similar to: asterisk PLAR

say I needed to send a broadcast message that I wanted every user to hear when the pick up thier phone? can I "Play,message" on a line just before they get dialtone? or maybe after they dial before ring? how about a "ringdown" to a voicemail box and on end return them to thier line for the dialout? can * do ringdowns? when a user picks up an extension it automagically

OK, here's a configuration challenge: I want to have certain line appearances able to be "interrupted" by various other line apperances elsewhere in the office. This is harder to describe than it is to demonstrate, so I'll do that: Let's assume I have Cisco 7960's on all desks. 1) Call comes from inbound line X destined on extension 1234 2) Phones A, B, C all

Dear R-users, These days, I have been struggling to install the package sna on my Ubuntu 8.04 laptop. My R version is currently R 2.7.2. And I had no trouble installing any other packages (including network). However when I run 'install.packages("sna")', compilation of C codes is ok, but a syntax error is reported when building help pages and sna does not want to install. I

Greeting, I am trying to setup PLAR signalling in asterisk. I have modified the FXSLS TX bits in dahdi-base.c on line 2580, and I can make calls. .sig_type = DAHDI_SIG_FXSLS, .bits[DAHDI_TXSIG_ONHOOK] = DAHDI_BITS_ABCD, /*changed by for PLAR*/ .bits[DAHDI_TXSIG_OFFHOOK] = (0), /*changed by for PLAR*/ .bits[DAHDI_TXSIG_START] = DAHDI_BITS_ABCD, /*changed by for PLAR*/ When I got to change

Hello, Could somebody tell me if I can connect CISCO 2600 router with support of H.323 to Asterisk ? If it is possible could somebody tell me how to do it. I would like to document it and put on some website so everyone can see it. Regards, -- bart -------------- next part -------------- An HTML attachment was scrubbed... URL:

Basically, I am looking to set up an extension which will be used as a "help-line". I want it to function kind of like the bat phone from the old Batman series, where Commissioner Gordon would pick up the extension in his office and it would ring the phone back at Wayne's mansion. Is there a way to duplicate this functionality with Asterisk? I just need asterisk to auto-dial an

Hi all, I have looked through the wiki for any information on how to make an extension autodial another extension when it goes off hook. Anyone done this or know how it's done. regards Dave

I'm running Asterisk 1.2.8 with Polycom ip501's xten softphones The only problem I'm experiencing is the following: I can't seem to get blind transfers to work with my Polycom 501 phones Either through the feature code or the soft keys. Feature code blind transfers: I set up a feature map in features.conf like this: blindxfer => # This works for all my

I'm running asterisk 1.2.8. I would like PSTN inbound calls to do the following: 1-once PSTN callers enter their desired extension; they have to record their name 2-recording then announces that it is trying to locate the user 3-asterisk calls local extension and announces callers recorded name 4-local recipient user can choose to take the call, send it to voicemail or transfer it to

We are doing a medium sized office in NYC with 80 phones. The customer originally requested Polycom 601 phones. The COO also authorized us to purchase 2 Grandstream GXP2000 phones for the mail room. We find these phones much easier to configure and work with asterisk . They support BLF & intercom right out of the box. They can also be centrally managed and provisioned. They also sound great

Hi, Im new, but trying real hard! I just need general direction, not details yet..i'll try to figure those...just looking to avoid brick walls...bottlenecks...inefficiencies etc upfront. Hardware: motorola vt2442 - trixbox Apps: Dot Net application that operates the Manager API and the FASTAGI interfaces. I have the 2442 set as a PLAR so as soon as the ext is off-hook, it dials into the *61

Is there any technical difference between a T1 cable and a cat5e patch cable as far as using them with Digium T1/E1 cards? Can PRI circuits terminating at a smart jack connect successfully to Digium cards using straight through CAT5e cables? If so, are they using all of the pins in the cable? Thanks in advance

I had some questions come up today during the Puppet demo here: 1. Is there any way to apply different intervals to different classes? For example, we might want a security baseline manifest to run once a month, package installation files once/day, etc. Since we''re a brokerage, we also don''t want ANY system changes during trading hours. I guess we could cron killing the

You have the session target as the IP address of the router's own ethernet interface. You probably want that to be the address of the Asterisk server instead. I also highly recommend you use full duplex ethernet, as voice packets don't really like to be restransmitted when a collision happens. -d > Message: 10 > From: "Bartosz Jozwiak" <bartek@cq-link.sr> >

I would think I could do this but for some reason I am stymied. I have a PRI from RCN connected to a cisco 3640 (in my day "cisco" is all lower case :-)). My config looks something like this on the cisco... --------------------------------------------------------- voice-card 3 dsp services dspfarm ! ip cef ! isdn switch-type primary-5ess ! controller T1 3/0 framing esf linecode

I was wondering if people have experienced issues with Cisco 7960G and Asterisk. Any feedback on people's experience deploying this phone in production environments would be appreciated. Thanks in advance

Running Asterisk 1.2.9. I just installed a TE110P card and configured zaptel.conf & zapata.conf. The config files look right to me but I'm getting the following error when trying to start asterisk: Asterisk died with code 1. Automatically restarting Asterisk. Does anyone have any idea what is wrong with this configuration?? Thanks in advance!!! Here's my config files: zaptel.conf

Does anyone know if sip phones from any of the major IP phone vendors support 802.1x authentication? Any feedback would be greatly appreciated. Thanks in advance. ====================== Jeronimo Romero EUS Networks Email: jromero at euscorp.com Cell: 917-332-7238 Office: 212-624-5943 Web: www.euscorp.com ====================== -------------- next part -------------- An HTML

don't read this. i'm a complete newbie, and i don't know how to start sending messages. if someone read this, then can answer me also how to delete a file with the console (really ashamed) y know that copy with cp, but delete??? thanks... ___________________________________________________________ 1GB gratis, Antivirus y Antispam Correo Yahoo!, el mejor correo web del

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