[asterisk-users] tellabs 2572 echo can and zaptel yellow alarm
I have a tellabs 2572 echo can wired and setup according to the instructions in http://www.voip-info.org/tiki-index.php?page=Tellabs+Hardware+Echo+Cancellers . I added the echo can between my zaptel card and my adit 600 channel bank. I used the straight thru and crossover cables as instructed. I get a Yellow alarm on my zaptel card (digium x100p) and on the echo can under *Send In* the rem light is yellow and under *Rcv In* the local light is red. Are there any changes I should be making to my adit 600 setup or my zaptel.conf? on the echo can option 20 is 5(ESF), 60 is 1(B8ZS), and 63 is 0 my zaptel.conf fxsks=1-8 fxoks=9-48 span=1,1,7,esf,b8zs span=2,1,7,esf,b8zs adit 600 -- SLOT A: Settings for DS1 1: Circuit ID: CAC DS1# A:1 Up/Down: UP Framing: ESF Line Coding: B8ZS Line Build Out: DSX-1 EQUALIZATION FOR 0-133 ft. (CSU 0dB) Loop Code Detection: ON Loopback:OFF FDL Type:None ___ --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] Camp on?
Why not just create a .call file when the number is busy? The .call
file tries to dial the destination with the retry interval and max
attempts you specify, when the call goes thru, dial that other number.
Here's what I have been using to produce a callback feature. It's not
pretty but it works
**
; menu callback initiated
exten = *,1,GotoIf($[ ${CALLERID(num)} 800 ]?:i|1)
exten = *,n,System(/usr/bin/ast_callback ${CALLERID(num)} ${CALLEDEXTEN} )
exten = *,n,AGI(tts_playback.sh,call back activated,li)
exten = *,n,Hangup()
**
The GotoIf statement is checking to see if the call is an internal call
The CALLEDEXTEN variable that I defined contains the extension number
you dialed, you could also have it be the channel (ie Zap/2) your dialing.
This is the /usr/bin/ast_callback script.
**
#!/bin/bash
#
# Initiate a call file to call back busy extension
callerid=` echo $1 | sed -e 's///g'`
called=` echo $2 | sed -e 's///g'`
CALLBACKFILE=$(cat -EOF1
Channel: local/[EMAIL PROTECTED]
MaxRetries: 3
RetryTime: 5
WaitTime: 5
CallerID: Call Back $callerid
Context: callback
Extension: $called
Priority: 1
EOF1)
echo $CALLBACKFILE /var/spool/asterisk/callback/c_back$callerid$$
mv /var/spool/asterisk/callback/c_back$callerid$$
/var/spool/asterisk/outgoing/c_back$callerid$$
**
This is the context that the ast_callback script call file 'dials' back
into:
**
[callback]
exten = _8XX,1(available),ChanIsAvail(${ext_${EXTEN}}|s)
exten = _8XX,n,System(/usr/bin/ast_callback_ok ${CALLERID(num)}
${ext_${EXTEN}} )
exten = _8XX,available+101,System(/usr/bin/ast_sleep 5)
exten = _8XX,n,Set(loops=$[ ${loops} + 1 ])
exten = _8XX,n,GotoIf($[ ${loops} 120 ]?available:)
exten = _8XX,n,Hangup()
**
I have all my extension channels defined as global variables (ie
ext_820 = Zap/13). So ${ext_${EXTEN}} maps extension number to channel.
If they the phone is no longer busy another call file is initiated where
the extension you were attempting to reach will be calling you back.
If it's still busy we wait 5 seconds and try again. There must be a
better way to do a 5 second wait but I used a script.
**
#!/bin/bash
#
# Initiate a call file to call back extension that is no longer busy
callerid=` echo $1 | sed -e 's///g'`
called=` echo $2 | sed -e 's///g'`
CALLBACKFILE=$(cat -EOF1
Channel: $called
MaxRetries: 2
RetryTime: 15
WaitTime: 15
CallerID: Call Back $callerid
Context: internal
Extension: $callerid
Priority: 1
EOF1)
echo $CALLBACKFILE /var/spool/asterisk/callback/c_back$callerid$$
mv /var/spool/asterisk/callback/c_back$callerid$$
/var/spool/asterisk/outgoing/c_back$callerid$$
**
**
#!/bin/bash
#
# sleep for x seconds
sleep $1
**
___
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Re: [Asterisk-Users] How to check if a phone / line is used?
In the dialplan you can use ChanIsAvail command Show channels? On Mar 31, 2006, at 2:09 AM, Ronald Wiplinger wrote: In the past I used SetGroup and CheckGroup to figure out if my allowed providers lines are all used or not. Since most of my provider have given me a single line anyway, I wonder if there is a way to check if this (provider) line is taken already. How can I do that? Same is with the phone. How can I see in CLI if a phone is now in use or not? Sip show peers shows me just if it is on-line, but not if it is in a call or not. In the dialplan I could dial the number and if it is busy, it would go to the Voicemail for unavailable or busy. I expect that there is just a test function as well, without trying to call. bye Ronald Wiplinger ___ ___ --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] Fresh checkout Zaptel will not compile?
I then compiled libpri fine and moved on to zaptel. Did a make clean then make install and get the following error: How about `make linux26' ? I am not up to speed on make or its errors, but it looks like to me that it is complaining about /usr/src/zaptel not being there or that modules is missing. AS you can see from the last line there is a /usr/src/zaptel directory. Or is it something with my 2.6 kernel and a modules directory or something. Have never gotten this error before, that is why I am asking for help. Robert do you have kernel sources installed? I just dealt with these same issue on a recent install on debian. You don't need the kernel sources, but you do need the kernel headers for the kernel image you're running. You need to have a sym-link named /lib/modules/`uname -r`/build/ which links to your kernel header directory. Zaptel wouldn't compile properly until I removed the 2.6 kernel source files, because I had the 2.6 version of the kernel sources and my kernel image and headers were 2.6.11. Marv Horst "dicovers" the kernel source ___ --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] detect SIP phone availability before dialing
Use application ChanIsAvail with the s option. This option only exists in CVS-HEAD version, the 1.0.x versions don't have this option. from documentation: If the option 's' is specified (state), will consider channel unavailable when the channel is in use at all, even if it can take another call. This is a pretty popular question. IIRC SIP phones can't tell you their statuses, you need to send a call to them and determine whether or not they're Busy Now... [EMAIL PROTECTED] wrote: Hello, I need to detect availability of SIP phone before dialing. I need to know if phone is BUSY, CHANUNAVAIL before dialing. If phone is free, then I will dial it. ___ --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] SOHO Survey / Creative Asterisk Solutions
You can do some really fun things interfacing asterisk with IO control equipment. We use Opto22 Snap Brains and asterisk for various purposes. Opto provides some nice linux API code examples to interface with their units. I have a 3D tube laser cutter interfaced to asterisk so that someone gets called or paged when the cutter runs out of material or an error occurs, we also control air compressors with asterisk, and can check on various equipment status. Brian McEntire wrote: Hehe... that's awesome :) I laughed out loud when I read it. Someone else replied that they are going to use * to control their entry gate system by cell phone. Nice. Thanks for the examples! While reading over at voip-info.org, I found the auto-dial feature that can be combined with .call files. That should be perfect for an idea I had -- use cron and POP3 to check my e-mail account for any new messages from the transit authority... if there are any, dial/ring home phones at 6am and playback a message to check e-mail for possible morning commute problems. Not quite as good as drunkdial though :) ___ --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] Wifi UT Starcom F1000: Raising Audio volume Level via asterisk?
[EMAIL PROTECTED] wrote: Hi all, I' m currently testing a Wifi UT Starcom F1000 and for my taste even with the audio volume level even on max. setting, the speaker is not loud enogh. Is there a way to raise the audio-level specifically for a concrete SIP device via a sip.conf setting or something like that ?? Check if you have the latest firmware, which is version 3.1 . It has increased the volume output somewhat. The manufacturer doesn't provide the firmware directly, you have to get it from the reseller. In my case that was voipsupply.com, who replied promptly to my email request. ___ 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] c++ class for agi?
Being a lazy person, I was wondering if anyone has a c++ class for interfacing with asterisk AGI? I am aware of the C library listed on the wiki. ___ 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] Hangup Faster
David Sampson wrote: Hello My single line extension users (connected via channel banks) need to be able to hang up faster. If they just flash the hook it doesnt disconnect right away. Any ideas on how to resolve this? Thanks, Dave In zapata.conf put this line. rxflash=50 This may prevent flash from working properly. I don't know for certain because I have those functions turned off. threewaycalling=no transfer=no ___ 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] app_valetparking.c
Try this
Since www.bkw.org seems not to exist anymore (getting response from
some hosting provider), does anyone happend to have a copy of
app_valetparking.c from www.bkw.org - the one that should work with *
stable 1.0.X ? If so please contact me.
One that can be downloaded from www.loligo.com dosn't compile with
1.0.X, and SuperValletParking (www.asterlink.com/svp/) seems to be for
* HEAD (1.1.X), so it wont do me any good.
/*
* Asterisk -- A telephony toolkit for Linux.
*
* Routines implementing call valetparking
*
* Copyright (C) 1999, Mark Spencer
*
* Mark Spencer [EMAIL PROTECTED]
*
* This program is free software, distributed under the terms of
* the GNU General Public License
*/
#include asterisk/lock.h
#include asterisk/file.h
#include asterisk/logger.h
#include asterisk/channel.h
#include asterisk/pbx.h
#include asterisk/options.h
#include asterisk/module.h
#include asterisk/translate.h
#include asterisk/say.h
#include asterisk/callerid.h
#include asterisk/channel_pvt.h
#include asterisk/parking.h
#include asterisk/musiconhold.h
#include asterisk/config.h
#include asterisk/cli.h
#include asterisk/app.h
#include asterisk/manager.h
#include stdlib.h
#include errno.h
#include unistd.h
#include string.h
#include stdlib.h
#include stdio.h
#include sys/time.h
#include sys/signal.h
#include netinet/in.h
#include pthread.h
#define DEFAULT_VALETPARK_TIME 45000
static char *valetparking = ValetParking;
static char *valetparkedcall = ValetParkCall;
static char *valetunparkedcall = ValetUnparkCall;
static char *valetparklist = ValetParkList;
/* No more than 45 seconds valetparked before you do something with them */
static int valetparkingtime = DEFAULT_VALETPARK_TIME;
/* First available extension for valetparking */
static int valetparking_start = 1;
/* Last available extension for valetparking */
static int valetparking_stop = 1;
static char *vpsynopsis = Valet Parking;
static char *vpcsynopsis = Valet Park Call;
static char *vupsynopsis = Valet UnPark Call;
static char *vlsynopsis = ValetParkList;
static char *vpdesc =
ValetParking(exten|lotname|timeout[|return_ext][|return_pri][|return_context])\n
Auto-Sense Valet Parking: if exten is not occupied, park it, if it is
already parked, bridge to it.\n\n;
static char *vpcdesc =
ValetParkCall(exten|lotname|timeout[|return_ext][|return_pri][|return_context])\n
Park Call at exten in lotname until someone calls ValetUnparkCall on the
same exten + lotname\n
set exten to 'auto' to auto-choose the slot.\n\n;
static char *vupdesc =
ValetUnparkCall(exten|lotname)\n
Un-Park the call at exten in lot lotname use 'fifo' or 'filo' for
auto-ordered Un-Park.\n\n;
static char *vldesc =
ValetParkList(lotname)\n
Audibly list the slot number of all the calls in lotname press * to unpark
it.\n\n;
struct valetparkeduser {
struct ast_channel *chan;
struct timeval start;
int valetparkingnum;
/* Where to go if our valetparking time expires */
char context[AST_MAX_EXTENSION];
char exten[AST_MAX_EXTENSION];
char lotname[AST_MAX_EXTENSION];
int priority;
int valetparkingtime;
struct valetparkeduser *next;
};
static struct valetparkeduser *valetparkinglot;
static ast_mutex_t valetparking_lock = AST_MUTEX_INITIALIZER;
static pthread_t valetparking_thread;
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int valetparking_count(void)
{
struct valetparkeduser *cur;
int x=0;
ast_mutex_lock(valetparking_lock);
for(cur = valetparkinglot;cur;cur = cur-next)
x++;
ast_mutex_unlock(valetparking_lock);
return x;
}
static int valetparking_say(struct ast_channel *chan,char *lotname)
{
struct valetparkeduser *cur;
int x=0,y=0,res=0;
int list[1024];
if(!lotname)
return 0;
ast_mutex_lock(valetparking_lock);
for(cur = valetparkinglot;cur;cur = cur-next)
if(cur-lotname !strcmp(lotname,cur-lotname))
list[y++] = cur-valetparkingnum;
ast_mutex_unlock(valetparking_lock);
for(x=0;xy;x++) {
ast_say_digits(chan,list[x], , chan-language);
res = ast_waitfordigit(chan,1500);
if(res != 0) {
res = list[x];
break;
}
}
return res;
}
static int ast_pop_valetparking_top(char *lotname)
{
struct valetparkeduser *cur;
ast_mutex_lock(valetparking_lock);
for(cur = valetparkinglot;cur;cur = cur-next)
if(cur-lotname !strcmp(lotname,cur-lotname))
break;
ast_mutex_unlock(valetparking_lock);
return cur ? cur-valetparkingnum : 0;
}
static int ast_pop_valetparking_bot(char *lotname)
{
struct valetparkeduser *cur,*last=NULL;
ast_mutex_lock(valetparking_lock);
for(cur =
Re: [Asterisk-Users] Re: app_valetparking.c for * STABLE (1.0.X)
Oops, I sent the wrong one. Here's one I modified to work with 1.0.X
Try again
Nope ! This is the one that tries to include PRE 1.0.X header file
parking.h.
It cannot compile on * 1.0.X (I have tried also to include
features.h instead of parking.h (as far as I know features.h is
successor to parking.h), but still without results).
Thanks anyway.
Nenad
Try this
/*
* Asterisk -- A telephony toolkit for Linux.
*
* Routines implementing call valetparking
*
* Copyright (C) 1999, Mark Spencer
*
* Mark Spencer [EMAIL PROTECTED]
*
* This program is free software, distributed under the terms of
* the GNU General Public License
*/
#include asterisk/lock.h
#include asterisk/file.h
#include asterisk/logger.h
#include asterisk/channel.h
#include asterisk/pbx.h
#include asterisk/options.h
#include asterisk/module.h
#include asterisk/translate.h
#include asterisk/say.h
#include asterisk/callerid.h
#include asterisk/channel_pvt.h
#include asterisk/features.h
#include asterisk/musiconhold.h
#include asterisk/config.h
#include asterisk/cli.h
#include asterisk/utils.h
#include asterisk/app.h
#include asterisk/manager.h
#include stdlib.h
#include errno.h
#include unistd.h
#include string.h
#include stdlib.h
#include stdio.h
#include sys/time.h
#include sys/signal.h
#include netinet/in.h
#include pthread.h
#define DEFAULT_VALETPARK_TIME 45000
static char *valetparking = ValetParking;
static char *valetparkedcall = ValetParkCall;
static char *valetunparkedcall = ValetUnparkCall;
static char *valetparklist = ValetParkList;
/* No more than 45 seconds valetparked before you do something with them */
static int valetparkingtime = DEFAULT_VALETPARK_TIME;
/* First available extension for valetparking */
static int valetparking_start = 1;
/* Last available extension for valetparking */
static int valetparking_stop = 1;
static char *vpsynopsis = Valet Parking;
static char *vpcsynopsis = Valet Park Call;
static char *vupsynopsis = Valet UnPark Call;
static char *vlsynopsis = ValetParkList;
static char *vpdesc =
ValetParking(exten|lotname|timeout[|return_ext][|return_pri][|return_context])\n
Auto-Sense Valet Parking: if exten is not occupied, park it, if it is
already parked, bridge to it.\n\n;
static char *vpcdesc =
ValetParkCall(exten|lotname|timeout[|return_ext][|return_pri][|return_context])\n
Park Call at exten in lotname until someone calls ValetUnparkCall on the
same exten + lotname\n
set exten to 'auto' to auto-choose the slot.\n\n;
static char *vupdesc =
ValetUnparkCall(exten|lotname)\n
Un-Park the call at exten in lot lotname use 'fifo' or 'filo' for
auto-ordered Un-Park.\n\n;
static char *vldesc =
ValetParkList(lotname)\n
Audibly list the slot number of all the calls in lotname press * to unpark
it.\n\n;
struct valetparkeduser {
struct ast_channel *chan;
struct timeval start;
int valetparkingnum;
/* Where to go if our valetparking time expires */
char context[AST_MAX_EXTENSION];
char exten[AST_MAX_EXTENSION];
char lotname[AST_MAX_EXTENSION];
int priority;
int valetparkingtime;
struct valetparkeduser *next;
};
static struct valetparkeduser *valetparkinglot;
AST_MUTEX_DEFINE_STATIC(valetparking_lock);
static pthread_t valetparking_thread;
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int valetparking_count(void)
{
struct valetparkeduser *cur;
int x=0;
ast_mutex_lock(valetparking_lock);
for(cur = valetparkinglot;cur;cur = cur-next)
x++;
ast_mutex_unlock(valetparking_lock);
return x;
}
static int valetparking_say(struct ast_channel *chan,char *lotname)
{
struct valetparkeduser *cur;
int x=0,y=0,res=0;
int list[1024];
if(!lotname)
return 0;
ast_mutex_lock(valetparking_lock);
for(cur = valetparkinglot;cur;cur = cur-next)
if(cur-lotname !strcmp(lotname,cur-lotname))
list[y++] = cur-valetparkingnum;
ast_mutex_unlock(valetparking_lock);
for(x=0;xy;x++) {
ast_say_digits(chan,list[x], , chan-language);
res = ast_waitfordigit(chan,1500);
if(res != 0) {
res = list[x];
break;
}
}
return res;
}
static int ast_pop_valetparking_top(char *lotname)
{
struct valetparkeduser *cur;
ast_mutex_lock(valetparking_lock);
for(cur = valetparkinglot;cur;cur = cur-next)
if(cur-lotname !strcmp(lotname,cur-lotname))
break;
ast_mutex_unlock(valetparking_lock);
return cur ? cur-valetparkingnum : 0;
}
static int ast_pop_valetparking_bot(char *lotname)
{
struct valetparkeduser *cur,*last=NULL;
ast_mutex_lock(valetparking_lock);
for(cur = valetparkinglot;cur;cur = cur-next) {
if(cur-lotname
Re: [Asterisk-Users] Options for Attendant Console.
www.quadrasoftware.com Kyle Will McCown wrote: We've been playing with Asterisk with an eye towards possibly replacing or augmenting our existing PBX serving about over 600 phones (and needing to expand). The one missing bit that I can't find any mention of is an Attendant Console. Are there any good solutions out there? I've considered that maybe one of the better softphones might suffice, but the ones I've looked at so far not geared to the need to handle and quickly dispatch large numbers of calls. www.asternic.org ___ 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 transfer questions
also tried the following without luck [featuremap] blindxfer = #1; Blind transfer disconnect = *0 ; Disconnect automon = *1 ; One Touch Record atxfer = *2 it still seems to want to accept only # as transfer I am running Asterisk CVS-v1-0-03/07/05-06:50:06 You are running V1.0.x stable of asterisk. Tthe attended transfer feature is only available in CVS-HEAD, which at some point (June ?) will become 1.1.x stable ___ 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] Hardphone deployment recommendation
I'm looking to purchase and deploy a bunch of hardphones for agent use. The phones will have to register with Asterisk and/or SER, depending on where the phones go. They need only one line, G729 codec, and no super fancy features. Preferrably something that is easy to provision. I would think the BudgeTone would be good, but then I've read so many people complaining about them, and some people seem to recommend the Sipura adapters. I'm looking to keep my cost down, and the BudgeTone is around $100 CDN, give or take. I just received my sipura SPA-841 phone (not an ATA), and it's a nicer than the BudgeTone for about the same price. I paid $85.00 for it. Features it has that the BudgeTone doesn't: * 2 line appearances (you can add 2 more for a total of 4) transfer and conference work when using more than 1 line appearance * headset jack * voicemail button It doesn't have a network switch or POE. ___ 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] Way to disable # as transfer and just take thekey.
This is quite possibly the most popular question on this mailing list. - Remove t/T options from the Dial command this only works in CVS-HEAD not 1.0x stable - Change the transfer-key in features.conf ___ 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] Help With Adit 600 Configuration
When you say regular do you mean straight through ?, and can you kindly confirm for me if the HyperTerminal settings I am using are OK?. I am definitely not using a null-modem cable, and yes the port works cos I use it every day on various Cisco gear I have called Carrier Access, I am just waiting for them to get in to the office, so I can register, any help in the mean time is still very much appreciated though Press Enter several times after connecting to the adit 600, because it doesn't send any login prompt. I copied these setup instructions from the adit manual: Setting up a CLI Connection If connecting via RS-232, the port settings should be set to: Bits per second: 9600 Data bits: 8 Parity: None Stop bits: 1 Flow control: None Set your Terminal Emulation to: VT100 NOTE: When using Tera Term TCP/IP, CLI commands will not be recognized until the following setup is completed. In Tera Term go to Setup/Terminal. Set the New-line/Transmit valueCR+LF. ___ 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 command 'stream file' not working?
. Specifically, X is not a digit, you must either use for no interuptions permitted or use 0123456789 for all digits available to interupt. I also 'discovered' that you cannot send a sequence of commands to asterisk without reading the results between each command submission. Similar to the use of the manager interface. Thanks Marv Horst ___ 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] agi command 'stream file' not working?
I'm running Asterisk CVS-v1-0-12/28/04
I use a short php agi script to get the temperature from an opto22 module.
STREAM FILE temperature does nothing, but SAY NUMBER works fine.
What am I doing wrong?
#!/usr/bin/php -q
?
$output = @shell_exec('eiocl 192.168.10.212 2001 10 44 r ap');
$words = explode( , $output);
echo STREAM FILE temperature X \n;
fflush( STDOUT );
echo SAY NUMBER .intval($words[count($words)-1]) . X\n;
fflush( STDOUT );
?
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Re: [Asterisk-Users] Out the box solutions?
I chose FC3 (Fedora Core) for the install, and now I'm sorry that I did. I setup an asterisk server on FC3 without any problems using with some help from the wiki. http://www.voip-info.org/wiki-Asterisk+Fedora+Core+3 It doesn't require a kernel compile, just a few conf file changes and a different make command when building the zaptel drivers, because of the 2.6 kernel udev system I'm not so interested in notifying these guys at lists.sourceforge.net, since I'm only interested in running asterisk. Once I commit to actually using linux I might participate in their forum, but not yet :) So ... the question: What flavor of linux does asterisk actually run on Out the box? I'm not scared to compile asterisk, but I'm not at all interested in recompiling a linux kernel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SuperValetParkCall Application Unable to Re-ParkCall
Just in case you didn't know, app_valetparking does have the ability to
park to a
specific park number. It doesn't have the VDial stuff in
app_supervaletparking.
Here is how I use valet parking:
exten = _5XX,1,Playback(beep)
exten = _5XX,2,ValetUnParkCall(filo|8${EXTEN:1})
exten = _6XX,1,ValetParkCall(auto|8${EXTEN:1:2}|60|6${EXTEN:1:2}|2|pbx_ext)
Thanks, for the feedback. It would be nice if the Valetpark application
didn't have that limitation. The park to a specific park number is a
nice feature.
-Original Message-
From: Paul Zimm [mailto:[EMAIL PROTECTED]
Sent: Friday, December 24, 2004 11:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SuperValetParkCall Application Unable to
Re-ParkCall
Kevin wrote:
After retrieving a SuperValetParkCall using the SuperValetUnparkCall
command, I am unable to re- SuperValetParkCall the call again. Can
anyone confirm if this is a bug or my configs may be incorrect.
Configs:
exten =
3,1,SuperValetParkcall(${EXTEN:1}|mylot|500|${EXTEN:1}|1|local)
exten = _3,2,Hangup
exten = _*3,1,SuperValetUnParkCall(${EXTEN:2}|mylot)
exten = _*3,2,Hangup
This is not a bug. I experienced the same problem, and took a peek at
the code.
Apparently app_supervaletparking uses it's own channel bridge function,
which
is very basic and doesn't process the # transfer command.
I need the ability to transfer a call after a SuperValetUnparkCall so I
went back to using app_valetparking. I wasn't using the new features
in
app_supervaletparking, just the park and unpark functions.
Marv Horst
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Re: [Asterisk-Users] SuperValetParkCall Application Unable to Re-Park Call
Kevin wrote:
After retrieving a SuperValetParkCall using the SuperValetUnparkCall
command, I am unable to re- SuperValetParkCall the call again. Can
anyone confirm if this is a bug or my configs may be incorrect.
Configs:
exten =
3,1,SuperValetParkcall(${EXTEN:1}|mylot|500|${EXTEN:1}|1|local)
exten = _3,2,Hangup
exten = _*3,1,SuperValetUnParkCall(${EXTEN:2}|mylot)
exten = _*3,2,Hangup
This is not a bug. I experienced the same problem, and took a peek at
the code.
Apparently app_supervaletparking uses it's own channel bridge function,
which
is very basic and doesn't process the # transfer command.
I need the ability to transfer a call after a SuperValetUnparkCall so I
went back to using app_valetparking. I wasn't using the new features in
app_supervaletparking, just the park and unpark functions.
Marv Horst
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Re: [Asterisk-Users] directed call pickup
Is anyone using directed call pickup? *8+exten to only pick up an extension if the phone is ringing. The wiki says asterisk supports it but it seems it does not work. What am I doing wrong? Directed call pickup for a specific extension is not currently part of Asterisk CVS. It is available as an add-on app see: http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002692 ___ 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] transfer not working with supervaletparking
I'm running Asterisk CVS-v1-0-12/09/04 with app_supervaletparking. The problem is, after I park a call and then unpark the call I can no longer # transfer the call on either end. Here is some debug from asterisk -- Executing SuperValetParkCall(Zap/11-1, auto|824|180|824|1|home) in new stack == Super Valet Parked Zap/11-1 on slot 1 -- Starting simple switch on 'Zap/9-1' -- Executing Answer(Zap/9-1, ) in new stack -- Executing SuperValetUnparkCall(Zap/9-1, fifo|824) in new stack -- Channel Zap/9-1 connected to SuperValet Parked call 1 in lot 824 At this point I have reconnected with the call but I can't transfer the call. The # key gets transmitted through to the other end of the call. The older version app_valetparking didn't have this problem, but that version apparently doesn't work with Asterisk. 1.0+ It's a great app :) please help! Marv Horst ___ 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: Top posting
So, that's how my tax dollars are spent? Outrageous, and certainly news-worthy. Good luck fighting off CNN and the like when this leaks out. Not at all, this is one of my favorite policies that has come from the performance improvement department. Yes that is right, it is official policy at my location to not deal with people who top-post. PI decided that with people moved around between positions it is always best for bottom-posting just as if on a mailing list even in two party communications as, if another person comes into the discussion, it is much quicker, and thus cheaper, to have a properly formatted communication to come up to speed. This is the same as the policy that businesses that send ill-formatted bussiness letters will not receive addition business when there is another suplier capable of delivering the product/service. Top-posting is even grounds for being written up if you later need to forward a copy of a message on to another department or person. It's no wonder that people gripe about dealing with government bureaucracy. Too pedantic in my opinion. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Hooking up a an Adit 600
Richard Reina wrote: Thank you very much for your response. I was wondering if it would be ok for me to ask you a couple of additional questions. 1. Do you think this woul work? http://www.phonegeeks.com/patpanwit25p.html yes, but be certain you are ordering the 2-conductor jack model. 2. If I use the 25 pair (Amphenol) for hooking up analog phones, what ports on the ADIT 600 do I use for hooking up my eight analog incoming phone lines? You need to replace one of your FXS cards with an FXO card for your incoming lines from the telephone company. They are more expensive. :( Thanks again for your help. If my questions are unclear (not suprising since I am completely clueless) feel free to call me toll free at 888-448-7874. Richard Reina. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Top posting
If someone provides me with an answer to a question or provides information to enhance my asterisk system, I don't care if they top-post or bottom-post. Marv ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Disable flash hook hold?
Currently, if I briefly press the flash hook on my phone, the caller is placed on hold. I would like for the channel to hangup if I do this instead, never placing a caller on hold (I'll be using call-parking instead). I disabled threewaycalling that is supposed to control this, but it doesn't make any difference: threewaycalling: If enabled, you can place a call on hold by pressing a hook flash, whereupon you get a dialrecall tone and can make another call. Default: no. Here are the relevent sip.conf statements. What am I doing wrong? I'm assuming you mean zapata.conf not sip.conf. These settings do not affect sip clients. The sip client manages its own flash settings, not asterisk. Also, when you modify the zapata.conf file you must shutdown and restart asterisk for the changes to be recognized. I have the following settings in my zapata.conf and it works fine for me. [channels] callwaiting=no callwaitingcallerid=no threewaycalling=no transfer=no echocancel=yes echocancelwhenbridged=yes echotraining=yes rxflash=50 The rxflash setting is to shorten the length of time the on-hook button needs to be depressed before a hangup is registered. [channels] callwaiting = no cancallforward = no callreturn = yes immediate = no callwaitingcallerid = no threewaycalling = no transfer= no echocancel = yes echocancelwhenbridged = yes echotraining= 800 adsi= no busydetect = yes busycount = 8 callprogress= no musiconhold = random relaxdtmf = yes usedistinctiveringdetection=no useincomingcalleridonzaptransfer=yes ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] reverse the selection order of zap channels for outgoing calls
exten = _9NXX,1,Dial(Zap/G1/${EXTEN})
Zap/g1 = hunts for the first available channel in group 1
Zap/G1 = hunts for the first available channel in reverse order in group 1
Is it possible, code wise, configuration wise, at all - to reverse the
order in which the available zap channels are used for *outgoing* calls?
Code wise, I looked at the channel structure and it appears as though
there is only a next pointer, not a previous pointer, so to 'easily' to
this in the code would require a change to the code that reads in
zapata.conf?
I know I could simply plug the wires in 'backwards' for POTS lines, but
I was just wondering if this could be done otherwise.
Thanks,
Chris
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Re: [Asterisk-Users] Overhead Paging
I am currently implementing a VoIP PBX, and need to deal with the paging situation. I would prefer to do paging via overhead speakers. My plan is to connect a "Paging Unit" to an FXS port of an IAD, and assign an extension to that port. I would then simply be able to call that extension, and have my call patched through to the overhead speakers. Has anyone implemented this type of setup, if so, what type of paging unit did you deploy, did you require an external amplifier or power supply, and how many speakers were you able to connect to the unit? As it stands, I will need between 4 and 8 speakers, and some of the speakers will be 400 feet from the main telco closet. Any thoughts, comments, and suggestions that you can shed on this topic would be much appreciated. If you have other methods of implementing overhead paging, I would also be interested. If you search the archives I think you'll find this discussed several times. One (of many) ways to accomplish it is simply based on using a Cisco 7940/7960 phone configured with paging, and pipe the audio to an amplifier input. If you're planning on deploying the Cisco phones already, then using that approach basically has built-in sparing covered. You can do these with the Grandstream Budgetone. They only cost around $75.00 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] uniden phones
Gary Carr wrote: James H. Thompson wrote:Who are the US wholesalers selling the uniden phones? www.thevoipconnection.com But unfortunately they are on backorder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Paging with Sipura 3000 FXO
I have a Sipura 3000 that I 'd like to use for paging thru the FXO port, but I keep getting Service Not Available. I guess that is because there is no loop current or tip/ring voltage on the line. Anyone know a work around for this? From manual ** 4.2.5. Determining the Availability of the PSTN line SPA determines that the PSTN line is not available if the one of the following conditions is true: - PSTN line is not connected (loop current is 0 or Tip/Ring RMS voltage is below 1V) - PSTN line is being used by another extension. Tip/Ring RMS voltage lower than the threshold set in Line-In-Use Voltage If the PSTN line is not available, the PSTN gateway function will be rejected; any VoIP caller requesting PSTN gateway functions will be turned down with a Service Not Available response. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] queue.conf default behavior?
If I don't have any settings defined for these 2 parameters, what are the default settings? ; How long do we let the phone ring before we consider this a timeout... ; ;timeout = 15 ; ; How long do we wait before trying all the members again? ; ;retry = 5 ; ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] unclean hangups can I turn off hook flash?
I'm having problems with unclean hangups (being read as a flash instead of a hangup?). Can I turn off hook flash recognition in asterisk, but still have the flash button on the analog phone operational? Could I use these settings in zapata.conf to fix my problem? *prewink*: Sets the pre-wink timing. *preflash*: Sets the pre-flash timing. *wink*: Sets the wink timing. *rxwink*: Sets the receive wink timing. *rxflash*: Sets the receive flash timing. *flash*: Sets the flash timing. *start*: Sets the start timing. *debounce*: Sets the debounce timing. The debounce settings in the Asterisk configuration affects how Asterisk handles hookswitch transitions on its FXO/FXS interfaces. -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Variable for extension that transferred the call?
Is there a variable available to use in my dial plan, that identifies
the extension or callerid that
transferred the call?
I use valet parking to stack calls for an extension in their own parking
lot.
I'd like the extension they use for valet park to be the same for all
phones. That way I can
program a memory button on the phone without worrying if it's hooked
up to the
correct extension.
My extensions are 3 digit and all begin with 8.
current . . . . . . . . . .
exten = _5XX,1,ValetParkCall(auto|8${EXTEN:1}|120|8${EXTEN:1}|1|default)
exten = *66,1,ValetUnParkCall(fifo|${CALLERIDNUM})
what I want . . . . . . . . . .
exten =
*55,1,ValetParkCall(auto|${extension_that_transferred}|120|${extension_that_transferred}|1|default)
exten = *66,1,ValetUnParkCall(fifo|${CALLERIDNUM})
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[Asterisk-Users] Manager originate command from SIP to Zap not working
I'm running Asterisk CVS-HEAD-06/07/04. When I try to originate a call from a SIP channel to a ZAP channel using manager everything works up to the point when I pickup the ringing ZAP phone. Originate ZAP to SIP works fine. This is the error from my asterisk debug. Jun 25 09:41:26 WARNING[770069]: chan_sip.c:1718 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) I don't have any problem calling directly(using only the phones) from SIP to ZAP. My SIP phone is a Grandstream and I'm using ULAW and ALAW only. This is from my sip.conf file [roger] type=friend disallow=all allow=ulaw allow=alaw host=dynamic username=roger [EMAIL PROTECTED] context=home callerid=roger 805 This is the manager command sent: --- Action: Originate Channel: SIP/ROGER Callerid: [EMAIL PROTECTED] Exten: 820 Context: home Priority: 1 These are the manager events received: -- Event: Newchannel Channel: SIP/ROGER-9f51 State: Down Callerid: Uniqueid: 1088170907.40 Event: Newcallerid Channel: SIP/ROGER-9f51 Callerid: [EMAIL PROTECTED] Uniqueid: 1088170907.40 Event: Newchannel Channel: SIP/ROGER-9f51 State: Ringing Callerid: [EMAIL PROTECTED] Uniqueid: 1088170907.40 Event: Newstate Channel: SIP/ROGER-9f51 State: Up Callerid: [EMAIL PROTECTED] Uniqueid: 1088170907.40 Response: Success Message: Originate successfully queued Event: Newexten Channel: SIP/ROGER-9f51 Context: home Extension: 824 Priority: 1 Uniqueid: 1088170907.40 Event: Newchannel Channel: Zap/9-1 State: Rsrvd Callerid: marvin 824 Uniqueid: 1088170908.41 Event: Newstate Channel: Zap/9-1 State: Ringing Callerid: [EMAIL PROTECTED] Uniqueid: 1088170908.41 Event: Newstate Channel: Zap/9-1 State: Up Callerid: [EMAIL PROTECTED] Uniqueid: 1088170908.41 Event: Link Channel1: SIP/ROGER-9f51 Channel2: Zap/9-1 Uniqueid1: 1088170907.40 Uniqueid2: 1088170908.41 Event: Unlink Channel1: SIP/ROGER-9f51 Channel2: Zap/9-1 Uniqueid1: 1088170907.40 Uniqueid2: 1088170908.41 Event: Hangup Channel: Zap/9-1 Uniqueid: 1088170908.41 Cause: 0 Event: Hangup Channel: SIP/ROGER-9f51 Uniqueid: 1088170907.40 Cause: 1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Manager command MailboxCount and MailboxStatus not working
I'm running Asterisk CVS-HEAD-06/07/04, and I'm having problems with the MailboxCount and MailboxStatus commands in the manager interface. I have some new messages in mailbox 824, but the manager command reports this: Response: Success Message: Mailbox Message Count Mailbox: 824 NewMessages: 0 OldMessages: 0 The MailboxStatus command also reports 0 for message status. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] sidetone noticeably loud on analog handsets on T100P
Has anyone experienced this? I did some googling on it through the archives of course, but don't see much discussion of sidetone issues with analog handsets. I'm wondering if there's some way I could be adjusting the sidetone in Asterisk or should I be looking at my FXS channel bank? I have the same problem with an Adit 600 channel bank that I've been trying to resolve without any success. The sidetone is loud and very annoying. The only way to avoid it is to keep the phone mic further away from your mouth. Unfortunately, I think it is a channel bank issue. I've disconnected my channel bank from the Asterisk server and the sidetone is still very loud. I've changed the gain on the channel bank but that hasn't made any difference. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] sidetone noticeably loud on analog handsets on T100P
I have the same problem with an Adit 600 channel bank that I've been trying to resolve without any success. The sidetone is loud and very annoying. The only way to avoid it is to keep the phone mic further away from your mouth. Interesting; I'm using a T100P connected to an Adit600 and I have no sidetone issues. I've used scenarios involving FXS ports connecting to trunk lines on a Norstart MICS KSU as well as regular old phones connected to the FXS ports. The Adit600 does have its own tx/rxgain settings; have you played with those? I have played with the tx/rxgain settings on the Adit600. They don't seem to have any effect on the sidetone issue. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] sidetone noticeably loud on analog handsets on T100P
I just received some new analog phones that sound great, the sidetone isn't noticeable. They are Easy Touch 76510's by NorthWestern Bell Phones. The problem still exists for my other analog phones. :( I figure it probably must have to do with my Rhino Equipment FXS channel bank guys... I'm in the process of researching if there's anyway to adjust it on the equipment. I also agree with what some of you guys are reporting re. the rx/txgain not making any difference. I think that's for the same reason that changing the same variables within Asterisk really didn't solve the problem. You get volume differences, but the *sidetone* itself seems unaffected by those changes. I have played with the tx/rxgain settings on the Adit600. They don't seem to have any effect on the sidetone issue. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Active extensions via Web
Isamar Maia wrote: Hi, There is already any CGI script to show the active online extensions through the web? Thanks, Isamar Maia http://www.asternic.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] powered analog phone and adit 600 feedback problem
I'm using an adit 600 channel bank with some powered analog phones that have caller id features, and some unpowered analog phones. The unpowered phones sound great, but the powered phones have a problem with loud feedback if you speak to close to the handset mic. I've determined that its not the settings in asterisk or the digium cards by disconnecting the T1 line from adit 600 to asterisk box. I turned the gain down as far as I could. Any ideas? Some adit settings: pbz-cbank1 show 2:5 SLOT 2: Settings for FXS: channel 5: Type:VOICE Signaling: LS RxGain: -9dB TxGain: -9dB LineLength: SHORT pbz-cbank1 status 2:5 FXSRx AB Tx AB Signal=T1 Sig T1 TP ---- - -- - -- 2:5 01 01 LS = LS Traffic N ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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:Remote Call Forwarding
I use this setup for users to set call-forwarding remotely to another
extension.
exten = *76,1,Read(extfrom,fwd-ext-from)
exten = *76,2,Read(extto,fwd-ext-to)
exten = *76,3,DBput(CF/${extfrom}=${extto})
exten = *76,4,Hangup
Marv Horst
Kekin Dand wrote:
Philipp,
I already have that call-forwarding feature set into asterisk.
What I am looking is how to set that feature remotely by calling into your
voicemail or any given no. so that person can set call-forwarding remotely.
Few of our sales people want this kind of feature, because if they are stuck
in traffic and expecting important call, so that, they can call from there
mobile into asterisk and set call-forward to there mobile.
With the current call-forwarding feature, person has to be there physically
to set this feature from there extension.
If somebody has any example, it would be great help.
Regards,
KD
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Re: [Asterisk-Users] TE405P + Adit 600 and FXO module - should this work?
I have an Adit 600 working with a TDM400p card. - check this link for info on Adit-600 configuration http://lists.digium.com/pipermail/asterisk-users/2003-June/013072.html -Make sure the pinouts for your T1 cable are correct: DigiumT1 - RJ-48(F) X-over Digium T1 Card Channel Bank RJ-48(F) Pin Pair#T/R Color Pin T/R 12 T white/orange 4 R1 22 R orange/white 5T 41 T blue/white 1 R 51 R white/blue 2 T1 3, 6, 7, 8 Not assigned Darren Nickerson wrote: Folks, I'm experimenting with bringing multiple (8) analog lines from our local telco into a Carrier Access ADIT 600 channel bank with an FXO module, then having this talk to Asterisk via the T1 TDM controller on the ADIT and a TE405P card. I don't know if this will work well (ie: give me decent echo cancellation, call disconnect supervision, caller ID etc) but I haven't had much luck getting this combination flying at all thus far. Should it work? I've been concentrating on the Adit T1 interface TE405P connection ... and have the following in zapata.conf: span=1,1,0,esf,b8zs fxsks=1-24 loadzone = us defaultzone=us At the end of my zaptel.conf, I have: signalling=fxs_ks group=2 callerid=Joe Schmoe (215) 555-1212 channel = 1-24 To get things flying, I do: modprobe zaptel modprobe wct4xxp which causes the TE405P card to activate, but show a single flashing red alarm on the configured span. The Adit's TDM controller also displays a solid red LED. Here's its status: Adit 600 status a:1 SLOT A: Status for DS1 1: Receive: Loss of Signal Transmit:RAI/Yellow Alarm Loopback:OFF Adit 600 I'm not sure I have the DS1 configured appropriately. It says: Adit 600 show a:1 SLOT A: Settings for DS1 1: Circuit ID: CAC DS1# A:1 Up/Down: UP Framing: ESF Line Coding: B8ZS Line Build Out: DSX-1 EQUALIZATION FOR 0-133 ft. (CSU 0dB) Loop Code Detection: ON Loopback:OFF FDL Type:None Can anyone familiar with the Adit 600 and/or TE405P see any obvious errors here? -Darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] PC based Switchboard application
me too [EMAIL PROTECTED] pat munis wrote: interested ... please send me some info. - Original Message - From: "Kyle Hagan" [EMAIL PROTECTED] Date: Thu, 15 Apr 2004 09:20:01 -0700 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] PC based Switchboard application Im interested can you send information? Kyle [EMAIL PROTECTED] - Original Message - From: "Pertti Pikkarainen" [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, April 10, 2004 2:26 AM Subject: Re: [Asterisk-Users] PC based Switchboard application We have switchboard application ( PC+browser+Java ) with quite a rich feature set. It talks to * via manager port. Works as a call center too. However, it is not open source. If you are interested in, please contact me directly. Best regards Pertti Keith D'Atrio wrote: Hello All I am looking for a PC based switchboard application. Cisco CallManager has a web attendant console that allows you to use the PC to transfer calls and the like and I was wondering if there was a similar program compatible with *. Thank you in advance Keith D'Atrio ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] ANNOUNCE: Flash Operator Panel
Hi, I am using Version .03, everything works fine except I can't transfer by drag and drop. It seems to be a problem with flash since the perl program is not outputting any debug info when I attempt drag and drop. -- Marvin Horst Paul B Zimmerman, Inc Nicolas Gudino wrote: Version .03 is on the website, case insenstive and more channel types supported. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Grandstream - firefly call translator problem
I've also included output for call from Zap channel to firefly that works fine here is output from iax2 debug == -- Executing Macro(SIP/mhorst-1f03, ext|IAX2/[EMAIL PROTECTED]) in new stack -- Executing DBget(SIP/mhorst-1f03, caller=CF/8030) in new stack -- DBget: varname=caller, family=CF, key=8030 -- DBget: Value not found in database. -- Executing DBget(SIP/mhorst-1f03, dnd=DND/8030) in new stack -- DBget: varname=dnd, family=DND, key=8030 -- DBget: Value not found in database. -- Executing Dial(SIP/mhorst-1f03, IAX2/[EMAIL PROTECTED]|15|Tt) in new stack Feb 26 14:18:16 WARNING[-1242121296]: chan_iax2.c:5112 iax2_request: Unable to create translator path for UNKN to G723 on IAX2[marvcomp]/1 -- Hungup 'IAX2[marvcomp]/1' Feb 26 14:18:16 NOTICE[-1242121296]: app_dial.c:527 dial_exec: Unable to create channel of type 'IAX2' == Everyone is busy at this time -- Executing VoiceMail(SIP/mhorst-1f03, b8030) in new stack Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: HANGUP Timestamp: 1ms SCall: 1 DCall: 0 [192.168.10.61:4569] Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 1ms SCall: 25111 DCall: 1 [192.168.10.61:4569] -- Playing 'vm-theperson' (language 'en') -- Playing 'digits/8' (language 'en') -- Playing 'digits/0' (language 'en') Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: HANGUP Timestamp: 1ms SCall: 1 DCall: 0 [192.168.10.61:4569] Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 1ms SCall: 25111 DCall: 1 [192.168.10.61:4569] == Spawn extension (macro-ext, s, 304) exited non-zero on 'SIP/mhorst-1f03' in macro 'ext' == Spawn extension (home, s, 1) exited non-zero on 'SIP/mhorst-1f03' Output from Zap channel to firefly = Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 1ms SCall: 25511 DCall: 3 [192.168.10.61:4569] Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACCEPT Timestamp: 1ms SCall: 25511 DCall: 3 [192.168.10.61:4569] -- Call accepted by 192.168.10.61 (format ULAW) -- Format for call is ULAW Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 1ms SCall: 3 DCall: 25511 [192.168.10.61:4569] Rx-Frame Retry[No] -- OSeqno: 001 ISeqno: 001 Type: CONTROL Subclass: RINGING Timestamp: 2ms SCall: 25511 DCall: 3 [192.168.10.61:4569] Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 2ms SCall: 3 DCall: 25511 [192.168.10.61:4569] -- IAX2[marvcomp]/3 is ringing Adam Hart wrote: strange, do a iax2 debug to see what codecs firefly is asking for. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream - firefly call translator problem
When I try to initiate a call from my Grandstream phone (ext 8010) to my firefly softphone (ext 8030) I get the following error messages, but I have no problem calling from firefly ext to grandstream ext. Calling from a Zap phone to firefly works fine also. Feb 26 07:25:47 WARNING[-1242334288]: chan_iax2.c:5112 iax2_request: Unable to create translator path for UNKN to G723 on IAX2[marvcomp]/3 -- Hungup 'IAX2[marvcomp]/3' Feb 26 07:25:47 NOTICE[-1242334288]: app_dial.c:527 dial_exec: Unable to create channel of type 'IAX2' == Everyone is busy at this time I have ULAW, ALAW, and GSM enabled on the firefly softphone. here are relevant configs. * iax.conf [marvcomp] disallow=all allow=ulaw allow=alaw type=friend host=dynamic username=marvcomp secret=mayhem context=home [EMAIL PROTECTED] callerid=marv 8030 ** sip.conf *** [mhorst] type=friend disallow=all allow=ulaw allow=alaw host=dynamic username=mhorst [EMAIL PROTECTED] context=home callerid=mhorst 8010 ** extensions.conf ** exten = 8010,1,Macro(ext,SIP/mhorst) exten = 8020,1,Macro(ext,Zap/2) exten = 8030,1,Macro(ext,IAX2/[EMAIL PROTECTED]) exten = 8040,1,Macro(ext,IAX2/[EMAIL PROTECTED]) exten = 8050,1,Macro(ext,SIP/roger-gs) ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Grandstream - firefly call translator problem
I also ran sip debug. The output is listed below.
=
Sip read:
INVITE sip:[EMAIL PROTECTED]:5060
SIP/2.0
Via: SIP/2.0/UDP 192.168.10.2;branch=z9hG4bKd1e65c5f319bffc5
From: "Marvin Horst"
sip:[EMAIL PROTECTED]:5060;tag=099422b3d98a1e89
To: sip:[EMAIL PROTECTED]:5060
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 57341 INVITE
User-Agent: Grandstream SIP UA 1.0.4.26
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Content-Length: 272
v=0
o=mhorst 8000 8000 IN IP4 192.168.10.2
s=SIP Call
c=IN IP4 192.168.10.2
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 2 15
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=ptime:20
12 headers, 13 lines
Using latest request as basis request
Sending to 192.168.10.2 : 5060 (non-NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format ULAW
Found audio format UNKN
Found audio format GSM
Found audio format UNKN
Found description format PCMU
Found description format PCMA
Found description format G723
Found description format G729
Found description format G726-32
Found description format G728
Capabilities: us - 2147483647, them - 285/0, combined - 285
Non-codec capabilities: us - 1, them - 0, combined - 0
Looking for 8030 in home
list_route: hop: sip:[EMAIL PROTECTED]
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.2;branch=z9hG4bKd1e65c5f319bffc5
From: "Marvin Horst"
sip:[EMAIL PROTECTED]:5060;tag=099422b3d98a1e89
To: sip:[EMAIL PROTECTED]:5060;tag=as6c82465a
Call-ID: [EMAIL PROTECTED]
CSeq: 57341 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
to 192.168.10.2:5060
-- Executing Macro("SIP/mhorst-5fd0", "ext|IAX2/[EMAIL PROTECTED]")
in new stack
-- Executing DBget("SIP/mhorst-5fd0", "caller=CF/8030") in new stack
-- DBget: varname=caller, family=CF, key=8030
-- DBget: Value not found in database.
-- Executing DBget("SIP/mhorst-5fd0", "dnd=DND/8030") in new stack
-- DBget: varname=dnd, family=DND, key=8030
-- DBget: Value not found in database.
-- Executing Dial("SIP/mhorst-5fd0",
"IAX2/[EMAIL PROTECTED]|15|Tt") in new stack
Feb 26 14:58:51 WARNING[-1242121296]: chan_iax2.c:5112 iax2_request:
Unable to create translator path for UNKN to G723 on IAX2[marvcomp]/1
-- Hungup 'IAX2[marvcomp]/1'
Feb 26 14:58:51 NOTICE[-1242121296]: app_dial.c:527 dial_exec: Unable
to create channel of type 'IAX2'
== Everyone is busy at this time
-- Executing VoiceMail("SIP/mhorst-5fd0", "b8030") in new stack
We're at 192.168.10.205 port 10514
Answering with preferred capability 2147483647
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.2;branch=z9hG4bKd1e65c5f319bffc5
From: "Marvin Horst"
sip:[EMAIL PROTECTED]:5060;tag=099422b3d98a1e89
To: sip:[EMAIL PROTECTED]:5060;tag=as6c82465a
Call-ID: [EMAIL PROTECTED]
CSeq: 57341 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Type: application/sdp
Content-Length: 111
v=0
o=root 5520 5520 IN IP4 192.168.10.205
s=session
c=IN IP4 192.168.10.205
t=0 0
m=audio 10514 RTP/AVP
to 192.168.10.2:5060
-- Playing 'vm-theperson' (language 'en')
Sip read:
ACK sip:[EMAIL PROTECTED]
SIP/2.0
Via: SIP/2.0/UDP 192.168.10.2;branch=z9hG4bKd1e65c5f319bffc5
From: "Marvin Horst"
sip:[EMAIL PROTECTED]:5060;tag=099422b3d98a1e89
To: sip:[EMAIL PROTECTED]:5060;tag=as6c82465a
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 57341 ACK
User-Agent: Grandstream SIP UA 1.0.4.26
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE
Content-Length: 0
11 headers, 0 lines
Adam Hart wrote:
strange, do a iax2 debug to see what codecs firefly is asking for.
- Original Message -
From: "Paul Zimm" [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, February 26, 2004 11:42 PM
Subject: [Asterisk-Users] Grandstream - firefly call translator problem
When I try to initiate a call from my Grandstream phone (ext 8010) to my
firefly softphone (ext 8030) I get the following error messages, but I
have no problem calling from firefly ext to grandstream ext. Calling
from a Zap phone to firefly works fine also.
Feb 26 07:25:47 WARNING[-1242334288]: chan_iax2.c:5112 iax2_request:
Unable to create translator path for UNKN to G723 on IAX2[marvcomp]/3
-- Hungup 'IAX2[marvcomp]/3'
Feb 26 07:25:47 NOTICE[-1242334288]: app_dial.c:527 dial_exec: Unable to
create channel of type 'IAX2'
== Everyone is busy at this time
I have ULAW, ALAW, and GSM enabled on the firefly softphone.
here are relevant configs.
* iax.conf *
