similar to: AgentRequest() and which agent id?

Hi, I am upgrading from Asterisk 1.4 to 12.4. I am able to authenticate the user and call AgentLogin. But after that when I call AgentRequest I keep getting Agent '1234' is busy. If I put a delay of 5 second or more before calling AgentRequest then it works most of the times. Here's my dialplan: [login] exten => s,1,Background(thank-you-for-calling) same =>

What the minimum Server Specifications do I need to run 200 concurrent channels at the time with .WAV recording (MixMonitor)? It will be connected via VOIP sip account. Codec will be ulaw. Which UK dedicated server provider do you recommend and how much bandwidth do I need? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm pleased to announce that Flash Operator Panel 0.26 has been released! FOP is a GPL'd switchboard type application for the Asterisk PBX. It runs on a web browser with the flash plugin. It is able to display information about your Asterisk box in real time. It is included in FreePBX, Asterisk@Home, DeStar, startShop, and several other projects both free and commercial. You can grab the

My last problem was nicely solved through this mailing list so hopefully this new problem will have the same happy outcome. My situation is that I have many extensions. Here is a sample: [client-phone](!) type=friend host=dynamic secret=XXXXXXXXXX dtmfmode=auto disallow=all allow=ulaw allow=gsm allow=g723 allow=ilbc subscribemwi=no [4165555555](client-phone) secret=xxxxxxxxxxxxxxxxxxxxxxxxxx

Hello, Can someone recommend me where is best place to find Asterisk Expert/Consultant for freelance work? If you are interested to work as a freelancer, you can email me directly. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150902/5a99cbfe/attachment.html>

I have this in my sip.conf: exten => *98,1,Verbose(0,CALLERID number is "${CALLERID(num)}") same => n,VoicemailMain(${CALLERID(num)}@LocalSets,s) same => n,Hangup However, my extensions are set up so that they always show the external number, not the extension: [foobar2](client-phone) secret=xxxxxxxxxxxxxxxxxxxxxxxxxxxxx callerid=Candace <5555551212>

Asterisk 11.1.0 One queue with strategy=leastrecent. (Full queues.conf below.) Occasionally (several times today), a caller will get "stuck" in the queue - there are operators available to take the call, but the caller stays in the queue for a long time. Any idea what might cause this, or where I can start looking to debug it? I'm going to start digging through the queue log

On 2017-04-18 02:42 AM, Pete Mundy wrote: > Try this: > > asterisk -r > core set verbose 10 > [get user to trigger fault] > [examine console output, and post to list if still unclear] > > If you don't solve it yourself, then we'll be able to help further once > we've seen the output. I can't see much more than at my previous debug level but here it is

Hello, I have Asterisk version: Asterisk SVN-branch-11-r420435 I have the following code: exten => 303,1,NoOp(Dialing ${EXTEN}) ? ? ? ? same => n,NoOp(DBKey = ${DBKey}) ? ? ? ? same => n,DB_DELETE(office/${DBKey}) ? ? ? ? ? same => n,Playback(auth-thankyou) ? ? ? ? same => n,Hangup() And I get the following error: [2014-08-09 18:00:30] WARNING[4338][C-00000067]: pbx.c:4869

On Fri, 12 Jun 2015 11:49:05 -0700 John Kiniston <johnkiniston at gmail.com> wrote: > Try this for CHAN_SIP: > > same => n,Set(Peer=${SIPCHANINFO(peername)}) ; Get the peer > same => n,Set(MailBox=${SIPPEER(${Peer},mailbox)}); Get the > mailbox same => n,VoicemailMain(${MailBox}@LocalSets,s) ; If we > have a mailbox defined log into it Perfect.

On 12/2/19 11:52 AM, Joshua C. Colp wrote: > So I know that AMI is listening and I can talk to it.  Here is the > main log" > > [Nov 27 06:16:00] VERBOSE[101155] asterisk.c: Remote UNIX connection > [Nov 27 06:16:00] VERBOSE[101245] asterisk.c: Remote UNIX connection > disconnected > [Nov 27 06:16:01] VERBOSE[101244] manager.c: Manager

This just started after upgrading to 11.11.0. After a call is completed (both ends hang up) the call still shows as active. # asterisk -x "core show channels" Channel Location State Application(Data) SIP/thinktel-0000000 (None) Up AppDial((Outgoing Line)) SIP/4164251212-00000 4165555555 at LocalSets Up Dial(SIP/thinktel/4165559999) 2 active

We have a template for extensions and voicmail. They look like this: exten => %ACCOUNT%,1,Verbose(0,Entering extension %ACCOUNT%) same => n(DialDesk),Verbose(0,${CALLERID(all)} Calling ${EXTEN}) same => n,Dial(SIP/%ACCOUNT%,30) same => n(VoiceMail),Set(CDR(userfield)=VoiceMail) same => n,Verbose(0,${CALLERID(all)} going into voice mail for %ACCOUNT%)

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

On 12/3/19 3:04 PM, Joshua C. Colp wrote: > >     The AMI command, after the login, looks like this: > > > >     Action: Originate > >     Channel: SIP/outgoing/%%(destination)s > >     Context: LocalSets > >     CallerID: Vybe Consulting Inc Fax Service <5555551212> > >     Exten: sendfax > >   

Hello, Let say all the SIP devices will be registered on the proxy like kamailio. Agent is a member of Support and Billings Queues on the asterisk servers. Support queue on "Server A" and Billings Queue on "Server B" for example. This will be done via RealTime Queue. I want Agent to dial 1234 on a sip device and it will prompt to enter a pin number to Login via

I recently upgraded from Asterisk 13.19 to 16.6.1. Everything is working fine with a few minor tweaks except outgoinf fax. Incoming works fine. I do outgoing faxing through an AMI call. Here is the output from the security log: [Nov 27 06:16:05] SECURITY[101222] res_security_log.c:

Hello, I am wondering has anyone used Live Recording (monitor or mixmonitor) on to Storage Server via network 1 Gigabit connection? Does it perform well, let say about 50 live recordings at the same time. I am planning to make some system changes at work. I would like to put Asterisk VM on a ESXi host and the datastore will be hosted on Storage Server. On a ESXi host, there will be a few

I would like to develop a Call Center Dialer (outbound and inbound calls) and it would use AMI method to communicate with Asterisk Server. A daemon would need to run in the background, would you recommend coding in PHP or Node.js? which would be much faster and stable. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am having troubles with sending faxes. I hope someone can help me work out a better method. Basically we have a special address that our users can send to. It winds up on our Asterisk server which runs a Python script that parses the message for attachments and the phone number from the recipient address. The attachments are converted to TIFF and stored in a folder with various information