Alejandro G wrote:
I have a problem with ATA-186 configured for silence supression (AudioMode
bit 0 = 1). When enabled and listening music on hold no sound is heared (if
I talk I began to hear the music and again mutes when I stop talking).
If I configure for silence supression off everything goes
hi friends !
i am facing a problem from one week and now required ur help urgently.
Actually, i want to configure asterisk for two groups javgroup and
linuxgroup.
i also have constraint to use only sip phone (esatara ). now, please help me
is it possible to configure astersik in that way or
Can you be more specific?
What are you trying to achieve with the creation of such groups?
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, April 04, 2005 3:50 PM
Subject: [Asterisk-Users] how to configure groups using a sip phone
hi friends
The term RTCache has never been mentioned in the WIKI or these forums. I
assume that it's some sort of function to speed up realtime db access by
keeping transactions in RAM and writing periodically? If so, I can
understand why this would need to be flushed.
- Original Message -
From:
Hi list,
We are running a CVS version of 03-30-2005 but also had this behaviour
on previous versions.
From time to time, after a period of not making calls (eg a night or
few hours), we have no dialtone when we want to call. SIP show peers
show EP registered with status OK but nothing happend.
AS5300 setup
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2005.04.04 09:37:31
=~=~=~=~=~=~=~=~=~=~=~=
sh runn
Building configuration...
Current configuration : 11599 bytes
!
! Last configuration change at 03:26:25 GMT Mon Apr 4
2005 by charles
! NVRAM config last updated at 03:06:50 GMT Mon Apr 4
2005 by
No.
- Original Message -
From: Alexandre Charles [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, April 04, 2005 3:48 PM
Subject: [Asterisk-Users] V92 modem with asterisk
Hi everyone,
I just install Linux and asterisk on one of my pc. I
want to run some basic
Hi everyone
I'm trying to setup this Welltech Wellgate 3701 box.
If I got to the proxy setup it seems to work but the Pstn incoming call
always got a voice prompt from the Wellgate.
Going to peer to peer mode seems to be better but I couldn't find any
working configuration inside Asterisk.
I
Over the last few weeks/months I have been testing
phones and ATAs from Grandstream (BT101, GXP2000, 286, 488), SNOM (190), Zyxel
(Piece of Crap), Sipura (SPA-2000, SPA-841) and I personally feel that the
Sipura SPA-841 is the best value, good quality phone that I have used. I haven't
used
Lee Lee wrote:
Hi everyone
Presently all our calls are channel to one provider and we would like
to change that based on LCR.
the following is what we have presently;
# Dial the requested number, if we got something from the caller.
if ($dialto != )
{
$AGI-exec('SetAccount',
My personal opinion is that the Polycom IP-300 is a slightly better phone than
the Sipura, but I would be happy to be proved wrong on that.
later,
PaulH
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon
Sent: Monday, 4 April 2005 5:00
You would use the caller ID to route the call.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nabeel
Jafferali
Sent: Sunday, April 03, 2005 10:17 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE:
Does anyone else have this problem? Is there a workaround?
Yeah, I had this problem when I added a lot of SIP register statements
and SIP peers. Changing the hostnames (FQDNs) to IP addresses solved the
problem. It seem * was getting stuck waiting for DNS lookups.
Nabeel
Mishehu
try 19750407
Also to get palmtool to work you need to play with the debug
settings on the phone first.
koltov
Clive
On 2 Apr 2005 at 0:30, I put the Who? in Mishehu wrote:
Hi guys,
I just got a Netweb 401 (AT-320) phone. It came with firmware 1.38 on
it, and it has since
I would like to get a notice by email, if we run out of gateways!
exten = _9011Z.,410,Busy
exten = _9011Z.,411,EMAIL = How to?
bye
Ronald
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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=\\
On Thu, 31 Mar 2005, Peter Svensson wrote:
It would not be very hard to add both features to libpri. Libpri already
has a function to decode and dump the time/date information. If I
remember correctly the time/date IE should be added to the SETUP
messages. I have been thinking about adding it,
Ronald Wiplinger wrote:
I would like to get a notice by email, if we run out of gateways!
exten = _9011Z.,410,Busy
exten = _9011Z.,411,EMAIL = How to?
-= Info about application 'System' =-
[Synopsis]:
Execute a system command
[Description]:
System(command): Executes a command
Hi ya-all.
Little question that has been bothering me somewhot.
Say I have only 2 out going analog phone lines.
Some1 in the office decides to call their a client...
so the Dial command it using a group and it will start at the first Zap
channel listed
in the group.
But now what if I disconnect
Nabeel Jafferali a écrit :
Does anyone else have this problem? Is there a workaround?
Yeah, I had this problem when I added a lot of SIP register statements
and SIP peers. Changing the hostnames (FQDNs) to IP addresses solved the
problem. It seem * was getting stuck waiting for DNS lookups.
On 04/02/05 10:11 Mike Mueller said the following:
I don't think an Asterisk box can generate enough calls to cause sockets
related performance penalties. Five packets per phone call. What's the
max call rate an Asterisk box can support?
i think that would require an OS dependent answer.
but
On 04/01/05 00:00 Matthew Boehm said the following:
Steve Underwood wrote:
And your EU bias is clearly demonstrated by this. I've never seen a
BRI product outside he EU. :-)
Come to Houston, TX. We were running a BRI for quite some time before
upgrading to a T1.
ahem, ISDN BRIs are fairly
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Hash: SHA1
Uhmm ...
maybe a connection plar from ccme to an * number (like 511 on my conf),
then a simple forward from 511 to 601 on ccme?
Something like:
exten = _511,1,Dial(SIP/601,45)
I need help ... :D
Andrea
-BEGIN PGP SIGNATURE-
Version: GnuPG
Hi Bacon
Thanks for the quick response.
Actually I want to confirm that whether it is possible to divide logical
channels into group just like physiacl channels in zapata.
Deepak Dhiman
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon
Sent:
administrator tootai wrote:
Nabeel Jafferali a écrit :
Does anyone else have this problem? Is there a workaround?
Yeah, I had this problem when I added a lot of SIP register statements
and SIP peers. Changing the hostnames (FQDNs) to IP addresses solved the
problem. It seem * was getting stuck
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Apr 4, 2005, at 10:07 AM, Andrea Riela wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Uhmm ...
maybe a connection plar from ccme to an * number (like 511 on my
conf), then a simple forward from 511 to 601 on ccme?
Something like:
exten =
Olle E. Johansson a écrit :
administrator tootai wrote:
Nabeel Jafferali a écrit :
Does anyone else have this problem? Is there a workaround?
Yeah, I had this problem when I added a lot of SIP register statements
and SIP peers. Changing the hostnames (FQDNs) to IP addresses solved
the
BRI's are in use in roughly 2/3 of the world with the US and I think
China being the main exceptions.
On Apr 4, 2005 9:37 AM, Dinesh Nair [EMAIL PROTECTED] wrote:
On 04/01/05 00:00 Matthew Boehm said the following:
Steve Underwood wrote:
And your EU bias is clearly demonstrated by
Hi,
QoS is nice (and important) but only works within a FULLY controlled end to
end link.
Inside a BIG enterprise LAN, on leased lines its OK.
Using end to end MPLS should also be ok
Mind that some provider sell MPLS but it is not their own MPLS end to end.
Going from one provider on MPLS to
Hello all,
I have a working Asterisk setup, also a working sipgate.co.uk account
(tested with a GrandStream ATA 486), but got stuck in forwarding calls
from local users to sipgate. Very frustrating, since I feel there's just
one silly error somewhere.. story follows:
REGISTER both of the local
On 4 Apr 2005, at 09:25, Shaoul Jacobson - TELLINK wrote:
Hi,
QoS is nice (and important) but only works within a FULLY controlled
end to
end link.
Inside a BIG enterprise LAN, on leased lines its OK.
Using end to end MPLS should also be ok
Mind that some provider sell MPLS but it is not their
Good day all
Did someone get the planet VIP 450 working
Thanks
Altus
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I guess I should have added that this is based on the European, and
specifically UK model, but I would have expected it to have been
deemed best practice by most operators.
On Apr 4, 2005 4:04 AM, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote:
Rod Bacon wrote:
This is quite interesting.
Razvan Cosma a écrit :
Hello all,
I have a working Asterisk setup, also a working sipgate.co.uk account
(tested with a GrandStream ATA 486), but got stuck in forwarding calls
from local users to sipgate. Very frustrating, since I feel there's
just one silly error somewhere.. story follows:
Hello there,
How do I configure any type of action based caller's extension and dialed
number? For example if someone on extension 1777 calls extension 1777 this
should be treated as accessing his voicemail box, so he won't need to call
voicemail and entering mailbox number and password.
I.N.
I am hoping someone in the * community has come across this problem before.
Problem:
Person SIP Phone A (SIPA)
Person SIP Phone B (SIPB)
SIP Phone C (SIPC PSTN Line)
SIPA calls a billable phone number via SIPC
exten = _123456/_1XX,1,SetAccount(${ACCOUNTCODE_COMPANYZ})
exten =
Nabeel,
Could you expand on your comments, or provide a link / paste in a
sample extensions.conf to show how this would be set up?
David
On Apr 4, 2005 12:57 AM, Nabeel Jafferali [EMAIL PROTECTED] wrote:
Dial(SIP/904)calls whoever logged on as john.
You could define a variable in
I tried to use ONE entry of my voicemail.conf to put into the database:
[other]
;602=1357,Ronald Wiplinger 602,[EMAIL PROTECTED]
INSERT INTO `voicemail_users` ( `uniqueid` , `customer_id` , `context` ,
`mailbox` , `password` , `fullname` , `email` , `pager` , `stamp` ,
`attach` , `saycid` ,
Hello, Alejandro!
AG I have a problem with ATA-186 configured for silence supression
Don't!
I.N.
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I am hoping someone in the * community has come across this problem before.
Problem:
Person SIP Phone A (SIPA)
Person SIP Phone B (SIPB)
SIP Phone C (SIPC PSTN Line)
SIPA calls a billable phone number via SIPC
exten = _123456/_1XX,1,SetAccount(${ACCOUNTCODE_COMPANYZ})
exten =
On 04/04/2005 12:46 PM, administrator tootai wrote:
according to your sip.conf, should be
[...]
exten = _9.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
in extensions.conf
Ye :) Thank you very much!
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Hi,
I'm not sure I totally agree.
Good, we do agree on some :)
I also agree with some of your remarks
(no flame war)
It is also useful if you control the narrowest pipe.
I agree. But I disagree about the definition of the narrowest pipe.
A well configured router there will slow outgoing
Hi all, when I try to transfer a call asterisk say me:
-- Executing SetCallerID(SIP/20012-cb87, Gallina Daniele
20012) in new stack
-- Executing Dial(SIP/20012-cb87, SIP/20013) in new stack
-- Called 20013
-- SIP/20013-034d is ringing
-- SIP/20013-034d answered SIP/20012-cb87
Hi
I am using CVS latest
Is it correct there is no jitter buffer for SIP (RTP)
Are there any plans for this?
prob a stupid question:
Is it required / do the endpoints handle this - if the
src and destination are both SIP and there is no
transcoding but asterisk is still in the media path?
I have come accoross the fact that * can't handle if there is no
dialtone
So out of interist, can you do Line hunting in * in a sequencial manner
and can you
also do so in a random fasion?
--
Kind Regards
Etienne
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On Mon, 4 Apr 2005, Tobias Jönsson wrote:
On Thu, 31 Mar 2005, Peter Svensson wrote:
It would not be very hard to add both features to libpri. Libpri already
has a function to decode and dump the time/date information. If I
remember correctly the time/date IE should be added to the SETUP
Ok - I was told that you set a group for Zap channels.
So I tried to make use of my Zap channels so the 2 I am interisted
in is channel 3 and channel 4.
I make Channel 3 in use bu calling a line... then I try to call another
line so expecting to have Zap channel 4
open and allowing me to
I'd think about using a prefix for each trunk as a form of password. At
home I have to dial 1 then the number to use one of my trunks, or 2
then the number for a different trunk. If you gave them a code of say
666 they would have to dial that then the number. If you had a code for
your
- Original Message -
From: Irakli Natsvlishvili [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, April 04, 2005 4:52 AM
Subject: [Asterisk-Users] Manipulation based on SIP extension
Hello there,
How do I
In article [EMAIL PROTECTED],
Kamran Ahmad [EMAIL PROTECTED] wrote:
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
For a start it should be
${EXTEN}
You have to realize that ALL variables look like that.
Dollar-open-curly-brackets-variablename-close-curly-brackets.
So it didn't see your text as a variable and it tried to call the number
$EXTEN on Zap/g2.
-Original Message-
From: [EMAIL
administrator tootai wrote:
Olle E. Johansson a écrit :
administrator tootai wrote:
Nabeel Jafferali a écrit :
Does anyone else have this problem? Is there a workaround?
Yeah, I had this problem when I added a lot of SIP register statements
and SIP peers. Changing the hostnames (FQDNs) to IP
In the Asterisk system I am testing for implementation at a small
luxury resort, there are four fax machines that the guests can use for sending
and receiving faxes. Because they require confidentiality, we cannot use
hylafax or other method than a stand alone fax.
I would just connect
Dope --- *sheepish grin*. Sorry. Thanks for the help.
Kind Regards
Etienne
Technical Support
Kingsley Technologies
Rob Scott wrote:
For a start it should be
${EXTEN}
You have to realize that ALL variables look like that.
Dollar-open-curly-brackets-variablename-close-curly-brackets.
So it didn't
I am using CVS latest
Is it correct there is no jitter buffer for SIP (RTP)
Are there any plans for this?
prob a stupid question:
Is it required / do the endpoints handle this - if the
src and destination are both SIP and there is no
transcoding but asterisk is still in the media
Hello list,
Newbie questions
Seems that nated sip peers/friends are not functional with RealTime
because the database peers/users are not kept in memory.
On the other side the dynamic config (MYSQL_FRIENDS) system does not
support the nat option.
Not sure but may be ast_data is the
[...]
See this as a short time fix. We need to make a better solution on
the REGISTER parsing to prevent this from happening, it's clearly a
bug.
Well noticed. Should I concider bugs #3850 and #3933 including this
matter or should I open a new one?
We had the same problem, on two different
On Monday 04 April 2005 6:23 am, Daniele Gallina - 3P System S.r.l. wrote:
Hi all, when I try to transfer a call asterisk say me:
-- Executing SetCallerID(SIP/20012-cb87, Gallina Daniele
20012) in new stack
-- Executing Dial(SIP/20012-cb87, SIP/20013) in new stack
-- Called 20013
Darren Wiebe wrote:
That capability is not there yet. I would personally recommend
using the 'Local' channel and routing your calls via the
extensions.conf file. This is totally up to you but I find it gives
me more flexibility. That would also make it easier to do something
like you are
Sorry for the delay, do you have any clue when realtime will get added
to stable? I never did get this working but before I go too much further
I'd like to run production on a stable version..
I'll try out SIP today and let you know, the reason I'm using IAX is
because everything SIP we do is
tim panton [EMAIL PROTECTED] writes:
On 4 Apr 2005, at 09:25, Shaoul Jacobson - TELLINK wrote:
Hi,
QoS is nice (and important) but only works within a FULLY controlled
end to
end link.
Inside a BIG enterprise LAN, on leased lines its OK.
Using end to end MPLS should also be ok
Mind that
What is your problem with IAX in realtime? I have it working (finally).
Wojtek
- Original Message -
From: Matt Schulte [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, April 04, 2005 9:01 AM
Subject: RE:
Message: 9
Date: Sun, 3 Apr 2005 23:52:39 -0500
From: * KAPIL * [EMAIL PROTECTED]
Subject: [Asterisk-Users] [EMAIL PROTECTED] Question
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1
Greetings!
This is my first post to the list...and
Greetings!
This is my first post to the list...and I'm kinda' new to Asterisk, so
please be kindI did a fair amount of Googling but was not able to
find an answer.
I am using [EMAIL PROTECTED] 0.8
I was wondering if there is a way to select the outbound trunk based
on the
Well, I made several posts. Basically realtime works fine on the system
you register to, if you try to contact that peer from another Ast server
(running realtime), it does a SELECT query and all finds the peer and
continues to say Unable to contact peer as if the user doesn't exist.
I even went
On 03-Apr-2005, Tim Pushor wrote:
I prefer PF's approach to security first, convenience second, and I
*really* like the fact that PF has a real parser. As the requements get
more complex, having everything in one file, and very readable and
structured is a huge plus. Also, the integration
Hi list, I'm getting the message...
Apr 4 15:13:09 WARNING[1069]: chan_zap.c:7512 zt_pri_error: PRI:
received SETUP message for call that is not a new call, wicked!!!
This is running Asterisk 1.0.6-BRIstuffed-0.2.0-RC7k.
These messages happen when someone calls from the Telco on a BRI line...
Alexandre Charles said:
Hi everyone,
I just install Linux and asterisk on one of my pc. I
want to run some basic functionality tests. Is it possible to use a v92
modem as a FXO or FXS card. If yes how do we configure and install the
card? What are the commands?
Thanks in advance for your
I'm not sure about QoS, but I do run ATLQ on FreeBSD/PF. In a SOHO
environment where there is likely to be DSL or cable, I find it very
useful (on the upload side at least, which is usually a problem on
asyncrhonous connections).
I can max out my pipe and hear no effect of it on the phone.
Rod Bacon wrote:
The term RTCache has never been mentioned in the WIKI or these
forums. I assume that it's some sort of function to speed up realtime
db access by keeping transactions in RAM and writing periodically? If
so, I can understand why this would need to be flushed.
RealTime
Hello,
The problem is - and i was wandering if anyone knows the solution - is
that When I dial from my windows machine,
to an external phone line through Zap, then the receiving party does not
hear my voice - but when the receiving party
calls me back, then we have voice on both sides. What
do you have any clue when realtime will get added to stable?
It won't.
Not to mention since
realtime doesn't support qualify= and NAT mode must be manually set,
Have you been using RTC? (RealTime Cache) It fixes the NAT/MWI problem.
-Matthew
Hello All!
I just got my IAXy.. Configured it.. Got it Up and Running
Calls OUT have no problems (that means from IAXy - Asterisk -
ZAP/SIPclient/IAXclient)
Calls IN do have problems (that means from ZAP/SIPclient/IAXclient -
Asterisk - IAXy)
On those incoming calls on my IAXy I hear the
Laurent FOULONNEAU wrote:
Hello list,
Newbie questions
Seems that nated sip peers/friends are not functional with RealTime
because the database peers/users are not kept in memory.
*sigh* I'm quoting this wiki page:
http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime+Sip
Hi,
I have a Linux Fedora 3 Asterisk only box (2 FXO 2 FXS ports) with no
GUI or WEB server running. I can get to it remotely using Putty but I
want to add the capability to at least do Dial Plan configuration via a
browser. Do any of the GUI based configurations support such a setup. I
At 15:36 04/04/2005, you wrote:
On 03-Apr-2005, Tim Pushor wrote:
I prefer PF's approach to security first, convenience second, and I
*really* like the fact that PF has a real parser. As the requements get
more complex, having everything in one file, and very readable and
structured is a huge
Hi,
I dial a number with following setting:
exten = _X.,1,Absolutetimeout(20)exten
= _X.,2,dial(SIP/[EMAIL PROTECTED]|L(30))exten
= T,1,BackGround(tt-weasels)exten =
T,2,Hangup()
I find Absolute time out is not working , is it
normal?
kaiser
David John Walsh wrote:
I guess I should have added that this is based on the European, and
specifically UK model, but I would have expected it to have been
deemed best practice by most operators.
On Apr 4, 2005 4:04 AM, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote:
Rod Bacon wrote:
Hi,
I know this can be done but I guess I am not understanding the few notes
I have seen on this - can SIP phones be tied to Asterisk using a PC mac
address instead of their IP address (obviously I am using DHCP). If
someone could please point to the right Wiki or How to I would greatly
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
To help to find my mistake, I've two debugs:
1) isdn -- connection plar to 5600 on * -- 601 on cme -- vm
call-forward to 5601 on *
ext.num 123456789 calls my ISDN number, on ccme there's a connection
plar to internal 5600 (on asterisk), that dials
can you send me a dump from SQL for this account?
I have it working both ways,
W
- Original Message -
From: Matt Schulte [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, April 04, 2005 9:34 AM
Subject: RE:
Chris Mason (Lists) said:
In the Asterisk system I am testing for implementation at a small luxury
resort, there are four fax machines that the guests can use for sending
and receiving faxes. Because they require confidentiality, we cannot use
hylafax or other method than a stand alone fax.
I
Hi,
Is it possible to distribute services used by Asterisk onto several
boxes - similar to Pingtel (Pingtel is not an option for me since I need
to tie analog phones into the system). The main service I want to
distribute is the voice mail. I know that Mysql (have not tried
PostGreSQL yet) can
Hi,
The wiki http://www.voip-info.org/wiki-Asterisk+RealTime+Extensions shows a
very trivial sample:
INSERT INTO `extensions_table` VALUES
(1, 'mycontext', '_574555', 1, 'Wait', '2');
but how would you 'translate' an old definition as :
exten = _9.,2,Dial(SIP/${EXTEN:[EMAIL
Hi,
I know this can be done but I guess I am not understanding
the few notes
I have seen on this - can SIP phones be tied to Asterisk
using a PC mac
address instead of their IP address (obviously I am using DHCP). If
someone could please point to the right Wiki or How to I
would
On Mon, 2005-04-04 at 12:29 +0800, Dinesh Nair wrote:
On 03/23/05 04:15 Jesse Guardiani said the following:
This should be has some issues. I do not consider
the FreeBSD zaptel support to be production quality
in any way. I experienced reproducible system hangs
(mostly after an asterisk
Hello,
How does md5 authentication works?
I have created a user on my sip.conf like this:
[dov]type=friendhost=dynamicusername=dovauth=md5; echo -n "dov:myhost.com.br:dov" | md5summd5secret=a72d3b44ea28fc6515d922b21970b83c
;secret=dov
Where myhost is the real that I normally use on my SIP
do you have any clue when realtime will get added to stable?
It won't.
why not?
Not to mention since
realtime doesn't support qualify= and NAT mode must be manually set,
Have you been using RTC? (RealTime Cache) It fixes the NAT/MWI
problem.
I haven't tried this yet because of
Where myhost is the real that I normally use on my SIP phone when I don't use
md5 authentication. The echo line is the command I used to convert my
user:realm:pwd into md5.
In my X-Ten phone I just enter my username dov and password dov as plain
text.
It doesn't log in as I thought it
Hallo.
At the begining I would like to use asterisk as a VoIP server for some internal
extensions inside one building without connection to external world. I planning
to use kphone as soft phones. I tried to use configureation description that is
described in
http://asterisk.net.au/tutorial/1/
Shaoul Jacobson - TELLINK wrote:
Hi,
The wiki http://www.voip-info.org/wiki-Asterisk+RealTime+Extensions
shows a very trivial sample:
INSERT INTO `extensions_table` VALUES
(1, 'mycontext', '_574555', 1, 'Wait', '2');
but how would you 'translate' an old definition as :
exten =
Title: Just a test
Just testing our new subscription.
Ping J
Rick
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In most call centers
I have worked in, the agents had the ability to change their status from "auto
ready" or "available" into an AUX of After call state, Aftercall basically works
like wrap time, in that the agent would not receive another call in the queue
until their status was manually
Okay, after talking with Sven today, it turns out my problem
description is wrong (I was combining to cases, one of which does work
in the current firmware):
- Multiple incoming calls (works already)
- Incoming call while dialing (or waiting for answer of) outgoing
call (doesn't)
--
yip, i think that is the best approach.
--Dalon
On Apr 4, 2005 6:33 AM, Giles Coochey
[EMAIL PROTECTED] wrote:
Greetings!
This is my first post to the list...and I'm kinda' new to Asterisk, so
please be kindI did a fair amount of Googling but was not able to
find an answer.
I
Hi,
Has anyone had problems registering an SJPone software phone. I get
lots of junk mail so I have some filters running in Thunderbird and I
have not seen my registration acknowledge come through. Do they
(www.sjlabs.com) use some other domain for registration?? Any one else
had this
On Mar 31, 2005 1:24 PM, Dana Olson [EMAIL PROTECTED] wrote:
On Thu, 31 Mar 2005 10:04:34 -0500, Kanuri, Seshu (Company IT)
[EMAIL PROTECTED] wrote:
Folks!
I want to let everyone know that I have been trying to migrate from
1.0.6 to 1.0.7 last few days and I have come across serious
Any FreeBSD/OpenBSD solutions we should add to the list at the
bottom of this page?
http://www.voip-info.org/tiki-index.php?page=VOIP+Routers
Jim
James H. Thompson[EMAIL PROTECTED]
- Original Message -
From:
Arnaud
PIGNARD
To: Asterisk Users Mailing List -
On Mar 31, 2005 1:44 PM, Dana Olson [EMAIL PROTECTED] wrote:
On Thu, 31 Mar 2005 11:37:19 -0600, Rich Adamson [EMAIL PROTECTED] wrote:
My understanding is that to an extent when we buy Sangoma
we're putting the dagger to Digium.
If anything puts the dagger to Digium it'll be
Matt Schulte wrote:
do you have any clue when realtime will get added to stable?
It won't.
why not?
Now, this has been answered many, many, many times...in fact..I believe
Olle answered this in his Welcome to Asterisk post he sent out over the
weekend.
To summarize: The stable
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