search for: dwildes

...n't need to page across multiple Asterisk servers but if you did the software wuld need to be smart enough to "know" which groups of extensions could be in a multicast and whci need to be bridged. Basically check to see if the SIP phone are on the same subnet. --- DUSTIN WILDES <dwildes@pabbankshares.com> wrote: > I feel this needs to be a separate application in Asterisk, like > app_sipintercom > The application would connect to all available auto-answer SIP > phones, play a short frequency tone for the intercom alert, only > allow one-way streaming to the phone...

Does anyone have the .bin file(s) to convert a Cisco 7940/7960 back to the CallManager image? I want to start playing around with the chan_skinny addition, but it seems the .exe's from cisco want to open a connection to a SQL server or CallManager (which I don't have).

Is there a way to get Music-on-Hold to use the Line input of the soundcard instead of playing MP3s? Some potential clients like the music-on-hold to be the local radio station, or from a satellite broadcast. Thanks!

I thought I would give RedHat 9 a try with Asterisk..I thought it would be a good idea to use the latest version.. Zaptel, Zapata and Libpri all appear to have compiled sucessfully.. But.. (Why is there always a but??) It seems Asterisk is having issues with 'termcap' or 'tgetent' whatever that is.. Here is the output from 'make install'.. --------Start-------- if [ -d

Hello, I'm new to Asterisk and would like to know SIP support status. Are there any testing been done with some widely deployed client (Cisco SIP IP phone, ...)? I was using Vocal but I'm now interested in Asterisk as it seems to offer more features...if it supports SIP. Thanks for your help. Francois.

I'm setting up a T1 for both voice and data and was wanting to know the steps involved. I've recompiled zaptel for ZAPATA_NET support and setup my /etc/zaptel.conf: Fxsks=1-10 Nethdlc=11-24 Modprobe wct1xxp works fine and my voice lines work. Now when I use the provided sethdlc, I use the following commands: Sethdlc hdlc0 mode ansi Ifconfig hdlc0 <ipaddress> <network>

I got it figured out now. In case anyone else comes across it - this is what I had to do: Sethdlc hdlc0 mode ansi Ifconfig hdlc0 up ***NO IP INFO SET*** Sethdlc hdlc0 create <dlci number> Ifconfig pvc0 <local ip> pointopoint <remote ip> up Then set your default gateway to the <remote ip> listed in pvc0. That's about it!! Hope it helps! -----Original

Newt is a separate library from asterisk that is used to easily create a ncurses based program. If you want to see some examples of how to use newt, look under the 'zaptel' source for zttool.c and/or under the asterisk/astman source for astman.c Newt pretty easy to code off of and can make a quick & easy frontend for things. Check out the /usr/share/doc/<newt directory> for more

Did anyone have a way to make codec negotiation work with Asterisk? This is something I would love to have working as well. I won't need PSTN -> G729 mixing. Just SIP -> SIP using G729 for calling remote offices via VPN, but everything else use G711. -----Original Message----- From: Brancaleoni Matteo [mailto:mbrancaleoni@espia.it] Sent: Wednesday, July 16, 2003 11:32 AM To:

I've been digging around with some cisco engineers for about a week & I finally got an encouraging response to the Auto-Answer issue with the SIP Phones. Here is their reply: =============== == FROM CISCO == =============== Auto-Answer feature is introduced in SIP IP Phone 6.0 version. This software version is expected to be available for customers shortly. Please let me know if you

Hey guys - hate to beg, but my Cisco ID has expired (yes - I'm renewing) and I can't get the latest ver 6.0 image for my SIP Phones - could anyone send me the .scp & .bin? Of course this email never happened! :-) Thanks!!

Hello all! I have an FXO port on a cisco router that is directly connected to our PBX. Our ATA-186 (firmware version 3) registers with asterisk, which connects to our cisco router's fxo port to give me a dialtone on our PBX from the ATA. How do I pass the flash button to the PBX? It seems the ATA-186 wants to control the flash by putting my first call on hold and prompts me to dial another

Hi all!! Does anyone have a short example or even better - a working AGI script that uses "GET VARIABLE' from a /var/spool/asterisk/outgoing call that uses "SetVar"? Here's what I've tried with no luck so far: sample.call ================= Channel: SIP/1000 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Application: Agi Data: playTasks.agi Callerid: Nightly Processor

This is Off-Topic for Asterisk, but I wanted to get some feedback on headsets for Cisco 7940/7960 phones. We have about 10-20 people who wants/needs a headset for their phone & was hoping to collect some real-world input. Thanks!!

I would like to add the ability for our users to be able to press "0" whenever reaching someone's voicemail box to re-reroute them to the auto-attendant. Here's a sample extensions.conf: [incoming] include => ciscophones exten => s,1,Wait,1 exten => s,2,Answer exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,15 exten => s,5,BackGround(auto-greeting)

I run a SIP-only shop with a 23 channel PRI and single T100P. Here are my configs: /etc/zaptel.conf: span=1,0,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us /etc/asterisk/zapata.conf [channels] context=default switchtype=dms100 signalling=pri_cpe pridialplan=unknown rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=no hidecallerid=no callwaiting=no

Does anyone have any recommendations on implementing Answering Machine detection for call generation programs? What I would like is * to determine what picks up the other line (Answering Machine, Voicemail, or Human) to determine which action to take. For example: If * detects Answering Machine or Voicemail, hangup call & the AGI will log (ANSWERING MACHINE DETECTED) and at that point,