search for: automixmon

Hi all, I have 2 quick question regarding the file location and post record command of the recording using automixmon in features.conf. With the normal monitor/mixmonitor applications you can change the location of where the recordings will be stored, by changing the MONITOR_FILENAME variable. I tried changing the TOUCH_MIXMONITOR_OUTPUT variable to include a path but it sill puts the recorded file in /var/spool/...

Hi folks, I was using the following featuremap: blindxfer => *1 disconnect => *9 atxfer => *2 parkcall => *7 automixmon => *0 and everything worked. Then the need arouse to use some features like automixmon during a conference, but MeetMet has the * key bound to the (admin) menu. Thus, in order to enable features like automon and transfers even during a conference, I tried to swap * fro # in the featuremap:...

Hi list! I'm trying to configure Asterisk to record incoming calls, if the called press *3. I added in features.conf: automixmon => *3 then, in my dialplan: exten => 1,n,Dial(SIP/00493511111111,20,RcxX) Well, if I **CALL** a number I'm able to record the call, but if I'll be called, and press *3 nothing happens... In the console I can't see anything, too. Could you suggest me what is wrong? Thanks Luca...

...I want to be able to stop it during a call and maybe restart it. I tried using the value defined in [featuremap] but it starts another MixMonitor application even if there already one instead of stopping it. Any idea on how I can stop the MixMonitor application while it is running? [featuremap] automixmon => *1 I tried also to use the [applicationmap]] but it doesn't seem to work. Pressing #1 do nothing. Here my dialplan: 9999 => { Set(__DYNAMIC_FEATURE=pauseMonitor); MixMonitor(test); Dial(SIP/1000 at srv01,30,TtX); } [applicationmap]...

Asterisk 13.8 Is CALLERID(all) supposed to wok for pjsip? When I do this: exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) same => n,Dial(PJSIP/phone123, 30) I expect the callerid to be as set, but is always seems to be "phone123", the name of the endpoint. Andrew -------------- next part -------------- An HTML attachment was scrubbed... URL:

...[DEADDEADBEEF] type=endpoint timers_sess_expires=1800 device_state_busy_at=0 dtls_cipher= from_domain= dtls_rekey=0 dtls_fingerprint=SHA-256 direct_media_method=invite send_rpid=false pickup_group= sdp_session=Asterisk dtls_verify=No message_context= mailboxes= named_pickup_group= record_on_feature=automixmon dtls_private_key= named_call_group= t38_udptl_maxdatagram=0 media_encryption_optimistic=false aors=DEADDEADBEEF rpid_immediate=false outbound_proxy= identify_by=username inband_progress=false rtp_symmetric=false transport=transport-udp rtp_keepalive=0 t38_udptl_ec=none fax_detect=false t38_udptl_na...

On 06/05/2017 at 11:30 AM, Joshua Colp wrote: > On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote: >> On 06/04/2017 at 01:41 PM Telium Technical Support wrote: >>> Just a guess (without knowing about your network), but are the two ends >>> points on public networks and visible to one another? If not the reinvite >>> may be passing an internal (nat'ed)

...hit '*1' to record call. filename: wav,auto-1250793354-24-22,m TOUCH_MONITOR_OUTPUT is set to "auto-1250793354-24-22" but /var/spool/asterisk/monitor/ does not contain any new file. Dial options are "twhx" and in features.conf there is: [featuremap] automon => *1 automixmon => *3 Is there anything else to follow to get monitoring working for asterisk 1.6 or is it a bug in the 1.6.2.0-beta4 ? Thanks in advance -- Stefan Tichy ( asterisk2 at pi4tel dot de )