similar to: Queue Dial Plan

Hello all, I was wondering if anyone has had much experience with Realtime Asterisk. I like the ability to setup my extensions and voicemail boxes in MySQL, but I have a huge worry. What if MySQL crashes. I played with rtcachefriends, but can't seem to find a way to have asterisk store the extension information to ensure the phones will continue to work even if MySQL has a hiccup. Any

Hello all, I think Lee has given me a head start, but I'm still running in a circle (at least i'm in the lead). The problem is with my queues. The phones go to their own voicemail after 5 rings. That's about the same time I allow the phone to ring before trying another phone in the queue. Is there a way to tell asterisk....? If this call is coming from a queue, do not follow a

Hopefully a fairly easy question for the group... I have a queue which should contain about 10 agents (it will be all the phones in the office). This office is remote, so I would like to add their sip phones into the queue remotely. Also, if the system ever gets reloaded or rebooted, I need those agents to remain in the queue. Question: 1) How do you remotely add agents to their respective

So I might have a bit of a more narrow question from my earlier one. Previous, I had been wondering what would cause a phone dialing into a DID that connects to the asterisk box to get a fast busy. I've noticed the following message: chan_zap.c: Ring requested on unconfigured channel 0/1 span 2 Any idea what would give me this error? And would this cause a fast busy? Thanks again everyone

We have a location that is having a really odd issue. We have a sangoma POTs card. We are running software echo cancellation with the card (through asterisk) to try to eliminate some major echoing problems. I've turned on both EC and echotrain, which seemed to have gotten rid of the echo for the most part. However, we are now running into an issue where the outside caller hears a star wars

I'm sure this was asked before, but I can't seem to make this work... If a customer dials one of our DIDs, and the operator transfers that call to another employee, the Caller ID doesn't seem to do what I would expect it to. I would expect it to show the original caller's ID. Example: John calls in from the outside using (213-555-1234) and he calls into the asterisk system

I'm having a problem getting my queue to function as it should. After 20 seconds or so, it should prompt the user with a message "thanks for holding..... press # to leave a message or stay on the line to continue holding". I set up the "context" in the queues.conf file, so if a user presses a digit, they should be able to leave. But I get a SIP BUSY message. Here are my

We are trying to use a softphone from a location that is behind a firewall. We are using x-lite as the softphone. So far, we've been able to get the phone to register with the asterisk server, and it can receive voice from the asterisk server (IE, voicemail, etc). However, asterisk can't hear anything from the softphone. We have used 2 different machines to test this on. We are watching

Hey all, I had a user complaining of calls which were dropping mid-conversation. I looked into the time of one of the calls, and saw the following: Apr 4 12:13:03 WARNING[6670] chan_zap.c: No D-channels available! Using Primary channel 28 as D-channel anyway! Apr 4 12:13:05 WARNING[6660] channel.c: Avoided initial deadlock for '0x82b8430', 10 retries! Apr 4 12:13:05 WARNING[6660]

I'm trying to duplicate a behavior we had with our old avaya system, and I've come across Auto Answer (Ring Answer). However, its not quite the same yet. Right now, when I dial **5053, it will add the SIP header for Ring Answer and it will call 5053. The phone auto pickups just fine. However, we need that call to be muted. If you were to call into a meeting, we wouldn't want them to

I've been able to page to a specific phone (intercom type of thing), but I'd like to have a macro or agi that pages all phones but first checks if their on the phone. It looked like there used to be a pageall.agi type of script on the wiki, but that link isn't valid anymore. Does anyone have that script, or something else that would work? I would just do SIP/1000&SIP/1001, but

Hello All, Hoping all of you might have an additional option for me to try at this point. :) My Goal: To have a paging option that does the following.... When I press **_XXXX it will send a ring-answer page to that person. The person on the other end should be muted, so if they are in a conference, you can't hear what is going on in the meeting. If that person hears me and decides they want

Hi all, Here's our setup: Asterisk 1.4.18 Agx-ast-addons 1.4.5 Problem: When accepting a fax, the fax itself comes through just fine, and it does successfully create a tiff file. However, the dialplan should be executing a system command right after that completes, but isn't due to hanging up early. I'm getting a cause 16 hangup, which I believe is a "Normal Hangup", but

Hoping someone out there has run into this or has some ideas for us. We currently have asterisk set up with Realtime (using mysql) for its extensions,sip and voicemail files. The problem we are trying to solve, is one of a failover mechanism. What if our mysql server went down. Can Realtime be set up with a secondary mysql server to get its data from. We can set up mysql to sync with its fellow

Hi! I want to have this behabior in my queue. When a call come in, if there are unavailable agents or and busy agents, the queue reject the call. Thanks ! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070711/a05ed825/attachment.htm

Dear Group, I need to write a couple of reporting tools for my Call Center Asterisks implementation. I have multiple call queues with multiple agents that can sign in and based on gain access to multiple queues based on their assignments. I would like to write a script to collect call statistics for the agents the queues and the calls, and to put these into MySQL for reporting purposes.

We're trying to get a SIP peer going between our asterisk box and our provider. It should then ring our phone. The call does come in and it does execute the extension in the dial plan. But the provider says they never get a 200 OK back and therefore they send another INVITE and then after a few seconds drop the call. Here's our setup: sip.conf [ngt-trunk] type=peer qualify=yes port=5060

<Correction in my zapata.conf file I used> Hey Everyone, So this is a problem I've been having for sometime now. I sent a few messages to the list with no luck. The problem is that when people dial into the Asterisk system using DID numbers, it works the first time or 2, then I get busy signals. A friend recommended I clear out the zapata and zaptel, start over, and recreate my

Hi everyone: I've searching for a while and haven't found what i need. The thing is that i have a tdm422p with the two fxo ports connected to the pstn. I want my sip users to be able to call other numbers(any number) in the pstn through my zap fxo channels. I have a big number of sip users so as you can imagine there will be congestion when some of them(more than two!!) want to call

Hey all, This should be an easy one. I have a few different queues and wanted to set up a standard macro to handle them, so I can shrink the dial plan down and stop having so much redundancy. But when I try to use it, i get a "no answer". Here's what does work (non macro): exten => 5054,1,Answer() exten => 5054,n,Ringing() exten => 5054,n,Wait(2) exten =>