similar to: Agent Queue, Silent Calls Problem

I have an issue with silent calls when an agent gets a call from the queue What happens is - The system dials a call (agent call) - The caller picks up - Asterisk sees the person picked up - Transfered to an agent - Agents phone automaticly picks up (sjphone auto accept on) -The user hears nothing says "Hello, Hello, Hello ???" - Asterisk sees agent as 'Available'

Hey guys, I have just developed this problem with my Windows XP box. I think it started since I installed XP SP2. Both SJPhone and Xlite does some kind of bridging with the speaker out port. When ever I make a sip call to where ever, the other party hears a lot of echoing. Well I noticed just now when I was playing mp3's via Winamp, the music was being played through my sip calls that I

I have a queue I configured as "strict" and a cron script I use to QueueAdd and QueueRemove agents according to my company's requirements. Usually I have 2 or 3 agents at a time and the ring strategy is ringall. These agents use non-open-source Windows softphones that do not let you configure it so that if they're on the phone, a second call will be rejected (agent busy).

AgentCallBackLogin is going to be deprecated, so I've decided to emulate chan agent using AQM and RQM funcions and Local channel. I use asterisk 1.2.13 and latest 1.2.x. zapata. I used example 2 from http://www.voip-info.org/wiki/view/Agents+without+agent+channel and example from queues-with-callback-members.txt from asterisk 1.4 doc directory. My dialplan is very similar to Digium's

I recently setup an Asterisk Server- I was able to follow a tutorial from http://www.automated.it/guidetoasterisk.htm#_Toc49248752 Until it told me to call another line, let it ring until voice mail picks up. My problem is the tutorial left out how to configure a SJPhone so that it connects to my asterisk server not directly FWD. I've tried everything I can think of, I must be missing

I had the same problem with snom 190 phones. Using the transfer with # instead of "Transfer Button on the phone" worked for me. In my configuration "REFER" was not send, so the transfer with the button on the phone did not work. Guido Hecken -----Urspr?ngliche Nachricht----- Von: steve szmidt [mailto:steve@szmidt.org] Gesendet: Sonntag, 26. Dezember 2004 17:14 An:

Good Day list, Anyone know if there is a way to have the AgentCallBackLogin function play a voice file after the agent picks up the phone? If this is not an available feature, any ideas on the difficulty in making this feature? Example: Extensions.conf exten?=>?700,1,AgentCallbackLogin(${CALLERIDNUM}|?AnnounceCAllQue-TechSu pport?); ....... exten => s,6,Queue(queue1)

Hello. I've got 1 queue setup with 2 possible agents. Agent 1 is logged in and awaiting a call via AgentCallback. Agent 2 has not logged in. An outsider (caller A) calls in and is placed in the queue, cytelcs. Agent 1's phone rings and Agent1 and A talk. While they are talking, caller B calls in. Caller B is correctly placed in the queue and hears music, however this shows up in asterisk

I have not been able to find anything that relates to this problem. The agents are using Cisco phones. Calls goes into a queue. but once an agent picks it up it cannot be transferred. However if they call directly to the agents extension it's not a problem transferring calls. It sounds like a misconfiguration but I cannot see what's wrong. Any takers? -- Steve Szmidt "They

Hi, Is there any way to have queue member interface answer automatically? Basically when agentA is called, his phone picks up with no intervention from his part? (assuming of course he's available and not on the phone, and not paused). I already manage this with the Page application (using exten => s,n,SIPAddHeader(Alert-Info: Ring Answer)) and Polycom phones. But how do I do this

Hello! We are having an interesting problem with the queue. What is happening is that no matter how many agents are logged into the queue, only one phone will ring at one time. So, for example, if we have two agents in the queue and two incoming calls. The first incoming call will ring on one of the agent phones, but the second agent's phone will not ring until the first agent picks up the

I was wondering if it was possible to run a macro once the agent/member picks up, I know I can do this with dial in the extensions.conf but wasn't sure about the queue. Basically I have a macro that identifies the caller and need that run. -- ~Shaun

Hi all I have a following problem. I want to use sjphone to connect to asterisk sip server and then I want asterisk to do a conversion to h323 and send this to h323 gateway. sjphone---sip----ASTERISK----h323-----GATEWAY Example: if someone from plane PSTN line dials 123456 the gateway will forward this to asterisk and asterisk will forward this to sjphone and the other way around. Could

Hi everyone, I am running asterisk on red hat linux 9 box. The sound card is Intel 82801db AC' 97 audio and the module is i810_audio. It runs well with other applications like xmms and the standard tests deliver a sound . I have also tried to record voice and that works well too. 1-)Now when i run asterisk and i dial out an extension to play any sound there is none. The same thing

Good evening all. I'm having this strange behavior when dialing two or more simultaneus calls via IAX to other * boxes. Sound starts to have more latency, wich increments until it's almost impossible to talk (6 or more seconds), I try this calling with two grandstreams, one grandstream one tdm410p, one tdm410p and sjphone, sjphone and one grandstream, the result are similar.

Hello I need to know who's the agent that picks up a call in a queue, preferably in real-time so I can invoke a script that sends info to the agent's PC. It doesn't seem easy to do unless I use the info on asterisk's log file (using swatch or something? is this possible?) ... is there any other way? Anyone? NOTE: I've succesfully tried to use firefly as an IAX2 extension to

Hi Guys I have an issue where a call is picked up from a queue. The caller asks the person who answered to attended transfer to extension 3082 (for argument's sake.) 3082 picks up the attended transfer and speaks with the outside caller picked up initially from the queue. A few seconds after 3082 has started speaking to the outside caller - 3082's call goes dead in their

configuration ISDN BRI card : ISDN Olitec PCI 128 (hisax gazel) internet connection by ISDN 64kb/s dynamic IP nom de domaine registered to : dyndns.org avec ddclient to register IP par ddclient asterisk (on internet gateway) configuration pour ISDN BRI par virtual modems /dev/ttyI* (modem.conf) logical telephone SIP "SJPHONE" on 2 local stations "windows" (i don't succeed

When an agent logs into a queue using AgentCallBackLogin, he should be ready to take calls until he logs out right? For some reason the first time a customer calls the queue, it rings the agent just fine but after the agent hangs up the phone and the next caller calls the queue, no more calls will be transferred to the agent. He shows as logged in, but the calls wait in the queue forever and

Hi All, I've been experimenting during this weekend with asterisk as a softswitch, talk about me being completely lifeless, but let not talk about that. I've been conducting some really funny tests, trying to get an optimal SoftSwitch functionality. Here is my current setup: Source: Windows XP Pro + SJphone Box 1: Asterisk running in PassThorugh mode Box 2: Asterisk running in