similar to: Set custom file name for automon recordings

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

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!

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.

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.

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]

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 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

I have been trying for a very long time to get asterisk to detect and utilize dtmf tones from my sip clients within my dial scripts. I have set automon=>#9 in my features.conf, I have Dial(....,gWw) in my dial scripts. I have Set(DYNAMIC_FEATURES=automon) as the first script in my extension. I can see the dtmf tones on the wire as SIP INFO packets. Using the Read() app I have verified that * is

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?

Hi list, The default automon (touch monitor) output file name format is: auto-epoch-caller-callee.wav A variable is available to modify the second half: auto-epoch-${TOUCH_MONITOR}.wav But, I can't modify the first half, 'auto-epoch-', with any variables that I know of, including ${MONITOR_FILENAME}. I want to immediately convert this output file to mp3, e.g. with

Roman: I created two bash scripts called Mail2Fax and Fax2Mail for use with the asterisk sever. They leverage the app_txfax and app_rxfax scripts, along with ast_fax. They make using these apps a lot easier, including being able to mail to fax@domain.ca for outgoing faxes and then extracting phone numbers from the subject line! (Makes it easy to use with Sendmail without complex rules /

I'm running * 1.4 and can successfully restart asterisk from the command line with: /usr/sbin/asterisk -r -x "restart gracefully" However, I have a cron job that tries to restart asterisk and gets this error: No such command 'restart gracefully' (type 'help restart gracefully' for other possible commands) Can anyone think of why this is happening? Thanks

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

Hi, I have latest Elastix 64 bit setup and running fine (Asterisk 1.6.2.13) I would like to customize the file name of call recordings: /var/spool/asterisk/monitor/20110511-110858-1305126538.912.wav I would like to include the extension number in the file name. Did a lot of googling but not much help. Pls advice. Thx Sans -------------- next part -------------- An HTML attachment was

I see a lot of attempts by hackers to call 00972595301123? or 011972595115207? or variations but that same 972595 is often present. Can someone break down that dial string with an explanation? The 011 look like an overseas call (from Americas), while the 972595XXXXXX is unclear... -------------- next part -------------- An HTML attachment was scrubbed... URL:

We've deoplyed a number of pure VoIP wireless (wifi & proprietary) phones, but not dect. Is there a simple overview of integrating DECT phones with Asterisk somewhere? I assume the DECT basestation has a multi-account SIP VoIP interface, and the handsets are just plain old dect? Can you push configuration info to individual phones? (Are they individually addressible / configurable

Is there a simple way to get call volume from a particular trunk within the dialplan (for conditional branching)? I suspect we will have to build an AGI script but I'm hoping something new in Asterisk 13 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150627/6774c750/attachment.html>

We're looking at a large wifi phone deployment, and we're looking for wifi phones that: 1. Are SIP compliant (Asterisk friendly) 2. Provision capable (ideally TFTP of own MAC address) 3. Industrial quality (no cheap plastic stuff). 4. Well documented (and none of the "only telco's get documentation" crap) Does anyone have a suggestion? Thanks, MD -------------- next