similar to: Clearing Queue Stats?

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..

I searched for "say number" in the * google archives and have not found reference to options for "say number". I would like to have * say digits instead of the hundreds and thousands. EG, "1234" would say one two three four.

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

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

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 everyone, I have done my reading & research and everything I try is coming to different errors, so I am going to beg & pray someone here can help me with my problem. I appericiate any help in advance! I am running Samba 2.2.11 on a RedHat Linux 7.3 Server, connected to a network of Windows 2000 & XP Machines. I want to configure Samba to be the Domain Controller for my other

All, I would like to play an announcement to the user on what external line a call came in, right before this call get bridged to this user. How would I go about implementing this in * ? Regards, Hans -- The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may

Anyone have any example scripts in PHP that connect to the manager? I'm not really a much of a programmer so I could use boost. Once I can figure out how to get it to login properly, I'll be ok from there. Thanks, Kevin _____________________________________________________________ Are you a Techie? Get Your Free Tech Email Address Now! Visit http://www.TechEmail.com

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

Hi. There is not much info on the FWD site about this. What codecs do they use? When I try to connect with X-Lite, it works with GSM. When I try to call out with *, it wants G729. I have disallow=all and allow=gsm in the sip.conf. I end up getting errors: Unable to find a path from G729A to GSM Unable to find a path from GSM to G729A What's up with that? I was able to make a call once

Thanks to ww and his patch on bug #104, I have successfully implemented Asterisk behind NAT without using STUN or anything crazy. It's quite straight forward. Until this gets tested enough and put into CVS, you will have to patch your chan_sip.c file to do this. I'm sure within the next few days this will get put merged into CVS if no one finds any problems. I tried this on chan_sip.c

Hi. I just tried updating asterisk and I guess I broke something. Here's the log: Unable to find a path from G729A to SLINR Unable to find a path from ULAW to G729A Any ideas on what I should try? I tried nuking all the zaptel stuff in the system and the source and started over agian. Also nuked the asterisk config files.... I saw this asked once before but there was no reply :-/

--- Jonathan Tew <jonathan@ultracart.com> wrote: We've got the app_queue configured to supposedly allow for a call to be transferred. When the call comes in and an agent answers it (using X-Lite Pro) and then decides to transfer the call (using the SIP phone interface) they get disconnected from their call and after left logged in to the queue system. Obviously we're doing