similar to: say number question

All, I know you can use, say, a 2620 w/2 port FXO card as a SIP gateway. Clearly you can use the 5300, 5800, and MGX8850 too. Does anyone know which cards, if any, exist for a 7206VXR to act in a similar capacity, either as a T1/PRI, DS3, or POTS FXO/FXS? What other Cisco routers can act as SIP gateways today? Thanks, Dave

I need to get your thoughts on something.. :) I am trying to create a system to process the CDR call logs for department accounting.. I think there are two ways of doing it.. Either I can create an AGI that will run on the "h" extension and will lookup the last entry that matches the account code of the call that just ended in the MySQL CDR and calculate the call cost immediately..

Is there a way to clear the Queue stats? That is with out restarting *? I'd like to reset them daily and don't see a way to do that. Unless the only way is maybe a cron restart asterisk like every weekday @ 04:00?

Hi everyone! I was wondering... Does the musiconhold quality improve if the mpg123 processes run with a negative priority? If so, is there a way to make them start like that, so I don't have to renice them? Regards, Evert

Hello, I've search through all of the lists and cannot find any descriptions of live monitoring (monitoring a phone call going on between an extension and a zaptel channel live from another extension while the monitoring phone is muted). I am aware of the monitor function which is actually a call recorder, but I'm looking for live monitoring from a muted extension. is this easily

A friend and I have recently added the ability to announce the callers position in the call queue every x seconds.. or even just inject an anouncement every x seconds. All setup in queues.conf and can be setup per queue. My next project is to add the ability to announce the callers estimated wait time. I want some feedback to see whats the best method to calculate that? What do you want just

I have a PRI link up and running between Asterisk and a Nortel Norstar MICS v4.1 . I'm having a problem getting the textual Caller Name across the link from Ast to Ns, however numeric Caller ID arrives and displays fine. From Ns to Ast both elements come through fine. I'm forcing dummy values for testing using: exten => s,1,SetCIDName(Test) exten => s,2,SetCallerID(1234561234)

Hello: I am testing Asterisk with oh323. My question is: can Asterisk route some calls thru a second h323 gateway (a h323 <-> PSTN gw)? - Asterisk ip: 192.168.1.10 - h323<->PSTN gw: 192.168.1.20 I've tried: exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20) or exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION@192.1.1.20) but it does not work at all. If my h323 client

Thank you for reading this, sorry to waste bandwidth otherwise. I am part of a US company looking for someone to setup a demo IVR system for us. I seem unable with my current knowledge to pull this off myself. The demo is the regular enter your id and validate/repeat/continue methodoligy you put up with in everyday life. I would like to have the validation and other parts done via database

Hey folks -- Some of you had asked about getting my modification to ZapBarge that lets you monitor active Zap channels, scanning through them with the # key. I had posted a version a few days ago that was pretty crude; it didn't check to see which channels were active and assumed you had 23. I've finished a new version that checks to see what channels are actually active and pulls

Anybody have any experience pushing XML display data to a Cisco 7960? I am pulling data from a webserver now, but I have stuff, like queue status info, I'd love to be able to push at particular phones. A "pick up the damn phone" push message would be cool too. :) Dave ===================================================================== David C. Troy [dave@toad.net]

Hey folks, I have a T100P card with a PRI; when doing outbound dialing over the PRI, I can use SetCIDNum(2024561414|a) to force caller ID to display as "The White House" on a land line. This is apparently done as a reverse lookup by Verizon, as I do not hand the PRI the words "The White House" -- 202-456-1414 is an actual White House number. Using SetCIDName("Flying

I am about to try our TDM400P E model from the developer kit (not the "Lite") we just got and noticed a large number of reported problems. I had the CVS from Sep 12 (or so the CVS/Entries file has in it). My drivers seem to modprobe fine. My card show up as "Found a Wildcard FXS: Wildcard S400P Prototype (4 modules)" in dmesg and no apparent errors...yeah. Is "S400P

All, I've been doing a lot of work on the Queue application and have some ideas I wanted to get some feedback on. Many people, myself included, have stated their desire for more/better queue statistics. We ought to be able to bring things up to a standard comparable to the established ACD apps. While some basic metrics (completed/abandoned calls, which agents last took a call, etc) can be

Hello all, A few days ago I noticed that the following seems to happen when upgradign OpenSSH on Linux-based (/etc/rc.d/init.d/sshd) based systems: - sshd is restarted with 'sshd restart'; however sshd serving in port 22 is not replaced. - you have to kill the old one (netstat -ltp | grep :ssh ; kill ...) first, then restart sshd. - This might happen only when performing the upgrade

If anyone is looking, I just ran accross an ebay auction for X100P Cards at what I thought was a very reasonable price. http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3046843672&category=48483&rd=1 --------------------------------- Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software -------------- next part -------------- An HTML attachment was

Anyone have a good source for BT-101 phones? I had a lead on some, but they've not materialized. I'm also interested in the ATA-286 (HandyTone) units as well. This is for my personal Asterisk/INOC-DBA setup, that has yet to materialize heh. --- Tom Sparks

I just ordered the Asterisk Developers Lite kit. My environment will be the RH9 Linux server and a Windows workstation with Samba. I also of course have analog lines and DSL. I am interested in SIP development. I already downloaded the Asterisk software. What else should I download. Is there a doc that basically tells you the steps to install Asterisk and get it up and running? I would like a

Another thing I am working on is to do the timeout people have been asking about. If they have been on hold for (X minutes) dump out of the Queue App, so that the next priority can be a common mailbox. And I forgot to add the patch to my last email!!! argh!!!! http://pbx.usedontmiss.com/queue_patch John

I think I solved the errors I was getting with my patch, sort of anyway. Brief over view: Tell all the callers their position in the queue. When they move, tell them their new position. I was receiving Thread xxx already blocked by xxx. I found that if I only tell caller 4 and above (Which becomes caller 3) that their position changed, I do not receive the errors.