similar to: Polycom not picking up phone transferred phone call.

Ok, that's done. Now after this the should compile and run : in En.rc MAIN_MENU is defined now like this : MAIN_MENU MENU { POPUP "&File" { MENUITEM "&New...", 0x100 En.rc is included by #include "En.rc" in rsrc.rc. And gcc complies about it when compiling : [syl@snoop notepad]$ make gcc -c -I. -I. -I../../include -I../../include -g -O2 -Wall

Hello, I have problem with interpretation of CDR entries. What happened? ------------- There was: 1. at 09:00:26 we received call from unknown caller (no callerid) to secretary with extension 17 2. secretary answered and after some conversation called to extension 18 to check if she could transfer customer call 3. at 09:02:55 extension 18 answered and call was forwarded 4. call was very long

Hi The first step I have to go through when users call into our IVR is to handle the case where users' PBX hides their CID number. In that case, I need to have them type their phone number (ten digits). OTOH, those who call without hiding their CID number are sent directly to the main menu. How would I go about prompting users for their phone number? Here's what I have at this point:

Hi, I was trying out call file just to see how they worked and my system does not seem to do anything with them, although asterisk *is* deleting the files that I put into /var/spool/asterisk/outgoing. 1. I nano'd a quick call file like so: Channel: SIP/axVoice/9105555555 CallerID : Leebo <5555555555> MaxRetries: 2 RetryTime: 30 WaitTime: 10 Context: main_menu Extension: s Priority:

Hi all, I have configured the queue below, but when I go into the queue, asterisk does not announce hold time: [support] musiconhold=>default strategy=ringall context=check_time timeout=20 wrapuptime=1 maxlen=3 announce-frequency=5 announce-holdtime=yes joinempty=no leavewhenempty=yes reportholdtime=yes I've tried changing timeout, announce-frequency, but still the same; queue works

Hi all, I have just setup a trixbox system and I am implementing hylafax/iaxmodem solution for the faxing. When i send a fax to it by phoning in listening to the IVR and manually pressing start to initate the fax, the call gets picked up correctly as a fax and everything works well. When I try sending a fax by entering the phone number and pressing start to initiate dialing it sounds like

Hello Instead of using PrivacyManager, I'd rather use my own dialplan to prompt the user for a ten-digit number if they called while blocking CID. This code does prompt the user, but 1) hangs up if the user didn't type the ten digits before the timeout 2) if the user did type the right number of digits, it still hangs up instead of Returning and then jumping forth to the "cid"

Hi all, Running 1.2.14 When I call a FastAGI script such as this script for an incoming call: [calldirect] exten=>s,1,Answer() exten=>s,2,AGI(agi://192.168.1.175/calldirect?check&${CALLERID(num)}) exten=>s,3,Goto(check_time,s,1) and the FastAGI server is not running (Asterisk gets "connection refused" TCP error), Asterisk just terminates the call like so: May 17

If you are using Windows to generate the .call files, make sure they are in Unix format (LF only at EOL, not CR+LF) - Notepad makes bad Unix files. Use Crimson Editor www.crimsoneditor.com to make the file, and click Document > File Format > Unix Format. I ran into this same problem, and it turns out my Asterisk install would not use Windows-formatted text files, it would just ignore them

Hello About Record(), ATFT 2nd Edition says that "if the filename contains %d, these characters will be replaced with a number incremented by one each time the file is recorded." Problem is, the documentation doesn't explain how to refer to this filename later in the dialplan :-/ In this particular example, I want to move the file to the web server's /htdocs so users can

[Sorry if this arrives more than once. I have sent this twice and it never arrived, despite other messages getting to the list O.K.] ----------- Hello, I would like an incoming caller to be able to choose from the menu options in my extension.conf below. Once They have dialled the appropriate digit, * should call two extensions simultaneously: one SIP phone on this * server, and one over a

Hi all, After reading great things about the OSLEC Echo Canceller (http://www.rowetel.com/ucasterisk/oslec) and seeing the reactions of people who have tried it on a recent Trixbox thread (http://www.trixbox.org/forums/trixbox-forums/open-discussion/need-people-echo-problems), it sounds like it is the "bees knees" for sorting out echo problems with cards like the x100p. Has anyone

Hi all, I've just setup a sip line with Telasip and when they route the calls to my asterisk box, they include an extension along with the context that is defined in sip.conf for that DID. At first, I couldn't figure why they were getting 404 error from my asterisk box, but then figured out that they are sending the call to an extension that matches my number with them, in the

Add a way to disable the timer IRQ routing check via a boot option. The VMI timer code uses this to avoid triggering the pester Mingo code, which probes for some very unusual and broken motherboard routings. It fires 100% of the time when using a paravirtual delay mechanism instead of using a realtime delay, since there is no elapsed real time, and the 4 timer IRQs have not yet been delivered.

Add a way to disable the timer IRQ routing check via a boot option. The VMI timer code uses this to avoid triggering the pester Mingo code, which probes for some very unusual and broken motherboard routings. It fires 100% of the time when using a paravirtual delay mechanism instead of using a realtime delay, since there is no elapsed real time, and the 4 timer IRQs have not yet been delivered.

Hello List, Im just dipping my feet into the asterisk world, and im having major fxo problems Im running Asterisk (from svn) + libpri (from svn) + asterisk-addons (from svn) + asterisk gui (svn 1.4 branch) + zaptel (svn 1.4) on a Debian Etch box, with 1gb ram, running all of the services for my home server (web / db / music server etc), and i would like to run my PSTN line from Kingston Comms,

Isolate some of the non-VMI timer related changes in Linux. This patch moves the cyc_2_ns conversion code into a common location, eliminating redundant code in hpet and tsc timer implementations, and introduces some macros that may be redefined by the sub-architecture to avoid dependence on APIC routing, CMOS time sync, and testing for broken time hardware (which presumably, does not happen in a

Isolate some of the non-VMI timer related changes in Linux. This patch moves the cyc_2_ns conversion code into a common location, eliminating redundant code in hpet and tsc timer implementations, and introduces some macros that may be redefined by the sub-architecture to avoid dependence on APIC routing, CMOS time sync, and testing for broken time hardware (which presumably, does not happen in a

hi, i''ve got main site navigation in the application.rhtml file. the links in this template correspond to the ''index'' page of various controllers. The controllers themselves each have navigation of their own, which i would like to appear in every view for a particular controller in *addition* to the main site navigation. The problem I am encountering is that

# HG changeset patch # User Ian Campbell <ian.campbell@citrix.com> # Date 1254298855 0 # Node ID 5215da46d60f95d57244e709cb3b189caffec50c # Parent 6472342c8ab0789b844714bcf557e9e5eeacca42 Disable HPET broadcast mode on kexec. Without this the new kernel cannot receive timer interrupts from the legacy sources. Hangs are observed in the second kernel''s "check_timer()"