similar to: Queue timeouts

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

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 list, I'm using an IVR where the caller chooses between 1. sales 2. support. When choosing 1 the caller is directed to the sales-queue when choosing 2 the caller is directed to the support-queue. Then the caller is directed to a free agent. I notice in the CDR-rapports that the destination is always '1' or '2', namely the DTMF-digit that the caller presses. How can I

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?

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]

I'm trying to fix a problem with the GrandStream Budgetone 102. I've been reading the source code, mailing lists and other resources. Here's the scenario and the approach I have been pursuing. I'm having some problems with the AGI calls and I hope someone can give me some clarification. PSTN <---> T1,PRI * <---> Grandstream BT 102 (12)

> Message: 5 > Date: Fri, 12 Sep 2003 13:24:13 -0400 (EDT) > From: "David C. Troy" <dave@toad.net> > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] 7206 as SIP->PSTN Gateway? > 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. > >

No More Cold Calling! We Get Prospects To Call You! Don't waste your time manually dialing potential customers. Almost 85% of your sales calls, or customer retention calls never reach a live person, and end up in voice mail. Most sales or customers service staff can only make 50 to 70 sales calls per day. VoiceMail Messenger liberates you from the repetitive monotony of out-bound cold

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

Hello all. I'm having a problem debugging an IVR I'm building. I can't see any reason this shouldn't be working. Firstly the asterisk version is: Asterisk SVN-trunk-r7230 built by root @ localhost.localdomain on a i686 running Linux on 2006-02-17 22:44:48 UTC Basically the problem is this. While the playbacks are happening you can push any one of the options and to happily

Asterisk 11.1.0 One queue with strategy=leastrecent. (Full queues.conf below.) Occasionally (several times today), a caller will get "stuck" in the queue - there are operators available to take the call, but the caller stays in the queue for a long time. Any idea what might cause this, or where I can start looking to debug it? I'm going to start digging through the queue log

I'm working on an IVR that gives the users the option (number 5 in the main menu) to dial by extension: exten => 5,1,Set(TIMEOUT(digit)=5) ; Dial Extension exten => 5,2,Set(TIMEOUT(response)=10) exten => 5,3,Background(LCL/prompt-60) exten => 5,4,WaitExten(15) When going option 5 you can dial some extensions such as 2802, it goes to the extension (all extens start with 28 on the

Hi, I'm trying to find (with some desperation now) a decent web based or application based UI that integrates with an Asterisk based PBX and is designed for a Serviced Office environment. Key features we're looking for: Concept of a client with multiple staff/associates Recognition of inbound DIDs with display of company details and greeting Support for multiple receptionists Some

Greetings, list! My boss would like some statistics on how many calls are answered out of specific queues during a given time period, and I'm not sure how exactly to obtain those stats. Here's how our queue system works. 1) Call comes in and enters our 'ring' queue where the phones ring for 15 seconds (caller hears the standard ring tone). 2) After 15 seconds, the caller

Hi, My client wants incoming callers who do not press a digit to go straight to the operator. Does anyone have an idea of how this could be done? I've looked for some examples, but I'm still not clear on it. Here's the relevant portion of my extensions.conf: ------- ; Wait 15 seconds for an answer (pick up the local phone) exten => s,1,Wait,2 ; Answer the phone exten =>

I am running asterisk 1.0.1 with OH323 compiled in. We have a 323 trunk to CallManager with a mgcp controlled pri router. When using sip phones (directly registered with asterisk) to call out the 323 trunnk to PSTN, calls timeout after 3 rings. If I answer b4 3 rings - no problem, otherwise I get "no one is available to answer at this time" on the consoel and it redirects to an

I have an Asterisk 1.6.0.28 system, with a queue called 'marketing'. Everything appears normal, but the status of the members never changes from 'not in use', even if they are being rang or are in a call. Members are added like so: queue add member SIP/1406 to marketing penalty 0 as SIP/1406 state_interface SIP/1406 And they are present as a hint: exten =>

Previously incorrectly sent to asterisk-dev list, sorry. I tried today while connected to a Jtec QSIG E1 card, with DAHDISendCallreroutingFacility with the following test dialplan: Extension 4888 is on the Fujitsu [incoming] exten => 8688,1,Answer() exten => 8688,n,Playback(connecting) exten => 8688,n,DAHDISendCallreroutingFacility(4888,8688) exten => 8688,n,Playback(goodbye)

Hi, I have a requirement, which I am not sure whether it can be implemented. I had done some searches but didnt find an answer to this. Kindly let me know if some one has an idea to implement this: I have two Queues - Sales & Booking I have 12 Agents who are added to both the queues Suppose there are 12 calls in the Booking Queue, and 6 calls in the Sales Queue. Only 8 calls in the