search for: mixmon

Dan et al; This looks like a perfect solution. However, I have one issue. If I initiate the macro manually (put it in the proper context/dialplan) it works. I see the *.wav file being created and growing in the /var/spool/asterisk/monitor directory. If I try to implement it adding the MixMonApp => *1,self/both,Macro,mixmon to the [applicationmap] in features.conf, I cannot get it to work. Steps. 1. added the example macro to the dialplan in extensions.conf 2. added the line MixMonApp => *1,self/both,Macro,mixmon to the features.conf file under [applicationmap]...

Hello Everyone; I am looking for a solution to record calls that come into our Asterisk server. I am hoping for something that is easy to use - however, if I have to modify it to make it easier to use, I do not mind. Does anyone know of any opensource or otherwise solutions out there that I can try out? Thanks much. Glen

....c: -- Channel 0/6, span 7 got hangup, cause 27 [Mar 18 17:04:15] VERBOSE[27353] pbx.c: == Spawn extension (incoming-pri, sw-30-218543080, 11) exited non-zero on 'DAHDI/189-1' [Mar 18 17:04:15] VERBOSE[27353] chan_dahdi.c: q931_hangup: other hangup [Mar 18 17:04:15] VERBOSE[27354] app_mixmonitor.c: == MixMonitor close filestream [Mar 18 17:04:15] VERBOSE[27353] chan_dahdi.c: NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null, hold-state Idle [Mar 18 17:04:15] VERBOSE[27353] chan_dahdi.c: NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null, hold-stat...

Hi, I have Asterisk 1.2 + Adit600 Channel bank(which gives analog output and also takes PSTN lines into the Asterisk system). Conversations recorded by the ASTERISK comes in two separate Files: xxxxxx.0-in (GSM Audio) for the Asterisk Extension Side of the conversation; xxxxxxx.0-out (GSM Audio) for the Caller's side of the conversation. I have Quick Time Player to playback the

...in Asterisk/VOIP/CM I would like to make sure that this system design can work: Cisco 2811 Voice Gateway - sip trunk1 - asterisk on linux box - s?p trunk2 - Cisco Call Manager 6.0 There is also a Siemens Hicom old pbx connected with QSIG to the Voice Gateway. I would like to record all calls with mixmon going through Asterisk. Is it possible? Also if there is call forward/find me/call transfer etc. in Cisco CM, how can I find the change in Asterisk? I'd like collect Asterisk CDR data in SQL, as I know there is also a CDR data in CM. Is there some kind of CALL ID which is exactly the same in...

...== Spawn extension (from-trunk, s, 6) exited non-zero on 'SIP/358-106-000000c0' -- Executing [h at from-trunk:1] Macro("SIP/358-106-000000c0", "hangupcall,") in new stack -- Executing [s at macro-hangupcall:1] GotoIf("SIP/358-106-000000c0", "1?endmixmoncheck") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s at macro-hangupcall:9] NoOp("SIP/358-106-000000c0", "End of MIXMON check") in new stack -- Executing [s at macro-hangupcall:10] GotoIf("SIP/358-106-000000c0", "1?nomeetmemon")...

Hello, I am not an asterisk expert but i am developing a web application that is using asterisk. I would like to know if it is possible to configure a Manager to only monitor a special extension, and of course how to do that. The application is written in java and is using asterisk-java. Right now i have one manager that i am connected to and i receive all the events but i would like to have

...'s Topics: > > 1. Re: asterisk-users Digest, Vol 81, Issue 27 (Steve Edwards) > 2. Re: Asterisk FOP (Doug Lytle) > 3. Re: Asterisk FOP (Flavio Miranda) > 4. Re: Asterisk FOP (Doug Lytle) > 5. Re: IAX2/0.0.29.199 (Satish Patel) > 6. Re: Call Recording using MixMonitor - close, but would like > some more words of wisdom. (Dan Journo) > 7. Re: Call recording - methodology (Dan Journo) > 8. Re: Asterisk FOP (Flavio Miranda) > 9. Re: send voicemail to multiple emails (vip killa) > 10. Ubuntu "*-server" kernels [was: Re:...

Hi this message give me when I calling a number than actually not busy: "Dial failed due to trunk reporting BUSY - giving up" max channel is unlimited and sometimes it dial number ok but most of the time it gives me this error. Please inform me how can solve this problem. thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

...(ONETOUCH_REC)=RECORDING") in new stack -- Executing [s at macro-one-touch-record:8] Set("SIP/1010-00000162", "MASTER_CHANNEL(REC_STATUS)=RECORDING") in new stack -- Executing [s at macro-one-touch-record:9] Set("SIP/1010-00000162", "AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack -- Executing [s at macro-one-touch-record:10] MixMonitor("SIP/1010-00000162", "2012/08/21/out-7190000000-1010-20120821-183119-1345595479.530.wav,a,") in new stack == Begin MixMonitor Recording SIP/1010-00000162 -- Executing [s at macro-one-...