search for: schochet

...p users mailing list that he sent this to. <sccp mailing list> 2006/2/28, picciuX In fact: the one you mention is not a config file; it is part of the "Locale-Installer for Cisco Call Manager". You need a valid service contract to download it. Sorry... picciuX 2006/2/28, Schochet, Wes <wes.schochet@selectcomfort.com>: I know that's true of firmware, there seems to be a lot of XML config file examples out there on just about every web site you find. of course, not these particular one that I am looking for.... -------------------------------------------------...

...receiving 1NXXNXXXXXX. Anyone have any suggestions of how to accomplish this? I can't get the soft phones to send the DTMF (the other digits go down the d-channel of our PRI). I also have not bee able to get the dial or the outgoing queue command to work. Anyone run into this? Wesley A. Schochet Senior Telecommunications Engineer Select Comfort Corporation *763-551-7757 *651-592-5441 <**************************************> *wes.schochet@selectcomfort.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-...

I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I make this thing into MGCP gateway or even a SIP gateway for asterisk? Seems like it should bee useful for something! I'm perfectly happy to do my homework, but also don't feel thee need to reinvent the wheel! So, links with relevant info would be appreciated. If there is a config for a 2621 being used as a gateway

Hello, here is the SIP-log from my VOIP-phone when getting an MWI message from asterisk: NOTIFY sip:105@192.168.10.11:2054;line=g2kiz8tz SIP/2.0 Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK33f02076 From: "asterisk" <sip:asterisk@192.168.0.1>;tag=as229cbc7c To: <sip:105@192.168.10.11:2054;line=g2kiz8tz> Contact: <sip:asterisk@192.168.10.1> Call-ID:

...an't figure out why. I do have a dialplan.xml file: <DIALTEMPLATE> <TEMPLATE MATCH="*" Timeout="5"/> <TEMPLATE MATCH="#..." Timeout="5"/> </DIALTEMPLATE> Anyone have any insights? Thanks Wesley A. Schochet Senior Telecommunications Engineer Select Comfort Corporation *763-551-7757 *651-592-5441 <**************************************> *wes.schochet@selectcomfort.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-...

Hi All- I am trying to create a post call survey application. I would like to: 1. ask the caller if they want to take a survey after their call completes 2. If no, just transfer the call 3. if yes, 4. bridge up another extension 5. wait for that extension to hang-up 6. have the system (not the user) transfer the call to different extension that administers an IVR based survey. Anyone

...; Mawhorr (1995)", "Hackler & Hagan (1974)", "Sadd, Kotkin, & Freidman (1983)", "New York State Division for Youth (1972)", "Spergel (2005)", "Bloom et al. (1997)", "Quigley et al. (1999)", "Thambidurai (1980", "Schochet et al. (2001)", "Gruenewald, Laurence, & West (1985)", "Cave et al. (1993)", "Elliot & Knowles (1976)", "Elliot & Knowles (1976)", "Porter (1967)", "Kawaguchi (1975)", "Minnesota Governor’s Commission (1973)", &q...

...o return. Is there like an error handler / trap type routine that I can use in a dialplan? I am going to try and dig through the source code of the dial command, but there has got to be a better way... Has anyone dealt with this before? Thanks in advance, Wes -----Original Message----- From: Schochet, Wes Sent: Friday, January 13, 2006 2:40 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Bridging app OK - this is great! However, I'm showing my lack of depth / newness here. Calls from internal SIP phones work perfectly. Calls from e...

We've been trying unsuccessfully to connect our Meridian Option 81C to a TE110P via PRI. We've followed the directions in asterisk-meridian-a1.pdf (link on http://www.voip-info.org/wiki/view/Asterisk+legacy+integration), but it doesn't seem to work on our 81C (even though many, many users report it works very well on Option 11's). Has anyone had any success in getting the above

I've been beating myself up over this script but clearly I'm missing something. If I enter an extension like 101 it rings through fine, but if I pick 2 for sales it hangs up with this message: == Spawn extension (sales, s, 1) exited non-zero on `Zap/1-1' Since I'm not sure what that exacly means I cannot take appropriate action. Any help would be appreciated. [default]

Is there a good Web Conferencing add-on or a compatible package for Asterisk? I know there are web based controls for the audio, but I am looking for PowerPoint or desktop sharing functionality similar to WebEx. Anyone using a package they like?

I have several instances where conference calls are not being torn down appropriately. My CDR shows 3000 minute calls, which are coming in on PRI. I know that the calls aren't really lasting that long. What could be causing this? IN fact, here is what shows now: asterisk*CLI> meetme Conf Num Parties Marked Activity Creation 138 0000 N/A

But for us? _____ From: William Boehlke [mailto:william.boehlke@signate.com] Sent: Wednesday, January 11, 2006 2:24 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Second edition of my * book has been released $39.95 retail. _____ From: asterisk-users-bounces@lists.digium.com