similar to: SIP registrations not staying registered

Hi, When a SIP client registers on Asterisk server, why it expires after certain amount of time?

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

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

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

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.

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

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

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 =>

> -----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

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

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

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

> -----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

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

Hi We are in a project where we will use asterisk as a residential gateway for IP phone service. We are aiming to replace the primary phone line so the service must be up as long as possible so we are looking at ways to avoid shut downs. We are looking for a solution to allow us to add/remove PRI cards without shutting down the system Is there such a thing as an asterisk compatible

Hi, I ve had some problems printing from NT, I had samba 2.0.3 and every time at the end of document there is a blank paper and I don't know how I can stop this ??? Regards, and thanks in advance for any help Amin.

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

Hello, ** Please CC replies to ian@ichilton.co.uk as I am not subscribed ** I have a Brother laser printer connected to a linux box running Samba. It is working fine from Win98 workstations, but now I have Win2K on my laptop, I am unable to print. This is because the linux box obviously doesn't have the win2k driver so I tried to install it as a local printer and then change it to a network

Hello, ** Please reply direct to ian@ichilton.co.uk as I am only subscribed to the list in digest mode ** I am trying to compile Samba 2.0.7 on my SGI Indy running IRIX 6.5 I did a: ./configure --with-smbmount (I want smbmount) then, after an hour compiling, it gave this: Compiling client/smbmount.c client/smbmount.c:27: asm/types.h: No such file or directory client/smbmount.c:28:

Hello, I have a problem with an LAN-Server behind an NAT-router. Asterisk Version 1.2.1 or 1.2.2 doesnt matter 10 minutes after starting Asterisk I loose all registrations at external SIP-proxys. The reason seemed to be that Asterisk send every second an request to every sip-proxy "Request: OPTIONS sip:sip.domain.tld". Every request is responded by the sip-proxy. After some minutes