similar to: Overhead Paging

Here's one last plea for help from the list before I give up on the totally cool software FAX modem concept in complete despair (for now, at least, until I have more time to dig into it).... I am simply not able to successfully receive a FAX no matter what device I transmit from or how I bring the call into Asterisk. My lastest effort was a call from a Pitney-Bowes FAX machine into an X100P

We have purchased an Asterisk based PBX solution that is completely setup with one exception, overhead paging. We have a powered Paging System and want to find a way to set an extension (i.e. 999) to use the sound card, which would in turn go out the paging system speakers. I've seen several references to this being possible, but no examples how to do it. The system we purchased does not

I've rummaged through the archives and documentation and have yet to find references to nbsd or mention of how to implement overhead paging using chan_oss as mentioned in the list previously. I suspect that one would use a soundcard in the PBX system and feed the output to speakers and/or PA system. Would someone please point me to some procedures or documentation to acomplish overhead paging?

Hello All, I've mostly solved my DID problem from a few days ago. Apparenly the lines weren't configured properly. Now heres the next question. 12 E&M wink lines from telco. I have them all plugging into an Adtran 750 with FXS cards. The Adtran ports are configured DPO. How do I signal this from Zaptel. I have them setup E&M in zaptel.conf and EM_W in zapata.conf. They

I've been trying to get a Win 2000 RAS server working with my asterisk PBX for quite some time, to no avail. I've googled, I've tried loads of configurations, I've rewired phone lines, and still I am not winning the battle. Here's my config. PRI->T400P->Asterisk->T400P->Adtran 750(L36 Firmware)->RAS Server. I have 4 Zap channels signalled FXO_KS to the 750

Has anyone on the list been able to locate and try out the 1.1.0 firmware? It was released in November, but I have yet to get my hands on it. The Polycom site has way more docs online, but the link to the firmware only brings up the release notes. -sb

I have a customer that has two buildings that are connected with a fiber link. We have a single Asterisk server to cover both buildings. Now the customer went and bought an overhead paging system for the remote building and they want to integrate it with Asterisk. Is there a device that can connect over IP or an ATA that has an audio output port? The buildings are about 500 meters apart so we

I have been testing * with some Cisco 7912G's, in hope to trash our Nortel system. One feature our Nortel system has that I will need to fiqure out on the * system is paging. Is it possible to page a group of phones (all phones) with announcements? We are a k-12 school and we use our current phone system to make announcements on the phones monitor speaker. Any direction I can be pointed in

Does anyone know if its possible to have more than one sound card in an Asterisk box and use each one as a paging zone? How about left and right channels of a single sound card? I'm looking to have 2 paging outputs if possible - I've read about using a Grandstream phone on autoanswer but I'd prefer to have the feeds come directly from the * box and go into a stereo amp feeding 2

Hello All, Quick question here on equipment selection. Is there any benefit to using a bunch of Adtran 850's with RCUs over 850s with BCU L1 to interface with the T400P. I'm looking at building a PBX with 7 channel banks. 6 x 24-port FXS and 1 x 24-port FXO. Would the RCU give me any advantage/functionality over the BCUL1? Thanks, Scott

I have several phones running Cisco's 6.0 SIP software release at this time. Two of the phones have not shown any abnormal behaviors, but one of them has an unsettling propensity to lock up after several hours, where the softkey labels disappear and the phone stops registering, requiring the standard *-6-settings reboot sequence. Otherwise, the phone seems to work OK except for a slight

Hi, I'm a newbie.Sorry for my question. I have RH9 & downloaded rpm of dovecot for RH9. I installed it: rpm -ivh nameOfthatrpm & when i ran #service dovecot start ,I recieved following lines: Starting IMAP daemon (dovecot): Fatal: Can't use SSL certificate /usr/share/ssl/certs/dovecot.pem: No such file or directory

Hello All, I am experiencing some intermittent problems with calls coming inbound on my DID trunk. I have 12 DIDs that come into an Adtran 750. From there T-1 to a port on T400P. The problem is that some calls that come in don't seem to bridge properly. Heres what happens. Call comes in on Trunk. Call Routed to correct Zap Channel. Phone Rings. Person Answers phone, but hears nothing but

Hi , I recently installed Fedora Core 1. Installed postfix on it and that is running fine. I am trying to get dovecot to run and I have changed only the following line from the default dovecot.conf : imap_listen = 127.0.0.1 since I'd like to install squirrelmail later on. Now when I try to start dovecot I get "Starting Dovecot Imap: Fatal: Can't resolve address

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Caller*ID used to work as some point, but I can't seem to get it going these days. The card is a x101p. I've tried going up and down the rxgain scale. Can the txgain effect it at all? When I plug in a phone into the line with a splitter it can decode caller id with no problems. Reading through the mailing list archives hasn't given me any move clues. Any ideas?

Hey all, I know you all saw the topic and let out a groan. However, I understand how to get an overhead paging system to work with respect *, however I am now looking for a small(?) paging amp, that I could hook 3 or 4 horns to. I would like to just have the * extention be routed to a soundcard and out an output, so I would like an amp that is voice signal activated. Has anyone found anything

> I have a customer that has two buildings that are connected with a > fiber link. We have a single Asterisk server to cover both buildings. > Now the customer went and bought an overhead paging system for the > remote building and they want to integrate it with Asterisk. Is there a > device that can connect over IP or an ATA that has an audio output port? > The buildings

Hello there, We want to move from UW-IMAP to Dovecot (mbox format), works like a charm, but UW-IMAP has that bloody habbit to make a stupid message in all his mailbox files : From MAILER-DAEMON Thu Apr 19 10:30:01 2001 Date: 19 Apr 2001 10:30:01 +0200 From: Mail System Internal Data <MAILER-DAEMON at mailhub2-vil.isdnet.net> Subject: DON'T DELETE THIS MESSAGE -- FOLDER INTERNAL DATA

Hi all, I installed the estara softphone and had no problem registering it with asterisk. I could make calls to other hardware SIP phones (Cisco 7960) from the softphone, but I couldn't call the softphone from the Cisco 7960s. The asterisk console gave me an error message saying "unable to create channel" to my softphone. What could be the problem? I searched the archive with no