similar to: DTMF integration pana d500

Hi, all ! I have a situation like this: [SIP Terminals] <-> [*] < -ISDN-PRI-> [Panasonic D500] <-> Telecom (conn to Telecom is with second PRI card in Panasonic and 16 POTS lines). Panasonic has 2 ISDN PRI cards (one to Telco, and second to Asterisk), 16 POTS lines to telco and 32 (advanced hybrid telephone type) extensions. Idea is to have possibility to have some users on

Hey all, Wondering if this is possible.. Incoming call is answered through X100P, then an extension is dialed using the same X100P card. Basically I want to dial in, enter 9 + <phone#> and have it do a flash then have it dial *08 <the same phone number> + # on the same PSTN line to have it transfer my call to another phone number. I realize this isn't very safe, but I would

Hey everyone. I'm wondering if anyone has any ideas on a way to limit the number of outbound calls at a time, and if the limit is reached a message is played when someone tries to place the next call. I've searched the wiki but have yet to come up with anything. Any help would be appreciated. Thanks!

Hey all, I'm trying to get my Asterisk server up and running on fwd.pulver.com just to get the hang of it until I get my FXO card in a couple of days. It seems to connect but that's about it. If I try to dial into it from another fwd # it says user is not online. In sip.conf I have the following added: register => xxxxxx:xxxxxx@fwd.pulver.com/489125 [fwd.pulver.com] type=friend

Hey, I've checked all over and can't find what I need to know, so I'm posting here. I want to use Asterisk with my Primus VoIP service but it seems I need a username and password to authenticate with at Primus. Has anyone had any experience with this? How did you get it? Is it stored somewhere in the D-Link gateway they gave me? Thanks in advance and sorry if this makes no sense.

Hi all, I'm not sure if this is the correct mailing list for this (I was going to send to Asterisk-Biz, but seems more for this one). Anyway, I'm having more problems with Broadvoice. I still can't get calls unless I comment out the secret= line in sip.conf, but now I can't even place test calls to it from my cell phone. When I do, I get a fax macine. Debugging SIP shows NO

Hey All, I have a question that I'm curious about. I want to set up a 4 phone system in my home with 2 actual lines coming into the house. Both or just regular lines (not sure of this matters?), one being VoIP and the other just a regular analog line. For now though I just want the VoIP line coming in, but would like the ability to expand to 2 lines in the future. What type of hardware is

Anyone have any luck with connecting Asterisk to the Panasonic KX-TD500. I have Asterisk connected via crossover to the TE110P. We are able to make internal calls into the Asterisk Box but the PBX vendor (I know nothing about the KX-TD500) tells us it is not possible route DID over the trunk. I find this hard to believe. Anyone have any luck with this? Thanks! -------------- next part

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Call for Participation - HPDC-2013 New York City, NY, June 17-21 Early registration is open! http://www.hpdc.org/2013 The ACM International Symposium on High-Performance Parallel and Distributed Computing (HPDC) is the premier annual conference for presenting the latest research on the design, implementation, evaluation, and the use of parallel and distributed

Guys (and Gals), FYI I also have the *same* message here. Wonder is it is related to my Compaq D500 Space Saver PIV 1.7 or the fact that I don't yet have a modem card in the * box. (Please don't shoot me, did try Google first) Many thanks, Wagner Gimenes -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf

Hello Everyone, We are looking for solutions where the transcoding is abstracted away from our * box (i.e., to the network layer) using some carrier grade gateway, or router. The reason for my post is to know about solutions people have used in the past, and how it fits into their overall architecture. Our transcoding needs consists mainly of u/alaw <-> g729, and gsm would also be good....

Hi all, I just upgraded to Asterisk 1.4.18 a few days ago and I don't use Broadvoice TOO often, however I have a Vermont number with them and so my mother in law calls it to talk to my wife once in a while, so that's why it took me so long to notice it wasn't working. Anyway, when she calls she gets a busy signal (as I've tested when calling it from my cell). When I enable

Call for papers *Grid 2012: 13th IEEE/ACM International Conference on Grid Computing* Beijing, China September 20-23, 2012 http://grid2012.meepo.org Co-located with ChinaGrid'12 Grid computing enables the sharing of distributed computing and data resources such as processing, network bandwidth and storage capacity to create a cohesive resource environment for executing distributed

Call for papers *Grid 2012: 13th IEEE/ACM International Conference on Grid Computing* Beijing, China September 20-23, 2012 http://grid2012.meepo.org Co-located with ChinaGrid'12 Grid computing enables the sharing of distributed computing and data resources such as processing, network bandwidth and storage capacity to create a cohesive resource environment for executing distributed

I have a weird problem.... Asterisk 1.4 E100P connected to a Panasonic TDA phone system Here is what I get SIP Ext -> Panasonic Extensions No Problems Panasonic Ext -> SIP Extensions No Problems SIP Ext -> VOIP Provider No Problems Panasonic Ext -> VOIP Provider Errors ---------- Working SIP -> VOIP -- Executing [903........@from-sip:1]

I''ve been messing around with Puppet on a VM on my personal desktop. It looks descent. I was wondering what kind of load this thing would have managing about 400 nodes. Does this thing require a beefy server? -- You received this message because you are subscribed to the Google Groups "Puppet Users" group. To post to this group, send email to puppet-users@googlegroups.com. To

> On Sun, 27 Mar 2005, Nenad Radosavljevic wrote: >> Only way I have managed to get Zap channel to reject a call on TE110P >> without answering it, is to dial number that is not handled in dialplan >> (I >> have a ISDN PRI with 100 number DID service, and about 30 of them are >> handled by dialplan). So far I didn't manage to reject call that are >>

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Hi, I'm trying to use the altmbr.bin that came with 3.80 on an ATA IDE drive. When I boot I get this error msg: > Missing operating system. > > No bootable devices--strike F1 to retry boot, F2 for setup utility. The first line is coming from syslinux, the final line is my BIOS offering for me to retry. =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- PC: Dell D500 notebook, P4 1.4GHz