Hi everybody In advance sorry for my bad english and if my problem was already exposed (I didn't find any tips in the mailing list archive. Bad luck) I have some questions about asterisk 1.6 release : 1) how can I do a n+101 priority jumping if a SIP canal is busy ? I read that the general parameter "priorityjumping" is depreciated in the 1.6 release and I already try the "j" option in dial() application but no way. Here a sample of my simple dialplan : exten => 101,1,Ringing exten => 101,2,Answer() exten => 101,3,Dial(SIP/quentin,10) exten => 101,n,VoiceMail(101 at default,u) exten => 101,n,Playback(vm-goodbye) exten => 101,n,Hangup() exten => 101,104,Playback(busy) exten => 101,n,Wait(3) exten => 101,n,VoiceMail(101 at default,b) exten => 101,n,Playback(vm-goodbye) exten => 101,n,Hangup() 2) about asterisk voicemail maximum message limit, is it possible to send a notification mail to an user if his vmbox is full ? How can i do that if it's possible. In advance, Thank. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090724/e34e133e/attachment.htm
harry R wrote:> Hi everybody > > 1) how can I do a n+101 priority jumping if a SIP canal is busy ? > I read that the general parameter "priorityjumping" is depreciated in > the 1.6 release and I already try the "j" option in dial() application > but no way.You'll want to use DIALSTATUS exten => 101,1,Dial(SIP/quentin,10) exten => 101,n,NoOP(Dial Status: ${DIALSTATUS}) exten => 101,n,NoOP(Hangup Cause: ${HANGUPCAUSE}) exten => 101,n,Goto(s-${DIALSTATUS},1) http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."
Here's how I think your dialplan should look: exten => 101,1,Ringing exten => 101,2,Answer() exten => 101,3,Dial(SIP/quentin,10) exten => 101,n,VoiceMail(101 at default,u) exten => 101,n,Playback(vm-goodbye) exten => 101,n,Hangup() exten => 101-BUSY,1,Playback(busy) exten => 101-BUSY,n,Wait(3) exten => 101-BUSY,n,VoiceMail(101 at default,b) exten => 101-BUSY,n,Playback(vm-goodbye) exten => 101-BUSY,n,Hangup() for question 2, this depends on how voicemail.conf is setup. By default, an asterisk mailbox is considered "Full" when 100 messages are stored (see voicemail.conf.samples), but this can be changed up to 9999. Therefore you would just need a job to look for the creation of the max file and send a mail accordingly. I'm not enough of a scripter to do this in bash, but here's a find and grep that would tell you when a user's box was full (based on default params): find /|grep msg0099.txt|grep INBOX _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of harry R Sent: Friday, July 24, 2009 7:44 AM To: asterisk-users at lists.digium.com Subject: [asterisk-users] dialplan tips Hi everybody In advance sorry for my bad english and if my problem was already exposed (I didn't find any tips in the mailing list archive. Bad luck) I have some questions about asterisk 1.6 release : 1) how can I do a n+101 priority jumping if a SIP canal is busy ? I read that the general parameter "priorityjumping" is depreciated in the 1.6 release and I already try the "j" option in dial() application but no way. Here a sample of my simple dialplan : exten => 101,1,Ringing exten => 101,2,Answer() exten => 101,3,Dial(SIP/quentin,10) exten => 101,n,VoiceMail(101 at default,u) exten => 101,n,Playback(vm-goodbye) exten => 101,n,Hangup() exten => 101,104,Playback(busy) exten => 101,n,Wait(3) exten => 101,n,VoiceMail(101 at default,b) exten => 101,n,Playback(vm-goodbye) exten => 101,n,Hangup() 2) about asterisk voicemail maximum message limit, is it possible to send a notification mail to an user if his vmbox is full ? How can i do that if it's possible. In advance, Thank. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090724/31996a84/attachment.htm
Hello, i?ve a question about the Meetme Options. How could i play a enter and leave sound but without recording the user name first. I just want a User joined conferenc and a user leaved. With the i or I Option i have to record the name first. Is there any way of doing this? As i can see in the Meetme help the background agi couldnt be used on non dahdi channel which i will have on this server cause there is no direct Pri link, just SIP. iam using asterisk 1.6.1.1 best regards steve F?r weitere Fragen stehen wir gerne unter voip at sil.at oder 059944 - 2440 zur Verf?gung. Mit freundlichen Gr?ssen -- Stefan Schmidt Sysadmin/VOIP // voip at sil.at // Tel 059944-2440// ------------------------------------------------- SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // -------------------------------------------------
Philipp Kempgen
2009-Jul-24 13:34 UTC
[asterisk-users] notification mail to an user if his vmbox is full (was: Re: dialplan tips)
harry R schrieb:> 2) about asterisk voicemail maximum message limit, is it possible to send a > notification mail to an user if his vmbox is full ? How can i do that if > it's possible.Write a cron job to check if one of the mailboxes is full (ls -l /var/spool/asterisk/voicemail/vmContext/mailBox/INBOX/msg*.txt | wc -l) or use the externnotify parameter in voicemail.conf (3rd arg tells you the number of messages in the INBOX folder) and send an email. You could read the email address from voicemail.conf or from the voicemail Realtime family/table depending on what you use. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de --
harry R wrote:> In advance sorry for my bad english and if my problem was already > exposed (I didn't find any tips in the mailing list archive. Bad luck) > I have some questions about asterisk 1.6 release : > 1) how can I do a n+101 priority jumping if a SIP canal is busy ? > I read that the general parameter "priorityjumping" is depreciated in > the 1.6 release and I already try the "j" option in dial() application > but no way. > Here a sample of my simple dialplan : > > exten => 101,1,Ringing > exten => 101,2,Answer() > exten => 101,3,Dial(SIP/quentin,10) > exten => 101,n,VoiceMail(101 at default,u) > exten => 101,n,Playback(vm-goodbye) > exten => 101,n,Hangup() > exten => 101,104,Playback(busy) > exten => 101,n,Wait(3) > exten => 101,n,VoiceMail(101 at default,b) > exten => 101,n,Playback(vm-goodbye) > exten => 101,n,Hangup()As has be alluded to by a couple other posters, don't use priority jumping. This is a very outdated method of handling busy status. You get much more flexibility with the dialplan when you use STATUS variables, such as ${DIALSTATUS} to perform routing logic per the status value. I would re-write your dialplan as follows. Note that I've removed all priority numbering (except for the necessary priority 1) as you should not be using priority numbering either. If all you want to do is a simple ring and voicemail, then you could use something like: ; I always like using something informative on the first priority, rather ; than performing a *real* action ; exten => 101,1,Verbose(1,Incoming call from ${CALLERID(all)}) ; ; I prefer using this method to answer a line as it starts sending the ; RTP audio right away, and thus saves the audio clipping for the first few ; milliseconds of call answer (this is even more useful in front of an IVR ; prompt) ; exten => 101,n,Playback(silence/1) ; ; Now you can call your device. ; exten => 101,n,Dial(SIP/quentin,10) ; ; If all you care about is unavailable and busy status, you can control ; which prompt on a single line ; exten => 101,n,Voicemail(101 at default,${IF($[${DIALSTATUS} = BUSY]?b:u)}) ; ; You don't need to playback vm-goodbye yourself, sine Voicemail() should ; handle all the prompts ; exten => 101,n,Hangup() There you have it -- same functionality, in only 4 lines.> 2) about asterisk voicemail maximum message limit, is it possible to > send a notification mail to an user if his vmbox is full ? How can i do > that if it's possible.I believe you would do this via the 'externnotify' option in voicemail.conf, which upon a voicemail being left, will run that script. From there, you should be able to perform logic that checks if the voicemail box is full, and if so, to send an email, or whatever other action you want to take. Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk
2009/7/24 Stefan Schmidt <sst at sil.at>> Hello, > > i?ve a question about the Meetme Options. How could i play a enter and > leave sound but without recording the user name first. I just want a > User joined conferenc and a user leaved. > > With the i or I Option i have to record the name first. > > Is there any way of doing this? As i can see in the Meetme help the > background agi couldnt be used on non dahdi channel which i will have on > this server cause there is no direct Pri link, just SIP. >By default join and leave sounds are played when someone joins and leaves the conference... d -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090803/ac6cd46a/attachment.htm