similar to: How can I configure to call from the console bymeans of a sip phone,

It's something like exten => 15,1,Dial(Console/DSP) -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Anthony Azzopardi Sent: Saturday, February 04, 2006 2:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] How can I configure to call from the consolebymeans of

I would like to make a suggestion and recommend that you put your Asterisk box on the outside and let it also pull duty as your firewall/nat router. The iptables overhead will be minimal on the system and you'll save yourself a lot of headaches in the long run. The biggest problem being that having an asterisk server behind a nat, and then also having sip phones trying to connect to said

Hi, Could you please help me!! I am trying to configure the Asterisk server. I have a analog phone connected to a FXS port of a Cisco 3745 router. This router is connected to a Asterisk server via Fast Ethernet interface. I am trying to make a call from the analog phone to a SIP client. This SIP client is registered to the Asterisk server. Analog phone number: 999 SIP client : 202 Sip client IP

Hello asterisk-users, I need to compile dahdi and then export it to another system. I managed to do this with DESTDIR=/root/destDir, then make a tar file and extract in / of the other system. However the module is not loading and /dev/dahdi is not created. Anyone done this? Thank you, Anthony. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is there a linux sip or iax phone that will autoanswer and connect to the console or soundcard? I found linphonec but it does not autoanswer from what I can tell. Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050916/8bc76bbb/attachment.htm

I have been told that my logic in extentions.conf is wrong in trying to configure a SIP software phone called Express Talk (remote) . I'd like to make outgoing calls and calls to local extensions. Could someone please look at my configuration files at http://pastebin.com/ajp62wqF and see what I did wrong? Thank you, Gary

Hi, How are you! I built a PXE server of MSDOS in Linux server, client can boot from network and login MSDOS, but by default, the output console is VGA, My target client has not VGA port, it can only output from serial port, so could you help to tell me how to boot a MSDOS client from network and keep the PXE client output console is serial port ? Below is the PXE configure file?s

I have some Snom 190's but the volume from is really low (speaker is ok). Is there any way this can be changed on the server? On the Snom I already set the volume to maximum. Thanks! Remco

Hi there We have a Cisco VOIP environment here, with hard and softphones. I have a softphone account/etc, but I'm a Linux user and (as far as I'm aware) there is no Cisco softphone for Linux. However I can run Asterisk. So I was wondering if there is a way to "convert" a SIP phone transaction into a SKINNY transaction so that the Cisco environment thinks it is a Cisco

How come I don't have the MeetMe application registered? Regards, Anthony.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello everyone, I'm relatively new to the subject - so pleace don't punish me for idiotic questions. ;-) Can Asterisk act like a normal Sip phone and e.g. connect to another sip-gateway? Background: There is a new german company at: http://www.sipgate.de (sorry German only page) They offer a a gateway between a real telephone number and

I've recently upgraded to R-2.0.1 on a Mac running OS X 10.3+ I am using the new Cocoa-based GUI. Everything was working well for a while. In the middle of an R session, I started "suddenly" to have a problem where code copied from an open editor window and pasted into the R Console gives a syntax error. It doesn't matter what the code is. If the same exact text is typed

I had this same problem with WINXP WinMESS, (what a name mess) I changed the Distro from Redhat 8.0 to Mandrake 9.1 and bam! It all works!! Does anyone know of a problem with this and RH 8.0???? Are you running Redhat?? I now have Messenger working fine as well as X-ten, Sipps, and some others. I have standardized on Mandrake 9.1 and asterisk seams to have NO problems. REDHat 8.0 proved as

Does this count have to be exact, or just an accurate approximation? The back-end may add/remove registers fairly late in the codegen process, so if you need an exact count you may need to run *just* before the assembly printer. Perhaps we could introduce a special machine node that represents a shared memory allocation. The node's value would be the shared address space pointer of the

OK, I have been running the UW-POP3 server which currently translates all login ids as lowercase (i.e. Anthony becomes anthony). Now the Dovecot server is very flexible and currently I have not found how to translate the upper case characters to lowercase. What configuration setting will translate the userids to lowercase? -- Albert E. Whale, CHS CISA CISSP Sr. Security, Network, Risk

Greetings! We are using cisco 7940 phone with SIP and asterisk. We would like to be able to have phone directories available on the phones that are sourced from active directory. Are their any scripts that can connect to the AD server via LDAP and then create the directory.html file for the phones? Thanks! Liz -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I have a server setup with an incomming PSTN line and a bunch of Grandstream BT100 phones. Is there a way for asterisk to foward an incomming callerID from the PSTN to the SIP phone that is setup as an extension? We have a Voice menu setup for incomming calls and I would like to recieve the caller ID of the calls we are recieving after the incomming caller reaches their final

After installing the timing source , what do I have to do to get meetme application registered? Do I have to recompile asterisk again ? I don't see the compiled meetme.so module in the directory. Regards, Sam -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kevin Bockman Sent: Friday, February 10, 2006

Is there any way you could approximate the register/instruction usage and perform live-range analysis in a higher-level LLVM IR pass? I'm not sure how useful NVPTXRegisterInfo would be anyway. Unlike backends that target "real" ISAs, these structures do not contain any special properties about registers or instructions, like cost or scheduling information. Are you trying to figure

Yes, global variables are the only way to access shared memory. I'm just trying to get an idea of what you're aiming to accomplish to see if we can improve on the interface here. A MachineFunctionPass runs after instruction selection and relying on doInitialization to run before instruction selection is an implementation detail that I do not believe is guaranteed anywhere (I could be