Hi
i am using call transfer feature between three
parties.
dial(sip/${EXTEN}||t)
it is working perfectly but the problem is that cdr is
incorrect.
here is the call senrio
A-B (A calls B, A and B connected)
B-C (B transfer call to C)
A-C (C got ringing, B Hangup, A and C connected)
in cdr
--- Kamran Ahmad [EMAIL PROTECTED] wrote:
Thanks alot for your answer Florian
I have a question in this case when call is
transfered
from loadbalancing-server to server01 or server02
what
will be media Path? media will be routed through
loadbalancing-server or it will not use
Thanks
Hi,
Kamran Ahmad wrote:
I have a question in this case when call is
transfered
from loadbalancing-server to server01 or server02
what
will be media Path? media will be routed through
loadbalancing-server or it will not use
loadbalancing-server anymore
EndPoint1--loadbalancing
--- Kamran Ahmad [EMAIL PROTECTED] wrote:
hi can we do autocreatepeer in iax.conf?
thanks
kAMRAN
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Thanks Russell
No, that option is not available for iax.conf.
--
Russell Bryant
Software Developer
Digium, Inc.
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-server--server01/02--EndPoint2
OR
EndPoint1--server01/02EndPoint2
any idea which one?
thanks
kAMRAN
Hi,
Kamran Ahmad wrote:
any idea how to loadbalance IAX2 trafic to multiple
asteirsk
Use app_random:
exten = _X.,2,Random(50:6)
exten = _X.,3,Dial(IAX2/server01/${EXTEN})
exten = _X.,4
hi can we do autocreatepeer in iax.conf?
thanks
kAMRAN
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hI
any idea how to loadbalance IAX2 trafic to multiple
asteirsk
thanks
kAMRAN
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--Bandwidth
HI
I want to route media directly to one Caller IAX Phone
to Called IAX phone
signaling
IAX Phone1-Asterisk---IAX Phone2
and media
IAX Phone1IAX Phone2
Is it possible ?
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I have followed these two for configuration of sangoma
A101
http://www.ss7box.com/s01_setup.html
http://www.ss7box.com/support_wancfg_1.html
on my side wanrouter star/restart is working fine
when i am tring to ztcfg -vvv i am getting
and when i am tring to load asterisk getting error No
such
I have followed these two for configuration of sangoma
A101
http://www.ss7box.com/s01_setup.html
http://www.ss7box.com/support_wancfg_1.html
on my side wanrouter star/restart is working fine
when i am tring to ztcfg -vvv i am getting
and when i am tring to load asterisk getting error No
such
HI all
I want to connect Asterisk(using realtime) with
Oracle. any one have any idea which one is the best
method for this. ODBC/ or some other interface modules
avaliable for directly connecting with oracle ?
thanks
Kamran Ahmad
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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 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
Hello
My asterisk is stoping. i am using asterisk with ser
on same mechine
here is the asterisk trace
-- Setting call duration limit to 3000 seconds.
Sep 2 15:58:12 WARNING[10334]: rtp.c:852
ast_rtp_new_with_bindaddr: Unable to allocate socket:
Hello
i am trying to use this exmple with SER-0.9.3
but still NATED Clients are not working any other
requirement
http://www.voip-info.org/tiki-index.php?page=SER+example+NAThelper
---
# $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $
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
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
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 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 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
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 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
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
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
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
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
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
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
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 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
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
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 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
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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install Authen::Radius
any other help full link i m new to perl
JD Austin wrote:
It's complaining that you don't have the perl module
installed or it is
not in your path.
Kamran Ahmad wrote:
hello
how to solve these errors
/var/lib/asterisk/agi-bin/agi-rad-auth.pl line 10
use Asterisk::AGI;
vi
Austin wrote:
It's complaining that you don't have the perl module
installed or it is
not in your path.
Kamran Ahmad wrote:
hello
how to solve these errors
/var/lib/asterisk/agi-bin/agi-rad-auth.pl line 10
use Asterisk::AGI;
vi /etc/asterisk/extensions.conf
exten =
_X.,1,agi,agi-rad-auth.pl
hello
how to solve these errors
/var/lib/asterisk/agi-bin/agi-rad-auth.pl line 10
use Asterisk::AGI;
vi /etc/asterisk/extensions.conf
exten =
_X.,1,agi,agi-rad-auth.pl|Routing=SIPAuthorizeBy=SIP
vi /etc/asterisk/modules.conf
load = res_agi.so
---errors
hello
i am trying to work with radiusclient form portaone.
but i have some problems in installation. when i am
trying to use example from
http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth
error sip debug
Can't locate Asterisk/AGI.pm in @INC (@INC contains:
hello
i m trying to use radius with asterisk
http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth
how to fix this patch
8. Make sure that your Asterisk includes all related
bug fixes and patches, namely:
- SIPGetHeaders for chan_sip (derived from chan_sip2 )
i m using
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
hello
i want to use SIPGetHeader application in
asterisk-1.0.9.
Jul 2 00:04:33 WARNING[19575]: pbx.c:1293
pbx_extension_helper: No application 'SIPGetHeader'
for extension (default, 2000, 1)
Any one using this
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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
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 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
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
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)
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
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 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
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 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 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 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.
-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
+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
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
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
?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 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
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
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
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
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 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
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 mailing list
Asterisk-Users
i have written app for billing with asterisk. what is
the problem in using radius.
kamran
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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
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 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
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 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
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
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
hello
how to register with irc. i want to connect to
#asterisk through x-chat
thanks
kamran
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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
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.
[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 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
1 - 100 of 120 matches
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