search for: silencethreshold

Hello there, I tried to found documentation about openloop disconnect on Asterisk/Zaptel. And up to now, I didn't find anything. Is openloop disconnect supported by zaptel/wcfxo drivers? Other question: maxsilence and silencethreshold don't seem to have any effect on voicemail. Could it be possible that in certain situation it just wont work? I tried adding some verbose in app.c to have the totalsilent variable printed out and it remains a 0 forever. Finaly, does callprogress=yes should be working down here in Cana...

...82284927 (null) (Fmt: 4) Apr 23 08:00:26 NOTICE[28319]: app_amd.c:134 isAnsweringMachine: AMD using the default parameters. -- AMD: initialSilence [3500] greeting [1500] afterGreetingSilence [300] totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence [50] maximumNumberOfWords [5] silenceThreshold [256] -- AMD: HANGUP any help is highly appreciated. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090423/e2d62015/attachment.htm

Hi all, i use asterisk 11.14.0 and I suspect that the voicemail application counts the time wrong. In my voicemail.conf: [general] minsecs=3 maxsilence=5 format=wav maxsecs=180 silencethreshold=140 [...cut..] In the asterisk-cli: [Jan 26 15:23:49] -- Executing [s at macro-voicemail:77]VoiceMail("SIP/XY-0005175a", "aNumber,su") in new stack [Jan 26 15:24:04] -- <SIP/YX-0005175a> Playing 'beep.gsm' (language 'de') [Jan 26 15:24:04] --...

...] logger.c: -- AMD: SIP/sip.broadvoice.com-098c4aa8 6079362172 (null) (Fmt: 4) Jan 24 17:01:37 VERBOSE[31455] logger.c: -- AMD: initialSilence [8000] greeting [1500] afterGreetingSilence [300] totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence [50] maximumNumberOfWords [5] silenceThreshold [256] Jan 24 17:01:44 DEBUG[31437] chan_sip.c: Auto destroying call ' 2223f71d5ccc73cc3f9b32a64fba2286@sip.broadvoice.com' Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command 'Command' Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command 'Command' Ja...

...4 at CC2-c93c,1", "") in new stack -- AMD: Local/0445540881644 at CC2-c93c,1 5540881644 (null) (Fmt: 64) -- AMD: initialSilence [2500] greeting [1500] afterGreetingSilence [800] totalAnalysisTime [5000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [5] silenceThreshold [256] == Spawn extension (CC2, 0445540881644, 2) exited non-zero on 'Local/0445540881644 at CC2-c93c,2' -- Executing [h at CC2:1] DeadAGI("Local/0445540881644 at CC2-c93c,2", "agi://localhost/updateCallStatus.agi?callStatus=hangupcc2") in new stack -- AGI Scri...

...gt; count 2. What can be the reason? It is not silence. > > Are you sure? Yes, im sure. I have looked at the time and talked for at least 4 seconds. In CLI log are 5-6 seconds visible between open to writing and Hang up. Nevertheless, Asterisk writes about two seconds. > The value for silencethreshold (140) is unusually large. It would be worth a try to set the value down. In asterisk 1.6 this value was still good. But that is far back again... I will write again. Greetings Dominique

hi, i am trying to do autoattendant but failing. as in the manual i inserted the background(welcome-mainmenu) file so that after the sound the caller can dial the extension he wants to call. i figured that the background sound wasn't coming in the asterisk. how do we do this without first loading the welcome message? for example after certain rings the caller can dial the extension no to

Hi, We have a system that recorded voicemail for about an hour after the caller hungup. I'm going to put a timeout on it but is there anything to look for that can help prevent this? The system is running on a telenet line in Belgium. The answer dialplan I used was: [macro-stddial] exten => s,1,Answer exten => s,2,Playback(transfer) exten => s,3,Dial(${ARG2},60) exten =>

...for voice mail. But the log says Invalid sender. Sender = asterisk@localhost.localdomain and not asterisk@somedomain.com as assigned in conf file. VM Config: [general] format=gsm|wav49|wav serveremail=asterisk@somedomain.com <----Actual file has a valid email. attach=no maxmessage=30 silencethreshold=128 maxsilence=10 fromstring=Asterisk Mail emailbody=${VM_NAME} <${VM_MAILBOX}>\n\nYou have received a ${VM_DUR} long message from ${VM_CALLERID}. The Message was left on ${VM_DATE} [bell] 100 => 1234,User1,user1@somedomain.com <------ Actual file has a valid email. Than...

hi all, you'll have to excuse the ignorance (i'm a software guy, not a telcom guy..) Is there any way to know if a channel has been answered by an automatic system (like voicemail) rather than a human being? Specifically, I want to use a .call to make a call on a channel and only do something if a person answers, not a machine of any kind. Is this even possible, or is an answered

...o apply. Below is the complete voicemail.conf file. Any help would be appreciated. David dfilion@dotality.com ============================== Begin ============================ ; ; Voicemail Configuration ; [general] format=wav49|gsm|wav serveremail=asterisk attach=yes skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 mailcmd=/usr/local/postfix/sbin/sendmail -t tz=eastern attach=yes saycid=yes dialout=default callback=default review=yes operator=no envelope=yes delete=no nextaftercmd=yes [zonemessages] eastern=America/New_York|'vm-received' Q 'digits/at' IMp central=America/Chic...

...+----------+-----------+-----------------+------------+---------+ 1 row in set (0.01 sec) mysql> \q Bye Voicemail.conf looks like this: ; ; Voicemail Configuration ; [general] ; Default formats for writing Voicemail format=wav49 serveremail=support@tss.net attach=yes skipms=3000 maxsilence=5 silencethreshold=128 maxlogins=5 dbuser=asterisk dbpass=xxxxxxx dbhost=localhost dbname=asterisk pbxskip=yes fromstring=Passnet Voicemail [zonemessages] eastern=America/New_York|'vm-received' Q 'digits/at' IMp central=America/Chicago|'vm-received' Q 'digits/at' IMp central24=Ameri...

...at CC2-dad7,1", "") in new stack -- AMD: Local/016566275538 at CC2-dad7,1 016566275538 (null) (Fmt: 64) -- AMD: initialSilence [3500] greeting [2500] afterGreetingSilence [800] totalAnalysisTime [5000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [5] silenceThreshold [256] AMD just stops and it takes over a minute until the line is dropped. The same number dialed through Zap works without a hitch. What could be the reason? If I dial the same number without AMD I can talk to the other person so I know the SIP line is fine. -- Telecomunicaciones Abiertas...

...verybody, I'm trying my Asterisk to send emails when a new message arribes to a voicemail user but no email arribes. my voicemail configuration is the following: VOICEMAIL.CONF: [general] format=wav serveremail=asterisk at mydomain.com attach=yes maxmsg=20 maxsecs=180 minsecs=3 maxsilence=10 silencethreshold=128 maxlogins=3 fromstring="My Asterisk" When I look at maillog file, this is what I get: * n7OCivth003603: from=root, size=5340, class=0, nrcpts=1, msgid=< Asterisk-1-227856683-222-3479 at MYSERVER>, relay=root at localhost * n7OCiw9W003604: from=<root at localhost.localdomain&...

Where are user's voicemail passwords stored and how does the asterisk administrator change them? TIA, Jeff Heath

...| 0 | 0 | voicemail.conf | default | minmessage | 5 | | 8 | 0 | 0 | 0 | voicemail.conf | default | maxsilence | 5 | | 9 | 0 | 0 | 0 | voicemail.conf | default | silencethreshold | 128 | | 10 | 0 | 0 | 0 | voicemail.conf | default | skipms | 3000 | | 11 | 0 | 0 | 0 | voicemail.conf | default | review | yes | | 12 | 0 |...

Hello Everyone, Can someone point me to reliable links on how to tweak Asterisk AMD I am calling a number and have to two files to play depending if it is a real person or an answering machine. Most everytime Asterisk calls it thinks it is an Answering Machine and it starts playing the AMD message, instead of the delivering the "1st real message" Any hints? Thanks in advance, -C

Hello, I have setup an Asterisk 1.2.7.1 system, with a working voicemail box: /etc/asterisk/extensions.conf exten => 83086921,1,Answer exten => 83086921,2,Dial(SIP/stefan,5,r) exten => 83086921,3,VoiceMail,u111 exten => 83086921,4,Hangup exten => 83086921,103,VoiceMail,b111 exten => 83086921,104,Hangup /etc/asterisk/voicemail.conf [default] language=de 111 =>

...e for '12' I have searched for solution a lot can Any one of you let me know how can I solve this issue do I need to apply any patch for asterisk Here is voicemail.conf config file [general] format=wav49|gsm|wav serveremail=asterisk attach=yes skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 saycid=yes [zonemessages] eastern=America/New_York|'vm-received' Q 'digits/at' IMp central=America/Chicago|'vm-received' Q 'digits/at' IMp central24=America/Chicago|'vm-received' q 'digits/at' H 'digits/hundred' M ...

...e pin in the voicemail.conf file there is no problem. I tried looking on internet for any information, but I found nothing useful. Does anybody have any insight on why I can't change my voicemail pin via the Sip phone? Thanks in advance. Here is my voicemail.conf file: maxsilence = 10 silencethreshold = 128 maxlogins = 3 emaildateformat = %A, %B %d, %Y at %r tz = central dialout = outbound sendvoicemail = yes callback = outbound review = yes nextaftercmd = yes [zonemessages] eastern = America/New_York|'vm-received' Q 'digits/at' IMp central = America/Chicago|'vm-received'...