similar to: features.conf / DTMF / automon hell

Actually the docs for the Queue application say: 'w' -- allow the called user to write the conversation to disk via Monitor 'W' -- allow the calling user to write the conversation to disk via Monitor couldn't get these to work tho. Does this mean I can do one touch recording with agents, or does it mean I can use the monitor() command? Very confusing... Doug.

I thought it was quite easy to implement but I cannot get one-touch recording to work. Here are the changes what I did: I restarted Asterisk after the change (because reload does not work for changes in features.conf). I press *1 on the Polycom IP600 phone to record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Any suggestions? Here is the console log:

Hi all. I posted this earlier but never got any advice that helped. If anyone knows how to get this going, I'd appreciate some advice. I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? In extensions.conf [globals] DYNAMIC_FEATURES=>automon [default] exten =>

I thought it was quite easy to implement but I cannot get one-touch recording to work. Here are the changes what I did: I restarted Asterisk after the change (because reload does not work for changes in features.conf). I press *1 on the Polycom IP600 phone to record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Test A: Outside line calling in

Hi, 1) I want to create add *1 call recording and wanted to know whether the file is created during recording or only after? I want to syncronise the recorded files with my web server (on a different machine (Windows)) so I need a way of telling when the recorded call has ended before copying it over. 2) I tried setting up *1 in features.conf but when I press *1, all that happens is

Hello asterisk-users, I setup my asterisk to support several features like automon,blindxfer,atxfer,parkcall etc. by using features.conf and the global variable DYNAMIC_FEATURES=automon#blindxfer#atxfer#parkcall#disconnect in extension.conf. Every Dial() command in my diaplan has the appropriate parameters out of {tTkWwW}. For calls from my SIP phones everything works fine. Pressing #1 will

I'm using Asterisk 1.4 but my "one touch recording" is not working: feature.conf automon => *1 extension.conf [globals] DYNAMIC_FEATURES=>automon exten => 117,1,Dial(SIP/117,30,jrwW) When I press "*1" on incoming call asterisk is not recording anything. Did I miss any setting? -- Joseph

Hi What causes deadlock? Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x82acb10', 10 retries! Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x8298160', 10 retries! Here is the portion of the log: Apr 5 14:02:42 NOTICE[23363] chan_zap.c: Got event 18 (Ring Begin)... Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing

Hi list, Can someone please explain how to get one touch recording (automon) to work reliably? I'm using Asterisk 1.4.14 on a Debian etch system. My current configuration includes the following settings: In /etc/asterisk/sip.conf: [2000] ; Siemens Gigaset S675 IP wireless SIP phone. type=friend secret=1234 context=phones-j dtmfmode=rfc2833 qualify=yes

automon is not working for me with asterisk 1.4.22.1 in extension.conf [globals] DYNAMIC_FEATURES=>automon dial is with "w" feature.conf automon => *1 -- Executing [11 at internal:1] Playback("SIP/218-007556b0", "transfer") in new stack -- <SIP/218-007556b0> Playing 'transfer' (language 'en') -- Executing [11 at internal:2]

Hello List, Maybe I'm mistaken, but, shouldn't the 'fname_base' variable of 'Monitor' application affect the file name generated through 'automon' feature? I initialized this variable with a value as follows: Set(fname_base=auto-${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}) a. Should I use 'fname_base' in uppercase (FNAME_BASE)?

Guys. How are filenames determined for automon and queue recordings enabled on queues.conf? I see the names have some tomestamps or something but is there a way to predefine the filenames to use? Thx!

On Wed, Aug 8, 2018 at 1:53 PM, Daniel Journo <dan at keshercommunications.com> wrote: > > Prior to a call entering a Queue, I set __DYNAMIC_FEATURES=NewRecordApp. > > AgentA answers and is able to use that feature code. > > If AgentA performs an attended transfer of a call from a queue to > AgentB, the > > feature code no longer works. > > > > It only

There is option automon => *1 in features.conf As I understand when *1 pressed during conversation => recording should begin. But unfortunately it doesn't work for me. I use CVS-HEAD-01/27/05 Does anyone has that feature working? Thanks -- Best Regards VladK

Hi, I think I've identified an issue and just want to check before completing a bug report. Prior to a call entering a Queue, I set __DYNAMIC_FEATURES=NewRecordApp. AgentA answers and is able to use that feature code. If AgentA performs an attended transfer of a call from a queue to AgentB, the feature code no longer works. Cases that do work are as follows... Calls using both Queue() and

Hi there, We used to record all the calls with the Monitor function. Now, I haveimplemented on-demand recording with automon instead... Everything is working fine apart from the generated filename, which as per all docs, should be auto-epoch-caller-callee....but in my case, it is auto-epoch-who_started_record-the_other_end. Thank you all in advance. Regards, Marta -------------- next part

I've got automon up and recording calls on demand from information I found in the list archives, however instead of ending up with one monolithic file, I've got a -in and -out version of the files in my monitor directory? Anyone have suggestions how I could end up with a monolithic file that does what I want?

Don't you just hate it when something was working and when you come to use it in anger it's broken :-( Something in the, fairly, recent series of Asterisk updates has broken DIGITAL call passthrough. I've an ISDN PBX behind my Asterisk Box (PRI ISDN comes into port 1 of a Digium Wildcard and the PBX it connected to port 2 via an ISDN crossover cable). This PBX used to be able to

Hi all, What's the best method to send automon call recordings (*1) to the voicemail box of the Asterisk user? Do you have to trap hangups, etc, or is there some global variable that can be set? Thanks! S.

Hi all, If I activate recording for an extension everything is OK. but If I activate call recording on demand i am non able to start recording In principle I should have to press *1, as indictaed in features.conf (I am using almost last asterisk code, updated 2 days ago from svn, version SVN-branch-1.2-r39379M ) Actually it produce no effect at all I am using FreePBX interface, and I saw