[asterisk-users] 1.6.2 ConfBridge suggestion
A very nice feature of another conferencing system that I've used is that the admin/moderator can press a star code to MUTE ALL OTHER USERS on the conference. This is great if you have several people on the call and one of the people puts the call on hold (and so the music/advertisement/your call is important/etc) message starts, or someone's cellphone handsfree unit in their car is making a bunch of noise, or someone's endpoint starts creating acoustic echo. To minimize disruption on the call, the admin should be able to press (*5) and this would mute all other users, and then he/she would just tell the participants that they can unmute their microphone by pressing *1. Just a suggestion for whoever is involved with evolving the ConfBridge app. Regards! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Where should I look for MWI settings if Aastra phones don't do it?
On Wed, Jul 14, 2010 at 10:04 AM, bruce bruce wrote: > Hi Guys, > Running Asterisk v1.4.26 (Elastix flavour) and have Aastra 9480i, 6757i, and > 6730i, but none of them indicate the voic-email. Where should I look for > trouble to find the root issue for MWI? (1) Check from the CLI> voicemail show users to ensure that the proper mailboxes have been set up and there is new mail in them. If this is not right, check the voicemail.conf entry for this mailbox. (2) Check the phone device configuration (in sip.conf) to ensure that the phone has a mailbox=xxx entry. for example: ;entry in sip.conf for extension 115 [115] context=yourcontext mailbox=115 ... Restart asterisk if you've made changes and re-test. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Logging codec used in CDR
Happy Friday everyone, Is there a way to log the negotiated codec that was used for each call in CDR or in a separate log file? This is for SIP-based calls, if that matters. Perhaps there is some variable that can be queried as part of the dialing script; Or is it possible to grab the codec name using the "exten =>h," after the call completes... Thanks in advance for all suggestions. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GoogleTalk to Asterisk - choosing voice menu options
GoogleTalk connects ok to Asterisk 1.6.2.7 but how can you choose voice menu options (press 1 for Bob, press 2 for Betty, ...) from the GT client? (There is no dial pad in the Windows GT client, but what you type in the message box does show up on the console as an incoming Jabber message.) Is there a way? Thanks all! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to get ConfBridge user count
I want to set up a conference call to be recorded automatically, so
I'd like the recording to start when the second caller joins the
conference (one caller already there). The recording would continue
until the last user hangs up.
How can you determine how many are already in the conference bridge?
[conferences]
exten => 66,1,Answer
exten => 66,n,Wait(1)
exten => 66,n,Authenticate(123456)
;
exten => 66,n,NoOp(-- ConfBridge 66 user count: ${count} --)
;<-- WHAT VARIABLE TO USE HERE?
;
exten => s,n,Set(MONITOR_EXEC=/etc/asterisk/monitor_exec.sh)
exten => s.,n,Set(DATETIME=${STRFTIME(${EPOCH},,%C%y-%m%d-%H%M)})
exten => s,n,ExecIf($[${count} =
1]?Monitor(wav,record-${CALLERID(num)}-${DATETIME},bm))
;
exten =>66,n,ConfBridge(66,Ms)
exten => 66,n,Playback(goodbye)
exten => 66,n,Hangup
Thanks for any info.
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[asterisk-users] [Asterisk-Users] Asterisk transfer to a conference using feature code?
Is it possible to use an Asterisk feature code to transfer a call to a specific extension? For instance, if you take a call, and the caller wants to go to a conference, it would be nice to use a feature code for this, rather than going through a longer transfer sequence. e.g.: - You have a meetme conference: [conferences] exten => 21,1,NoOp(MeetMe Conference) exten => 21,n,MeetMe(50,pM) ;p=prompt for pin, M=music for first caller exten => 21,n,Hangup - You then want to define a feature code *5 in features.conf which will blind transfer the caller to (conferences,21,1) Any suggestions/examples as to how to set this up? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Call Recording *1 Status Indication
When you press *1 in Asterisk (1.6.2.7) to start/stop call recording, the console CLI> shows: > User hit '*1' to record call. filename: wav,auto-1273791789-103-5551212,m Is it possible to play a sound to back to the person who pressed *1 to indicate to them that recording has actually started or stopped? Something like "Recording" / "Record Off", or else sounds like people are used to hearing when they plug/unplug a USB device into a PC. Also, I'd also like to have the completed recording go to the person's voicemail box as a message if that's possible when the recording stops by toggling *1 or the parties hang up, if anyone has suggestions for doing that or can point to a link. Thanks much! S. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Improving CLI Help - was [Re: 1.6.2 No "soft hangup"?]
On Tue, Apr 20, 2010 at 10:49 AM, Steve Edwards wrote: > On Tue, 20 Apr 2010, Tilghman Lesher wrote: > >> On Tuesday 20 April 2010 11:05:07 Steve Johnson wrote: >>> I wanted to force a hangup of a SIP to SIP call from the Asterisk CLI> >>> prompt, and found references on using the command "soft hangup >>> ", but as you can see below, the "soft hangup" command >>> does not seem to exist, and there is no mention about it in the >>> UPGRADE*.txt documents. >>> >>> Can anyone shed light on what would replace "soft hangup" in 1.6.2.x ?? >>> (This asterisk server is strictly SIP/IAX2, no DAHDI hardware) >> >> "channel request hangup " > > How obvious. > > Kind of makes me wish I still used 1.2 -- oh wait, I do. > > Seriously though, IMNSHO, with every release the CLI gets more obtuse. > > I'd like to see a more natural and intuitive interface following a "verb > noun" model like Oracle, MySQL, or even GDB. > > hangup [sip|iax|dahdi] channel > > seems so "obvious." > > -- > Thanks in advance, > - > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello Further to Steve Edward's comment, I think things would be more obvious if the help system was improved slightly, for instance: If you were trying to figure out the commands dealing with peers, you would be able to type: *CLI> help peer No "peer" command found. Possible alternatives: iax2 show peer Show details on specific IAX peer iax2 show peers List defined IAX peers sip show peers List defined SIP peers sip show peer Show details on specific SIP peer (and so on, maybe using the "[More]" option to help it be readable) In this case, if I could use the "help" system to search on all occurrences of the word "hangup" in the available commands, I would probably have found it myself instead of bothering the list. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.6.2 No "soft hangup"?
Hello asteriskers, I wanted to force a hangup of a SIP to SIP call from the Asterisk CLI> prompt, and found references on using the command "soft hangup ", but as you can see below, the "soft hangup" command does not seem to exist, and there is no mention about it in the UPGRADE*.txt documents. Can anyone shed light on what would replace "soft hangup" in 1.6.2.x ?? (This asterisk server is strictly SIP/IAX2, no DAHDI hardware) Thanks! Ref: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg185265.html Asterisk 1.6.2.7-rc2, Copyright (C) 1999 - 2010 Digium, Inc. and others. .. *CLI> sip show channels Peer User/ANR Call ID Format Hold Last Message Expiry 1.2.3.4 104 b45f77d8-f7789d 0x100 (ulaw) No Rx: ACK 5.6.7.8 1866*** 37e7735e0a294a2 0x4 (ulaw) No Rx: ACK *CLI> core show channels Channel Location State Application(Data) SIP/isp-00 (None) Up AppDial((Outgoing Line)) SIP/104-0012 1866...@outbound: Up Dial(SIP/isp/1866*** 2 active channels 1 active call *CLI> soft hangup SIP/104-0012 No such command 'soft hangup SIP/104-0012' (type 'core show help soft hangup' for other possible commands) *CLI> core soft hangup SIP/104-0012 No such command 'core soft hangup SIP/104-0012' (type 'core show help core soft' for other possible commands) *CLI> sip soft hangup SIP/104-0012 No such command 'sip soft hangup SIP/104-0012' (type 'core show help sip soft hangup' for other possible commands) *CLI> core stop now (This stopped the call of course, but also killed asterisk in the process) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk Dictate?
Google is your friend. You should use it. Search for: asterisk extensions.conf dictate or asterisk extensions.conf dictate example Some results: http://www.asteriskguru.com/tutorials/dictate.html and http://www.voip-info.org/wiki/view/Asterisk+cmd+Dictate On Wed, Dec 30, 2009 at 6:36 AM, hadi motamedi wrote: > Dear All > Can you please give me more hint on how Asterisk Dictate() works? > Thank you > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing for incoming call
If you try just this, what does the caller hear? It should be ringing for the first 20 sec, and then maybe the congestion tone afterwards. exten => s,1,Wait(20) exten => s,n,Hangup You shouldn't need/use the Ringing() command at all, as the initial ring before your system answers would be generated by the provider. If "wait ... answer" doesn't work for you, you'll have to provide more output from the CLI and tell us more about your configuration. On Fri, Dec 18, 2009 at 10:29 PM, Bob Smither wrote: > > On Fri, 2009-12-18 at 19:58 -0600, Steve Johnson wrote: >> Try putting the wait before the Answer. >> >> ... >> exten => s,n,Wait(10) >> exten => s,n,Answer >> ... > > Thanks Steve. I tried that: > >> On Fri, Dec 18, 2009 at 5:10 PM, Bob Smither wrote: >> > Dear All, > > > >> > >> > ... >> > exten => s,n,Answer >> > exten => s,n,Ringing() >> > exten => s,n,Wait(10) >> > exten => s,n,BackGround(sound file) >> > ... >> > >> > I have also tried moving the Answer app to right before the BackGround >> > app. > > > > i.e., after the Wait, but still no joy. > > Anything else I need to look at? > > Thanks, > -- > Bob Smither > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing for incoming call
Try putting the wait before the Answer. ... exten => s,n,Wait(10) exten => s,n,Answer ... On Fri, Dec 18, 2009 at 5:10 PM, Bob Smither wrote: > Dear All, > > I am using Asterisk 1.4 on CentOS 5. I have an incoming DID provided by > Vitelity. When the number is called it goes to my Asterisk box. The > protocol is SIP. This all works just fine if I answer the call and > begin a playback. > > I want to let the number ring for a few seconds before it is answered, > and would like the caller to hear it ringing. I have tried: > > ... > exten => s,n,Answer > exten => s,n,Playtones(ring) > exten => s,n,Wait(10) > exten => s,n,StopPlaytones() > exten => s,n,BackGround(sound file) > ... > > also > > ... > exten => s,n,Answer > exten => s,n,Ringing() > exten => s,n,Wait(10) > exten => s,n,BackGround(sound file) > ... > > I have also tried moving the Answer app to right before the BackGround > app. > > In all cases when I call the number I never hear it ringing. After the > 10 second delay, the BackGround app does run. Connecting to the CLI > does not give me any useful information - for example the Ringing app is > shown to run, but the caller does not hear it. > > Any suggestions? > > Many thanks! > > -- > Bob Smither > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rewrite calling number of incoming call
How about:
exten => 977,1,ExecIf($[${CALLERID(num)} =
733025975]?Set(CALLERID(num)=0317998975))
exten => 977,n,ExecIf($[${CALLERID(num)} = 1234]?Set(CALLERID(num)=317998977))
exten => 977,n,ExecIf($[${CALLERID(num)} = 5678]?Set(CALLERID(num)=317998978))
[..]
exten => 977,n,Dial(SIP/0317998977)
On Mon, Dec 14, 2009 at 12:21 PM, Magnus Benngård
wrote:
> Hi!
>
> Trying to figure out how to rewrite calling number of an incoming call...
>
> A cell phone (0733025975) dials a X-Lite (977).
> X-Lite "shows" 733025975 at the display, but I want it to be 0317998975.
> I thought i could do something like:
>
> exten => 977/733025975,1,Set(CALLERID(number)=0317998975)
> exten => 977,n,Dial(SIP/0317998977)
>
> [Dec 14 19:07:43] NOTICE[20731]: chan_h323.c:2272 answer_call: Dropping call
> because extensions '977', 's' and 'i' doesn't exists in context
> [inputinterior.se]
>
> Rewriting of outgoing is working... snip
>
> exten => _0X!/0317998975,1,Set(CALLERID(number)=317998975)
> exten => _0X!/0317998977,1,Set(CALLERID(number)=317998977)
> exten => _0X!/0317998978,1,Set(CALLERID(number)=317998978)
> exten => _0X!/0317998985,1,Set(CALLERID(number)=317998985)
> exten => _0X!/0317998987,1,Set(CALLERID(number)=317998987)
> exten => _0X!,n,Dial(H323/0${ext...@avaya)
>
> Can someone guide me on the correct track?
>
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Re: [asterisk-users] G729 with IAX
Of course, as long as your endpoints support it. Read more about it and purchase G.729 channel licenses for Asterisk from Digium: http://www.digium.com/en/products/g729codec.php Once you have the codec properly installed, enable it for your peer in your iax.conf file "allow=g729". Restart asterisk and go to it. Also, Google is your friend. Search: g729 iax for lots of information and examples. On Tue, Dec 8, 2009 at 1:03 AM, DHAVAL INDRODIYA wrote: > dear All, > > can I use G729 with IAX trunk or IAX calls > > regards > Dhaval > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Automon -> Voicemail
Thanks. One more feature/automon question:
Is there some way to provide audible feedback (play a specific sound
file) to the initiator that a feature has been activated -- in this
case that recording has started (like "bee-yup") and stopped
("bee-dah"), when you enter in the *1?
On Mon, Dec 7, 2009 at 2:58 PM, Doug Lytle wrote:
> Steve Johnson wrote:
>> Hi all,
>>
>> What's the best method to send automon call recordings (*1) to the
>> voicemail box of the Asterisk user?
>>
>>
>
> I've picked up the following off the list a while ago. Works pretty
> good. I do a mysql lookup to see if the user has the ability or not:
>
> __features.conf:__
>
>
> [applicationmap]
>
> recordtovm => *8,self,Macro,recordtovm
>
>
> __Dial plan entry:__
>
> ;
> ; Call recording, initiated by *8
> ; after hangup, send recording to
> ; callers voice mail box
> ;
>
> [macro-recordtovm]
>
> exten => s,1,MYSQL(Connect connid localhost username 'supersecret'
> call_recording)
> exten => s,n,GosubIf($["${MYSQL_STATUS}" = "-1"]?mysql_failed,s,6)
> exten => s,n,MYSQL(Query resultid ${connid} SELECT allowed FROM
> Indianapolis WHERE extension = ${CALLERID(number)})
> exten => s,n,MYSQL(Fetch fetchid ${resultid} results)
> exten => s,n,MYSQL(Disconnect ${connid})
> exten => s,n,MYSQL(Clear ${resultid})
> exten => s,n,Set(RECORDING.OK=${results})
> exten => s,n,GotoIf($["${results}" = "Y"]?9:15)
> exten =>
> s,n,Set(MONITOR_FILENAME=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${CALLERID(num)})
> exten => s,n,Set(ORIG_DATE=${STRFTIME(${EPOCH},,%c)})
> exten => s,n,Set(ORIG_TIME=${STRFTIME(${EPOCH},,%s)})
> exten => s,n,Set(ORIG_CID=${ARG2})
> exten => s,n,Playback(local/stutter)
> exten =>
> s,n,MixMonitor(${MONITOR_FILENAME}.wav,b,/usr/local/bin/recordtovm.pl
> ${CALLERID(num)} ${MONITOR_FILENAME}.wav "${ORIG_DATE}" "${ORIG_TIME}"
> "${ORIG_CID}")
> exten => s,n,NoOP(User allowed to record calls? ${results})
>
>
>
> __Perl script__
>
> cd /usr/local/bin
>
> cat recordtovm.pl
>
> #!/usr/bin/perl -w
> #
> use strict;
>
> my $monitordir="/var/spool/asterisk/monitor/";
> my $vmdir="/var/spool/asterisk/voicemail/sip/";
> my $vmfolder="INBOX";
> my $vmbox=$ARGV[0];
> my $vmpath=$vmdir."$vmbox/"."$vmfolder";
> my $monitorfilename=$ARGV[1];
> my $orig_date=$ARGV[2];
> my $orig_time=$ARGV[3];
> my $orig_cid=$ARGV[4];
>
> opendir (DIR, $vmpath);
> my @files = grep(/\.txt$/,readdir(DIR));
> closedir(DIR);
> my @sortedfiles = sort {$b cmp $a} @files;
> my $vmid;
> if ($sortedfiles[0] =~ /^(msg)(\d\d\d\d)(.txt)/)
> {
> $vmid=$2;
> $vmid++;
> }
> else
> {
> $vmid="";
> };
>
> open VMFILE,"> $vmpath/msg$vmid.txt";
> print VMFILE ";\n";
> print VMFILE "; Message Information file\n";
> print VMFILE ";\n";
> print VMFILE "[message]\n";
> print VMFILE "origmailbox=$vmbox\n";
> print VMFILE "context=\n";
> print VMFILE "macrocontext=\n";
> print VMFILE "exten=s\n";
> print VMFILE "priority=\n";
> print VMFILE "callerchan=\n";
> print VMFILE "callerid=$orig_cid\n";
> print VMFILE "origdate=$orig_date\n";
> print VMFILE "origtime=$orig_time\n";
> print VMFILE "category=\n";
> print VMFILE "duration=\n";
> close VMFILE;
>
> if ($ARGV[1])
> {
> system("mv $monitordir"."$monitorfilename $vmpath/msg$vmid.wav");
> };
>
>
> --
>
> Ben Franklin quote:
>
> "Those who would give up Essential Liberty to purchase a little Temporary
> Safety, deserve neither Liberty nor Safety."
>
>
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[asterisk-users] Automon -> Voicemail
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. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] local channels
> My Dial() command is Dial($LOCAL_DIAL)
Perhaps you should be using:
Dial(${LOCAL_DIAL})
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[asterisk-users] Determining extension's sip.conf default mailbox
Hello list,
How can you obtain the default mailbox for a SIP extension (as stored
in sip.conf and shown with "sip show peer ")? Is there a
function to extract it?
Why? Some extensions have shared mailboxes and others do not and I
don't want to duplicate logic, just use the extension's default
mailbox as coded in sip.conf.
sip.conf
--
[100]
mailbox=100
[102]
mailbox=102
[103]
mailbox=100
I want a function which I can use in the dialplan (1.6) that works like:
DefaultMailbox(100) -> 100
DefaultMailbox(102) -> 102
DefaultMailbox(103) -> 100
for example:
exten s,n,VoicemailMain(DefaultMailbox(${CALLERID(num)}))
Suggestions?
Thanks!
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Re: [asterisk-users] Maximum cable length for analog phone from FXS port
For long distances, a wireless point-to-point might be more economical than trenching. e.g: Carlson Wideband CDMA Spread Spectrum Phone Line Extender http://www.oksolar.com/communications/phone_line_ext.htm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Where are 2 letter language values defined?
Also check out: http://www.w3.org/International/questions/qa-lang-2or3.en.php On Wed, May 6, 2009 at 1:22 PM, Steve Johnson wrote: > They are 2-letter ISO country codes. > > http://www.iso.org/iso/english_country_names_and_code_elements > > On Wed, May 6, 2009 at 1:05 PM, Steve Edwards > wrote: >> I've googled for way too long, where are the 2 letter language values >> defined? >> >> I know: >> >> en = English >> es = Spanish >> fr = French >> >> but what about Croatian, Russian, Serbian, Vulcan, etc? >> >> Is there a list documented for Asterisk or is it "just use the 2 letter >> country code Internet TLD?" >> >> Thanks in advance, >> >> Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST >> Newline Fax: +1-760-731-3000 >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Where are 2 letter language values defined?
They are 2-letter ISO country codes. http://www.iso.org/iso/english_country_names_and_code_elements On Wed, May 6, 2009 at 1:05 PM, Steve Edwards wrote: > I've googled for way too long, where are the 2 letter language values > defined? > > I know: > > en = English > es = Spanish > fr = French > > but what about Croatian, Russian, Serbian, Vulcan, etc? > > Is there a list documented for Asterisk or is it "just use the 2 letter > country code Internet TLD?" > > Thanks in advance, > > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] one-button call parking/pickup on Asterisk with Polycom phones?
Anyone want to talk briefly about one-button call parking/pickup on Asterisk with Polycom phones? Does anyone use it or know to do it? On many phone systems there are 2 or 3 park buttons, and you can park a call onto an unlit park button, and then the light flashes. You can go to any other phone, and press the park button with the flashing light to pick up the call. Super easy from the user's point of view. How to do with Asterisk? Hints? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Patch to dahdi Chans.pm
Software: dahdi-linux-complete-2.1.0.3+2.1.0.2.tar.gz asterisk-1.6.1-rc1.tar.gz Hardware: 4-port fxs card Example: # /etc/init.d/dahdi status ### Span 1: WRTDM/0 "wrtdm Board 1" (MASTER) 1 FXSFXSKS (In use) 2 FXSFXSKS (In use) 3 FXSFXSKS (In use) 4 FXSFXSKS (In use) Use of uninitialized value in string eq at /usr/lib/perl5/site_perl/5.10.0/Dahdi/Chans.pm line 221. 5 unknown Use of uninitialized value in string eq at /usr/lib/perl5/site_perl/5.10.0/Dahdi/Chans.pm line 221. 6 unknown Use of uninitialized value in string eq at /usr/lib/perl5/site_perl/5.10.0/Dahdi/Chans.pm line 221. 7 unknown [..] Use of uninitialized value in string eq at /usr/lib/perl5/site_perl/5.10.0/Dahdi/Chans.pm line 221. 23 unknown Use of uninitialized value in string eq at /usr/lib/perl5/site_perl/5.10.0/Dahdi/Chans.pm line 221. 24 unknown Problem: $self->type is not being checked to see if it is defined. Add a line just above line 221 to fix this, such as: return undef unless defined $self->type; After the fix, the above command executes without error. # /etc/init.d/dahdi status ### Span 1: WRTDM/0 "wrtdm Board 1" (MASTER) 1 FXSFXSKS (In use) 2 FXSFXSKS (In use) 3 FXSFXSKS (In use) 4 FXSFXSKS (In use) 5 unknown 6 unknown 7 unknown 8 unknown 9 unknown 10 unknown 11 unknown 12 unknown 13 unknown 14 unknown 15 unknown 16 unknown 17 unknown 18 unknown 19 unknown 20 unknown 21 unknown 22 unknown 23 unknown 24 unknown FYI... S. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail message is dialtone
Here's also an example snip from the debug log:
[07:42:20] -- Executing [...@mainmenu:15] Dial("Zap/1-1",
"SIP/105|18|tKk") in new stack
[07:42:20] -- SIP/105-08571180 is ringing
[07:42:39] -- Nobody picked up in 18000 ms
[07:42:39] -- Executing [...@mainmenu:16] Answer("Zap/1-1", "") in new stack
[07:42:39] -- Executing [...@mainmenu:17] Wait("Zap/1-1", "1") in new stack
[07:42:40] -- Executing [...@mainmenu:18] Playback("Zap/1-1",
"silence/1") in new stack
[07:42:40] -- Playing 'silence/1' (language 'en')
[07:42:41] -- Executing [...@mainmenu:19] BackGround("Zap/1-1",
"please-leave-a-message") in new stack
[07:42:41] -- Playing 'please-leave-a-message' (language 'en')
[07:42:47] -- Executing [...@mainmenu:20] VoiceMail("Zap/1-1", "105|s")
in new stack
[07:42:48] -- Playing 'beep' (language 'en')
[07:42:48] -- Recording the message
[07:42:48] -- x=0, open writing:
/var/spool/asterisk/voicemail/default/100/tmp/UNCGT6 format: wav49,
0x8570eb8
[07:42:48] -- x=1, open writing:
/var/spool/asterisk/voicemail/default/100/tmp/UNCGT6 format: gsm,
0x8227430
[07:42:48] -- x=2, open writing:
/var/spool/asterisk/voicemail/default/100/tmp/UNCGT6 format: wav,
0x8259b10
[07:46:42] -- Recording automatically stopped after a silence of 10 seconds
[07:46:42] -- Playing 'auth-thankyou' (language 'en')
[07:46:43] == Auto fallthrough, channel 'Zap/1-1' status is 'NOANSWER'
[07:46:43] -- Hungup 'Zap/1-1'
On Fri, Jan 16, 2009 at 12:07 PM, Steve Johnson wrote:
> Hello all,
>
> I have one Asterisk 1.4.21 system connected to a North American POTS
> line. Normally hangup detection works fine, and Asterisk hangs up
> properly if you are talking to a caller and they hang up; but
> occasionally a call comes in (typically from a US telemarketer) where
> the caller hangs up just as voicemail recording is starting, and you
> get a voicemail recording of dialtone (then congestion and off-hook
> warning tones) for almost 4 minutes before asterisk gives up the line.
>
> zapata.conf [channels] options are:
>
> language=en
> context=default
> rxwink=300
> usecallerid=yes
> hidecallerid=no
> cidsignalling=bell
> callwaiting=no
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=no
> transfer=yes
> canpark=yes
> cancallforward=no
> callreturn=no
> musiconhold=default
> echocancel=yes
> echocancelwhenbridged=yes
> immediate=no
> faxdetect=no
> relaxdtmf=yes
> hanguponpolarityswitch=yes
> progzone=us
> signalling=fxs_ks
> channel => 1
>
>
> Any suggestions for voicemail detecting/rejecting messages when there
> is only dialtone on the other end?
>
> Thanks!
>
> S.
>
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[asterisk-users] Voicemail message is dialtone
Hello all, I have one Asterisk 1.4.21 system connected to a North American POTS line. Normally hangup detection works fine, and Asterisk hangs up properly if you are talking to a caller and they hang up; but occasionally a call comes in (typically from a US telemarketer) where the caller hangs up just as voicemail recording is starting, and you get a voicemail recording of dialtone (then congestion and off-hook warning tones) for almost 4 minutes before asterisk gives up the line. zapata.conf [channels] options are: language=en context=default rxwink=300 usecallerid=yes hidecallerid=no cidsignalling=bell callwaiting=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=no transfer=yes canpark=yes cancallforward=no callreturn=no musiconhold=default echocancel=yes echocancelwhenbridged=yes immediate=no faxdetect=no relaxdtmf=yes hanguponpolarityswitch=yes progzone=us signalling=fxs_ks channel => 1 Any suggestions for voicemail detecting/rejecting messages when there is only dialtone on the other end? Thanks! S. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Variables for dial plan
One of these methods will work:
exten => s,n,ExecIf($["${dialplan}" = "NZ"]|Set|NAT="0")
exten => s,n,ExecIf($["${dialplan}" = "NZ"]|Set|INT="00")
-or-
exten => s,n,GotoIf($["${dialplan}" != "NZ"]?not-nz)
exten => s,n,Set(NAT="0")
exten => s,n,Set(INT="00")
exten => s,n(not-nz),more_dialplan_stuff
On Mon, Dec 15, 2008 at 3:26 AM, Michael wrote:
> On Mon, 15 Dec 2008 21:31:56 you wrote:
>> Use setvar=variablename=value
>>
>> Eg: under [client1]
>> setvar=dialplan=NZ
>>
>> Then just reference ${dialplan} in your extensions.conf
>>
>> Cheers
>> Andy
>
> Thanks, now how do I achieve the following logic?
>
> if ($dialplan == NZ) {
> $NAT = 0;
> $INT = 00;
> };
>
> Michael
>
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[asterisk-users] Park buttons on Polycom IP501/601
Is anyone using fixed Park buttons (some of the ones on the left side of the screen) on a Polycom phone? Here's what I mean: - Call is received and parked, by the user pressing an unlit park button (e.g. 701) and the call is parked there. - The call can be picked up at any other extension by pressing the flashing park 701 button. - Once the call has been picked up, the 701 park slot is idle and the light goes off. For a small site, only a couple of Park buttons would be needed. Can you give an example of how to do this? Thanks, S. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Alternate names in Directory (dial-by-name)
Hi everyone, What creative methods are used to support dial-by-name functionality for people who go by more than one name? e.g.: Rebecca/Becky, Margaret/Peggy, William/Bill, Liz/Elizabeth, etc. We'd like to use the "f" first name option of the Directory function, as the particular phone system has multiple members of the same families (same last names). In the first example, the caller should be able to key either R-E-B or B-E-C and get to the same person/mailbox. Any suggestions? Thanks, S. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VOICEMAIL OPTIONS help needed
Hi everyone, We have a particular user on our Asterisk 1.4.x system who always listens to his voicemail messages via email. - Is there some way to send the voicemail ONLY to email and not retain them on the phone? - Alternatively, can the voicemail system only keep, say, just the last 10 messages (as backup in case of email delivery failure or a message getting deleted in email accidentally before it is heard), purging out the oldest when a new one is received? (If we set the option maxmsg=10 on his mailbox in voicemail.conf, I think it will stop accepting voicemails after 10 messages, not turf the oldest one and accept a new one in its place). Everyone else uses the normal voicemail options on their phones, so the solution should be just for this single user. Thanks for any suggestions. S. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Polycom: DHCP/boot server supporting 2 models of phones
On Fri, Mar 28, 2008 at 12:05 AM, Paul Hales <[EMAIL PROTECTED]> wrote: > > Can't you just use the same bootrom for all your polycom phones? > > PaulH > > > > > On Fri, 2008-03-28 at 15:38 +1100, Lee, John (Sydney) wrote: > > I have a question about DHCP and boot server supporting more than 1 > > model of Polycom phones. > > > > According to Polycom "standards", Polycom phone boots up to get a DHCP > > address and at the same time getting a boot server string (with username > > and password) to logon to boot server to download SIP, bootROM and etc. > > > > That is okay if there is only one type of phone (that requires a > > specific SIP and bootROM release). > > > > What about if the environment has to support two or more models of > > Polycom phones? > > > > On the boot server side, I can define another home directory like > > /home/polycom1 and /home/polycom2 to store different SIP and bootROM > > releases. However, the issue is how different polycom phone model can > > get a different user account and password to log on to different home > > directories. > > > > I understand the issue here is DHCP and not the boot server but I am a > > bit of a newbie here. > > > > Can anyone help please? > > As someone earlier pointed out, different models of polycom phones can be pointed to the same set of configuration files. With the standard ISC dhcpd server, the phones can be told where to look by using a directive like: option tftp-server-name "ftp://polycom:[EMAIL PROTECTED]/"; This would require a user account on the ftp server like: polycom:x:501:501:Polycom Phone Provisioning:/etc/asterisk/polycom/ftp/:/bin/bash and the configuration files would be placed in the /etc/asterisk/polycom/ftp/ directory. So if you wanted to have separate configurations for certain phones (for upgrade testing, etc., it is easily possible. SJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is encrypted iax safe and secure?
Of course *it would be nice if* the IAX2 authentication parameters were also encrypted, so that there was no danger of a 3rd party hijacking your connection and generating a bunch of extra charges. S. On Fri, Feb 15, 2008 at 11:31 AM, Kevin P. Fleming <[EMAIL PROTECTED]> wrote: > Tim Panton wrote: > > > The NEW frame doesn't _have_ to contain a dialed number, the digits > > can be sent later > > (I forget the frametype), but later means within the encrypted > > session :-) > > It's the DIAL command that you are thinking of. I'm considering > implementing this, but it has one major caveat: to really do the job > right, we wouldn't want any caller information (CLID or CNAM) to be in > the NEW message either, it would have to be added as IEs to the DIAL > command. Unfortunately no existing implementations are going to be > prepared to receive that information as part of DIAL, so they would > process this sort of call with an empty CLID and CNAM. We can of course > enhance chan_iax2 to understand this method of doing things, but it > won't be backward compatible with previous versions of Asterisk or any > other IAX2 clients. > > -- > Kevin P. Fleming > Director of Software Technologies > Digium, Inc. - "The Genuine Asterisk Experience" (TM) > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help: dtmf mode
Please post your sip.conf entry for your phone and also describe your calling path. Are you having a problem with internal calls (e.g.: to voicemailmain) on the same switch, or are you referring to calls to PSTN destinations via pots/pri/sip/? Also, which versions of Asterisk, Zaptel, linux, etc. are you using? S. On Jan 24, 2008 12:43 PM, Jarga Jallow <[EMAIL PROTECTED]> wrote: > > > > > Hi, > > I am having trouble making a selection when I call a number and need to make > a selection to go to an extension with my polycom phones 301. Anybody have > an idea how to fix this problem? > > Thanks in advance. > > > > > Jarga Jallow > > Technical Support Engineer > > 2985 S. Hwy. 360 > > Grand Praire, Texas 75052 > > Direct: 972-206-1212 ext# 29 > > Mobile: 214-669-9046 > > Fax:972-999-4113 > > Toll Free: 1-877-801-5511 ext 34 > > Toll Free: 1-877-926-2288 > > > > www.2mcctv.com > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom-SIP response 500
I have just retested and agree that this error eventually does clear itself. However, in this test it took about 35 minutes and each Polycom phone produced between 1000 and 1300 error message lines at 1 to 0 second intervals (which I captured to the debug log). Once one phone starts flagging an error, all Polycom phones that are buddy-watching join right in. I triggered the problem by simply restarting asterisk: /usr/sbin/asterisk -rx "restart when convenient". Sometimes (but not all the time) it will also start if you reload. All suggestions appreciated. S. On Jan 22, 2008 9:23 AM, Steve Johnson <[EMAIL PROTECTED]> wrote: > I am using Polycom's SIP 2.2.0047 (the current release) and am seeing > this. It seems to occur less often with "extensions reload" rather > than just "reload", but it would be nice to fix this. > > Tx. > > > > On Jan 22, 2008 8:30 AM, Steve Davies <[EMAIL PROTECTED]> wrote: > > On 1/22/08, Steve Johnson <[EMAIL PROTECTED]> wrote: > > > Hi list, > > > > > > There are many Polycom experts on this list -- hopefully someone has a > > > solution. > > > > > > With *several* versions of Asterisk 1.4.x, doing a reload of Asterisk > > > causes the Polycom 601 phones to start dumping these messages to the > > > CLI. > > > > > > -- Incoming call: Got SIP response 500 "Internal Server Error" > > > back from 192.168.2.x > > > > [snip] > > > > I have not seen this problem here since upgrading to 2.1.2 firmware. > > Or perhaps it was 2.2.0, one or the other. The phones now seem to > > recover on thier own when Asterisk returns. > > > > Cheers, > > Steve > > > > ___ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom-SIP response 500
I am using Polycom's SIP 2.2.0047 (the current release) and am seeing this. It seems to occur less often with "extensions reload" rather than just "reload", but it would be nice to fix this. Tx. On Jan 22, 2008 8:30 AM, Steve Davies <[EMAIL PROTECTED]> wrote: > On 1/22/08, Steve Johnson <[EMAIL PROTECTED]> wrote: > > Hi list, > > > > There are many Polycom experts on this list -- hopefully someone has a > > solution. > > > > With *several* versions of Asterisk 1.4.x, doing a reload of Asterisk > > causes the Polycom 601 phones to start dumping these messages to the > > CLI. > > > > -- Incoming call: Got SIP response 500 "Internal Server Error" > > back from 192.168.2.x > > [snip] > > I have not seen this problem here since upgrading to 2.1.2 firmware. > Or perhaps it was 2.2.0, one or the other. The phones now seem to > recover on thier own when Asterisk returns. > > Cheers, > Steve > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom-SIP response 500
Hi list, There are many Polycom experts on this list -- hopefully someone has a solution. With *several* versions of Asterisk 1.4.x, doing a reload of Asterisk causes the Polycom 601 phones to start dumping these messages to the CLI. -- Incoming call: Got SIP response 500 "Internal Server Error" back from 192.168.2.x They continue on until we force the devices to reboot from the CLI with a "sip notify polycom-check-cfg 140 141 142 ..." command. Of course this is "unpleasant" especially during the day. I have scoured the archives, google and the wiki and have found that although many experience this problem, no one has posted a solution. Help please! Thanks much! Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIPAddHeader in .call file
Sorry to answer my own post, but I have found a solution which perhaps
others can use too...
In the .call file, instead of specifying a channel line as:
chan: SIP/140 (for example)
use instead:
chan: Local/[EMAIL PROTECTED]
and put in extensions.conf
[polycom-paging]
exten => _1XX,1,NoOp(Paging Ext ${EXTEN})
exten => _1XX,n,SIPAddHeader(Alert-Info: Ring Answer)
exten => _1XX,n,Dial(SIP/${EXTEN},20,L(6))
exten => _1XX,n,Hangup
Steve Johnson wrote:
> Hi everyone,
>
> How can I add the equivalent of:
>
>exten => s,n,SIPAddHeader(Alert-Info: Ring Answer)
>
> in a .call file? This is to support paging to Polycom phones...
>
> Thanks for all info!
>
> Steve
>
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[asterisk-users] SIPAddHeader in .call file
Hi everyone, How can I add the equivalent of: exten => s,n,SIPAddHeader(Alert-Info: Ring Answer) in a .call file? This is to support paging to Polycom phones... Thanks for all info! Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Call Recording
Here's what I would suggest. You should insert some NoOp() statements
and watch the CLI as you dial your 555 extension so that you can see
whether it's working or not.
Your example (which you mentioned you want to run under Asterisk 1.4):
> [test]
> exten =>
> 555,1,SetVar(CALLFILENAME=outgoing/${TIMESTAMP:0:4}/${TIMESTAMP:4:2}/${TIMESTAMP}-${EXTEN})
> exten => 555,2,Monitor(wav,${CALLFILENAME},m)
> exten => 555,3,Dial(IAX2/ics.iax-trunk/${EXTEN})
> exten => 555,4,Hangup()
In Asterisk 1.4 (read the UPGRADE.txt file in the source directory):
- SetVar() has been replaced by Set()
- ${TIMESTAMP} no longer exists.
- (CALLERID usage has changed in 1.4 also).
so you first have to fix that stuff, with something like:
[test]
exten => 555,1,Set(DATETIME=${STRFTIME(${EPOCH},,%C%y-%m%d-%H%M)})
exten => 555,n,NoOp(DATETIME: ${DATETIME})
;
; Tweak the callfilename until you're happy with it...
; Note that the default recording directory is /var/spool/asterisk/monitor
exten => 555,n,Set(CALLFILENAME=${CALLERID(num)}-${DATETIME}-${EXTEN})
exten => 555,n,NoOp(CALLFILENAME: ${CALLFILENAME})
;
exten => s,n,Monitor(wav,${CALLFILENAME},b)
;
; Remove this next line after the determining that you have the filename right
; by checking the console progress...
exten => 555,n,Hangup()
;
exten => 555,n,Dial(IAX2/ics.iax-trunk/${EXTEN})
exten => 555,n,Hangup()
My script assumes that the monitor files are in the default directory,
so adjust it if necessary after you get the above working. When you
run it, the .mp3 stereo files should be produced.
In a production environment, I'd imagine that you'd want to run the
combine.sh script periodically as a scheduled cron job.
Don't forget to follow the legal standards for call recording in your
jurisdiction (and be nice).
Have fun,
S.
On Jan 14, 2008 12:53 PM, Mike Hammett <[EMAIL PROTECTED]> wrote:
> Does what I have in the dialplan look right or am I way off base with being
> able to use that script?
>
>
> -
> Mike Hammett
> Intelligent Computing Solutions
> http://www.ics-il.com
>
>
>
> - Original Message -
> From: "Steve Johnson" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>; "Asterisk Users Mailing List -
> Non-Commercial Discussion"
> Sent: Monday, January 14, 2008 10:51 AM
> Subject: Re: [asterisk-users] Asterisk 1.4 Call Recording
>
>
> > You might take a few ideas from this combine.sh script which works for
> > me. It uses the combine_wave program from
> > http://panteltje.com/panteltje/dvd/combine_wave-0.3.tgz and the lame
> > program to convert to mp3.
> >
> > It converts the entire directory /var/spool/asterisk/monitor/*-in.wav
> > files to mp3 where the mp3 file doesn't already exist.
> >
> > S.
> >
> >
> > File: combine.sh
> > ---
> > #!/bin/sh
> >
> > cd /var/spool/asterisk/monitor
> >
> > for f in *-in.wav
> > do
> >in=$f
> >out=`echo $f | sed -e 's/-in.wav/-out.wav/'`
> >tmpwav=`echo $f | sed -e 's/-in.wav/-both.wav/'`
> >mp3=`echo $f | sed -e 's/-in.wav/.mp3/'`
> >
> >if [ -e "$mp3" ]
> >then
> >continue
> >fi
> >
> ># combine the two tracks into one stereo file
> >/usr/local/bin/combine_wave -l $in -r $out -o $tmpwav 2>/dev/null
> >
> >/usr/bin/lame --silent -h -b 96 $tmpwav $mp3
> >
> ># Remove temporary .wav files
> >test -w $tmpwav && rm $tmpwav
> >
> ># Remove input files if successful
> >test -s $mp3 && rm $in $out
> > done
> >
> > exit 0
> >
>
>
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Re: [asterisk-users] Voicemail check
The user will receive email notification if you have configured the user's email address in /etc/asterisk/voicemail.conf . See: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+voicemail.conf Also check the externnotify option which lets you run an external script when new voicemail is received. S. On Jan 14, 2008 11:52 AM, Gilberto Nunes <[EMAIL PROTECTED]> wrote: > Hi all > > Someone knows how can I do to send any notify to user, when he received a new > message in your mailbox on voicemail? > > Thanks for any help > > -- > Gilberto Nunes > > Itajaí - SC > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Call Recording
You might take a few ideas from this combine.sh script which works for me. It uses the combine_wave program from http://panteltje.com/panteltje/dvd/combine_wave-0.3.tgz and the lame program to convert to mp3. It converts the entire directory /var/spool/asterisk/monitor/*-in.wav files to mp3 where the mp3 file doesn't already exist. S. File: combine.sh --- #!/bin/sh cd /var/spool/asterisk/monitor for f in *-in.wav do in=$f out=`echo $f | sed -e 's/-in.wav/-out.wav/'` tmpwav=`echo $f | sed -e 's/-in.wav/-both.wav/'` mp3=`echo $f | sed -e 's/-in.wav/.mp3/'` if [ -e "$mp3" ] then continue fi # combine the two tracks into one stereo file /usr/local/bin/combine_wave -l $in -r $out -o $tmpwav 2>/dev/null /usr/bin/lame --silent -h -b 96 $tmpwav $mp3 # Remove temporary .wav files test -w $tmpwav && rm $tmpwav # Remove input files if successful test -s $mp3 && rm $in $out done exit 0 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 330 beep on new VM
This is pretty easy to suppress using the configuration files. Check: http://www.voip-info.org/wiki/view/Polycom+SoundPoint+IP+MWI+audio On Dec 21, 2007 11:55 AM, Ugo Bellavance <[EMAIL PROTECTED]> wrote: > Hi, > > I have a Polycom 330 that emits a beep every 30s or so when there is a > message waiting. Is there a way to disable that? It is pretty annoying. > > Regards, > > Ugo > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DNS broken for www.voip-info.org ??
The DNS for www.voip-info.org seems to be non-responsive. Is there a mirror of this invaluable resource site? Tx, Steve dig www.voip-info.org ;; Got SERVFAIL reply from xxx.xxx.xxx.xxx, trying next server ; <<>> DiG 9.4.1-P1 <<>> www.voip-info.org ;; global options: printcmd ;; Got answer: ;; ->>HEADER<<- opcode: QUERY, status: SERVFAIL, id: 61402 ;; flags: qr rd ra; QUERY: 1, ANSWER: 0, AUTHORITY: 0, ADDITIONAL: 0 ;; QUESTION SECTION: ;www.voip-info.org. IN A ;; Query time: 4724 msec ;; SERVER: 127.0.0.1#53(127.0.0.1) ;; WHEN: Sat Dec 15 11:54:57 2007 ;; MSG SIZE rcvd: 35 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Asterisk-users] Show calls in progress
Is there an Asterisk CLI> command to show a list of calls in progress (for all channels: Zap/SIP/IAX2 etc). "Restart when convenient" waits until the system is idle, but is there an obvious way of seeing what's going on at the moment? Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk server and DSCP QOS
Thanks, Darryl, To clarify: in /etc/asterisk/sip.conf you have the lines: tos_sip=cs3; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. and in your Polycom configuration [I'm using Polycom's sip 2.2.0] you have something like (this is the one I'm uncertain about): Thanks again! Steve Darryl Duncan wrote: We're using 184 here (aka TOS 5/EF). Not set by iptables though, instead it is set in sip.conf (tos_sip/tos_audio) and on our Polycom/Cisco phones. -Original Message- Subject: [asterisk-users] Asterisk server and DSCP QOS Can anyone comment on the DSCP quality of service settings on your Asterisk server? The network we're setting up has data on the default VLAN, Asterisk server and phones on VLAN 4, and we're using Polycom phones with a PC hooked up to the phone's pass-thru port. What iptables settings are you using on the Asterisk server for DSCP? What are your Polycom DSCP settings? We're using the Linksys SGE2000P POE switch which supports QOS via DSCP. Thanks a lot for any info. Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk server and DSCP QOS
Can anyone comment on the DSCP quality of service settings on your Asterisk server? The network we're setting up has data on the default VLAN, Asterisk server and phones on VLAN 4, and we're using Polycom phones with a PC hooked up to the phone's pass-thru port. What iptables settings are you using on the Asterisk server for DSCP? What are your Polycom DSCP settings? We're using the Linksys SGE2000P POE switch which supports QOS via DSCP. Thanks a lot for any info. Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Requiring a login to a phone
Hi List, We have a remote asterisk SIP phone at the cottage. I'd like it to have minimal privileges when it first registers with Asterisk. Ideally it should be in a restricted context. Dialing any number would intercept the call and tell the person to log on. This way, if the phone was stolen or someone got into the cottage, we wouldn't have a bunch of surprise charges on our phone bill... :-) Once the phone has been authenticated, it should go into a context with normal privileges. After a couple of days of non-use, it should auto-logout to the restricted context. How can I change the sip context of a phone on the fly, based on authentication login? Any ideas? Thanks, Steve sip.conf: --- ; phone at the cottage [155] context=restricted-155 ... extensions.conf [restricted-155] exten _X.,1,NoOp(All Calls filter through this if not logged in on 155] exten _X.,n,Answer exten _X.,n,Wait(1) exten _X.,n,Playback(You must log in to use this phone) exten _X.,n,Authenticate(65535) // if the person authenticates sucessfully, change the context of ext 155 // from restricted-155 to sip-phones.(HOW???) [sip-phones] ; normal sip phone outgoing context ... ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
