I am sure, after I have it setup once, everything will be cristal clear.
I could not find a documentation, maybe together we can make at least a
README for the next user ;-)
1. Brands:
Brand Name
Create a brand name you want to market your pre-paid card
Language
Choose one of the
Hi All,
I've installed digium TDM02B. The PSTN line
connected to this card. At the IP network side, I have
SIP phones registered sucessfully to asterisk server.
How do I configure asterisk, so once there is incoming
call to the TDM channel from PSTN, the caller will
hear another dial tone from
dir sir;
If you want to got informations about that, so i can
give an architecture combining Hardware and software
for this...
sincerly
--- mohammad [EMAIL PROTECTED] a écrit :
Hi;
Any idea of how to have different ringing tone on
called party for
Running Asterisk CVS-HEAD-12/21/04-20:17:06.
When i Login with AgentCallBackLogin the agent will
have status unavailable until I do a reload on Asterisk, then the
Agent is logged in. If the Agent logs out and then logs in again, I will have
to reload Asterisk again before the Agent can
On Sunday 26 December 2004 11:13 am, steve szmidt wrote:
I have not been able to find anything that relates to this problem. The
agents are using Cisco phones.
Calls goes into a queue. but once an agent picks it up it cannot be
transferred. However if they call directly to the agents
On Sunday 26 December 2004 02:57 pm, Hecken, Guido wrote:
I had the same problem with snom 190 phones.
Using the transfer with # instead of Transfer Button on the phone worked
for me.
In my configuration REFER was not send, so the transfer with the button
on the phone did not work.
Guido
Hi list!
I'm trying to install bristuff on a newly installed box. The box is
running a rebuild of RedHat Enterprise linux with the latest kernel. The
architecture of the box is x86_64 (Athlon 64), the os is 64 bit with 32
bit compat libs installed. Also the kernel source is installed as well as
How do I configure asterisk, so once there is incoming
snip
This and a thousand other secrets are revealed in several available
free documents a few of which are listed below:
http://asteriskdocs.org
http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
On Tue, Aug 24, 2004 at 03:34:34PM +1000, david kwok said:
I have been bugging by a telemarketer who does not take any cue at all.
So I look up the Asterisk Handbook and send his call with the respect
caller id to my voicemail.
Has any one implemented any of this feature with database for
On Mon, 27 Dec 2004, Remco Barende wrote:
Hi list!
I'm trying to install bristuff on a newly installed box. The box is running a
rebuild of RedHat Enterprise linux with the latest kernel. The architecture
of the box is x86_64 (Athlon 64), the os is 64 bit with 32 bit compat libs
installed. Also
Here is where the problem is.
When the call comes in, it will be ringing on 2 of the FXO ports,
and all the other phones in the office. I would like various / all
the IP phones to ring, however asterisk must not answer the call
while that is happening or else the normal extension would
Hi all,
Is there a way to change the hookflash
timing with the TDM400P?
Allready been searching the mailing
list/google etc but i can't find anything ;-(
I tried flash= in zapata.conf, but that
only works with the T1/E1 cards.
Greetz,
Caspar
___
How do you set up Asterisk to allow SIP call requests from specific IP
addresses? We have no control over what account (From: header) is used. We
want to be able to allow calls based on the IP address the INVITE comes
from, not the account. Is there a way to do that?
Hi:
We have got SIP clients connecting to our Asterisk fine with a DSL
connection behind router (NAT), but when we bring the Sipura 2000 ATA
to a Rogers Cable connection behind a Netgear router (NAT), the SIP
clients aren't able to reach the Asterisk at all.
We enabled the SIP debug in Asterisk,
want to be able to allow calls based on the IP address the INVITE comes
from, not the account. Is there a way to do that?
In iaxy.conf this looks like this
permit=0.0.0.0/0.0.0.0
It may be identical in sip.conf (or not!)
___
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permit=0.0.0.0/0.0.0.0
Obviously, the above permits all... Needs to be adapted
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Try setting the SIP signalling port in your client to something other than
5060 (eg ) and run tethereal on your Asterisk box to see if you're
getting packets on .
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of K Wong
Sent: Monday, December 27, 2004
Hi all,
i'm trying to configure MoH for OptiPoint400std SIP, but it doesn't work. If i try with a softclient moh works fine. Somebody experience with OptiPoint400? Is there additional settings on the Optipoint?
Any help would be much appreciated!!
thx.
Do you Yahoo!?
Meet the all-new My
How do you set up Asterisk to allow SIP call requests from specific IP
addresses? We have no control over what account (From: header) is used. We
want to be able to allow calls based on the IP address the INVITE comes
from, not the account. Is there a way to do that?
Something like this in
I compiled and installed zaptel on an x86_64 box (bristuffed).
The compile when fine but when I try to modprobe the module i get loads of
errors:
/lib/modules/2.4.21-27.301.EC/misc/zaptel.o: Possibly is module compiled
without -mcmodel=kernel!
/lib/modules/2.4.21-27.301.EC/misc/zaptel.o:
Are you guys running Debian Stable or what? If so, apt-get install asterisk
will install an outdated version. Are you using a different apt source for
asterisk?
I am looking at this, and we want stability, and even though I use Sid every
day at home and it's fine, I don't think it's smart to
If youre using
G.711 make sure youve got silence suppression turned OF, seems that the
phone only receives rtp while sending it.
(or, do as we did, throw
the OptiPoint out of the window J)
Van:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens richard Coco
Verzonden:
Has anyone put together a list of basic modifications recommended
settings to optimize a TCP/IP (mostly wireless)network for VOIP?
Specifically G.711 IAX protocol, and the Vonage and Packet8 services.
My network is mostly StarOS (which uses linux cbq and ipchains) and
Mikrotik and I'd like to
Hello there
Can Asterisk send me a Mail wenn a voice mail
arrived for me ?
if yes how can i do this ?
Thanks
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Hi Erik,
thx for the feedback. I turned off the Silence Suppression but unfortunately it doesn't work.
(i don't think that throwing it out of the window will resolve my problem... but it will think about it... to be continued;-))"E. Versaevel" [EMAIL PROTECTED] wrote:
If youre using G.711
Hello there
Can Asterisk send me a Mail wenn a voice mail arrived for me ?
if yes how can i do this ?
Thanks
YES IT DOES.
SEARCH: GOOGLE
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Good morning,
I just got asterisk up and going
and set it up to communicate to test calling out to a PSTN phone number. My
configuration is as follows:
My Diax client -- IAX2 on my asterisk server
-- IAX2 (voice pulse server) -- PSTN --my PSTN phone.
I don't have any ZAP hardware. I'm
hi
check the voicemail.conf
Attach=yes
Attach causes Asterisk to copy a voicemail message to an audio file and send it to the user as an attachment in an e-mail voicemail notification message. The default is not to do this. Attach takes two values yes or no.
extension_number =
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
put this in sip.conf
;externip = x.x.x.x; Address that we're going to put in outbound
SIP messages ( official ip address)
;localnet=10.0.0.0/255.255.255.0; if we're behind a NAT
add
nat=yes to every sip account which is behind of NAT
BR Hanson
steve szmidt wrote:
If you terminate the T's in the Asterisk box and then put patch cables between
the Asterisk box and your Comdial, you can probably accomplish these things.
You might need to detect what your Comdial does to talk to a VM system and
then configure Asterisk to answer properly.
Toggle the break key in the web config on your snom and then the
break/transfer key will actually be the transfer key.
On Sun, 2004-12-26 at 09:09 -0500, steve szmidt wrote:
On Tuesday 21 December 2004 10:36 pm, Tracy R Reed wrote:
I am having a hell of a time with transfers.
First the
Hi!
Is there a way to avoid being at the middle of communications between
two SIP endpoints? So that we can avoid loosing bandwidth with it?
Is there a way to forward the authentication to a IAX provider and
transfer the call to it, avoiding using my own bandwidth?
I've tested it with SER
VOIP pioneer predicts a roiling 2005 for IP telephony
Eetasia.com (subscription) - USA
Open source software communications will begin to influence the VoIP market
in a big way next year, according to VoIP pioneer Jeff Pulver. ...
http://www.eetasia.com/article_content.php3?article_id=8800354924
OS: Redhat 9.0
Kernel: 2.4.20-8
Asterisk: latest CVS head
Hi all,
I try to compile the latest chan_capi, but it won't work. I also tried
the patch that is arround there from frank sautter and I have
installed the asterisk header files.
Asterisk is working very good :) and the Fritz AVM pc Card
On Mon, 27 Dec 2004 08:08:28 -0700, Matt Larsen
[EMAIL PROTECTED] wrote:
Has anyone put together a list of basic modifications recommended
settings to optimize a TCP/IP (mostly wireless)network for VOIP?
Specifically G.711 IAX protocol, and the Vonage and Packet8 services.
My network is
Seems to me that you are not running most recent version of asterisk.
-Matthew
- Original Message -
From: Steve Beaumont [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, December 26, 2004 2:14 PM
Subject: [Asterisk-Users] Asterisk realtime load error
Anyone know
On Monday 27 December 2004 10:53 am, Andrew Thompson wrote:
steve szmidt wrote:
If you terminate the T's in the Asterisk box and then put patch cables
between the Asterisk box and your Comdial, you can probably accomplish
these things.
You might need to detect what your Comdial does to
Hi ;
Here isthe dial-plan for distinctiv ring
tonesfor differnet caller-id (incoming call) :
exten = 206,1,GotoIf($[${CALLERIDNUM} = 303]?2:3)
exten = 206,2,SetVar(_ALERT-INFO=Bellcore-dr4)
exten = 206,3,Dial(SIP/206 |20| Ttr)
If callerid is 303 play different ring tone , else
play the
Hi Kristian,
I managed to lose the link you sent me earlier. Would you mind sending
it to me again for this retailer? I would greatly appreciate it.
Thank you for all your help
Sean
Sean Kennedy wrote:
Hi Kristian,
What extra configuration is needed for the one touch voicemail? I
have an
Sean - We have plenty of IP500's in stock at competitive prices, we can ship
today.
http://www.voipsupply.com/product_info.php?cPath=3_64products_id=251
Cory J Andrews
Partner / Purchasing
^
VOIPSupply.com
454 Sonwil Drive
Buffalo, NY 14225
^
voice
Thanks that did the trick!
Thanks again,
Jon
- Original Message -
From: Michael Bielicki [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, December 24, 2004 12:18 AM
Subject: Re: [Asterisk-Users] TE410P X100P
I'm somewhat new to Asterisk and am tasked with having it perform some automated functions. Is there a way with the current system (and/or extra modules out there to:)
1. Launch something from a command line (on the Asterisk server) to:
2. Dial an extension
3. Issue some DTMF sequences,
4.
Hi Fletcher,
Good morning,
I just got asterisk up and going and set it up to communicate to test
calling out to a PSTN phone number. My configuration is as follows:
My Diax client -- IAX2 on my asterisk server -- IAX2 (voice pulse
server) -- PSTN --my PSTN phone.
I don't have any ZAP
I recently swapped 2 FXO modules on to what had previously been a 4 FXS
version of the TDM400 board. The FXS ports are recognized - the FXO
ports don't appear to be recognized (ie modprobe wcfxo and ztcfg both
say channel 1 isn't there). Has anyone experienced this problem? All
software is
Hi, Im running into issues receiving faxes which,
from what I have read, may be caused by frame slips. While I can find
many posts saying to investigate it, I cant find any that describe *how* to debug the problem. Tried searching
this list as well to no avail.
Any pointers would be
Thanks for the response.
Show version indicates:-
foxy*CLI show version
Asterisk CVS-v1-0-12/24/04-20:33:21 built by [EMAIL PROTECTED] on a i686 running
Linux
foxy*CLI
Incidently, as a workaround to get the console up and running, I have noload
in for the realtime module, not clever i guess,
I
think Ive managed to figure out that there are two ways to transfer a Zap
call, using hookflash (defined in zapata.conf) or the # key (the t and T
options of the Dial command in the dialplan), but not why there are two ways to
do this, nor what the difference is between them. Is there
I love it when people prove themselves wrong. hehe. show version should say
this:
Asterisk CVS-HEAD-12/15/04-21:47:35 built by [EMAIL PROTECTED] on a i686
running Linux
Note: CVS-v1-0 is not CVS-HEAD
You are not running most recent version.
-Matthew
- Original Message -
From: Steve
Robert Rozman wrote:
Hi,
I plan to connect to remote Asterisk that will terminate calls to ISDN
primary channel. I'd certainly like to secure this type of service, so would
kindly ask for any advice on how to secure this authentication as much as
reasonably possible.
What are you trying to secure?
Warren Burstein wrote:
I think I've managed to figure out that there are two ways to transfer a Zap
call, using hookflash (defined in zapata.conf) or the # key (the t and T
options of the Dial command in the dialplan), but not why there are two ways
to do this, nor what the difference is between
Is
the power connector on the TDM400P only needed for line and dial voltage, or do
you also need it if it has all FXO lines?
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Warren Burstein wrote:
Is the power connector on the TDM400P only needed for line and dial
voltage, or do you also need it if it has all FXO lines?
In my experience, you only need the power connector for FXS modules. If
you have all FXO modules, you should be okay without it.
--
Kristian
As you have a TDM400 you should load wcfxs module ( modprobe wcfxs or
modprobe wctdm for new versions) regardless to modules you have installed on
your board. You should also check the signalling specified in zaptel.conf
according to your modules and the order they are placed on your TDM400.
Best
Hello everybody and merry xmas.
I have a problem with sip phones calling each other inside the same
network (no nat, no firewall).
You can make and receive calls and pick them up, but you cannot hear
anything on any side of the call. But if you press hold/unhold or you
transfer the call, then
Mamadou Lamine KA wrote:
As you have a TDM400 you should load wcfxs module ( modprobe wcfxs or
modprobe wctdm for new versions) regardless to modules you have installed on
your board. You should also check the signalling specified in zaptel.conf
according to your modules and the order they are
Joe Presto wrote:
Hi, Im running into issues receiving faxes which, from what I have
read, may be caused by frame slips. While I can find many posts saying
to investigate it, I cant find any that describe **how** to debug the
problem. Tried searching this list as well to no avail.
Joe Hi,
Warren Burstein wrote:
Is the power connector on the TDM400P only needed for line and dial
voltage, or do you also need it if it has all FXO lines?
In my experience, you only need the power connector for FXS modules. If
you have all FXO modules, you should be okay without it.
--
On Mon, 27 Dec 2004, Brent Goran wrote:
I'm somewhat new to Asterisk and am tasked with having it perform some
automated functions. Is there a way with the current system (and/or
extra modules out there to:)
1. Launch something from a command line (on the Asterisk server) to:
2. Dial an
does anybody know what these log messages mean?
what ever it is, asterisk needed a restart to become active
again. (server was not rebooted and remained live to ssh
and other network functions.)
no outgoing calls can be made.
The system was just sitting idle over night and trying to make
a call
Thanks,
I love it too if I understand.
Please enlighten me I guess CVS-HEAD is the development version and
CVS-v1-0 satble version ?
Sorry to ask dumb questions, but like the old saying they are only dumb if
you know the answer.
Best regards
Steve B
-Original Message-
From: [EMAIL
The lable 'CVS' just means that you got your version of asterisk using cvs.
HEAD is devel version and v1-0 is stable.
-Matthew
- Original Message -
From: Steve Beaumont [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent:
Steve Beaumont wrote:
Please enlighten me I guess CVS-HEAD is the development version and
CVS-v1-0 satble version ?
Correct.
Don't forget to bookmark http://www.voip-info.org
--
Cheers,
Matt Riddell
___
Daily Asterisk News:
hi
Username : 112
Codecs : 0x11a (gsm|alaw|g726|g729)
Codec Order : (gsm|g729|g726|alaw|ulaw)
the above is from SIP SHOW PEER 112, and as it clearly shows, g.729
is preferred before alaw. If I dial this SIP - * - SIP from a phone
with G.729 enabled, it uses G.729.
Thanks, so I download with -r head.
Sorry to be thick, I'm network engineer not a developer :-)
I'll stop creating noise, this is a busy enough list as it is.
Cheers and thanks again.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Matt
Sent: 27
I am reading the WiKi page for the The Asterisk Manager API and I am
interested in the functionality provided by the Redirect with
ExtraChannel event. There is minimal explaination on the WiKi page. Can
anyone help/direct me to more doc on the subject.
regards
moe smadi
Steve Beaumont wrote:
Thanks, so I download with -r head.
If you want to get the head version, you do not need the -r tag.
--
Cheers,
Matt Riddell
___
Daily Asterisk News:
http://www.sineapps.com/news.php for html
http://www.sineapps.com/rssfeed.php for
Hi,
- Original Message -
From: Andrew Thompson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, December 27, 2004 7:32 PM
Subject: Re: [Asterisk-Users] How to connect two Asterisks as secure
aspossible without too
OpenVPN
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Robert Rozman
Sent: Monday, December 27, 2004 2:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] How to connect two Asterisks
Brian West wrote:
OpenVPN
What happened to AES in IAX2?
/O
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I have 2 asterisk boxes: asterisk-alpha (running 1.0.3) and dev-asterisk
(running latest CVS).
I am the only SIP user on dev, everyone else in the office is on alpha.
If someone dials my extension, it should go IAX to the dev server and the
dev server should ring me.
Here is what I see on the dev
Lyle,
Can you point me to the location on the Uniden site where I can find UIP200
firmware upgrades.
Thanks,
Ed McKinnon
http://www.crmi.com
*** REPLY SEPARATOR ***
On 12/23/2004 at 9:43 AM Lyle Giese wrote:
Firmware v 4.63 has been released on the Uniden website. No docs
I would love to see it.. but we need to get that codec ordering stuff tony
and I worked on but nobody seems to be even remotely interested in.
http://bugs.digium.com/bug_view_page.php?bug_id=0002971
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
We use this in the astGUIclient to transfer an active conversation(both
parties) to a meetme room:
Action: Redirect
Channel: Zap/73-1
ExtraChannel: SIP/199testphone-1f3c
Exten: 8600029
Context: default
Priority: 1
where 8600029 is a meetme room.
Works very well.
Sadly like most obscure
Help, Any ideas ? I guess I missing something.
make[1]: Entering directory `/usr/src/asterisk/utils'
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat
ions -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686
Yes, that is possible,
you configure that in voicemail.conf
Thorben
Fra:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] P vegne af Tasos Daskalopoulos
Sendt: 27. december 2004 16:12
Til:
asterisk-users@lists.digium.com
Emne: [Asterisk-Users] mail
function
Hello there
I just updated my asterisk box and now it's giving me this error I
looked it up on the internet found no solutions
any other information that you need please ask.
[EMAIL PROTECTED] root]# modprobe wcfxo
/lib/modules/2.4.22-1.2115.nptl/misc/wcfxo.o: init_module: No such device
Hint: insmod errors
Hi there, i installed RxFAX/TxFAX with some troubles but i did it, so i have some problems when i try to receibe a FAX, i got this error:
" Executing RxFAX("IAX2/[EMAIL PROTECTED]/16385", /var/spool/asterisk/incoming/16227743.tif") in new stack
Dec 27 15:30:54 NOTICE[1141895616]: channel.c:1731
Paste your extensions.conf section that is relevant.
-Matthew
- Original Message -
From: Greg - Cirelle Enterprises [EMAIL PROTECTED]
To: asterisk-dev@lists.digium.com
Sent: Monday, December 27, 2004 4:32 PM
Subject: [Asterisk-Dev] realtime voicemail
Let me clarify my last message.
Carlos Medina wrote:
Hi there, i installed RxFAX/TxFAX with some troubles but i did it, so i
have some problems when i try to receibe a FAX, i got this error:
Executing RxFAX(IAX2/[EMAIL PROTECTED]/16385,
/var/spool/asterisk/incoming/16227743.tif) in new stack
Dec 27 15:30:54
James Moran wrote:
I just updated my asterisk box and now it's giving me this error I
looked it up on the internet found no solutions
any other information that you need please ask.
[EMAIL PROTECTED] root]# modprobe wcfxo
/lib/modules/2.4.22-1.2115.nptl/misc/wcfxo.o: init_module: No such device
Matt wrote:
James Moran wrote:
I just updated my asterisk box and now it's giving me this error I
looked it up on the internet found no solutions
any other information that you need please ask.
[EMAIL PROTECTED] root]# modprobe wcfxo
/lib/modules/2.4.22-1.2115.nptl/misc/wcfxo.o: init_module: No
At 05:49 PM 12/27/04, you wrote:
I have a similar problem with my *.Works fine but after some number of
hours, nothing works with no apparent reason. Restarting * fixes
everything.
I hope someone comes up with some suggestions!
Norm Z
I just downloaded a new cvs to see if that helps,
At 06:12 PM 12/27/04, you wrote:
James Moran wrote:
I just updated my asterisk box and now it's giving me this error I looked
it up on the internet found no solutions
any other information that you need please ask.
[EMAIL PROTECTED] root]# modprobe wcfxo
Hi,
I have a Budgetone 100, and a X-Lite soft phone, both registered with
Asterisk. when a call is made between them (regardless of which way) the
Soft phone can hear both ways, but the Budgetone can't hear the soft phone.
Any ideas?
--
Richard
___
Hi,
How can I give prompt/options to dial-in users for which extension to
connect?
Regards,
Norman Zhang
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As far as I know, Asterisk/Zaptel does not support analog DID service.
--Eric
I am thinking that there is some confusion about DID's.
The only important difference between an analog DID and a POTS line is
that when you pick up a DID, you are sent via DTMF the number (or a
portion there of)
Hi all,
I am wondering how to get PassThrough working in a NAT environment.
Many Thanks.
Vincent
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On Mon, 2004-12-27 at 17:16 -0800, Norman Zhang wrote:
Hi,
How can I give prompt/options to dial-in users for which extension to
connect?
By first looking over the documentation. Or even the example
extensions.conf file.
Hi Vincent,
Yes it works. I have tried that before with 2 BT101 talking at G723.1
through Asterisk where nat=yes
David
[EMAIL PROTECTED] wrote:
Hi all,
I am wondering how to get PassThrough working in a NAT environment.
Many Thanks.
Vincent
___
Just a note on this. I tried using an external device with the TDM400
configured as 4 FXO to ring even with asterisk. But no matter how I
configured it, asterisk always picked up. and the external device
didn't ring (just the first ring for CallerID to come in).
On Mon, 27 Dec 2004 07:04:00
On Mon, 2004-27-12 at 13:34 -0800, Edward J. McKinnon wrote:
Lyle,
Can you point me to the location on the Uniden site where I can find UIP200
firmware upgrades.
Thanks,
Ed McKinnon
http://bcs.uniden.com/
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Yes I do. Post the zapata.conf, zaptel.conf and the extensions.conf,
i'll see what I can do.
On Fri, 24 Dec 2004 18:49:27 -0500, Jeff [EMAIL PROTECTED] wrote:
Hey folks;
I am a old Dialogic telephony hack and have tried asterisk a while back on
a laptop with the OSS module and I liked what
Hi everyone,
I am trying to connect 2 asterisk servers via IAX, but it just
fails to do so.. I'm using SIP to connect the IP phones on the
LAN at the 2 different physical locations where each server
resides and I'm able to communicate on my LAN via SIP without
any issues. The problem is that
Hello All,
I have a problem that is alien to me and obvious for some of you
:). I have asterisk setup with few sip clients.
In a proper context, I have mentioned extensions 107 as
[EMAIL PROTECTED]
Asterisk Server-simputer(sip ua)
I can make calls from sipua to
Hello *'s,
Hi, I've just tried to enable MYSQL Friends in CVS HEAD. But i cannot
find this option.On wiki i found this.
To enable this, you need to edit the Makefile in the channels directory
of your source tree and enable MYSQL_FRIENDS. This enables database
definition of both IAX2 and SIP
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;
Mike, thanks so much.. here are the results - unfortunately, not even a new
build on a new system is solving this issue.
* See my notes to the tests below. I ended up trying 3 computers, the last
being a Dell 1600 server with a fresh install following the AMP instructions
at
Hello everybody,
im newbie in VoIP, but find this project asterisk very interesting, i
tried to install and its a great sw, i really get sorprised about all of
its functions, we need to use the asterisk server in conjunction with
cisco callmanager.
We have a Cisco Callmanager 4.1 and the clients
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