search for: localsettings

https://bugzilla.samba.org/show_bug.cgi?id=3432 Summary: rsync -azv --cvs-exclude forgets "LocalSettings.php" Product: rsync Version: 2.6.6 Platform: x86 URL: http://pto.linux.dk/albackup.tgz OS/Version: Linux Status: NEW Severity: major Priority: P3 Component: core AssignedTo: wayned@samba.org...

My last problem was nicely solved through this mailing list so hopefully this new problem will have the same happy outcome. My situation is that I have many extensions. Here is a sample: [client-phone](!) type=friend host=dynamic secret=XXXXXXXXXX dtmfmode=auto disallow=all allow=ulaw allow=gsm allow=g723 allow=ilbc subscribemwi=no [4165555555](client-phone) secret=xxxxxxxxxxxxxxxxxxxxxxxxxx

Asterisk 11.1.0 One queue with strategy=leastrecent. (Full queues.conf below.) Occasionally (several times today), a caller will get "stuck" in the queue - there are operators available to take the call, but the caller stays in the queue for a long time. Any idea what might cause this, or where I can start looking to debug it? I'm going to start digging through the queue log

I have this in my sip.conf: exten => *98,1,Verbose(0,CALLERID number is "${CALLERID(num)}") same => n,VoicemailMain(${CALLERID(num)}@LocalSets,s) same => n,Hangup However, my extensions are set up so that they always show the external number, not the extension: [foobar2](client-phone) secret=xxxxxxxxxxxxxxxxxxxxxxxxxxxxx callerid=Candace <5555551212>

On 2017-04-18 02:42 AM, Pete Mundy wrote: > Try this: > > asterisk -r > core set verbose 10 > [get user to trigger fault] > [examine console output, and post to list if still unclear] > > If you don't solve it yourself, then we'll be able to help further once > we've seen the output. I can't see much more than at my previous debug level but here it is

On 2017-04-18 08:17 PM, Pete Mundy wrote: >> On 19/04/2017, at 7:58 am, D'Arcy Cain <darcy at VybeNetworks.com >> <mailto:darcy at VybeNetworks.com>> wrote: >> >> <snip> >> Everything looks the same as another one that works except for two >> things. The one that works doesn't have the "Probation passed" lines. >> I am

Try this for CHAN_SIP: same => n,Set(Peer=${SIPCHANINFO(peername)}) ; Get the peer same => n,Set(MailBox=${SIPPEER(${Peer},mailbox)}); Get the mailbox same => n,VoicemailMain(${MailBox}@LocalSets,s) ; If we have a mailbox defined log into it If you are using PJSIP it's more complex same => n,Set(EndPoint=${CHANNEL(endpoint)}) ; Get the peer same =>

On Fri, 12 Jun 2015 11:49:05 -0700 John Kiniston <johnkiniston at gmail.com> wrote: > Try this for CHAN_SIP: > > same => n,Set(Peer=${SIPCHANINFO(peername)}) ; Get the peer > same => n,Set(MailBox=${SIPPEER(${Peer},mailbox)}); Get the > mailbox same => n,VoicemailMain(${MailBox}@LocalSets,s) ; If we > have a mailbox defined log into it Perfect.

Hello, I have Asterisk version: Asterisk SVN-branch-11-r420435 I have the following code: exten => 303,1,NoOp(Dialing ${EXTEN}) ? ? ? ? same => n,NoOp(DBKey = ${DBKey}) ? ? ? ? same => n,DB_DELETE(office/${DBKey}) ? ? ? ? ? same => n,Playback(auth-thankyou) ? ? ? ? same => n,Hangup() And I get the following error: [2014-08-09 18:00:30] WARNING[4338][C-00000067]: pbx.c:4869

Hello, I'm trying to create a H.323 trunk between Asterisk 11 and Avaya. I have done this before between Asterisk 1.6 and Avaya but had some issues placing external calls from the Asterisk to the Public network which is connected to Avaya. I'm trying to create that trunk on Asterisk 11 because the 1.6 is outdated and has no support. On the Asterisk side I have Aastra 6731i SIP phones

This just started after upgrading to 11.11.0. After a call is completed (both ends hang up) the call still shows as active. # asterisk -x "core show channels" Channel Location State Application(Data) SIP/thinktel-0000000 (None) Up AppDial((Outgoing Line)) SIP/4164251212-00000 4165555555 at LocalSets Up Dial(SIP/thinktel/4165559999) 2 active

On 12/3/19 3:04 PM, Joshua C. Colp wrote: > >     The AMI command, after the login, looks like this: > > > >     Action: Originate > >     Channel: SIP/outgoing/%%(destination)s > >     Context: LocalSets > >     CallerID: Vybe Consulting Inc Fax Service <5555551212> > >     Exten: sendfax > >   

On 12/2/19 11:52 AM, Joshua C. Colp wrote: > So I know that AMI is listening and I can talk to it.  Here is the > main log" > > [Nov 27 06:16:00] VERBOSE[101155] asterisk.c: Remote UNIX connection > [Nov 27 06:16:00] VERBOSE[101245] asterisk.c: Remote UNIX connection > disconnected > [Nov 27 06:16:01] VERBOSE[101244] manager.c: Manager

Hello list, I'm hoping that you could read through this mail and give me some tips on how to improve my setup (functionality, security, really anything). It's my first Asterisk installation and meant for simple home use. I installed Asterisk 11 on an OpenWrt Barrier Breaker router. Currently it's configured for Ekiga so I can test. In a few weeks I'll change to a Telco SIP

We have a template for extensions and voicmail. They look like this: exten => %ACCOUNT%,1,Verbose(0,Entering extension %ACCOUNT%) same => n(DialDesk),Verbose(0,${CALLERID(all)} Calling ${EXTEN}) same => n,Dial(SIP/%ACCOUNT%,30) same => n(VoiceMail),Set(CDR(userfield)=VoiceMail) same => n,Verbose(0,${CALLERID(all)} going into voice mail for %ACCOUNT%)

Hi, If agents is already logged in via AgentLogin() and users dialled extension 300 which will be placed in Queue(support-queue). How to find out which agent is available I can put their Agent id in AgentRequest() ? If this is not a good approach then how it should be done? Agent should automatically get next call when he/she is available. extensions.conf [LocalSets] exten =>

I recently upgraded from Asterisk 13.19 to 16.6.1. Everything is working fine with a few minor tweaks except outgoinf fax. Incoming works fine. I do outgoing faxing through an AMI call. Here is the output from the security log: [Nov 27 06:16:05] SECURITY[101222] res_security_log.c:

*Dear List* Plz help, i am not much experienced with asterisk. i configured it on ubuntu 12.04. no problem when i call any mobile no(0091XXXXXXXXXX) but when i call on my local asterisk no.(101,102 or 105) it is not connecting giving error "Auto fallthrough, channel 'SIP/lucknow-0000006f' status is 'CHANUNAVAIL' *while when i call 200 it is playing audiofile successfully.

Dear all, When receiving a fax, the extension is "spawned", despite nothing but positive messages (see below) The sending fax considers it a success & the verbose output of asterisk gives a "FAX_SUCCESS" and a "NO_ERROR" error in the ReceiveFax command. The problem is that all the next steps (conversion of the fax to pdf & sending it to a mailbox) are never

Hello everyone, A few days ago I had a problem with a couple of extensions. I have about 12 Aastra 6731i phones, 6 are at our main office and 6 more on remote branches. We use VPN to communicate to our branches so there's no NAT involved any where. The problem I had was that I couldn't call from two extensions located at two branch offices. But I could call to them just fine. On any call