prefer not to on cost grounds. Has anyone tried Asterisk and Sphinx (bonus points if in a production environment)? If so, what's your opinion on quality of recognition, stability, resource usage, etc? Anyone have any
2009 Nov 10
4
RAIDs and JBOD?
Hey Guys, I have some questions?regarding?a new home server I am going to build in the hopefully very near future (ASAP, I just need to finish planning everything and this is the penultimate?hurdle), I will be creating a software RAID... Lets say I have three drives "knocking" around which are all 1TB SATA II drives but each made by a different manufacturer. I am going to guess that
2005 Jan 11
3
requiring logon for SIP users
Hello there, I am playing around with Asterisk the first time and it really looks great. ;-) However, I have one problem: Any SIP device can connect to my PBX. How can I requre logon for SIP users and deny access in the case of wrong or missing credentials? Thanks Florian
2004 Dec 14
2
SIP registrations not staying registered
Hi, I have several SIP registrations on my Asterisk box. Sometimes, I try to call in the inbound number from 1 and find it doesn't work. When I do sip show registry, it's showing Unregistered (and sometimes there are several which are showing Unregistered). If I type reload, it registers and works fine straight away. Is there something I can set to keep registrations connected all the
2005 Feb 17
1
(Kphone) Registration Failed: Forbidden
I just can't get kphone to register with asterisk, i can make calls to the demos and even get into the mailbox but kphone cannot register. Here's my story. Can you help me?? Please I have installed asterisk on debian using apt-get install asterisk. I have configured an extension in extensions.conf as follows exten => 8003,1,Dial(Sip/8003,${RINGTIME},rt) exten =>
2005 Feb 25
1
Seting up for afirst time -- can not call
Hi, all I am setting up Asterisk for the first time and have some problems. Setup is very simple -- Astersik box and two Polycom SP300 phones. I will add bells and whistles as I go, at the moment things are very simple. No TFTP servers, so phones run with their default configuration. I set up IP addresses, netmask and gateway IPs manually on the phones. Now, I have read of problems with
2005 Mar 09
3
voicepulse "silence" during conversations
Hi all, I'm running Asterisk 1.0.0. I am a customer ( and supporter ) of voicepulse. For me, it works perfectly, but one of my customers noticed a small problem: During a conversation, when the otherside isn't talking, it's almost like the mic turns off. Not that big of a deal I know, and the more I think about it, the more this seems a voicepulse issue. But in the off
2005 Mar 03
4
[OT] - Why should I answer a Newbie question, therethick!
If you really want to do this the asterisk list is based off of mailman. You can learn all about mailman here: http://list.org/ But really, what are the odds that newbs will know to go there first? Are you going to moderate it? Someone has to actually answer the questions you know, if a newb only list is going to exist. Look, don't answer lame questions if you don't want to. Flaming a
2005 Feb 24
2
OT - C structure question
I hae tried searching the web for the answer, but, man is there a lot of pages ... :( in the language I develop in, if I have a structure I can dynamically refer to the contents of a field of the structure like so: MESSAGE SomeStructure:Field(SomeFieldName):Value where SomeFieldName is either a quoted constant or a variable expressions In "C", I beleive that you can refer to