similar to: Asterisk IAX Trunk and Queues

Hello all, I've seen that the application AgentCallbackLogin()has been set to deprecated in version 1.4. So I've done some tests based on the tutorial "queues-with-callback-members.txt" coming with version 1.4. What's not clear for me is what is happening to agents.conf, it seems that it's no longer needed, and I have to define my agents using variables in

I'm running Asterisk CVS-04/28/04-13:22:35 (fairly current) Today when I setup queues for the first time (with one member in my default queue), I got some really strange behaviour, aside from my hysterical laughing after hearing the default MOH =) I only have one SIP hardphone I'm testing with right now, so I tested using DIAX, Firefly(IAX) and XLite(SIP). My hardphone is an analog

Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digiuim card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over IAX trunk from which calls get routed to third server (C) (1.6.0.9) again via IAX trunk. SIP clients are connected to third server. A is the PSTN termination server, B runs the menu and AGI and C is

Hi, How do you folks handle really large queues (350+ simultaneous callers) in your Asterisk PBXes? We're going to be bringing in around 16 PRIs' worth of inbound callers, doing skills-based routing, and queuing them up for approximately 200 agents. What's the best way to handle all of these callers? We want to record the calls and we'll probably use the ramdisk method that has

Hello all, we're using asterisk 1.2.12.1 in an Inbound callcenter using the queue application. If there are many calls in the queue, it sometimes takes up to 30 Seconds before a call is distributed to an agent. For example there are 10 callers in the queue, an Agent is finishing a call and it takes up to 30 seconds before his phone rings again. We're already set the

It seems that I am having a heck of a time explaining my attempts at distributing ACD Queues to the list. Here's one little problem, that's only a piece of the puzzle. dundi.conf: 180q => global_dundi_q_pbx1,100,IAX,dundi1:${SECRET}@${IPADDR}/${NUMBER},nopartial 180q => global_dundi_q_pbx2,200,IAX,dundi2:${SECRET}@${IPADDR}/${NUMBER},nopartial 180q =>

I have something going on that I don't fully understand after a weekend of looking for answers. I have an iax account with Tesco that works flawlessly with the Zoiper client - but is giving me trouble with inbound calls in Asterisk 1.6. After some playing I have ended up with an iax.conf file that looks like this: [general] calltokenoptional = 77.75.0.0/255.255.248.0 maxcallnumbers = 16382

We are receiving phone-calls at one Asterisk box, it forwards them to a different location over IAX. Calls are answered on the other end of this connection. However, the callerID seems to be lost in the process. The 1st box shows the original callerID, on the 2nd box callerID shows the callerID of the 1st box. Apparently the 1st Asterisk box replaces the original callerID with its own. Is

Hello all, I?ve found another issue with the queue application. Assuming I?ve configured a queue with a long periodic announcement and have two queue members assigned. Both queue members are busy at a time, while another caller is joining the queue. After a while the periodic announcement is played back to the caller, in that case it takes about 40 seconds to be played back. If then one of the

Really I am thinking in something and I donot know if it can be used or not: Is it possible to use the Database (to be located in a server) so whenever the agent is login via any server, this will be logged in the database, so all the queues in all the boxes can check with this database to see if the agent is logged in and if it is available or not. Is it possible to be like this? About DUNDi,

Hi, I'm a little surprised, up until 1.4.22 my agents where using an IAX channel to ZoIPer Softphone, however since after the upgrade to .22 we experienced a problem with hangup failure between zoiper and asterisk (look like bug http://bugs.digium.com/view.php?id=13184) i made them switch to SIP instead. Weird thing is that the 'Not In Use' warning message keep showing

Hi all, Can someone help with with Asterisk, SER, and Asterisks Queues? I have three servers: Server A: Asterisk with TE410 connected to PSTN Server B: Asterisk connected to Server A via IAX2 trunk Server C: SER where SIP agents register/connect to What I wanted to do is configure Server A so that it would route certain DIDs to specific UA that are registered in Server C. I don't think

Hi, I do not understand the difference between SIP and IAX, is it only two different protocols or something more special. The problem I have is that I've created two users Aix.conf register => users1:passwd1 register => user2:passwd2 [user1] type=user context=default secret=passwd1 host=dynamic [user2] type=user context=default secret=passwd2 host=dynamic extensions.conf exten

Hi All - Well, after happily existing in a one office environment with asterisk for a few months, I've now decided to start adding in our other offices with their own * boxes and IAX connections (over VPN). Unfortunately, I'm an idiot and I can't get it to work. I'm having some kind of problem with codecs, I guess, but I don't understand what or why. When trying to use

Dale, Sorry for taking so long to answer, I've been traveling. Thanks so much for the suggestion, your solution worked perfectly. I'm not sure why I didn't notice that the IAX trunk was working in the other direction. Once again, thanks for your help. Mitch Date: Mon, 25 Jun 2012 05:44:37 -0500 From: Dale Noll <dnoll at wi.rr.com> Subject: Re: [asterisk-users] IAX Trunk

I would especially look at the CHANUNAVAIL dial status Since it sounds like you are probably qualifying your IAX trunk, that status will be the quickest way to overflow from IAX to TDM. On Sat, May 30, 2015, 11:35 PM Ashwin Surendran < Ashwin.Surendran at now-health.com> wrote: > Hi Matt, > > > > I was a bit concerned on the delay if there might be any when my iax link

The command he gave you was in Asterisk. Why do you not want to call it to try it? Then you can fail over to the other trunk if the IAX link is down. Kind regards, Matt > On May 30, 2015, at 2:03 AM, Ashwin Surendran <Ashwin.Surendran at now-health.com> wrote: > > Many Thanks Carlos, I was hoping to check whether the remote server is > available before I issue the dial in

I'm testing a few IAX trunk scenarios in a controlled lab. From server2 extension 5000 (Server IP Address 172.16.200.212) I dial 6001 which goes across the IAX trunk to server 1 (IP address 172.16.200.210). Instead of ringing the 6001 phone, it plays tt-weasels (the s extension). When I dial 6099 it also plays tt-weasels as it's supposed to, but it's not the tt-weasels under its

Hello, maybe is a stupid question but I'd like know if I'm doing some mistake on IAX trunk configuration. I have two Asterisk (1.6.2.14) with this configuration: A (iax.conf) register => serverb:password at 192.168.142.246 [servera] type=friend host=dynamic trunk=yes secret=password context=phones deny=0.0.0.0/0.0.0.0 permit=192.168.142.246/255.255.255.255 qualify=yes B

On 5/29/15 1:16 PM, Ashwin Surendran wrote: >> Hi, > I have multiple Asterisk servers in various parts of the world all > connected using dedicated VPN?s. > > Each of these servers have iax and dahdi TRUNK configured on them. > > Occasionally the VPN?s fail. > > What I want to be able to do is on my dial plan, use IAX if the asterisk > server can reach the remote