similar to: Asterisk Queue Call Transfer

HI, My Asterisk is registed with my SER. My client are connected to asterisk when they dial any no like 62222 asterisk passes this is ser and then again ser passes this no 2222 (strip 1) back to my asterisk. but insted of ringing this exten it says loop detected. can some one tell me what is wrong. thanks arun -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I'm just posting this questions to both forums as its related to both. In hope to get some help on below issue: Asterisk 1.4.x CCM = 4.x CME = 4.x codec = g711ulaw Here is my setup: 600X Phones ----> Asterisk ---- SIP Trunk ----> Call Manager -----> CME -----> 461X Phones 461X Phones ----> CME -----> my dial peer points to Asterisk IP for 600X Phones so in

Hi, I've an PRI coming to my asterisk ,calls are coming fine and my agents are able to answer no prob. but I've an agreement with my telco with some incoming no if the no of calls on these no are more then 3 then send to another no. they use busy signal to divert call on another number so I'm sending the call to Congestion() if no of calls in this group are more then 3. But my

HI I bought 20 license from Digium and install in my server and b'coz of some problem I've to change my server is it possible that I can use those lice and register again in my new server ? Is it possible that I'll be able to use those lice in my old box also ? thanks arun -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm running a php script to generate calls using Asterisk Manager and its working fine. this script call a specified land line number if the phone is answered then It will connect to an extension and play an IVR. But I see in Asterisk CLI its placing the call and it shows channel answered but I don't receive call on my land line and it starts playing the IVR. Please guide me how to

Hi Im trying to install my TC400B trans coder card when I do: modprobe wctc4xxp tail -f /var/log/messages says: May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Encoder' with 92 transcoders (srcs=0000000c, dsts=00000101) May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Decoder' with 92 transcoders (srcs=00000101, dsts=0000000c) May 13 14:56:36 pbx2

Hi I've few queues configured in * box is there any what that before sending call to a particular queue can we get the status of the queue that is how many agents are available in this queue (logged in, paused, busy, unavailable). thanks arun -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, In my dial plan I've configured two trunks to make outbound calls (trunk1 and trunk2) to same service provider but I want when any of my exten starts with _2. should goto trunk2 and there should be some kind of disturbance (like some noise or some background noise) when my calls goes to trunk2 to make the call quality bad. Mainly I want to achieve bad call quality on trunk2 by adding

Hi sorry for asking the same question again: here is my details: I've 50 exten in my sip and I'm using snom300 to my asterisk box this asterisk box is connected to another asterisk box using IAX trunk over 1MB full duplex line. I'm using g729 as the preffered codec. Can you please tell me how many calls can go at the same time without causing the any type of problem. thanks arun

Hi, in my dial plan I've configured two trunks to make outbound calls (one for national calls and other international). I want to allow only 2-3 extension to make use of my international trunk to make outbound calls so I want some kind of auth. based on their callerid . Please guide. thanks arun -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I ve setup a queue with 2+ agents for managing our inbound calls from customer. Using Asterisk 1.2.18 in a CentOS box. Agents login using AgentCallbackLogin application and I use a BASH AGI to accomplish this as there are some validations done with MySQL DB. Im aware that transfer could be done with option 't' in the queue() application and I was able to successfully transfer

Hi I've two servers : 1. UK 2. Pakistan Pakistan * server has ISDN30. Pakistan(ISDN30) <====> UK ===> User Im planning to setup an IAX2 trunk between these two server ? so , how much bandwidth I need for 30 simul. calls ? Im planning to use G729 on both my server ? to support 30 calls over IAX2 do I've to change some setting during compile time or not ? pls suggest.

HI Im getting strange message on asterisk console WARNING[26853]: app_queue.c:2321 try_calling: Announcement file 'custom/announce-adslsetupnatrate' is unavailable, continuing anyway... thanks arun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070603/6564c117/attachment.htm

In my * box I've configured two queues and incoming number and whenever any one calls those number call comes to my *box and it sends call to my agents in queue. but if no agent is available it still answer the call. Is there any why when my agents are not available I don't want call to get answered. Here is my dialplan: exten => xxxx,1,GotoIfTime(*|*|20|dec?ccagents,xxxx,6) exten

Hi All; It look like there are some free (open source) tools that are used for Asterisk reporting special for call center (to see number of agents logged in, number of calls now, .. etc), and to be used as dashboard. Can someone direct me for something really is suitable and stable? Regards Bilal

Hello, I thought to post this here before my manager starts his own coding project to give us a workaround. My situation I'm running into is as follows: 1. A call comes into our Asterisk system, it's trunked from one office to another via IAX. 2. Call enters a queue and is picked up by one of the agents. 3. That agent has to transfer the call, could be for a number of reasons the client

Dear All, I have six different campaign and 5 different agent have login on that campaign.*Same thing i have done using agi and database,i never use queue management on this scenario. Agent** can also shuffling one campaign to anther campaign. * Now i want to do some work with queue.I want to use single queue to managing this. Eg: campaign Agent Login A a_1,a_3

Hi, do I have a chance to use iax trunking on OpenBSD where there is no zaptel driver or ztdummy available? Do I can use sth. else as timing source? kind regards Sebastian

Hi list, Using Asterisk 1.2.10 I am getting seriously confused by Queues and Agents. So far I configured my queue and agents, had my agents login using agentcallback. Call enters queue agent gets a call, if agent doesn't answer after 20 seconds a flag is set in AstDB (thanks to: Leo Ann Boon), call is returned to queue and the cycle continues. If the same agent doesn't

Hello, I have three queues (F1,F2,F3) with default queue weight and three agents (A1,A2,A2) with default agent penalty. If the three agents are busy and tt same time a caller (C1) enter in the queue F1, and after 20 seconds a second caller (C2) enter in the queue F2. So, few seconds later, the agent (A1) state comes to availabe. In this case the asterisk deliveries the caller (C2) to agent (A1),