hello
i want to rout my calls to h.323. i have registered my
asterisk with GnuGatekeeper. but it is not routing my
call to h.323 channel. he is saying Internal channel
initialization failed. Bad binary?
can any one check my settings what is problem here
thanks in advance
kamran
hello
i try to call from sip phone on asteris to open phone
on GnuGK.
can any one tell me why it is saying
chan_oh323.c:2501 ast_oh323_new: Internal channel
initialization failed. Bad binary?
Mar 16 13:28:46 WARNING[5963]: chan_oh323.c:2727
oh323_request: Failed to create new H.323 private
hello
i was searching for solution to problem (sip-h.323).
any one from this list asterisk mailing have any idea
how to fix it.
i am getting error when i try to call from sip to
h.323 user
i am successfully registering my asterisk box with
gnugk. but when i try to call to h.323 openphone on
Executing Dial(OH323/R11429, OH323/40923335224005)
but i want him to dial
Executing Dial(OH323/R11429, OH323/923335224005)
Kamran Ahmad
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hi
Any one give me any hint how to start radius with
asterisk.
Is there any addon available for asterisk+radius.
Please provide me helpfull link which could help me.
i am new to radius.
regrads
kamran
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,Dial(OH323/${EXTEN:2})
Thanks
Kamran Ahmad
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Asterisk-Users
i have written app for billing with asterisk. what is
the problem in using radius.
kamran
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hello pongco
if you are talking about disconnecting a call session
at his credit time. then you have to look at
ast_channel-whentohangup
kamran
On Fri, 2005-03-18 at 14:10, Paul P. Pongco wrote:
Hello,
Im actually deciding if I will use asterisk+radius
for AAA purposes
or
use logging
hi list
i know i am asking question out of the scope of this
list. actualy i cant find any place to ask question
like this. may be someone using ODBC with asterik.
actualling i want to make ODBC connection for asterisk
on my new fedora core 2. i have tried every thing.
tried rpms. compiled
hello
i dont know why unixodbc is not working. i am trying
to make odbc connection. yesterday my odbc connection
was working with mysql on my one mechine but now it is
not working. is there any problem in code.
/etc/odbc.ini
[test]
Description = My test dsn
Trace = Off
TraceFile = stderr
Driver
hello
can any one tell me what is the problem in my odbc
connection.
here is my sql.log connection with mysql is working
and with freetds is giving me error jawad is one
windows server having MS Sql server
#isql kdsn
src/tds/login.c: tds_connect: jawad:1433: Connection
refused
[ISQL]ERROR: Could
gcc -shared -Xlinker -x -o cdr_odbc.so cdr_odbc.o
-lodbc -L/usr/lib/pgsql
gcc -pipe -Wall -Wstrict-prototypes
-Wmissing-prototypes -Wmissing-declarations -g
-Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6
-march=i686 -DASTERISK_VERSION=\1.0.7\
-DINSTALL_PREFIX=\\
hello
any one tell me how to make asterisk stateless only
for handshaking.
UAC--(sip)ASTERISK--(sip)UAS
UAC---(RTP)--UAS
UAC---(SIP_BYE)--UAS
what are the configuration needed
Thanks in advance
Kamran
hello
i have a prblem in routing call from ser to asterisk.
i have the following senrio.
UA is registered at ser
when UA calls another UA ser try to look for the user
not found then forword the call to other side
asterisk.
problem i am facing that ser is not forwording request
to asterisk
?menu=features
Codecs
* ADPCM
* G.711 (A-Law #956;-Law)
* G.723.1 (pass through)
* G.726
* G.729 (through purchase of commercial license
through Digium)
* GSM
* iLBC
* Linear
* LPC-10
* Speex
what is the meaning of G.723.1 (pass through)
Thanks
Kamran Ahmad
hello
how to pass G723.1 to other side is there any
softphone using g723.1. i want to use G723.1 in my
voice communication.
regrads
Kamran
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hello
i am using phone with g723 and gw is complient for
g723.then why after 200 oK i am getting this.
can any one tell me why i am getting.
Apr 8 16:14:05 NOTICE[5750]: channel.c:1833
set_format: Unable to find a path from g723 to slin
Apr 8 16:14:05 WARNING[5750]: channel.c:2263
hello
Any one know how to resolve NAT issue.
PublicIp(UA)-Asterisk on
publicIP--privateIP(UA) its not working
PrivateIP(UA)-Asterisk on
publicIP--publicIP(UA) its working
how to reslove this issue
Thanks
Kamran
__
Do you
is working
with NAT, which I
gather is your problem.
-Andy
FWD:428725
On Apr 8, 2005 7:06 AM, Kamran Ahmad
[EMAIL PROTECTED] wrote:
hello
Any one know how to resolve NAT issue.
PublicIp(UA)-Asterisk on
publicIP--privateIP(UA) its not working
PrivateIP(UA)-Asterisk on
publicIP
Can any one help me in understanding REGISTER message
when i send REGISTER message to asterisk it is
replying 407
with header
Proxy-Authenticate: Digest
realm=asterisk,nonce=1011592446
i want password for my user so i entered secret in
sip.conf against userid
can any one tell me how to handle
i have gone through RFC 2617 and able to run the test
program
but problem i that when i try these values
this is not matching with my asterisk proxy response
can any one tell me what should be pszMethod here
i dont have Qop here
char * pszNonce = 539b02d7;
char * pszCNonce = ;
i found that here Method is REGISTER
char * pszNonce= dcd98b7102dd2f0e8b11d0f600bfb0c093;
char * pszCNonce = 0a4f113b;
char * pszUser = Mufasa;
char * pszRealm = [EMAIL PROTECTED];
char * pszPass = Circle Of Life;
char * pszAlg = md5;
char szNonceCount[9] =
it is sometime generating wrong response
can any one help me in this
#include stdio.h
#include digcalc.h
void main(int argc, char ** argv) {
char * pszNonce =
dcd98b7102dd2f0e8b11d0f600bfb0c093;
char * pszCNonce = ;
char * pszUser = 6000;
char * pszRealm = asterisk;
hello
i want to use mysql database server with my asterisk
PBX. i have installed mysql on linux mechine. i have
already installed asterisk on same mechine. now i want
to know what is the way to connect asterisk to mysql.
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i have problem in compiling asterisk-addons 1.0.1
-
[EMAIL PROTECTED] asterisk-addons-1.0.1]# make
cc -fPIC -I../asterisk -D_GNU_SOURCE
-I/usr/local/mysql/include -c -o
cdr_addon_mysql.o cdr_addon_mysql.c
../asterisk: Not a directory
now it is giving another error
-
[EMAIL PROTECTED] asterisk-addons-1.0.1]# make
cc -fPIC -I../asterisk -D_GNU_SOURCE
-I/usr/local/mysql/include -c -o
app_addon_sql_mysql.o app_addon_sql_mysql.c
app_addon_sql_mysql.c:31:25:
asterisk -cv
Jan 31 18:03:20 WARNING[13145]: cdr_addon_mysql.c:264
my_load_module: Unable to load config for mysql CDR's:
cdr_mysql.conf
[app_addon_sql_mysql.so] = (Simple Mysql Interface)
[pbx_dundi.so]Jan 31 18:03:20 WARNING[13145]:
i want to know how we can add extension to mysql
database. i am using asterisk_addons and i have
checked that mysql database is connected with asterisk
here is trace
[skipping app_intercom.so]
[codec_ilbc.so] = (iLBC/PCM16 (signed linear)
there is a problem in compiling asterisk-addons
any one have fixed this problem. i want
res_config_mysql.so any one help me
-
[EMAIL PROTECTED] asterisk-addons]# make
cc -fPIC -I../asterisk -D_GNU_SOURCE
-I/usr/include/mysql -c -o
prbolem still there
first of all i have these two(asterisk,
asterisk-addons) working on my system i got these
packages from asterisk.org
then i recompiled asterisk-addons because i want
res_config_mysql.so module for real time database
i got this addon by following command
i am getting 403 Forbidden message from asterisk when
it try to register my user agent. i am basically
useing mysql through ODBC. i hvae checked ODBC
connecteion with
'ODBC Show' command.
--
*CLI odbc show
Name: mysql1
DSN: asteriskdsn
Connected:
i am using asterisk-1.0.5 latest available stable
version downloaded from www.asterisk.org
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-
Matthew what do you meant by this
*bamn* damn..hate it when i'm right. you are
attempting to use
asterisk-1.0.5 with asterisk-addons-CVS.
insert *VERY LOUD buzzer*
-
i have compiled
can any one tell me what is the problem with my
asterisk configuration it is not connnecting with
mysql. it is giving the follwing error message.
--
Asterisk Dynamic Loader Starting:
[cdr_addon_mysql.so] = (MySQL CDR Backend)
[chan_modem.so] =
i am working on asterisk. i am using fedora core 2 on
my asterisk mechine. when i was working on stable
version my voicemailmenu was working well. i can
lissten to menu and send dtmf to control menu now i
have compiled CVS version of asterisk. now when i
configure my voicemail for any extension
can anyone tell me how to add extension to extension
table
i think this is the main prblem. any one to guide me.
++-+-+--+--+-+
| id | context | exten | priority | app
| appdata |
at first it was not answering (there was complete
silence after 200 Ok and ACK). i dont know what was
the reason. but now it is answering me(asking for
mailbox then password). but the problem that is is not
authenticating me to check mailbox i have defined
mailbox and 1234 password (it is
hello
any one using cvs version of asterisk(realtime
addons). i have defined two users 2000 and 3000 in
sip.conf. after that when i try to call 2000 from 3000
or try to call 3000 from 2000 it is giving me 404 Not
Found error.
Found user '2000'
Looking for 3000 in default
Reliably Transmitting
hello
any one using cvs version of asterisk with realtime
mysql addons. i am having a problem with it. i have
defined two users 3000 and 2000. when i try to call
3000 from 2000 it is giving me '404 Not Found' and
saying Found user '2000' and Looking for '3000'
but when i try to call 2000 from
[3000]
type=friend
dtmfmode=INFO
insecure=yes
canreinvite=no
auth=plaintext
host=dynamic
allow=ulaw
[2000]
type=friend
dtmfmode=INFO
insecure=yes
canreinvite=no
auth=plaintext
host=dynamic
allow=ulaw
[default]
;
; By default we include the demo. In a production
system, you
; probably don't want to have the demo there.
;
include = demo
exten = 3000,1,Dial(SIP/${EXTEN})
exten = 2000,1,Dial(SIP/${EXTEN})
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thanks
it is register and receiving the invite. some time my
user agent (i am sjphone) is sending invalid address
in his contact and SDP. then i try to call form
another ua it i transmitting invite to invalid
address.
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Hello
Any one using asterisk-prepaid with mysql. i want
asteirsk-prepaid for fedora core 2. i have installed
mysql-devel. but after that i am unable to compile the
asterisk-prepaid it is giving me error for
libmysqlclient. i already have this library in my
/usr/lib/mysql. i am using asterisk-CVS.
hello
i was using CVS Head version for realtime mysql it was
working well. now i want to use odbc connection for
realtime database it is not working i am using it with
stable release. i have checked everything my conf is
ok odbc connection is working. any one working with it
res_conf_odbc.conf
hello
how to register with irc. i want to connect to
#asterisk through x-chat
thanks
kamran
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me what is the
reason. Is this a bug or what
Kamran Ahmad
--
*CLI sip debug
SIP Debugging Enabled
*CLI
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP
hi
If i remove _. from my dialplan(extensions.conf).
application is invoked only once. otherwise
application is invoked again and again. any one know
what is the problem and how to make (global) dialplan
for all user agents.
thanks
Kamran
hello all
i am having a problem in compiling openh323.
[EMAIL PROTECTED] openh323]# ./configure
checking for g++... g++
checking for C++ compiler default output... a.out
checking whether the C++ compiler works... yes
checking whether we are cross compiling... no
checking for suffix of
hello
i have tried
http://www.oinko.net/astrecipes/index.php?action=artikelcat=270174id=10artlang=en
but failed same error while compiling openh323
---
g++: Internal error: Terminated (program cc1plus)
Please submit a full bug report.
See
hello
i am using asterisk-oh323-0.7.1. i want to convert sip
call to h323 (h323 sjphone or h323 proxy). what could
be the best way for this. i am successfull in
converting h323-sip by using asterisk as gateway.
help required on sip-h323.
kamran
i am using gnugatekeeper. i have three things
gatekeeper ip, account, accountpassword how to set
account and password in oh323.conf
gatekeeper=gnu gatekeeper ip
gatekeeperPassword=accountpassword
accountCode=account
is this ok any example how to use this i want to rout
my sip call to this
HELLO
i am using gungk gatekeeper from a provider. he has
given me a account,password,ip now i want to connect
to it with asterisk.
1. i want to call to my sip phones registered on my
local area network working. ok
2. i want to divert PSTN call to gun gatekeeper (from
service provider company).
hello
now i am using my own gnugatekeeper. asterisk is
registering successfully with Gnugatekeeper. but it is
not transfering call to gnugk.
i am running 1234 user of OpenPhone with GNUgatekeeper
when i try to call from sip User agent 3000 to 3211234
asterisk is not forwarding it to GnuGK it
i am sending this message to my asterisk
first message:
INFO sip:172.16.0.32 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.21
From: sip:[EMAIL PROTECTED]
To: sip:172.16.0.32
Call-ID: [EMAIL PROTECTED]
CSeq: 21 INFO
Contact: sip:[EMAIL PROTECTED]
Content-Type: application/dtmf-relay
ContentLength: 26
Hi all
I am using Asterisk 1.2.2 on frdora core 4. i have two
sip UA. if i put canreinvite=yes voice Ok on both
sides. and if i change canreinvite=no there is no
voice (media through asterisk)
one thing more if i try to use playback application
for playing some sound file it is also working
hi
i am using asterisk-1.2.4 + asterisk-addons-1.2.1 on
2.6 kernal. i have added user in sip_buddies and
followed
http://www.voip-info.org/wiki-Asterisk+RealTime+Sip
but my ip phone is not registring properly.
asterisk is just sending SIP/2.0 404 Not found. i
think it must check DB table for
hi
i am using asterisk-1.2.4 + asterisk-addons-1.2.1 on
2.6 kernal. i have added user in sip_buddies and
followed
http://www.voip-info.org/wiki-Asterisk+RealTime+Sip
but my ip phone is not registring properly.
asterisk is just sending SIP/2.0 404 Not found. i
think it must check DB table for
Hi
i am using call transfer feature between three
parties.
dial(sip/${EXTEN}||t)
it is working perfectly but the problem is that cdr is
incorrect.
here is the call senrio
A-B (A calls B, A and B connected)
B-C (B transfer call to C)
A-C (C got ringing, B Hangup, A and C connected)
in cdr
hello
how to drive SIPGetHeaders from chan_sip2 as described
in
http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth
thanks,
Kamran Ahmad
Sell on Yahoo! Auctions no fees. Bid on great items
hello
how to drive SIPGetHeaders from chan_sip2 as described
in
http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth
thanks,
Kamran Ahmad
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hello
http://www.voip-info.org/tiki-index.php?page=Asterisk+SIP+chan_sip2
where can i download chan_sip2.c
thanks
Kamran
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hello
i am trying to develop perl application for asterisk
with radius accounting how can i debug that weather
callback is working when call is stoped.
how can i check this
syslog('info', 'hello Asterisk!');
thanks
Kamran
hello
i am getting this error while trying to run
ast-rad-acc.pl
my $ast_connected = 1;
while( 1 ) {
if( $astman-connect ) {
$ast_connected = 1;
syslog('info', 'Connected to Asterisk!');
$astman-setcallback('DEFAULT', \status_callback);
hello
i am using ast-rad-acc.pl from portaone connected with
asterisk manager.
my (%cdr) = @_;
$cdr{'CALLERID'},
$cdr{'DNID'},
these are empty
why these two variables are not working on hangup
any comments
thanks
Kamran Ahamd
__
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hello perl experts
i am working with ast-rad-acc.pl from
http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth
i dont know why $cdr{'DNID'} and $cdr{'CALLERID'}
under 'sub send_acc {' are empty. i m successfully
connected with asterisk manager and when call i hangup
my perl
hello
how can i install meetme application without Zaptel
interface. and if this is not posible then how to
install zaptel module.
any helpful link
thanks in advance
Kamran
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hello
Can we use SER in front of 10 Asterisk for load
balancing. any idea
Thanks in advance
Kamran
Start your day with Yahoo! - make it your home page
http://www.yahoo.com/r/hs
hello
is there any way to register all user without
declaring them in sip.conf. because i want all users
to auth.
thanks in advance
Kamran
Start your day with Yahoo! - make it your home page
http://www.yahoo.com/r/hs
hello
any one please tell me if there is a way to define a
range of users in sip.conf
suppose i want to create 1000 user from 500 to
5000999 with no password from
thanks
Kamran
Start your day with Yahoo! - make it your
hello
i am using asterisk-1.0.9. i have a NAT problem.
without NAT registration is ok. and if user is bhind
NAT it is registring on asterisk. but SJPhone is
showing not registered. i think asterisk is properly
sending request to UA. any commentsthis
sip.conf setting was working
problem
At firewall/NAT you have to do port forwarding.
If your phone is at port 5060, NAT device will
receive a
connection and has
to know that it is destined for your SIP phone.
So, forward
port 5060 to the
phone.
Rudolf
- Original Message -
From: Kamran Ahmad
SIP phone.
So, forward
port 5060 to the
phone.
Rudolf
- Original Message -
From: Kamran Ahmad [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, August 14, 2005 6:52 AM
Subject: [Asterisk-Users] Why NAT problem
hello
i am using asterisk-1.0.9
hello
i am trying to use res_odbc for sipuser. my connection
is working. i have checked using isql. even cdr_odbc
is working but i hav problem in res_odbc. i have
created user in sip_buddies table but asterisk is no
getting user from this sip_buddies table.
/etc/asterisk/extconfig.conf
hello
why asterisk starts listening on all ports
and he is trying to listen messages from 5060.
/etc/asterisk/sip.conf
bindport=5070
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hello
i m getting follwing messages in asterisk-1.0.9 what
is the reason can u pls tel me how to solve this
Aug 20 13:06:09 WARNING[7706]: rtp.c:829 ast_rtcp_new:
Unable to allocate socket: Too many open files
Aug 20 13:06:09 WARNING[7706]: channel.c:311
ast_channel_alloc: Alert pipe creation
hello
i m getting follwing messages in asterisk-1.0.9 what
is the reason calls are not going out. can u pls tel
me how to solve this
Aug 20 13:06:09 WARNING[7706]: rtp.c:829 ast_rtcp_new:
Unable to allocate socket: Too many open files
Aug 20 13:06:09 WARNING[7706]: channel.c:311
hello
i m using asterisk-1.0.9. i want to connect to db
through odbc. isql is working. but asterisk is not
getting user information from this table. can any one
pls check this
/etc/asterisk/extconfig.conf
[settings]
sipusers = odbc,mysql1,sip_buddies
sippeers = odbc,mysql1,sip_buddies
sip.conf
hello
i m using asterisk-1.0.9. i want to connect to db
through odbc. isql is working. but asterisk is not
getting user information from this table. can any one
pls check this
odbc connection is working properly is there some
thing required
/etc/asterisk/extconfig.conf
[settings]
sipusers =
hello
i m getting follwing messages in asterisk-1.0.9 after
small interval. And i have to restart asterisk because
after these errors asterisk cannot do any call. what
is the reason calls are not going out. can u pls tel
me how to solve this.
http://www.voip-info.org/wiki-file+descriptors
i
Bob Goddard you are right but i said in my previous
mail that i am still getting this problem
some body replied me and i have followed this link but
still same problem and asterisk is stoping.
http://www.voip-info.org/wiki-file+descriptors
On Wednesday 24 Aug 2005 13:40, Kamran Ahmad wrote
hi list
i am using Quintum gw for pstn. sipPSTN call
when i iniating call quintum is replying me
183 Session Progress
asterisk starts calculating CDR
actually it should start from when both side starts
RTP after 200 ok and ACK.
if callee (PSTN) receives call after 10 seconds these
10 sec
hello
Any help.
CDR duration starts from 183 Session Progress. cdr
duration should start from 200 OK when both parties
are inside session.
i am using Quintum gw for PSTN Calls.
here is the call flow between Asterisk and
QuintumGateway.
ASTERISK GW
1
+billing
duration: Total time in system, in seconds (integer),
from dial to
hangup
What are you looking for (from my point of view) is
billsec: Total time call is up, in seconds (integer),
from answer to
hangup
-b
- Original Message -
From: Kamran Ahmad [EMAIL PROTECTED
-info.org/wiki-Asterisk+billing
duration: Total time in system, in seconds
(integer),
from dial to
hangup
What are you looking for (from my point of view) is
billsec: Total time call is up, in seconds
(integer),
from answer to
hangup
-b
- Original Message -
From: Kamran Ahmad [EMAIL
hello
i need help on PSTN Calls via quintum gateway. i have
a simple problem when i am try to send INVITE to PSTN
quintum gw. it is replying me 183 session progress and
call duration is starting at this point. after this he
is sending ringing then 200 OK. Billseconds are
incorrect in this case.
hello
i want to get extension from ivr
its not working
exten =6000,1,ResponseTimeout(5)
exten =6000,2,Background(enterexten)
exten =6000,3,SetVar(myexten=${digitstack})
exten =6000,4,Wait(5)
exten =6000,5,Goto(default,myexten,1)
Kamran
__
hello
i am using a call file. i want to insert delay before
execution of this call file. any idea how to do this
Channel: SIP/2000
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Context: default
Extension: 6000
Priority: 1
i am making a callback system.
when person rings to callback number this call
hello
i want to insert delay into callfile execution.
UA6000(callbackNumber) this will create call file
UA---asterisk(callfile)
how to insert delay into this callfile execution.
thanks
Kamran
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hello
i am trying to make a callback solution.
client will call callback number and call is
terminated.
now callback server will create a call for that
client.
actually i have a problem in this process. that server
is creating call to client (UA) when previous call is
not disconnected yet.
hello
he is still not replying after correct time
this is the sip debug
May 16 21:41:02 WARNING[3902]: chan_sip.c:730
retrans_pkt: Maximum retries exceeded on call
76fa142e2805cc9a5d44ba4564165b1e@ for seqno 102
(Critical Request)
May 16 21:41:02 NOTICE[3902]: pbx_spool.c:234
attempt_thread:
hello
can any one tell me
Channel: SIP/[EMAIL PROTECTED]:5060
MaxRetries: 1
# Retry in 5 min
RetryTime: 60
WaitTime: 30
Context: default
Extension: 6000
Priority: 1
why this is not working
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hello
sip.conf
bindport=5070
i am trying to register at ser 5060. but why i am
getting request at asterisk 5070.
thanks
Kamran
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hello
I am using ser with asterisk
asterisk on 5070 (on back end)
ser on 5060 (on front end)
i am getting all requests at asterisk.
i tried by changing asterisk port
bindport=5090
but still getting all requests from sjphone at
asterisk.
can any one tell what is the reason
regrads
Kamran
hello
like if 6000 is the main exchange number. any one dial
to 6000 will be asked for pressing his desired
extension then he can press his desired extension then
his number is diled
exten=6000,1,Background(enterdesiredexten)
exten=6000,2,Wait(2)
i know there is example in extension.conf
but that is not working in my case
i am unable to get the extension pressed by user after
listening menu
like how to get when 2000 pressed.
because it is not dialing 2000
exten = 6000,1,Background(k-enterexten)
exten = 6000,2,Wait(2)
3000, start ringing it, wait for 20 seconds,
and then hangup,
if it is never answered. If it is answered, it will
hangup after the
user does, just like you would expect.
Good luck.
Ben
On 5/25/05, Kamran Ahmad [EMAIL PROTECTED] wrote:
i know there is example in extension.conf
hello
i have a small problem in installation of asterisk can
any one tell me what is the solution
gcc -shared -Xlinker -x -o app_zapscan.so
app_zapscan.o
gcc -pipe -Wall -Wstrict-prototypes
-Wmissing-prototypes -Wmissing-declarations -g
-Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6
hello
can u tell me what is the problem in my asterisk or
linux why i am getting this error while make.
PTEL_OPTIMIZATIONS
-DASTERISK_VERSION=\CVS-HEAD-05/26/05-20:43:39\
-DASTERISK _VERSION_NUM=99 -DINSTALL_PREFIX=\\
-DASTETCDIR=\/etc/asterisk\ -DASTLIB
DIR=\/usr/lib/asterisk\
Hello
i have two GWs and some uas. i want if ua (bw 3000 to
4010) is calling any number then this call will be
routed to first GW and if ua (bw 4020 to 5000) want to
call any number this call will be routed to second GW.
Gateways=GW1,GW2
UAs=3000 to 5000
if 3000 wants to call any number ip or
hello
i am trying to follow
http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth
can any one tell how to install this
2. Install Asterisk::AGI and Asterisk::Manager
(unfortunately it is not on CPAN yet!)
thanks in advance
Kamran
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