similar to: Multi-Level Queue

I've got a very strange problem and I can't figure it out. I have a Cisco PRI gateway connected to * via SIP. When I debug on the Cisco, I see callerID name, but it is not getting to * via SIP. I am running * 1.4.2 and the latest Cisco IOS for my router. Here is what is happening: A call comes into the gateway. It sends a SIP INVITE to * with "pending" as the callerID

Does anybody know the sql type for the "call-limit" field under sip peers? Everything on voip-info is missing that entry.

I have noticed that asterisk will only try one interface per queue at a time. Is there any way get get it to dial say three at a time and connect the first one that it reaches. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070405/a510dd31/attachment.htm

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Hi, is it possible to have the information stored in /var/log/asterisk/queue_log realtime in mySQL? thanks

Does anybody have realtime queue members working? Not the queues themselves, just the members. I have realtime working for voicemail and sippeers, but I can't get queue members to work. Here is what I have: res_mysql.conf: [general] dbhost = 127.0.0.1 dbname = ASTERISK dbuser = myuser dbpass = mypass dbport = 3306 dbsock = /tmp/mysql.sock queues.conf: [general]

Is there a limitation to the number of variables you can set from a PHP agi script? I have a simple example and I can't get it to let me set more than 1. I am pretty sure I am just missing something, but I've searched all over an can't find the answer. Here is the extensions.conf part: exten => _XXXXXXXXXX,1,AGI,diallocal.agi exten => _XXXXXXXXXX,n,NoOp(${ISLOCALCONTEXT})

Is there any way to see the called number when a call gets redirected to the 'a' extension from voicemail? Say x123 calls x456 and it rolls to voicemail. x123 hits * and gets dumped into the 'a' extension in the original context. I need some logic in 'a' to do a database lookup based on the original called number (x456). Any ideas? When I do a test, it appears

Hi all, does anybody know any way that when it run "reload app_queue" in the asterisk cli it don't lose the informations from "show queue (queue name)" ? I'm passing for this trouble, because I need this informations (http://www.voip-info.org/wiki/index.php?page=asterisk+cli+command+show+queue) that asterisk cli command "show queue (queue name)" show me

Is anybody using 1.2 (or 1.2.1) in a production network using Realtime (voicemail, sip or extensions) with 100+ SIP phones? If so, what are your experiences? We've been running 1.0.3 for about a year and it's been rock-solid. We'd like to upgrade to Realtime and 1.2, but I'm afraid of killing our stability. Obviously, we'd do it in stages (upgrade to 1.2, then realtime

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 =>

Hi All I would like to be able to have an announcement played to an operator advising them of the queue the call came from before the call is pasted over to them, so they know how to greet the customer. Does anyone have any ideas or can point me to some resource which details this? Many Thanks in Advance. SP

I have 20-30 GXP2000's connected to * over a T1 line. Neither end is NAT'd and there is plenty of bandwidth available over the line. The GXP's are 1.1.5.15, which is the latest. I have a problem where the phones keep dropping off of * and I get a "failed to register" message in the log of *. Sometimes they eventually connect and sometimes, I have to reboot them to

I have queue set up in realtime on Asterisk 1.4.2. Below is the senario that is happenening :: I have created a test queue with only one agent. Once I call the test queue the agents phone rings if the aagent is logged on. everything till here is fine. Now if the agent does not pick up the call, the call automaticaly disconnects after 15 secs as set for the queue, till here also it is fine. But

When phone A registers, I want phone B to ring, when picked up, it should call phone A and connect the phones. Translated: When GF in Mexico powers up laptop where soft iax-phone registers automatically, I want to talk to her asap :-) How to? Leif

Hi All; Which is better (to have more stable or release versions) of zaptel, libpri and asterisk: to use cvs or svn? In case of using cvs, why I need to type: export CVSROOT=:pserver:anoncvs:anoncvs at cvs.digium.com:/usr/cvsroot In other words: what is the use of pserver, anoncvs, ... with cvs checkout? Note: How can I know all the variables needed for cvs checkout so I might need to do

I'm sure this has been discussed many times, but I have a question about hoteling. My understanding would be this: A phone sitting on a desk. A user hits 9000 and it asks what extension you'd like to become. You type "1001" and then it asks for your password. You type 1234, and it says you're "logged in". You now are accepting calls at your phone and you're

Hi, this is to inform everybody that the translation of my new book (unstable version) is online at http://www.the-asterisk-book.com The book is a GNU FDL project. So everybody who wants to participate is welcome to do so. Also, everybody who needs material for his own work, feel free to take it as long as the new material will become GNU FDL too. I am glad that Stephen Bosch (who you

We've had a few people complain that the "beep" before leaving a voicemail is not loud enough and too short. Does anybody have a recorded beep that they can share, that is a little louder and a little longer? We've had this box in production for 2+ years, so I hate to mess with the gain on the PRI or anything like that because everything else works fine. I know nothing

I am using realtime on two boxes, one running 1.4.10.1 and one running 1.4.11. Everything works fine except for when I make a database change, such as a phones password. I change the DB, I prune the peer, I see it is gone and then I see it show up again in "sip show peer xxxx", but everything is not being updated. The phone will not register even though the DB and the phone have