Re: [Asterisk-Users] Shared call recordings with ARI!
On 5/9/06, Dan Littlejohn <[EMAIL PROTECTED]> wrote: On 5/9/06, Kevin P. Fleming <[EMAIL PROTECTED]> wrote: > Mimmus wrote: > > > Where is the problem? Asterisk or ARI? > > Since Asterisk has no control over permissions, where the files are > located or the users' names/passwords, it can't possibly be Asterisk. > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > It sounds like a configuration problem, but may be a bug. How do you have the variable $CALLMONITOR_ADMIN_EXTENSIONS set in main.conf.php? If you have all of the extensions set as admin, then everyone will be able to see everyone elses conversations. If you do not, then provide a snapshot of A or B ARI on the page showing the conversation. If you have the extension or callerid the same for A B C and D then they will be able to see all the conversations. Dan 512.791.0137 www.littlejohnconsulting.com I understand the problem better now. This is related to call file matching when not using a uniqueid in the cdr database. Since a uniqueid cannot be matched, a time range is used for matching, because the call recordings of on-demand calls do not start when the call starts. That means, if a call starts at 12:00:00 and ends at 12:01:00, all on-demand calls from 12:00:00 to 12:01:00 will match call log entries (example 12:00:37). The only way to fix this is to use the uniqueid matching method. The DMYSQL_LOGUNIQUEID must be set in the asterisk-addons module and in the ARI main.conf.php file, the setting $CALLMONITOR_ONLY_EXACT_MATCHING should be set to "1" I would default the asterisk-addons module to always set the uniqueid setting, but this is beyond my control. Dan 512.791.0137 www.littlejohnconsulting.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Shared call recordings with ARI!
On 5/9/06, Kevin P. Fleming <[EMAIL PROTECTED]> wrote: Mimmus wrote: > Where is the problem? Asterisk or ARI? Since Asterisk has no control over permissions, where the files are located or the users' names/passwords, it can't possibly be Asterisk. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users It sounds like a configuration problem, but may be a bug. How do you have the variable $CALLMONITOR_ADMIN_EXTENSIONS set in main.conf.php? If you have all of the extensions set as admin, then everyone will be able to see everyone elses conversations. If you do not, then provide a snapshot of A or B ARI on the page showing the conversation. If you have the extension or callerid the same for A B C and D then they will be able to see all the conversations. Dan 512.791.0137 www.littlejohnconsulting.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ODBC Storage for voicemail messages in database
Seems like other postings tend to think that saving recordings as files and not as blobs in the database are a more reliable way to go. Opinions on this? Looking at supporting it for ARI and judging interest. Dan 512.791.0137 www.littlejohnconsulting.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] what are these and can they be fixed?
Mar 21 15:56:25 DEBUG[18402] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Mar 21 15:56:25 DEBUG[18402] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Mar 21 15:56:25 DEBUG[18402] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Mar 21 15:56:25 DEBUG[18402] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Mar 21 15:56:26 DEBUG[18402] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Mar 21 15:56:26 DEBUG[18402] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Mar 21 15:56:27 DEBUG[18402] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Monitor
On 1/16/06, Matt Riddell (IT) <[EMAIL PROTECTED]> wrote: > Simon Faulkner wrote: > > Does anyone know of a web based live call monitor for *? > > > > I would have thought this was an ideal application for Ajax? > > There's the flash operator panel but nothing much using Ajax. We're > doing some chat room stuff but other than than I haven't seen much. > Sounds like a fun project :) > > -- > Cheers, > > Matt Riddell > ___ > > http://www.sineapps.com/news.php (Daily Asterisk News - html) > http://freevoip.gedameurope.com (Free Asterisk Voip Community) > http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > I saw this post about a week ago when I was trying to see if anyone else was trying out AJAX with Asterisk Applications. Happy to announce, that ARI is now AJAX enabled and that the voicemail and call monitor pages self update. You can take a look at it here www.littlejohnconsulting.com/ari and it has been checked into FreePBX svn. Dan www.littlejohnconsulting.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Web-Based Voicemail?
On 2/26/06, Sean Cook <[EMAIL PROTECTED]> wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > what about ARI, it gives web based access to the voicemail and is pretty > good at it... the default vmail.cgi is probably not the best as it has a > gaping security hole that allows anyone to listen to anyone elses > messages :) > > Sean > > Martin Joseph wrote: > > > > On Feb 26, 2006, at 12:57 AM, Alexander Burke wrote: > > > >> Hello, list! > >> > >> After Googling and checking out the voip-info wiki, I haven't had much > >> luck in locating a decent web-based voicemail system for Asterisk to > >> check your VM while you're away from the office without using a phone. > >> > >> Can anyone make any recommendations for such packages/applications? > > > > I like the emailing of messages. I email them directly to an > > account(IMAP) that has a webmail access, then I can view and listen from > > anywhere. > > > > This also creates a "backup" of the voicemail messages on a separate > > drive, which I also see as a positive. > > > > my 2c (us) > > Marty > > > > > > > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > -BEGIN PGP SIGNATURE- > Version: GnuPG v1.4.2 (MingW32) > Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org > > iD8DBQFEAhrEy9wPyZpnL2URAntcAJ4xWVmk3WTE8kWh+LIXjOnhNw2QfACbBybe > 7COlKpOrWR92IQWJt1h6kDs= > =2n+0 > -END PGP SIGNATURE- > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ARI is bundled with AMP or you can get it here. http://www.littlejohnconsulting.com/ari Dan 512.791.0137 www.littlejohnconsulting.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What to know for installing ARI
On 2/11/06, Zach A <[EMAIL PROTECTED]> wrote: > Hi everybody, > > I have an Asterisk box and I want to install just ARI on it for > monitoring the calls. Installing [EMAIL PROTECTED] utilizes too much > resources and > memory and also takes away freedom of configuration asterisk. I like > using asterisk on its CLI. But just for recorded calls I need to use > ARI. What I need to do for that. As I understand I need to install > Apache and MySQL on the same machine. What else I need to do. Is there > any step by step guide about it, or can somebody help me on this? > > Thanks, > > Zach > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > Hi Zach: Sorry for the slow response, but you can get ARI here: http://www.littlejohnconsulting.com/ari There are instructions for setting it up there as well. Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ARI 0.06
On 2/17/06, Jean-Marc Salsa <[EMAIL PROTECTED]> wrote: > Hi ! > > I always use your ARI through AAH, and indeed nice job ! > > A few comment : > - I have seen that we could use ARI only for the Call Monitor by setting a > value. would it be possible to do the same for "only Voicemail" ... indeed, > we are using Asterisk only for Voicemail, and this would be so good only to > present this tab to people ... ( And in Settings page also, hiding the Call > Monitor Settings part here too ) > - Same for help ( to show it or not ) > > I have installed it on our AAH 1.3 version and here are the error messages I > get : > > > Call Monitor Page (Only the first message on each page shows the "Play" > link): > Warning: is_dir(): Stat failed for > /var/lib/asterisk/bin/archive_recordings/ (errno=20 - Not a > directory) in > /var/www/html/recordings/includes/bootstrap.inc on line 113 > > Settings Page (Didn't try to apply new settings): > Warning: Invalid argument supplied for foreach() in > /var/www/html/recordings/modules/settings.module on line > 434 > Warning: Invalid argument supplied for foreach() in > /var/www/html/recordings/modules/settings.module on line > 473 > Warning: Invalid argument supplied for foreach() in > /var/www/html/recordings/modules/settings.module on line > 577 > > I hope you won't take these comments as critics, > you are really doing a GREAT job ! > Asterisk was really lacking this application part ! > > Thanks again, > > And all the best ! > > > Jean-Marc > > On 2/17/06, Dan Littlejohn <[EMAIL PROTECTED]> wrote: > > > > ARI (Asterisk Recording Interface) has reached another milestone. > > The project is starting to become a full featured user portal and > > handle all the common errors that people seem to have. This release > > supports: > > > > call monitor page – new features include column sorting and filter > > small duration calls > > in addition to the > ability to listen > > to call monitor recordings > > voicemail page – allows voicemail message listening and management > > handset feature code help page - I can never remember them all > > user settings web interface - that allows setting call fowarding, > > voicemail email and > >pager, > voicemail > > password, and call monitor recording > > > > There are also alot of i18n translations now, although with all the > > rework of the code many are now somewhat broken and need to be > > updated. If you speak one of the following, email and I will send you > > the page to translate or updating to the appropriate ari.po page and > > returning it to me would be very helpful. > > > > German > > Greek > > Spanish > > French > > Hebrew > > Hungarian > > Italian > > Portuguese > > Swedish > > > > If you would like to translate ARI into another language, I would be > > happy to support it. > > > > Loaded into AMP CVS and also here: > > www.littlejohnconsulting.com?q=ari > > > > If you have a chance, take a look. Comments and suggestions are welcome. > > > > Dan > > 512.791.0137 > > www.littlejohnconsulting.com > > > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > Jean-Marc: Thanks for the feedback. I have addressed these issues they are available on my website and have been checked into AMP cvs. I have added a setting to the /recording/includes/main.conf file. $ARI_DISABLED_MODULES = ""; allows for individual modules to be disabled (they are true modules though, and you can just delete them from the /recordings/modules directory) the is_dir error is a PHP bug. http://groups.google.com/group/mailing.www.php-dev/browse_frm/thread/1b5b94e775b70cdb/877e4406600a8121?lnk=st&q=Warning%3A+is_dir()%3A+Stat+failed+for+errno%3D20+-+Not+a+directory&rnum=1&hl=en#877e4406600a8121 But, I think I was able to suppress the error. The settings page errors have been corrected. Thanks; Dan 512.791.0137 www.littlejohnconsulting.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ARI 0.06
ARI (Asterisk Recording Interface) has reached another milestone. The project is starting to become a full featured user portal and handle all the common errors that people seem to have. This release supports: call monitor page – new features include column sorting and filter small duration calls in addition to the ability to listen to call monitor recordings voicemail page – allows voicemail message listening and management handset feature code help page - I can never remember them all user settings web interface - that allows setting call fowarding, voicemail email and pager, voicemail password, and call monitor recording There are also alot of i18n translations now, although with all the rework of the code many are now somewhat broken and need to be updated. If you speak one of the following, email and I will send you the page to translate or updating to the appropriate ari.po page and returning it to me would be very helpful. German Greek Spanish French Hebrew Hungarian Italian Portuguese Swedish If you would like to translate ARI into another language, I would be happy to support it. Loaded into AMP CVS and also here: www.littlejohnconsulting.com?q=ari If you have a chance, take a look. Comments and suggestions are welcome. Dan 512.791.0137 www.littlejohnconsulting.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with ARI and seeing voicemail...
On 2/6/06, Chuck Bunn <[EMAIL PROTECTED]> wrote: > Hi, > > I have tried both the stable version ARI-00.04.006 and the development > version ARI-00.05.018 with the same results. I can see call detail > records just fine but I cannot see any voicemail. I am using the > voicemail extension and password to log in but I still do not see > anything. If I log in as Admin with ari_password I see all of the call > detail but still no voice mail. Any ideas where I might look for my > problem. Voicemail is working since I can call the voicemail extension > and retrieve messages. I am not using AMP and I have set the standalone > flag to true. > > Thanks > > Chuck Bunn > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > Just for the archive. The fix was a permissions problem Changing the permissions of /var/spool/asterisk/voicemail fixed the problem, except this does not work for any new voicemails. The permanent fix is to add apache to the asterisk group. Dan www.littlejohnconsulting.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Lockups since upgrade 1.2.3 - anyone else? Any ideas?
On 1/27/06, Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote: > These modules are not part of the standard 1.2.3 release - did you also > install the 1.2.3 release of the asterisk-addons package ? > > If * is loading older modules (which it probably is because of your > config files) then it may cause grief ;) > > My .2p worth. Probably not helpful, but maybe, just maybe > > Julian > > Dan Littlejohn wrote: > > On 1/27/06, Noah Miller <[EMAIL PROTECTED]> wrote: > >> Hi Brent - > >> > >>> Boy oh boy. This blows. I upgraded to 1.2.2 from 1.0.9, and of course had > >>> the timebomb bug. Immediately after upgrading to 1.2.3 we were ok, for 24 > >>> hours or so. > >>> > >>> Since upgrading to 1.2.3, though, the whole system has locked up twice. > >>> Once > >>> on Thursday, and then about a half hour ago. The server would reply to a > >>> ping, but no ssh login, no local console login - just locked up. This > >>> ain't > >>> good for business. > >> > >> We've been doing fine with 1.2.3 so far. No problems reported, though I > >> only have it deployed in a small office. Definitely no lock-ups. > >> > >> On the asterisk side, just a basic question - did you make sure to remove > >> the old modules so the new 1.2.3 versions got installed? > >> > >> As far as the lockups, maybe it is coincidental? I've never had asterisk > >> (even the crazy CVS versions) lock a whole OS like that. I have had > >> machines running asterisk lock up, but it always turned out to be caused by > >> something else like bad hardware, or unrelated network problems. > >> > >> - Noah > >> > >> ___ > >> --Bandwidth and Colocation provided by Easynews.com -- > >> > >> Asterisk-Users mailing list > >> To UNSUBSCRIBE or update options visit: > >>http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > > > > > I was confused about the modules. > > > > Got this warning when upgrading to 1.2.3 even when using the most > > current asterisk-addons and even svn asterisk-addons. > > > > WARNING WARNING WARNING > > > > Your Asterisk modules directory, located at > > /usr/lib/asterisk/modules > > contains modules that were not installed by this > > version of Asterisk. Please ensure that these > > modules are compatible with this version before > > attempting to run Asterisk. > > > >app_addon_sql_mysql.so > >app_rxfax.so > >app_saycountpl.so > >app_striplsd.so > >app_substring.so > >app_txfax.so > >cdr_addon_mysql.so > >chan_modem_aopen.so > >chan_modem_bestdata.so > >chan_modem_i4l.so > >chan_modem.so > >format_mp3.so > >res_config_mysql.so > > > > WARNING WARNING WARNING > > > > Do not understand how to fix this? Do not know if that would also be > > related to the ops crashing. > > > > Dan > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > There is no asterisk-addons 1.2.3. Only 1.2.1 and I tried that and svn and still get this warning? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Lockups since upgrade 1.2.3 - anyone else? Any ideas?
On 1/27/06, Noah Miller <[EMAIL PROTECTED]> wrote: > Hi Brent - > > > Boy oh boy. This blows. I upgraded to 1.2.2 from 1.0.9, and of course had > > the timebomb bug. Immediately after upgrading to 1.2.3 we were ok, for 24 > > hours or so. > > > > Since upgrading to 1.2.3, though, the whole system has locked up twice. Once > > on Thursday, and then about a half hour ago. The server would reply to a > > ping, but no ssh login, no local console login - just locked up. This ain't > > good for business. > > > We've been doing fine with 1.2.3 so far. No problems reported, though I > only have it deployed in a small office. Definitely no lock-ups. > > On the asterisk side, just a basic question - did you make sure to remove > the old modules so the new 1.2.3 versions got installed? > > As far as the lockups, maybe it is coincidental? I've never had asterisk > (even the crazy CVS versions) lock a whole OS like that. I have had > machines running asterisk lock up, but it always turned out to be caused by > something else like bad hardware, or unrelated network problems. > > - Noah > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > I was confused about the modules. Got this warning when upgrading to 1.2.3 even when using the most current asterisk-addons and even svn asterisk-addons. WARNING WARNING WARNING Your Asterisk modules directory, located at /usr/lib/asterisk/modules contains modules that were not installed by this version of Asterisk. Please ensure that these modules are compatible with this version before attempting to run Asterisk. app_addon_sql_mysql.so app_rxfax.so app_saycountpl.so app_striplsd.so app_substring.so app_txfax.so cdr_addon_mysql.so chan_modem_aopen.so chan_modem_bestdata.so chan_modem_i4l.so chan_modem.so format_mp3.so res_config_mysql.so WARNING WARNING WARNING Do not understand how to fix this? Do not know if that would also be related to the ops crashing. Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Manager API and ZapBarge or ChanSpy
Am trying to monitor and record an in-process phone call using a
remote computer and the Asterisk Manager API. Have something that is
working, but the call recording volume is to low to be usable.
dialplan
exten => 8159,1,ZapBarge(Zap/1)
remote application with Asterisk Manager API
$telnet->print("Action: Originate\nChannel:
Local/[EMAIL PROTECTED]: ChanSpy\nData: |q\nPriority:
1\n\n");
$telnet->waitfor('/Response: Success/');
# get all the local channels and look for the extension in use
$telnet->print("Action: Command\nCommand: Local Show Channels\n\n");
$telnet->waitfor('/Response: Follows/');
while (($line = $telnet->getline) && ($line !~ /END COMMAND/i)) {
push(@channels,$line);
}
# start the monitor
while ($line = pop(@channels)) {
$pattern = "Local\/" . $exten;
if ($line =~ m/$pattern/i) {
print $line;
# start monitor
$recording = $timestamp . "-" . $uniqueid;
print $recording;
$telnet->print("Action: Monitor\nMix: 1\nFormat:
wav49\nChannel: " . $currentChannel ."\nFile: " . $recording .
"\n\n");
$telnet->waitfor('/Response: Success/');
}
}
What I think is happening is that a call is originated for the 8159
extension, which then executes the dialplan zapbarge on in process zap
channel call, then the chanspy listens in. This barely works, but the
call volume is just not usable. I am pretty sure I need to get rid of
the zapbarge or chanspy, but I am not sure how to go about originating
the call so it will work. Any advice would be appreciated.
Thanks;
Dan
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[Asterisk-Users] Re: TDM400 (TDM11B) configuration
On 1/9/06, Dan Littlejohn <[EMAIL PROTECTED]> wrote: > I have fixed this before, but I cannot for the life of me remember how I did > it. > > I have a TDM400P with one fxo module and one fxs module. I setup > Asterisk @Home and everything works fine, except when I try and call > out. If I call out with a SIP phone it calls the zap extension and > not the pstn line? If I take the zap extension offhook and call with > the SIP phone it dials out the pstn line fine. I am not sure why the > zap extension is being included in the group, but I cannot find where > to change it in AMP or the .conf files. Any help would be > appreciated. > > ; Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" > signalling=fxo_ks > ; Note: this is an extension. Create a ZAP extension in AMP for Channel 1 > context=from-internal > group=1 > channel => 1 > > ; channel 2, WCTDM, inactive. > ; channel 3, WCTDM, inactive. > signalling=fxs_ks > ; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 4 > context=from-pstn > group=0 > channel => 4 > > > Dan > Bad form to post to your own message, but I figured it out and thought anyone else interested would want to know. It is apparently some problem with how the zapata module loads. I switched the fxs and fso lines and all is well. Maybe something is inherited, etc. Seems like a bug to me. signalling=fxs_ks ; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 4 context=from-pstn group=0 channel => 4 ; Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" signalling=fxo_ks ; Note: this is an extension. Create a ZAP extension in AMP for Channel 1 context=from-internal group=1 channel => 1 Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400 (TDM11B) configuration
I have fixed this before, but I cannot for the life of me remember how I did it. I have a TDM400P with one fxo module and one fxs module. I setup Asterisk @Home and everything works fine, except when I try and call out. If I call out with a SIP phone it calls the zap extension and not the pstn line? If I take the zap extension offhook and call with the SIP phone it dials out the pstn line fine. I am not sure why the zap extension is being included in the group, but I cannot find where to change it in AMP or the .conf files. Any help would be appreciated. ; Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" signalling=fxo_ks ; Note: this is an extension. Create a ZAP extension in AMP for Channel 1 context=from-internal group=1 channel => 1 ; channel 2, WCTDM, inactive. ; channel 3, WCTDM, inactive. signalling=fxs_ks ; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 4 context=from-pstn group=0 channel => 4 Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chanspy options in Asterisk Manager API
The syntax for the options in chanspy are not well documented. How do I use multiple options? I am using the Asterisk Manager API and am using ChanSpy(|q) but would like to include volume ChanSpy(|q,v3) ? Any insight would be appreciated. Dan Littlejohn www.littlejohnconsulting.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Re: How to record a call
On 12/22/05, Blake Krone <[EMAIL PROTECTED]> wrote: > I'm running AAH 2.2 and *1 works from my eyebeam sip phones to do on demand > recording. > > You need to set the DIAL_OPTIONS of wW in order to utilize this feature. > lower case w means called person can initiate, upper case means callee can > initiate, I think that is the order. > > They show up as auto---.wav in > /var/spool/asterisk/monitor > However, they will NOT show up in ARI, I modified the code to show them and > sent the modification to Dan to implement if he chooses. > > -Blake > Thanks very much Blake. I have updated the software so it will now correctly recognize the on-demand call monitor recordings. It is available in AMP cvs and at the website http://www.littlejohnconsulting.com/?q=ari Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What's the best opensource web interface for customer portal
On 12/9/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > I'm looking for a good web interface for a customer portal for a residential > Voip business. It should give the customer the ability to set check > voicemail, set call handling options (forwarding, blocking, do not disturb, > etc), check usage, pay bills etc. I would like it if it were comparable to > the user portal for companies like Broadvoice, Vonage, Voicepulse, or > Sunrocket. > > Does anyone know of a good opensource solution for this? Please don't > suggest commercial packages unless they are really cheap (under $500), and > head and shoulders above the opensource solutions. > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ARI does most of what you need, although it does not yet have all of those features. http://www.littlejohnconsulting.com/?q=ari You could develop a spec and sponsor development if you wanted to stay with an open source solution. Dan Littlejohn 512.791.0137 littlejohnconsulting.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk GUI/web interfaces that don't change config files
On 10/28/05, Dustin Wildes <[EMAIL PROTECTED]> wrote: > Stay tuned for PhoneCALL's 2.7-RC1 release scheduled soon. We're adding > a new Security Manager that allows you to set the levels of editing for > your users/admins. > > > Chris Bagnall wrote: > > >Hello all, > > > >I'm trying to find an Asterisk web interface (or windows gui interface) to > >asterisk that won't allow users to go making changes to config files. I've > >trawled through the very extensive list in the wiki, but there doesn't seem > >to be a clear defining line between applications that are purely status > >viewers and ones that will allow config changes. > > > >I'm looking for the user to be able to do fairly simple things like see the > >last few people who called them, find out if other extensions are busy, add > >entries to the CLID directory and so on. > > > >Thanks in advance folks. > > > >Regards, > > > >Chris > > > > > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ARI allows for a user level experience. http://www.littlejohnconsulting.com/?q=node/11 Dan Littlejohn [EMAIL PROTECTED] www.littlejohnconsulting.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: web management interface
Steven: I believe so. If you recieve database errors you will have to disable those parts of ARI as well, but it might run out of the box without problems. Dan On 10/26/05, Steven <[EMAIL PROTECTED]> wrote: > Will this work if I am using text file configs? > > I started with AMP, but didn't like the limitations. > I disabled the DB config parts, but still use the other features of AMP. > > -- > -- > Steven > > May you have the peace and freedom that come from abandoning all hope of > having a better past. > ---- --- - - - -- - - -- - - - --- - -- - > - --- - - -- - -- -- - -- > "Dan Littlejohn" <[EMAIL PROTECTED]> wrote in message > news:[EMAIL PROTECTED] > Chris: > > ARI has been recently expanded into this space for end user > configuration. Don't know if you looked at it. > http://www.littlejohnconsulting.com/?q=node/11 > > It is works well coupled with AMP, but can be run stand alone as well. > Installation very easy and there are few dependencies. Just unzip > the tarball into /var/www/html and configuring the file > /var/www/html/recordings/includes/main.conf > > Features: > web Call Monitor recordings access > web Voicemail access > For end user settings > i18n language setting > voicemail password setting > voicemail audio format playback setting > call monitor settings > > I just spent a good bit of time over the last couple of weeks changing > the architecture to make it easy to add custom modules (contact me if > you would like to write one) and will be adding alot of end user > features shortly. Sure there is room for both solutions, this one is > PHP based. > > Regards; > Dan Littlejohn > [EMAIL PROTECTED] > www.littlejohnconsulting.com > > On 26 Oct 2005 15:29:39 -0400, astgroups <[EMAIL PROTECTED]> wrote: > > Common requests from my customers include; > > > > -MACs (moves,adds,changes) on extensions (sip, zaptel,CID) > > > > -Voice Prompt recording/modifying > > > > -CDR Access on the fly > > > > -Reboot/halt option > > > > -The Multi-tenant functionality would be very nice also.Big market for > > that. > > > > Hope this helps. Good luck! > > > > On Wed, 2005-10-26 at 13:59, snacktime wrote: > > > I'm finishing up a first version of a web interface for end users. > > > It's focus is specific for our own uses, but I plan on releasing it > > > under an open source license and would appreciate any feedback while I > > > wrap up the first version. > > > > > > The interface is designed for end users without any real technical > > > knowledge of asterisk except for some basic concepts of how things > > > relate to each other. Such as contexts in a dialplan and how they > > > relate to the context assigned to a sip/iax user, etc.. The interface > > > is for day to day management of areas such as the dialplan and > > > configuring new providers and phones in sip.conf and iax.conf. Things > > > that an end user would want to change on their own. It also includes > > > a nice voicemail interface for voicemail users, and some ability to > > > manage/monitor asterisk via the manager api. > > > > > > One of the main features is the ability to write canned scripts that > > > have associated configuration pages. A script is a text file with the > > > script, and a YAML definition file. In the text file you can put > > > variable placeholders, and in the YAML file you define the variables. > > > The web interface then builds an html form based on the text file and > > > the YAML definition. This way it's easy to add configurable sections > > > in extensions.conf without having to change any of the base code. For > > > instance providing canned scripts for extensions, call routing, voice > > > menu's, etc.. If you have a script that needs a more custom web > > > interface you can do that also by just creating the html form by > > > hand. The same template approach is also used for configuring phones. > > > > > > Since we will be using this for local and remote installations, we > > > also needed multi tenant capability. A basic multi tenant feature > > > set is built in, so multiple businesses can be maintained on one copy > > > of asterisk. > > > > > > Another requirement we had is to be able to coexist with an existing > > > asterisk installation, instead of requring that the management > > > interface take over
Re: [Asterisk-Users] web management interface
Chris: ARI has been recently expanded into this space for end user configuration. Don't know if you looked at it. http://www.littlejohnconsulting.com/?q=node/11 It is works well coupled with AMP, but can be run stand alone as well. Installation very easy and there are few dependencies. Just unzip the tarball into /var/www/html and configuring the file /var/www/html/recordings/includes/main.conf Features: web Call Monitor recordings access web Voicemail access For end user settings i18n language setting voicemail password setting voicemail audio format playback setting call monitor settings I just spent a good bit of time over the last couple of weeks changing the architecture to make it easy to add custom modules (contact me if you would like to write one) and will be adding alot of end user features shortly. Sure there is room for both solutions, this one is PHP based. Regards; Dan Littlejohn [EMAIL PROTECTED] www.littlejohnconsulting.com On 26 Oct 2005 15:29:39 -0400, astgroups <[EMAIL PROTECTED]> wrote: > Common requests from my customers include; > > -MACs (moves,adds,changes) on extensions (sip, zaptel,CID) > > -Voice Prompt recording/modifying > > -CDR Access on the fly > > -Reboot/halt option > > -The Multi-tenant functionality would be very nice also.Big market for > that. > > Hope this helps. Good luck! > > On Wed, 2005-10-26 at 13:59, snacktime wrote: > > I'm finishing up a first version of a web interface for end users. > > It's focus is specific for our own uses, but I plan on releasing it > > under an open source license and would appreciate any feedback while I > > wrap up the first version. > > > > The interface is designed for end users without any real technical > > knowledge of asterisk except for some basic concepts of how things > > relate to each other. Such as contexts in a dialplan and how they > > relate to the context assigned to a sip/iax user, etc.. The interface > > is for day to day management of areas such as the dialplan and > > configuring new providers and phones in sip.conf and iax.conf. Things > > that an end user would want to change on their own. It also includes > > a nice voicemail interface for voicemail users, and some ability to > > manage/monitor asterisk via the manager api. > > > > One of the main features is the ability to write canned scripts that > > have associated configuration pages. A script is a text file with the > > script, and a YAML definition file. In the text file you can put > > variable placeholders, and in the YAML file you define the variables. > > The web interface then builds an html form based on the text file and > > the YAML definition. This way it's easy to add configurable sections > > in extensions.conf without having to change any of the base code. For > > instance providing canned scripts for extensions, call routing, voice > > menu's, etc.. If you have a script that needs a more custom web > > interface you can do that also by just creating the html form by > > hand. The same template approach is also used for configuring phones. > > > > Since we will be using this for local and remote installations, we > > also needed multi tenant capability. A basic multi tenant feature > > set is built in, so multiple businesses can be maintained on one copy > > of asterisk. > > > > Another requirement we had is to be able to coexist with an existing > > asterisk installation, instead of requring that the management > > interface take over all the asterisk config files. All you have to do > > with asterisk is add one include line in each .conf file you want to > > manage. > > > > And last but not least, another reason we couldn't use any of the > > existing interfaces is that almost without exception all of them were > > too difficult to install. Or more correctly unnecessarily difficult. > > We need to have something we can hand our clients and know they will > > be able to install the thing and run it with little difficulty. > > Since this interface uses ruby on rails, it includes a built in > > webserver, and the installation is a matter of untarring the > > distribution into a directory, changing the ownership of the directory > > to something asterisk can read, and running the start script to bring > > up the webserver. If we can work out a bug in tar2rubyscript that > > makes it fail on freebsd, then the distribution will be just one > > single executable that you can run as is. > > > > > > I would be very interested in hearing about what features people would > > like in a tool like t
Re: [Asterisk-Users] ODBC Voicemail WEB Retrieval
On 9/20/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > Ok. > > I was sucessful in installing ODBC storage > > I'm using MySQL in the backend as it is. but all my drivers are now ODBC. > > I am running asterisk-cvs head as of last night 9/19/05 > > My question is this... the old voicemail.cgi script that allowed checking > voicemail no longer works etc, and never did work for me without a static > voicemail.conf file. > > Anyways.. that aside... how does one retrieve the longblob object from the > database and present it to the user (upon authentication) via a website. > > I'd be happy to help someone with the www/php/mysql integration but I just > dont know how to get blob's out and save to a temp file out of a database. > > Thanks > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > I would like to add Asterisk Realtime Support to ARI (www.littlejohnconsulting.com). Please contact me off list. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI + Ruby
On 9/12/05, joe heitzeberg <[EMAIL PROTECTED]> wrote:
> Hi Seshu,
>
> RAGI communicates with your Asterisk server over a socket, allowing
> you to create your call handling scripts in Ruby or Ruby on Rails.
> This allows you to build complex routines that using object models and
> database lookups in Ruby [on Rails], and not have to split your
> business logic across your app server and Asterisk's config files such
> as extension.conf. It makes it easy to do things like:
>
> user = User.find("phonenumber", connection.getVariable("callerid")
> if (user.balance()) < 5.00)
>connection.playSound("your-account-balance-is-low")
>#send email, send sms
>#update something in the database, etc
> end
> # The above is the same code and object model as used in your web app
>
> Joe
>
>
>
> On 9/12/05, Kanuri, Seshu (Company IT) <[EMAIL PROTECTED]> wrote:
> > What can RAGI do additionally that AGI or FastAgi and DeadAgi cannot do
> > which is already available under Asterisk?
> >
> > Seshu Kanuri
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of joe
> > heitzeberg
> > Sent: Sunday, September 11, 2005 12:31 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] AGI + Ruby
> >
> > Hi,
> >
> > We have created RAGI (Ruby Asterisk Gateway Interface) for the open
> > source community so that Ruby and Ruby on Rails can be used to easily
> > and effeciently create Asterisk-based applications. Examples: IVR,
> > call center apps, Asterisk management consoles, etc.
> >
> > RAGI includes a set of objects to interface over AGI to Asterisk for
> > handling inbound calls and outbound dialing, and includes a server
> > component, documentation and a sample apps to get you going quickly.
> >
> > Please see: http://ragi.sourceforge.net/
> >
> > The prelimenary release is available now on
> > https://sourceforge.net/projects/ragi
> >
> > We welcome input and development participation in the effort.
> >
> >
> > thanks,
> > Joe Heitzeberg
> > SNAPVINE
> >
> >
> >
> > On 8/24/05, Innocent Evil <[EMAIL PROTECTED]> wrote:
> > > I would like to write AGI script in Ruby Would anybody please show me
> > > right direction..
> > >
> > >
> > > Thanks___
> > > Asterisk-Users mailing list
> > > Asterisk-Users@lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > To UNSUBSCRIBE or update options visit:
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > ___
> > --Bandwidth and Colocation sponsored by Easynews.com --
> >
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > NOTICE: If received in error, please destroy and notify sender. Sender
> > does not waive confidentiality or privilege, and use is prohibited.
> >
> ___
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>
Joe:
You could do the same thing using PERL and strong classing.
sub new {
my ($class) = @_;
my $this = {};
bless($this,$class);
return $this;
}
Why would you pick Ruby over this? (not flaming, just trying to
understand the advantages) Granted, classes in Perl or PHP are not
perfect, but If you have to setup a socket and have the overhead, what
are the advantages?
Dan
___
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Re: [Asterisk-Users] CRM software
On 8/18/05, Lee Archer <[EMAIL PROTECTED]> wrote: > > > Can anyone recommend CRM software with a link into Asterisk? I would like a > pop up on caller ID if possible. I've played with the FOP and SugarCRM but > can't get them working together. > > Regards > > Lee > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > Don't know if you looked at this, but I fixed the script this morning and included the index.php file I am using for SugarCRM in the tarball. http://www.littlejohnconsulting.com/?q=node/15 Note that this only works with accounts. It uses the clid to look at the account phone number and then the phone numbers of any of the contacts in that account. If you would like it to work for just contacts it will need to be modified. Regards; Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo calibration with ztmonitor and a test line from a telco
On 8/15/05, Ken Dresdell <[EMAIL PROTECTED]> wrote: > Hello everyone, > > Does anyone have experience with echo calibration for TDM card with > rxgain and txgain (Zapata.conf) and ztmonitor (CVS Head)? > > I have found very few information about it and what I have found makes > me confused. I have a phone number provided by my TelCo(1004 hz at 0db) > and from what I saw, I am supposed to calibrate my rxgain to get a 14800 > value with ztmonitor . > > Here is the information I found: > > > http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.ht > ml > > > Does anyone have successfully reduced echo with this procedure? > > My main problem is that when I get 14800 with ztmonitor, I have now a > rxgain=14 and it seem to be too high for asterisk and I cannot dial out > anymore. > > Any suggestions? > > > Thanks in advance for your pointers > > Regards > > Ken > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > I have been doing a bit of this too lately. This was also useful. http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Wildcard - Hassle free clone?
On 8/9/05, Dan Littlejohn <[EMAIL PROTECTED]> wrote: > On 8/9/05, Douglas Logan <[EMAIL PROTECTED]> wrote: > > Now that the X100P is no longer being offered by Digium, what is the > > best solution? I seem to have run into a few posts where people talk > > about problems they've had with their X100P clone cards (dropping > > calls, echos, etc) other people seem to not have any problems. > > > > Of the three chipsets that will work: Intel 537EP, Ambient MD3200, and > > Motorola 62802 (as seen here > > http://www.voip-info.org/tiki-index.php?page=X100P+clone ) Is one of > > them more stable than another? What modem cards have you had the best > > luck with? Are there any ones to stay away from? > > > > I'll be using this for home use, so absolute reliability is not > > necissary. As a result I'd like to stay ~20 or less, and get the best > > quality I can for this price range. Thanks! > > > > Doug Logan > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > Doug: > > If you find that you like intel chipsets, I bought a compatable card > and did not need it. > http://www.newegg.com/OldVersion/app/ViewProductDesc.asp?description=25-180-004&DEPA=0 > > I'll sell it to you for $6 plus $3 shipping. > > Dan > Sorry about the list posting for this. It was not intensional. Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Wildcard - Hassle free clone?
On 8/9/05, Douglas Logan <[EMAIL PROTECTED]> wrote: > Now that the X100P is no longer being offered by Digium, what is the > best solution? I seem to have run into a few posts where people talk > about problems they've had with their X100P clone cards (dropping > calls, echos, etc) other people seem to not have any problems. > > Of the three chipsets that will work: Intel 537EP, Ambient MD3200, and > Motorola 62802 (as seen here > http://www.voip-info.org/tiki-index.php?page=X100P+clone ) Is one of > them more stable than another? What modem cards have you had the best > luck with? Are there any ones to stay away from? > > I'll be using this for home use, so absolute reliability is not > necissary. As a result I'd like to stay ~20 or less, and get the best > quality I can for this price range. Thanks! > > Doug Logan > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > Doug: If you find that you like intel chipsets, I bought a compatable card and did not need it. http://www.newegg.com/OldVersion/app/ViewProductDesc.asp?description=25-180-004&DEPA=0 I'll sell it to you for $6 plus $3 shipping. Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Monitoring
> > > On 7/27/05, Giorgio Incantalupo <[EMAIL PROTECTED]> wrote: > > Hi, > > if the file format is a problem, try Wavepad, it could help you. > > > > Giorgio > > > > Ian Bert Tusil wrote: > > > > > Can anyone help me how to open recorded converstations in asterisk? > > > > > > > > > > > >___ > > >Asterisk-Users mailing list > > >Asterisk-Users@lists.digium.com > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > >To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > -- > > > > > > GIORGIO INCANTALUPO > > Tel. +39 02 9350 4780 (104) > > > > FG&A Software > > 20017 Rho - Via Puccini, 8 > > > > E-Mail : > > [EMAIL PROTECTED] > > Internet: > > http://www.fgasoftware.com > > > > ___ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > I built a web interface named ARI (Asterisk Recording Interface). Download it here: http://www.littlejohnconsulting.com/?q=ari Place it in /var/www/html/recordings. AMP is including it in their distribution and I will make updates there and on my website. Regards; Dan Littlejohn (512) 791-0137 www.littlejohnconsulting.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: @Home AMP call recording documentation
Bad form to post to your own mailing, but I found the flash panel docs (http://www.asternic.org/) Oh well (I was blind or something) If someone could point me to the incoming/outfoing call recording feature for AMP it would be greatly appreciated. Dan On 6/16/05, Dan Littlejohn <[EMAIL PROTECTED]> wrote: > I have been looking through what I can find on @Home and AMP (wiki, > coalescentsystems.ca, maillists) and cannot find any documentation on > the incoming/outfoing call recording feature. If someone could some > point me to some I would be grateful. (and also for the flash panel, > like the default password). > > Thanks; > Dan > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] @Home AMP call recording documentation
I have been looking through what I can find on @Home and AMP (wiki, coalescentsystems.ca, maillists) and cannot find any documentation on the incoming/outfoing call recording feature. If someone could some point me to some I would be grateful. (and also for the flash panel, like the default password). Thanks; Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HT-488 vs. SPA-3000?
Perhaps there is something else going on the the Sipura 3000. Its voice quality and volume so poor/low that the device FXO port is not usable. However, same everthing and the TDM400P card works perfectly with excellent voice quality and volume. My experience, obviously just one data point. On 6/15/05, Tarpo, Louie <[EMAIL PROTECTED]> wrote: > I'm curious what other standalone FXO adapters work with Asterisk. At > everything from the default to the maximum in positive and negative values, > and combination of gain settings, we still get unacceptable distortion and > echo. I've checked the phone lines, they work normally with a regular phone. > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Rich > Adamson > Sent: Wednesday, June 15, 2005 1:05 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] HT-488 vs. SPA-3000? > > > The majority of the audio level issues seem to be on the fxo port > and setting the transmission levels (gain) to compensate for the > cable loss to the central office. Eg, setting the pstn gain values > to what should be appropriate causes echo, etc, not unlike the TDM > card. (I have both in use.) > > In other words, the further the spa3000 (or TDM card) is from the > central office, the more difficult it seems to be to set gain values > that are acceptable. That's apparently why many people find its use > is okay while others seem to think its objectionable. > > > > We have 6 SPA3000s. The device is extremely configurable and works > > inbound/outbound with > Asterisk with the latest firmware update with little trouble. However, we've > yet to resolve > sound volume and quality issues. The PSTN to SPA gain and SPA to PSTN gain > along with FXS Port > Input Gain and Output Gain settings have had no positive effect. The problem > is entirely with > the analog line adapter. VoIP calls from the analog phone to other VoIP > destinations are > perfect. We also have several SPA-1001s and SPA-2000s that have been running > perfect since day > 1. > > > > Also Sipura support is nonexistant. Just our experience. > > > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] Behalf Of Dan > > Littlejohn > > Sent: Wednesday, June 15, 2005 9:55 AM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] HT-488 vs. SPA-3000? > > > > > > I have only had experience with the Sipura 3000 and I would agree with > > the voice volume problems. I have given up on it working properly > > (adjusted gains, impedences, firmware, etc), the voice quality is just > > to low to actually use. I actually purchased a second one thinking > > that the first might be defective. > > > > Would not recommend it because of the low sound volume problem. > > Talking on the phone is actually the point of the device so who cares > > how configurable it is if you cannot hear anything. I purchased a > > Digium TDM400P and have had very good luck with it. > > > > Dan > > > > On 6/15/05, Rich Adamson <[EMAIL PROTECTED]> wrote: > > > > Just want to tap the collective wisdom of this list as to experiences > > > > pertaining to the Handytone HT488 and the Sipura SPA-3000 adapters... > > > > > > I've not played with the ht488, but I believe others have posted this > > > device does not provide access to the pstn-fxo port. The spa3k does > > > provide that access (if you want it). > > > > > > > Basically I'm looking for a FXO/FXS/LAN ATA and these two seems to be > > > > the top of the pick..Any comments and experiences esp. with Asterisk > > > > compatibility would be great, before I plonk in the bucks. > > > > > > The spa3k works fine with asterisk as many have posted. However, once > > > in awhile it does act a little strange in two different ways: > > > 1. the spa3k will sometimes interpret some voices as tones which cause > > > a little disturbance to any conversation going on. It is sort of like > > > the old telephony "talk off" that existed years ago. Doesn't happen > > > all that often and seems to be more sensitive to female voices based > > > on my one-year of experience. > > > 2. sometimes it seems to operate in half-duplex mode, where if you try > > > to talk at the same time as the other end is talking, the other end > > > won't hear you. > > > >
Re: [Asterisk-Users] HT-488 vs. SPA-3000?
I have only had experience with the Sipura 3000 and I would agree with the voice volume problems. I have given up on it working properly (adjusted gains, impedences, firmware, etc), the voice quality is just to low to actually use. I actually purchased a second one thinking that the first might be defective. Would not recommend it because of the low sound volume problem. Talking on the phone is actually the point of the device so who cares how configurable it is if you cannot hear anything. I purchased a Digium TDM400P and have had very good luck with it. Dan On 6/15/05, Rich Adamson <[EMAIL PROTECTED]> wrote: > > Just want to tap the collective wisdom of this list as to experiences > > pertaining to the Handytone HT488 and the Sipura SPA-3000 adapters... > > I've not played with the ht488, but I believe others have posted this > device does not provide access to the pstn-fxo port. The spa3k does > provide that access (if you want it). > > > Basically I'm looking for a FXO/FXS/LAN ATA and these two seems to be > > the top of the pick..Any comments and experiences esp. with Asterisk > > compatibility would be great, before I plonk in the bucks. > > The spa3k works fine with asterisk as many have posted. However, once > in awhile it does act a little strange in two different ways: > 1. the spa3k will sometimes interpret some voices as tones which cause > a little disturbance to any conversation going on. It is sort of like > the old telephony "talk off" that existed years ago. Doesn't happen > all that often and seems to be more sensitive to female voices based > on my one-year of experience. > 2. sometimes it seems to operate in half-duplex mode, where if you try > to talk at the same time as the other end is talking, the other end > won't hear you. > > Neither one of those have been all that objectionable to me, but they > happen and others have posted roughly the same issues. I've not heard > of anyone that has found a way to minimize those two issues. > > The down side of the spa3k right now is that Cisco bought the company > and there likely won't be much advancement of the code until after the > ownership (and development efforts) are sorted out by both companies. > (The same kind of product delays has been seen with their Linksys > purchase, as well as when other companies are bought/sold.) > > Its fairly common knowledge that ex-Cisco folks started Sipura for the > sole purpose of selling the company for a hugh profit. Their success > in accomplishing that objective could only be measured in terms of > producing Sipura products that had at least some acceptance of those > products by end users. With those previous objectives accomplished, > how will Cisco handle the Sipura products in the future? (It's any- > one's guess at this point since Cisco also has at least some track > record of mismanaging purchased companies for whatever reason.) > > >From an internal Cisco strategic perspective, they now own the assets > that can make a major dent in the mass-market end-user voip product > arena, and hopefully they'll take that in a positive direction. > > Given the price of the spa3k, I don't have any issue with purchasing > more of them right now. Excellent choice for the one-to-three pstn-fxo > market space. > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * @ Home: All Circuits busy
I got these errors and my hardware is working so I do not think they are an issue Hint: insmod errors & Removing zaptel module: zaptel: Device or resource busy What about the ztdummy module? Dan On 6/8/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > Dean, > > Here are the results of the genzaptelconf -s -d. As you can see, it is > throwing some errors, but I am a bit of a newbie so any help you could provide > would be greatly appreciated! > > [EMAIL PROTECTED] /]# genzaptelconf -s -d > > > STOPPING ASTERISK > Asterisk ended with exit status 0 > Asterisk shutdown normally. > > Disconnected from Asterisk server > Asterisk Stopped > > STOPPING FOP SERVER > FOP Server Stopped > Hint: insmod errors can be caused by incorrect module parameters, including > invalid IO or IRQ parameters. > You may find more information in syslog or the output from dmesg > Hint: insmod errors can be caused by incorrect module parameters, including > invalid IO or IRQ parameters. > You may find more information in syslog or the output from dmesg > Hint: insmod errors can be caused by incorrect module parameters, including > invalid IO or IRQ parameters. > You may find more information in syslog or the output from dmesg > Hint: insmod errors can be caused by incorrect module parameters, including > invalid IO or IRQ parameters. > You may find more information in syslog or the output from dmesg > Hint: insmod errors can be caused by incorrect module parameters, including > invalid IO or IRQ parameters. > You may find more information in syslog or the output from dmesg > Hint: insmod errors can be caused by incorrect module parameters, including > invalid IO or IRQ parameters. > You may find more information in syslog or the output from dmesg > Hint: insmod errors can be caused by incorrect module parameters, including > invalid IO or IRQ parameters. > You may find more information in syslog or the output from dmesg > Hint: insmod errors can be caused by incorrect module parameters, including > invalid IO or IRQ parameters. > You may find more information in syslog or the output from dmesg > Unloading zaptel hardware drivers: > Removing zaptel module: zaptel: Device or resource busy >[FAILED] > Loading zaptel framework: [ OK ] > Waiting for zap to come online ...OK > Loading zaptel hardware modules: > Running ztcfg: [ OK ] > > SETTING FILE PERMISSIONS > Permissions OK > > STARTING ASTERISK > Asterisk Started > > STARTING FOP SERVER > FOP Server Started > > ** SIP/200 in position 2 > ** SIP/201 in position 3 > ** SIP/202 in position 4 >Chan Extension Context Language MusicOnHold > pseudofrom-pstn en > Verbosity is at least 3 > [EMAIL PROTECTED] /]# > [EMAIL PROTECTED] /]# > > > Thanks, > Marc > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * @ Home: All Circuits busy
i am a newbie, but have you tried genzaptelconf -s -d Dan On 6/8/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > All, > > I have an [EMAIL PROTECTED] installation with a TDM40B card. I can make > internal > IP calls with no problems, but when I try to dial out I get a message that > "All > Circuits are Busy". I looked into the Zapata.conf files and such but see no > modifications. Is there a step that I am missing?? Does anyone have > documentation of step-by-step config for this TDM40B card? > > Thanks, > Marc > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Please comment on Dvorak's troll
The guy named himself after a keyboard (his pen name) and I have not finished one of his articles in a while. He is selling advertising though shock articles. Here is another gem. http://64.233.167.104/search?q=cache:qtQlMzEk9AYJ:linux.slashdot.org/article.pl%3Fsid%3D05/02/25/162243%26tid%3D109%26tid%3D106+dvorak+site:slashdot.org&hl=en I think you can safely discard this one too. Dan On 6/6/05, Colin Anderson <[EMAIL PROTECTED]> wrote: > > http://www.pcmag.com/article2/0,1759,1812887,00.asp > > Specifically, his assertion that ISP's would sniff traffic and block, say, > the SIP port. You could play wack-a-mole with port numbers, no? > > Also a community based, Freenet style of encryption implementation for > "free" VoIP traffic would address this issue. > > I raise this to the list because I'm sure there's a grain of truth in what > he's saying. ILEC's would be crazy to not consider this kind of lock in, > since it's pretty obvious that packet voice networks are going to supplant > circuit networks completely in, say, 20 years. Maybe sooner. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P vs SIP3000 x2
I just got through trying to set up a Sipura 3000 and am still looking for answers. There is a low volume problem (caller is underwater) on the FXO port that I wish someone would have told me about and I would have gone the other route. (even after upgrading firmware and adjusting gain settings) More details here. http://voxilla.com/index.php?name=PNphpBB2&file=viewtopic&t=3500&highlight=vol+volume http://voxilla.com/index.php?name=PNphpBB2&file=viewtopic&t=1249&highlight=vol+volume Maybe they work great and this one is defective, but others appear to have a simular problem and this was my experience. Dan On 5/26/05, Brian Roy <[EMAIL PROTECTED]> wrote: > On 5/26/05, Andres Paglayan <[EMAIL PROTECTED]> wrote: > > > > > I am about to start building my first ever * production server and would > > be nice to have some input from the list. > > My personal vote would be for the Sipura's. > > Pro's - > > It would make failing over to standby box much easier. > You could run a small 1u box and not have to worry about PCI requirements. > Lightens the load (especially interrupts) on the * box > PSTN doesn't have to be located by the * box, just by an ethernet port > > I think if you poll the archives, you would find problems with both of > them. I run a SPA3k and have had no problems with it at all. > > Just my .02 > > -Chuji > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura 3000 sound problems
I have configured a Sipura 3000 for FXO and FXS which appears to work well, except for an odd sound problem. Sometimes when the phone is off the hook the sound will fluctuate sometimes to the point where it cannot be heard. I am assuming that it is a line voltage problem or a bad box, but I have not been able to find anything to troubleshoot the problem. I have seen adjustments to gain etc, but not for the spa3000. In the US (Texas). Any help would be appreciated. Thanks; Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
