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Evening, Has anyone seen a how-to on getting Asterisk to work with Google Talk and Google Voice? Thanks

...ppens from windows xp home (even though they can't be domain members) the xp home computer can auth to pdc, bdc but not member server. the member server logs report: auth/auth_domain.c:domain_client_validate(246) domain_client_validate: unable to validate password for user username in domain MESSINET.COM to Domain controller HOME. Error was NT_STATUS_WRONG_PASSWORD. this error is reported many times successively on the member server, and nothing at all show up in the pdc or the bdc logs. i am thinking that i have misconfigured some parameter, but all of these configs worked prior to the 3.0.2...

...arameter here >> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate >> >> How can I pass callerid to following: >> >> exten => 6003,n,Originate(SIP/6003 at asterisk,app,meetme,"6003,x") > > I use this patch > > https://messinet.com/rpms/browser/asterisk/asterisk-12-app_originate_callerid.patch Thank you! I'll try it. > > because of https://issues.asterisk.org/jira/browse/ASTERISK-23016 Unfortunately , get The issue you are trying to view does not exist. on this link :-( > > -A > > > --------...

Hi, To check telco billing, I'm usinfg Asterisk-Stats from http://www.areski.net/asterisk-stat-v2/about.php . How can you tweak this application to display graphics and data that use Billsec instead of Duration ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is there a variable for the filename that is created by the FollowMe application when "a" is specified as an option to record the caller's name? I'd like to clean up the recorded name files after the call is complete. Thanks -Anthony -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part. Url : http:/...

Hello! I want to change call files, which has caller id in them, to call originate from dial plan. But I don't see such parameter here https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate How can I pass callerid to following: exten => 6003,n,Originate(SIP/6003 at asterisk,app,meetme,"6003,x") Thank you!

Hi all, I have done some extensive searching, and drawn a blank so far... Nothing odd is reported in samba logs, or in the syslog file. However, if I try to play an avi straight off the samba server, on an XP client with MP10, it brings the whole deal to its knees after a few mins at the most. I have to hard reset the server. Other than this, all my other uses are flawless (game server,

Hi All, Who do I need to poke to get the yum repository / RPM files updated? The dahdi RPMs are not up to date with the CentOS kernel versions any more, it's making doing an installation a bit tricky due to dependancies, I'd rather not roll back / remove new kernels if I don't have to.. Cheers Steve

Does anyone know if there are any plans to update the dahdi-linux kernel module code? It no longer compiles with recent kernels, and the last release of dahdi-linux appears to have been around March of 2016. I am currently running 4.6.3-300.fc24.x86_64 (on a Fedora system obviously) and the dahdi-linux-complete-2.11.1+2.11.1 release builds and runs under this kernel, but if I try to build it under

Using 1.6 on Fedora Core 9 I'm trying to receive faxes. I've got this far: [incoming-fax] exten => s,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d%H%M)}-0${CALLERIDNUM}) exten => s,2,ReceiveFAX(${FAXFILE}.tif) exten => s,3,Hangup() exten=>h,1,System(/usr/local/bin/fax2mail --cid-number "0${CALLERIDNUM}" --cid-name "home fax"

...ld proceed in the dialplan. asterisk]# cat /proc/zaptel/1 Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" (MASTER) 1 WCTDM/0/0 FXOKS (In use) 2 WCTDM/0/1 3 WCTDM/0/2 FXSKS (In use) 4 WCTDM/0/3 Where do I go with this? -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part. Url : http:/...

Hello, I'm using Asterisk 1.4.14, and I've noticed that the emails that are sent out when a user gets a voicemail don't have the timezone set in the header, so they're appearing in the user's email clients at the wrong time. Has anyone else seen this? I didn't find any bug reports or other info with Google. -- Jason Martin Metrix Matrix, Inc. 785 Elmgrove Road,

...install: cannot create regular file '/usr/lib/hotplug/firmware': No such file or directory make[1]: *** [hotplug-install] Error 1 make[1]: Leaving directory `/builddir/build/BUILD/dahdi- linux-2.9.1/drivers/dahdi/firmware' make: *** [install-firmware] Error 2 -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 198 bytes Desc: This is a digitally signed message part. URL: <htt...

...s - which one ? (Julian Lyndon-Smith) 17. Re: GSM gateways - which one ? (Michael Graves) 18. Re: Anonymous callerid (Tilghman Lesher) ---------------------------------------------------------------------- Message: 1 Date: Fri, 28 Nov 2008 15:46:53 -0600 From: Anthony Messina <amessina at messinet.com> Subject: Re: [asterisk-users] Any 1.6 SendFAX example ? To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Message-ID: <200811281546.54070.amessina at messinet.com> Content-Type: text/plain; charset="utf-8&qu...

...e I would expect that any extension that starts with at least one number, but includes a literal "*" followed by 1 or more numbers would match. This is not the case, and it matches any extension that starts with a number. Thank you in advance for your assistance. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part. Url : http:/...

...videosupport" isn't listed at www.voip-info.org. Do the order of the columns matter, etc.? Where do I go to find the latest Realtime table information for a specific Asterisk version? (The same questions apply to IAX Realtime.) Thank you in advance for your help. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part. Url : http:/...

...an. > But I don't see such parameter here > https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate > > How can I pass callerid to following: > > exten => 6003,n,Originate(SIP/6003 at asterisk,app,meetme,"6003,x") I use this patch https://messinet.com/rpms/browser/asterisk/asterisk-12-app_originate_callerid.patch because of https://issues.asterisk.org/jira/browse/ASTERISK-23016 -A -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E -------------- next part --------...

...I don't see such parameter here > https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate > > How can I pass callerid to following: > > exten => 6003,n,Originate(SIP/6003 at asterisk,app,meetme,"6003,x") > I use this patch > > https://messinet.com/rpms/browser/asterisk/asterisk-12-app_originate_calleri > d.patch > > Thank you! I'll try it. > because of https://issues.asterisk.org/jira/browse/ASTERISK-23016 > > Unfortunately , get > The issue you are trying to view does not exist. > on this link :-( Sorry fo...

Hi guys, I have 1 zap channel in my house shared among couple people. If someone dials 911, I want that zap channel to be disconnected right away to make way for the 911 call. I dug through voip-info.org and didn't find much. Any hints? kel

Greetings! We are using cisco 7940 phone with SIP and asterisk. We would like to be able to have phone directories available on the phones that are sourced from active directory. Are their any scripts that can connect to the AD server via LDAP and then create the directory.html file for the phones? Thanks! Liz -------------- next part -------------- An HTML attachment was scrubbed... URL: