similar to: Record call ends in 10min

In my dial plan I have these two lines: exten => _NXXXXXX,n,Set(recordfilename=${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},MST,%C%y-%m-%d-%H%M)}.wav) exten => _NXXXXXX,n,MixMonitor(${recordfilename},b) How to add "if" statement to execute these line only after let say 5pm. To record conversation only after 5pm. -- Joseph

I have following line in a context: ... exten => _587NXXXXXX,n,Set(recordfilename=${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},MST,%C%y-%m-%d-%H%M)}.wav) exten => _587NXXXXXX,n,MixMonitor(${recordfilename},b) ... It records the conversation but it ends prematurely, after 10min. Why? Where is the setting to records until a user hangup the handset. -- Joseph

Hello list ! I have some configuration issues with queues, but I'm sure they are minor and for someone who has already configured queues it could be trivial. This is my queue configuration : [VC_support_queue] musicclass = default strategy = ringall timeout = 20 retry = 5 wrapuptime=15 autofill=yes autopause=no maxlen = 0 setinterfacevar=yes announce-frequency = 0

Hi , I;ve Asterisk 1.6.0 with static agents (sip softphones with extns 100 & 101 ) in a queue..When a caller arrives in queue , it lands on first 100 , 100 then does a blind transfer to 101 .. so that the caller can converse with 101 .. strangely enough the queue_log shows : 1253814090|1253814090.12|55365|NONE|ENTERQUEUE||98221232123

I have two server Server_A(outbound call) for agent login and agent make a outbound call from here and pass into server Server_B call extension.conf exten => _91XX.,1,AGI(agi://127.0.0.1:4577/call_log) exten => _91XX.,n,Dial(SIP/${EXTEN}@192.168.53.197,,tToR) exten => _91XX.,n,hangup() Server_B[192.168.53.197] for call forwarding extension.conf exten =>

Dear Folks, [Test_Context] exten => _911.,1,AGI(agi://127.0.0.1:4577/call_log) exten => _911.,2,Set(CALLERID(num)=xxxxxxx) exten => _911.,3,Set(CALLTIME=${STRFTIME(${EPOCH},Asia/Calcutta,%d-%b-%Y-%H-%M-%S)}) exten => _911.,4,Set(RECSUBDIR=${STRFTIME(${EPOCH},Asia/Calcutta,%d%b%Y)}) exten => _911.,5,Set(${CALLERID}=${CALLERID(num)}) exten =>

I am using Elastix. Asterisk is used for PBX application in Elastix. I want to make customize MOH. But not able to use MOH. I make simple extension in asterisk conf file but no success :( Below are the details of configuration files. Even default MOH is also not working.... *Asterisk Version 1.6.2.17.2 * *1) Extension.conf* [incoming] exten => 6000,1,Answer exten =>

I am using Elastix. Asterisk is used for PBX application in Elastix. I want to make customize MOH. But not able to use MOH. I make simple extension in asterisk conf file but no success :( But when I used Vanilla Asterisk then All things are working.... Below are the details of configuration files. Even default MOH is also not working.... *Asterisk Version 1.6.2.17.2 * *1) Extension.conf*

I am having a strange problem with MOH. Say I have two users, A and B. I can set MOH in the extension for B and if A calls B and B hits hold, A will hear B's hold music. If however A hits hold, it goes to the default music. If I pull the setmusiconhold from extensions.conf and use musicclass in sip.conf under the peer A, I get the same thing. Peer A has musicclass set and A calls B and B

Does anyone know how to set the music on hold class with the Manager Interface in 1.8? Here is what I am using but I end up just getting no music when I put this in place, when I remove it the default is back. The classes I am setting work elsewhere just fine. I did not include the opening of the socket, logging in etc because that's all working fine along with other things I am doing within

Hello list, what musiconhold class has priority : - field "musiconhold" of the SIPaccount and field "musiconhold" of a queue or - Set(CHANNEL(musicclass)=) ?? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110201/b27a0534/attachment.htm>

Todd How do you have the context in the phones sip configs set? Bryant From: "Todd Reese" treese65 at gmail.com Hi all, I've been have problems with getting this system on line and would like to acquire some help with the extensions.conf. My current problem is that the phones won't dialout.on the VOIP lines listed as dialout1, dialout2, dialout3. This version of asterisk

Hello About Record(), ATFT 2nd Edition says that "if the filename contains %d, these characters will be replaced with a number incremented by one each time the file is recorded." Problem is, the documentation doesn't explain how to refer to this filename later in the dialplan :-/ In this particular example, I want to move the file to the web server's /htdocs so users can

Hi, Has anybody else noticed that MOH does not play on parked calls in 1.6.2.14? Or is it just my setup? MOH seems to work in every other respect (Call Held or in-Queue), but when a call is parked, the logs show MOH being started, but the parked party hears nothing. The verbose logs show the following. Any thoughts on whet to check next? Thanks, Steve ### Call comes in here and is answered

Hi, Since SetMusicOnHold() is being deprecated, how do we set the channel musicclass from an AGI script? Last time I checked you can't call dialplan functions from AGI. Thanks. -- James -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141005/03df3f3b/attachment.html>

Hi everyone, I'm having a hard time figuring this issue out, we just switched from a T1 PRI to a SIP trunk provider and that's when the issue started. Now when someone forwards all calls on their phone to a cellphone, when a customer calls in, Asterisk correctly calls the cellphone and connects the call, but there is a long delay before the audio starts, basically for the first 6-10

Hi... I just had a real quick and simple question... I have a asterisk implementation setup w/ real time off of a mySQL database for SIP peers and queues, voicemail, agents etc... I after the upgrade to asterisk 1.4.3 there are some new configuration features i would like to use. I was wondering if i could just add to the database table a column for the new config option? if this will work or

A call comes in and goes into the queue, the queue dials a sip channel using a macro. The macro plays a set of options to the callee and if the callee presses 3 it sets MACRO_RESULT=CONTINUE and the macro ends. For some reason the caller goes back into the queue rather than continueing on in the dial plan. Why is this, i could have sworn in 1.2 if i set MACRO_RESULT=CONTINUE that the

Hi All!! Prompt how to record all calls passing through certain span? --- Thanks...

I have a customer that needs to record all calls coming in and out. The problem I am having is when a call comes in to the operator and it is transferred to another extension. The first mixmonitor begins recording when the operator picks up but the recording stops when the call is transferred. I need to have a single recording for the complete call no matter how many times it is transferred.