in stun, does
that mean i can start configuring my clinets (xten and sipura) to use stun
server?
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asterisk
server. There's nothing else that needs to
be configured on the STUN server side. It's pretty much either running
or it's not.
Just start plugging in the server to your clients and give it a whirl.
It should work.
N.
Rizwan Hisham wrote:
Hi all,
This is the first time i am using stun
machine as an Asterisk server
and see nothing in terms of load increase. STUN's footprint is rather
negligible.
N.
Rizwan Hisham wrote:
Ok thanx. One more thing to ask is: does asterisk has a stun server
implemented in it or not. i mean does asterisk contain a stun server
and provides
into it.
However, we run a STUN server on the same machine as an Asterisk server
and see nothing in terms of load increase. STUN's footprint is rather
negligible.
N.
Rizwan Hisham wrote:
Ok thanx. One more thing to ask is: does asterisk has a stun server
implemented in it or not. i mean
applications running
on the same computer where softphone is also running.
On 8/2/07, Gordon Henderson [EMAIL PROTECTED] wrote:
On Thu, 2 Aug 2007, Rizwan Hisham wrote:
hi again.well i have been trying to know what is the relationship
between asterisk and stun. what i mean is, i understand
visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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=g729
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. Can anybody explain?
On 8/9/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hi all,
I am using an asterisk as a client to connect to another asterisk server
by registering with the register string. Registration is done without any
hassel, but after sometime my asterisk loses the registration
dont know, and would very much like to know, is what is the purpose
of this parameter in sip packets?
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] wrote:
- Rizwan Hisham [EMAIL PROTECTED] wrote:
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk,
nonce=584760da
Authorization: Digest username=bernart48, realm=asterisk,
algorithm=MD5, uri=sip:[EMAIL PROTECTED]:9060, nonce=584760da,
response=948d3923bf2df47eca17c572713af2c7
to the
changing nonce value.
- Brad
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rizwan
Hisham
Sent: Wednesday, August 15, 2007 7:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
. I
have little knowledge of sip events. So if anybody knows a good link plz
share. And if u know how to fix the bad event message then plz tell also.
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call limit on peers by commands or do i have to restart my asterisk server?
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to see if they have
any dead/zombie channels, which you can remove with soft-hangup?
What version of * are you running?
What kind of phones?
What config options are you using in SIP (or other tech) to limit the
calls?
On 8/17/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hi all,
Some of my
extensions which i am not using.
So what r hints?
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/mailman/listinfo/asterisk-users
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/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
On 8/23/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
im having a strange problem related to call-limit for peers. well im not
sure if its related
is UNREACHABLE it knows about that user and
does not bother to send sip packets to that user anymore. This way channel
is not even initialesed if sip invite is recieved for that channel (and goes
directly to voicemail) and uninitailised channels cannot get stuck.
On 8/24/07, Rizwan Hisham [EMAIL
hi all,
Is there any default timeout for manager connection. If its configurable
then plz tell me how.
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which
sniffs sip packets coming for asterisk and detect for multiple register
requests coming from different IPs for the same username. Can anybody help?
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the host= to configure the allowed IP in sip.conf
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rizwan
Hisham
Sent: 11 September 2007 11:30
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Prevent
using his credentials.
On 9/11/07, Atis [EMAIL PROTECTED] wrote:
On 9/11/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
We cant do that. Thats becoz the original user may change his/her
location
which will result in change of ip address. We have to set host=dynamic
for
allownig the original user
mailing list
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. Is that
possible? if yes then how do i turn them on?
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Software Engineer
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that was it. Thanx
On 9/13/07, Mark Michelson [EMAIL PROTECTED] wrote:
Rizwan Hisham wrote:
i connect remotely. I have tried both of these cases but no warnings
or mesages still.
It could be that your logger.conf file doesn't know to send debug
messages to the cli. Make sure
it.
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/07, Matt [EMAIL PROTECTED] wrote:
Your best bet may be to write your own. That's what we ended up doing and
it isn't that hard.
On 10/24/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hi all,
I need a fully developed web based remote provisioning system. I cant
find anything reliable
://lists.digium.com/mailman/listinfo/asterisk-users
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Hi all,
i need an XML file format which is used in remote provisioning of different
spa devices. Please somebody tell me the format or tell me where can i find
it on the internet. I also need a list of parameters which are configured
using auto-provisioning.
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i have spa 2100. tried to access the file but got 404 not found. Any clues
why?
On 10/29/07, Per Jessen [EMAIL PROTECTED] wrote:
Rizwan Hisham wrote:
Hi all,
i need an XML file format which is used in remote provisioning of
different spa devices. Please somebody tell me the format
[13520]: chan_sip.c:8117 check_auth: username mismatch, have adf,
digest has abc*
Any solutions to this problem?
On Wed, Feb 27, 2008 at 4:36 PM, Igor A. Goncharovsky [EMAIL PROTECTED] wrote:
Rizwan Hisham wrote:
I am having a strange problem. I am using my asterisk server AST1 to
register
Adding fromuser option in trunk declaration in AST1 has solved all
problems though.
On Wed, Feb 27, 2008 at 4:36 PM, Igor A. Goncharovsky [EMAIL PROTECTED] wrote:
Rizwan Hisham wrote:
I am having a strange problem. I am using my asterisk server AST1 to
register with another asterisk server
. The
From header show the name and number which i set before dialing but on cli
it shows only name:
From: salman sip:[EMAIL PROTECTED]:5238;tag=as5100f7b2
Any one knows what should i do to solve this problem?
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of the call or it just transfers the call and
then all of the responsibility of the call is handled on the other server?
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time to
30,000 mili seconds. Can it be done in the dialplan? or should i jump into
the code?
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Hi,
Does anyone know the purpose of /n attached at the end of the dial
command below
Dial(Local/[EMAIL PROTECTED]/n Local/[EMAIL PROTECTED]/n)
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tell me how to take
out the conf files then it will also be very helpfull.
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Also it will be great if anybody can tell where i can find the explanation
of all the warnig codes and error codes of asterisk if there is any.
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, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
Why is it doing so?
On Thu, Apr 17, 2008 at 2:36 AM, Tony Mountifield [EMAIL PROTECTED]
wrote:
In article [EMAIL PROTECTED],
Rizwan Hisham [EMAIL PROTECTED] wrote:
I have been seeing a lot
asterisk
cards. if someone knows where to buy cards in pakistan, plz tell me about
it.
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not
exist. Can anybody tell me why its doing so, coz i can see on cli that the
channel exists. If i try to set the variable without stating the channel
then it sets the global variable, but i want to set the channel variable.
Anybody has a solution to this problem?
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same is the case in 1.6, i cant set the variable still.
On Thu, May 8, 2008 at 8:43 PM, Tzafrir Cohen [EMAIL PROTECTED]
wrote:
On Thu, May 08, 2008 at 07:39:39PM +0500, Rizwan Hisham wrote:
Hi all,
I am using a simple perl script to connect with ast manager api. the
script
tries to set
Thanx a lot.that was it. will never forget to remove the new
character again. Now its working fine.
On Fri, May 9, 2008 at 4:31 PM, Tony Mountifield [EMAIL PROTECTED]
wrote:
In article [EMAIL PROTECTED],
Rizwan Hisham [EMAIL PROTECTED] wrote:
same is the case in 1.6, i cant set
Can anybody help in parsing the manager events efficiently? Any ideas?
On Fri, May 9, 2008 at 5:07 PM, Gunārs Grundāns
[EMAIL PROTECTED] wrote:
On 5/8/08, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, May 08, 2008 at 07:39:39PM +0500, Rizwan Hisham wrote:
Hi all,
I am using a simple
mailing list
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, but i could
not find a satisfactory explanation for the this statement. Can anybody help
me understand the switch statement?
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can solve my multiple cdr problem?
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in a dialplan.
I am using asterisk1.4.2 and asterisk addon1.4.0 package for mysql
connectivity.
Thanx in advance
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have done it, and its working fine. but still expecting to receive some new
ideas.
On Wed, Aug 6, 2008 at 2:12 PM, Rizwan Hisham [EMAIL PROTECTED]wrote:
hi all,
i just finished developing some incoming call features in a macro. that
macro gets executed everytime an incoming call is received
. .$calltime.\n;
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, Anthony Francis [EMAIL PROTECTED]wrote:
Rizwan Hisham wrote:
Hi all,
asterisk is giving me tough time. its been 3 days I am trying to
originate outgoing call using manager api/callfiles.
I would say remove the @TRUNK-OUT part and make sure that the context
you send the call to knows
there is no transfering)
I googled a little on strict and loose routing but i did not get it. maybe
someone here can help me solve this problem.
VERSIONS
asterisk 1.4.2
zaptel and libpri 1.4.0
I can send you core debug if you want it.
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callerid=adf xyz 123
accountcode=6:0:adf
amaflags=default
disallow=all
allow=g729
allow=ulaw
allow=alaw
allow=gsm
am i doing something wrong here?
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hi all,
Is there an option of dtmf passthru mode in asterisk. If yes, how can i do
it?
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Register
at 6:05 PM, Alex Balashov [EMAIL PROTECTED]wrote:
What is DTMF passthru?
DTMF is regenerated by default. If the DTMF mode is inband, it's simply
part of the audio stream. If it uses named RTP events, those are
regenerated on the other call leg.
Rizwan Hisham wrote:
hi all
with a username.
Rizwan Hisham wrote:
Hi all,
I am having a problem with sip uri incoming calls. I have 2 asterisk
servers both are 1.4.2. http://1.4.2. i dial sip uri from one asterisk
server which sends the call to the other asterisk server by seeing its
domain name in the uri. Invite
of asterisk but they are not, already downloaded
and checked in asterisk 1.4.21.
How can i install these applications. Are there anyother components required
to make my asterisk a fax-passthru system.
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The fax is originated from a fax machine connected to an ata which supports
t38.
On Wed, Sep 24, 2008 at 11:54 PM, C F [EMAIL PROTECTED] wrote:
On Wed, Sep 24, 2008 at 5:43 AM, Rizwan Hisham [EMAIL PROTECTED]
wrote:
Hi all,
Sorry to interrupt. I need some help regarding fax passthru mode
[Apr 17 09:14:45] WARNING[11234]: translate.c:675 __ast_register_translator:
plc_samples 160 format 6
Hi all i ned to know what the above warning is trying to say. I have a
slight idea that its about some audio conversion, maybe. but can anybody
tell me for sure whats it about?
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that message?
Astawerks
VoIP Hardware sales and consulting
http://www.astawerks.com
614-495-1400
--
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Rizwan Hisham
*Sent:* Tuesday, April 17, 2007 10:11 AM
*To:* Asterisk Users Mailing List - Non-Commercial
/rizwan) ;Secondary did for user rizwan
exten= 1714,2,Hangup
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Hi all,
i have changed it myself inside the code. so if anybody wants the solution
for the above problem, just ask.
On 4/19/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hi guys,
i just came to know that CDR(dst) field is set to current extension
instead of the dialed no. i need to set it to DNID
named 'users'. i have tried putting all user extensions
in a single file but couldent solve the problem.
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. This is a big
problem for me as i have to charge the forwarded calls also and all calls
are charged based on account code. If accountcode is empty, i cant make a
decision how to charge the call.
Can anybody fix this for me or do i have to jump back to asterisk 1.4.2?
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Software
/07, Steve Murphy [EMAIL PROTECTED] wrote:
On Thu, 2007-05-03 at 16:47 +0500, Rizwan Hisham wrote:
Hi all,
i just updated to asterisk 1.4.4 from 1.4.2. i was doing this to
forward an unanswered call in 1.4.2
exten= 1,1,Dial(SIP/123,,Ttg)
exten= 1,2,Gotoif($[${DIALSTATUS}=ANSWERED]?:10)
exten
Nops. removing res_features doesnt work.
On 5/4/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
Nops. Its not working. i have restored to original chan_local file. Im
also having another problem now (in asterisk 1.4.4).
The call originates fine, ringing is done, call is accepted, channels
bridged
://lists.digium.com/mailman/listinfo/asterisk-users
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Oooh. i already have this book (Asterisk The future of Telephony). its not
about the code.
On 5/4/07, Iban Lopetegi Zinkunegi [EMAIL PROTECTED] wrote:
Hi Rizwan,
You can find the book in the next web page,
http://www.oreilly.com/catalog/asterisk/
Iban
From: Rizwan Hisham [EMAIL PROTECTED
Steve,
I didnt mean to say that your patch did that. Actually i did saw this error
before applying your patch. i just mentioned it here. So is this problem
fixable?
On 5/5/07, Steve Murphy [EMAIL PROTECTED] wrote:
On Fri, 2007-05-04 at 15:25 +0500, Rizwan Hisham wrote:
Nops. removing
can i do this. i have used
the g flag in dial which tell asterisk to execute remaining extensions even
after hangup but its not doing in the above described case.
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the call counter to zero from the dialplan? or is there anyway to know that
the user is registered or not before dialing that user?
I am using asterisk 1.4.2
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Rizwan Hisham
Software Engineer
AXVOICE Inc.
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Hi all,
i never understood that why is there 2 branches of asterisk going on
parallel. asterisk 1.2.* and asterisk 1.4.*, i also heard about beginning of
another branch which will be 1.6.*. so whats the difference between these 2
or 3 versions, can anybody plz tel me?
--
Rizwan Hisham
Software
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Rizwan Hisham
Software Engineer
AXVOICE Inc.
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I did it anyway. i used another way around to do it:
suppose 88777 is your number
exten= 88777,1,Dial(SIP/you)
exten= 88777/88777,1,VoiceMailMain()
but in this case you will have to make a separate vm extension for every
user.
On 5/22/07, Rizwan Hisham [EMAIL PROTECTED] wrote
as there is for
asterisk (Asterisk the future of telephony). i need a book or any helpfull
link which should reveil the basic concepts of openser, how to install it,
how to program it, and how to manage it.
Thanx in advance
--
Rizwan Hisham
Software Engineer
AXVOICE Inc
PROTECTED] *On Behalf Of *Rizwan Hisham
*Sent:* Tuesday, May 22, 2007 5:14 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Caller ID matching
I did it anyway. i used another way around to do it:
suppose 88777 is your number
exten= 88777,1,Dial
Hi all,
sorry to ask you something not related to asterisk, but i really want to
know whether the Nokia N95 cell phone is released in the USA or not? if
somebody from USA knows, plz reply.
--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
___
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ringing
at the start so that user hears only beep beep beep if the called user is
busy. I have used the R and r options in Dial application but they dont
work.
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Rizwan Hisham
Software Engineer
AXVOICE Inc.
___
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] wrote:
You should (must!) remove any r/R parameter from your command. If you do
that, no false ring will be generated anymore...
Att, Ricardo.
Rizwan Hisham escreveu:
Hi all,
when a user dials any number, asterisk automatically generates ringing
which caller can hear, and after 2 - 3 rings
Maybe its a bug in asterisk 1.4.2
On 5/30/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
There is no R/r option in my dial application.im only using gM option
here is the dialplan:
exten= _1X.,1,NoOp(Dialing Local!!!)
exten= _1X.,2,Dial(Sip/[EMAIL
PROTECTED],,gM(payasyougo^${CDR(accountcode
then the actual
ringing starts. like this tone -- tone -- totone -- tone, and if the callee
is busy then tone -- tone -- tobeep beep .
does anyone know what this means: -- Call on SIP/CARRIER-OUT-007d0310 left
from hold
On 5/30/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
Maybe its a bug in asterisk 1.4.2
and flash again to put out of hold. But I'm realy not sure about it.
Rgds, Ricardo Martins
Rizwan Hisham escreveu:
Here is my CLI output:
Called [EMAIL PROTECTED]
-- SIP/CARRIER-OUT-007d0310 is ringing
-- Call on SIP/CARRIER-OUT-007d0310 left from hold
-- SIP/CARRIER-007d0310 is making
Anybody who can help?
--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
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