similar to: Best practise for ISDN Video Conferencing..

Hi, Is there a variable that contains the Sip Peer name? I was using ${CALLERID(num)} for outgoing calls, but when a call is being transferred, that variable contains something else. I need a variable that is always set to the SIP Peer's name. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

I received this email link from Macromedia today http://www.macromedia.com/newsletters/edge/february2005/index.html?secti onIndex=3&trackingid=AWQX They are now selling Breeze video conferencing server by the minute as an asp, this service is powered by the macromedia communications server which is just a sip server like asterisk but for video. I'm still offering a $2,000 bounty

hi, If i try a backup with tar / tape sdlt320 on a debian-sarge / 2.6.12 based XEN 3 System, the System freeze. If i test this by knoppix the backup works fine. Test with xen 2.07, Kernel 2.6.11.12 on xen 3.0.0, Kernel 2.6.12 This System in a Interserver with an LSI scsi-Controller and a Tandberg SDLT320 Streamer. Are thereany known issues with this config and xen ? regards Rene

I'm trying to use ZapRAS to enable ppp connection through my E1. After the ZapRAS command is executed, all sound is crappy on all lines! The only solution is to reboot the machine (or halt it, and then power it on since Digium's hardware doesn't like reboots). Anyone know how this can happen?! I'm using * 1.0.6 on Dell PowerEdge 1850 which are told (too late though) not to work

Hi, I've a problem configuring my Asterisk. What I try to reach is to interconnect a Tandberg Visioconference (SIP) world with my Asterisk (SIP) with 1 constraint I can't change : "every RTP flow needs to pass THROUGH Asterisk, and are NOT nated" What I observe : - a call made from a SIP Phone registred in Asterisk to Tandberg works (voice and video bidirectionnal) - a call

Hi, I'm trying to get ZapRAS working but not getting very far.. Asterisk CLI shows: WARNING[3355]: app_zapras.c:173 run_ras: wait4 returned -1: No child processes and /var/log/messages shows: using the plugin option requires root privilege Can anyone shed any light on this and any fix? Googling the error doesn't find much.. I'm not sure what 'plugin' it is talking about,

hi all Where can I find documentation about how to setup ZapRAS? What I want to do (optimally) is to allow for automatic dial-up to external sites, each having an ISDN router. Today we use a small ISDN router for this, but it'd be a lot better, IMHO, to have asterisk do this (functioning as a ISDN router), as we may cancel our BRIs then. Is this possible? And if so, how can I do it? I

We have here a TandBerg "videoconferencing system" connected in asterisk with a Beronet card BN4S0 (4 BRI ports). I`m trying to make a videoconferece (video + audio) with this Tandberg to another Tandberg using the ISDN channels through the BN4S0 BRI Card. But, i'm only obtaining audio calls, the video not appears. When i try to make a 64K calls (data calls), the call does not

Has anyone had any luck getting Asterisk 1.6.2.13 to register to a Tandberg Gatekeeper? The logs on the Asterisk end seem to show that the registration request is sent, and the Tandberg Gatekeeper responds. However, the response doesn't seem to be what Asterisk was expecting. Here is my ooh323.conf, followed by the relevant portion of the h323_log: [general] port = 1720 bindaddr =

My need to do this through asterisk is simply the ability to provide me access with no additional cost to my customer. It seems like a nice thing to include as long as authentication is done well. I have worked on a dozen or more types of switches and all of them have supported this or had the capabilty through hardware or licensing. I am trying to get around opening and closing the firewall,

hi! I'm trying to get zapras working in GSM csd network. Whenever a dialup call is initiated from the mobile to the * gateway the following appears in the log and zapras terminates. Phone gives the error dialup not answered. ==> /var/log/messages <== pppd[2310]: Plugin zaptel.so loaded. pppd[2310]: Zaptel Plugin Initialized pppd[2310]: Using zaptel device 'stdin' pppd[2310]:

Hello all! Sirrix AG, Saarbr?cken (manufacturer of the Sirrix.PCI4S0 4-port ISDN card for Asterisk) has released the new Asterisk dialplan command PPPD (app_pppd). It allows to connect a Linux PPP daemon to an arbitrary digital (ISDN) Asterisk channel to provide RAS dialin and dialout. The PPPD command has successfully been tested with Sirrix.PCI4S0 cards and a standard ISDN4Linux ipppd "on

Hi All, We have a Tandberg VCS System for Video conferencing and a customer running AsteriskNow (Asterisk 1.6 + FreePBX) for Audio conferencing. Problem Statement: How do we integrate the 2 systems such that Audio SIP calls are seamlessly passed between the two. Sorry we're just starting up so a bit of general advice, or a link to any document would be great! If anybody has done this -

Hello, We are working on implementing a solution for a medium service provider. They were previously using a Cisco AS5300 gateway with some PRI trunks to receive modem calls, then route them out the Internet. The Telco they were buying the trunks to discovered this configuration and restricted them due to legal conventions, and stated that in order to continue doing this, they would have to talk

Hi, For some time now (starting with the 3.10 or 3.11 kernels, but I'm not sure) I have the following problem: I have a machine with 6 SATA slots and two SAS controllers, one onboard HBA and one RAID controller in a PCIe slot. The problem is that the order of the SAS controllers changes randomly after a reboot, so the SCSI addresses of the devices change. One time lsscsi prints (empty

Is Asterisk capable of handling video conference? I am wondering if there is anybody in the list who tried it with NetMeeting(s). If it is possible, is the * required to register in the GK for this purpose? or making it as h323gw only is enough. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, We need to start obtaining some billing files from BT via a dial-up ISDN connection and I'm wondering if Asterisk is capable of doing this? I need to make an ISDN dial-up CHAP connection and, once connected, grab some files over FTP. Currently, our Asterisk box is connected to an ISDN30 with a Zaptel card. This may be a downright stupid question but I'm trying to find out if

Friends, SIPit is the main interoperability event for all things SIP. It's organized by the SIP Forum and creates good feedback to the IETF. Asterisk has been participating in SIPit during many years and in many variants - videocaps, Marc Blanchet's IPv6 branch and the standard Digium releases. All these tests has lead to a large amount of improvements for Asterisk and have helped us

Hello, I was wondering if it was easy to setup a dialup internet account system in Asterisk somehow(like PPP) laptop with modem -> PSTN line -> Asterisk server -> Server's broadband internet connection Has anyone done this? Thanks, MATT---

Anybody connecting to on-premise modems by dialing in to Asterisk and using an extension to reach the modem? How? Sipuras, FXO cards, other? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040629/92ee113a/attachment.htm