similar to: Really wacky problem with internal extensions.

Displaying 20 results from an estimated 120 matches similar to: "Really wacky problem with internal extensions."

2006 May 23
0
Wacky Failover Situation w/SIP - Bug?
I have a weird situation. A polycom phone is configured to use system pbx1 as the primary outgoing 'proxy', followed by systems pbx3 and pbx2. All three systems have identical sip.conf files. The phone is registered on pbx1. I shut down the Asterisk application on pbx1. I make a call. The phone sends an INVITE to pbx2. Pbx2 sends back a 407 Proxy Auth message to the phone. The phone sends
2005 Feb 25
0
[Bug 2389] New: block/character devices on Solaris yield wacky rsync stats
https://bugzilla.samba.org/show_bug.cgi?id=2389 Summary: block/character devices on Solaris yield wacky rsync stats Product: rsync Version: 2.6.4 Platform: All OS/Version: Solaris Status: NEW Severity: normal Priority: P3 Component: core AssignedTo: wayned@samba.org
2000 Mar 14
1
Really Wacky Problem!
Hello, I'm getting a really wacky problem with samba running on solaris 2.7 ./smbclient -L zeus -N -- gives: session request to ZEUS failed (Called name not present) session request to *SMBSERVER failed (Called name not present) The NT side give "\\Zeus is not accesible." "network name was not found" messages The same smb.conf file works on a solaris 2.5 and 2.6
2005 Feb 25
3
[Bug 2389] block/character devices on Solaris yield wacky rsync stats
https://bugzilla.samba.org/show_bug.cgi?id=2389 ------- Additional Comments From woodd@deshaw.com 2005-02-24 16:43 ------- Created an attachment (id=979) --> (https://bugzilla.samba.org/attachment.cgi?id=979&action=view) Proposed syscall.c patch -- Configure bugmail: https://bugzilla.samba.org/userprefs.cgi?tab=email ------- You are receiving this mail because: ------- You are the
2009 Jul 31
1
asterisk 1.6 call forwarding
Dear All, I'n trying to make a simple call forwarding, however I have small problem when evaluating an expresion. Here is my extensions.conf ... ; Unconditional Call Forward exten => _#21*X.,1,Set(DB(CFIM/${CALLERID(num)})=${EXTEN:4}) exten => _#21*X.,2,Hangup() exten => #21#,1,Set(ignored=${DB_DELETE(CFIM/${CALLERID(num)})}) exten => #21#,2,Hangup() ... exten =>
2006 Mar 03
2
Does an entry in AstDB stay after reboot?
I set up a call forwarding script in extensions.conf which uses the AstDB but I'm wondering if I reboot the server, will the entry in AstDB still reside? What the script does is when a call comes in, it check to see if there is a null value or a call forward number. If null, it will call the local office connections. If there is a number, it calls that. Now I just need to know if I reboot
2005 Jul 06
0
re: help debugging dialplan
hello all, another desperate request for help debugging my dialplan... from a certain extension i do the following: DBput(CFIM/${CALLERIDNUM}=${CALLERIDNUM}) a NoOp to the console says DBput: family=CFIM, key=2122022001, value=2122022001 and database show says /CFIM/2122022001 : 2122022001 so far, so good. but in a macro, when i try to get the data, exten
2004 Nov 23
0
Problems with MACRO_EXTEN variable
Hei! I have a little problem with the subject. I use Asterisk CVS-HEAD-09/06/04-12:42:56 as a production *, but I do tests with a newer version Asterisk CVS-HEAD-11/18/04-10:01:32. Ok the problem is: in extension.conf I use macro for redirection, found on wiki pages: [macro-stdexten] ; ; Standard extension macro (with call forwarding): ; ${ARG1} - Extension(we could have used ${MACRO_EXTEN}
2006 Mar 08
1
Calls forwarding to numbers only in user's context
Hello, I'm trying to do call forwarding based on this: http://www.voip-info.org/wiki/view/Asterisk+call+forwarding In the extensions.conf file I have several context defined (local, longdistance, mobile, international and so on). Each user can be associated with different context (so can make only i.e. local calls). How to set calls forwarding only to numbers that are available in
2008 Sep 27
1
[PATCH 2/6 v3] PCI: add new general functions
Centralize capability related functions into several new functions and put PCI resource definitions into an enum. Cc: Jesse Barnes <jbarnes at virtuousgeek.org> Cc: Randy Dunlap <randy.dunlap at oracle.com> Cc: Grant Grundler <grundler at parisc-linux.org> Cc: Alex Chiang <achiang at hp.com> Cc: Matthew Wilcox <matthew at wil.cx> Cc: Roland Dreier <rdreier at
2008 Sep 27
1
[PATCH 2/6 v3] PCI: add new general functions
Centralize capability related functions into several new functions and put PCI resource definitions into an enum. Cc: Jesse Barnes <jbarnes at virtuousgeek.org> Cc: Randy Dunlap <randy.dunlap at oracle.com> Cc: Grant Grundler <grundler at parisc-linux.org> Cc: Alex Chiang <achiang at hp.com> Cc: Matthew Wilcox <matthew at wil.cx> Cc: Roland Dreier <rdreier at
2005 May 15
1
Old DBGet/DBPut vs. new Set(var=${DB(...
Hello I upgraded to CVS head yesterday (due to the lack of zaptel drivers working with 2.6.10) And noticed that now DBGet and DBPut have been deprecated in favour of the new Set/DB one. In the UPGRADING.txt in Asterisk it says: * The applications DBGet and DBPut have been deprecated in favor of functions. Here is a table of their replacements: DBGet(foo=family/key)
2005 Jan 10
0
Problems calling between two local SIP extensions
Hi, I have two local SIP extensions (both bt100). One is on remote location behind another nat (16), but everyithing seems to be setup correctly as it can register and is listed as OK(57ms). However I can only call in one direction between those two. Extensions are defined in same context: exten => 11,1,Macro(oneline,SIP/11) exten => 16,1,Macro(oneline,SIP/16) both using same macro
2009 Apr 17
1
how to call forward on 1.6
Hello, I want to enable call forwarding for asterisk 1.6.0.6 I couldnt seen any config or option on gui or extensions.conf about it. I found some dialing plans to enable it on web as follows: [apps] ; Unconditional Call Forward exten => _*21*X.,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:4}) exten => _*21*X.,2,Hangup exten => #21#,1,DBdel(CFIM/${CALLERIDNUM}) exten => #21#,2,Hangup ;
2004 Jul 07
0
Audio cuts off 10 minutes into calls
Hello list, We run Asterisk CVS-HEAD-06/02/04-11:25:18 built by root@Gate01 on a i686 running Linux. All works fine except Audio is lost 10minutes into the call. This happens for every call PSTN-SIP, SIP-PSTN, SIP-SIP Example of one call setup using Snom200 and Grandstream 486: -- Executing Macro("SIP/xxxx1251-d638", "CFW|xxxx1251|SIP/xxxx1253") in new stack -- Executing
2004 Oct 05
0
loggedoff extension - why does * say "is onthephone"
Same here, I just changed the b to u. Unavailable message is more generic, but it beats it saying busy when its not. -Tim -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Henry Devito Sent: Tuesday, October 05, 2004 8:31 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE:
2004 Dec 09
1
No ring signal when calling internal extensions ?
Hi, I have attached configuration settings and cannot get ring signal when calling internal extensions. I'm probably doing something wrong so would kindly ask for a tip how to do it properly : exten => 11,1,Macro(oneline,SIP/11) Calling 11 (this is the same with BT or iax softphones) doesn't give me a ring - what is missing ? Thanks, Rob. [macro-oneline] ; ; Standard extension
2005 Jan 30
0
Setting call forward for Agent's in a Queue
Hi!, I'm trying to set up a Queue (which works fine now :-) Sip clients can login in to the Queue with dialing 91 on there phone. And as soon as there are customers the Queue calls the agents back. I would like that the queue calls the agents also if it's phone is call-forwarded. With agents (sip clients) are added with the following extensions: exten => 91,1,AddQueueMember(myqueue)
2005 Feb 11
0
Playing Dialtones
In AU we have a number of different dialtones defined for various purposes. >From indications.conf: au <ringcadance> 400,200,400,2000 au dial 413+438 au busy 425/375,0/375 au ring 413+438/400,0/200,413+438/400,0/2000 au congestion 425/375,0/375,420/375,0/375 au callwaiting 425/200,0/200,425/200,0/4400 au
2004 Oct 05
1
loggedoff extension - why does * say "is on the phone"
Hi, I have following one-line macro extension: ------------------------ [macro-oneline] ; ; Standard extension macro (with call forwarding): ; ${ARG1} - Device(s) to ring ; #exten => s,1,AGI(misterhouse.agi,"CallerID") exten => s,1,NoOp exten => s,2,DBget(temp=CFIM/${MACRO_EXTEN}) ; Get CFIM key, if not existing, goto 103 exten => s,3,Dial(Local/${temp}@default/n) ;