Is there a way to force asterisk to take care only of sip signaling without
forcing it to take care of rtp traffic?
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Probably I did not read well the information
I am concerning, if I am going to use ARA for the SIP
and I have
register = user:secret:[EMAIL PROTECTED]:port/extension
how I should input that line?
If I am going to delete it from the DB I am forced to reload everything or
there is a way to tell
Hi
after having installed asterisk 1.4.15 my voicemail does not work anymore.
Am I the only one?
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Probably I was not able to explain myself properly
however, for some measge this what happen
-- Local/[EMAIL PROTECTED],2 Playing
'/var/spool/asterisk/voicemail/default/300/Old/msg0003' (language
'it')
== Spawn extension (servizi, , 1) exited non-zero on
'Local/[EMAIL PROTECTED],2'
I
for
the message playing)
- Asterisk hangups.
I'm not always able to replicate the problem but, as Il Neofita, I'm using
the italian prompts... could be a problem related to that?
Bye and regards
Marco Signorini.
Il Neofita wrote:
-- Local/[EMAIL PROTECTED],2 Playing
'/var/spool
Hi,
with some messages the voicemailmain after give me the information
about the call (Days, hours and minutes) it finish.
Whant can I check for solve this problem?
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Hi
additional information if I am going to wait at least 3 seconds after
the voicemail starts to give me the instruction I am able to listen my
messages.
But why I need to wait?
On Nov 12, 2007 2:28 PM, Il Neofita [EMAIL PROTECTED] wrote:
Hi,
with some messages the voicemailmain after give me
-0500 schrieb Il Neofita:
Hi
additional information if I am going to wait at least 3 seconds after
the voicemail starts to give me the instruction I am able to listen my
messages.
But why I need to wait?
On Nov 12, 2007 2:28 PM, Il Neofita [EMAIL PROTECTED] wrote:
Hi,
with some
Hi
I believe that
exten = s,7,GotoIf($[${trips}=4]?,8)
the , should be :
On 10/24/07, Phil Knighton [EMAIL PROTECTED] wrote:
Hi
I have a situation where I want to be able to count how many times a
caller goes round a loop of Please hold..., please continue to hold. I
have found an
There is a way to force the order of the codecs in the sip.conf since the
allow seams to let know only the accepted codec.
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Hi,
I update from asterisk 1.2 to 1.4 and I have some problems.
In the extensions I used DIAL(SIP/100SIP/101,30,tTr) if I receive a call
from an external providers
now in 1.4 I recieve only one ring
What can I do to solve this problem?
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Hi,
I installed gtalk on asterisk 1.4.12.1, I change on rtp.conf the port from
1000 to 4
If I calling asterisk with GTALK in english everything is ok, however, some
of my friends with the italian version of gtalk they cannot have the audio.
Is it a bug? Or I did some mistake
Thank you
I need to wait the international version of gtalk
On 10/11/07, Philippe Sultan [EMAIL PROTECTED] wrote:
If I calling asterisk with GTALK in english everything is ok, however,
some
of my friends with the italian version of gtalk they cannot have the
audio.
Audio problems might
Hi
is there a tool to know what was the maximum calls that asterisk managed?
Thank you
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Hi,
I have a client (xlite) connected to my server, on the server I have
type=friend and siptimeout=60, canreinvite=yes and dial with tT option, the
server is listening on port 5060.
However, xlite is connect to a router where the port 5060 is blocked,
therefore, I am using 5065 and I have an
Thank you I will try tonight
On 9/10/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote:
Am Montag, den 10.09.2007, 05:14 +0200 schrieb Il Neofita:
On 9/9/07, Anselm Martin Hoffmeister [EMAIL PROTECTED]
wrote:
Am Sonntag, den 09.09.2007, 20:16 +0200 schrieb Il Neofita
Hi,
my ATA has two phones attached and the possibility to set different
accounts.
I put two account of my asterisk server, however, it is able to call only
with the second one in order to the sip.conf and the first it gives me 403.
And idea how to solve it?
On 9/9/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote:
Am Sonntag, den 09.09.2007, 20:16 +0200 schrieb Il Neofita:
Well, it seems there are differences between those accounts then.
You might want to post your sip.conf, and -if that is possible- the ATA
conf file; or at least
Hi,
I am using my asterisk server like a gateway and one provider ask me to pass
an extra field with the IP of the peer that is using the connection,
probably to have more control on the authentication. I was wondering how I
can implement this.
Thank you
Hi
when I turn on my PC I able to load the drivers and start my card,
if I reboot the PC I have the following error
ztcfg -vvv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel
PROTECTED] wrote:
On Thu, Mar 01, 2007 at 06:21:38AM -0500, Il Neofita wrote:
Hi
when I turn on my PC I able to load the drivers and start my card,
if I reboot the PC I have the following error
ztcfg -vvv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart
Hi,
today with my asterisk during a call I had a very strange noise, it was the
typical noise that you can have when your device uses a bad power supply.
I change phone and I had the same behavior after I while I tried again and
the noise was disappeared.
What can I check?
Yes, but I would like to try a number and after to try a second one.
Any Idea how to avoid this.
On 2/18/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
C F wrote:
Asterisk supports this directly by issuing the hangup command before
the answer command. However, when using an analog
On 2/14/07, Yuan LIU [EMAIL PROTECTED] wrote:
Well, you'll have to decide how you want to hang up the caller: Do you
want him/her to be ignored, or to be told that you are not available (like
an answering machine)? You also need to tell Asterisk how to determine
if
the next invite comes from
Ok thank you a lot!!!
On 2/15/07, Yuan LIU [EMAIL PROTECTED] wrote:
From: Il Neofita [EMAIL PROTECTED]
Date: Thu, 15 Feb 2007 03:37:14 -0500
But I tought that hangup was suppose to close the call, however, is not
the
case and a really did not catch why.
Now I see where the confusion comes
I have this simple context
I am register to an external provider and when I am not home I would like to
transfer the phone outside
The problem that the call goes in loop
I cannot understand why.
Can you figure out my error?
Thank you
sip.conf
register = user:[EMAIL PROTECTED]/400
[inside]
.
On 2/14/07, Yuan LIU [EMAIL PROTECTED] wrote:
From: Il Neofita [EMAIL PROTECTED]
Date: Wed, 14 Feb 2007 19:30:51 -0500
I have this simple context
I am register to an external provider and when I am not home I would like
to transfer the phone outside
The problem that the call goes in loop
I
Sei riuscito?
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Hi,
I tried thousands of time and finally I am a step closer to the solution.
I recompile iksemel with the option --prefix=/usr
I erase my zaptel-1.4, asterisk-1.4 and asterisk-addons-1.4, re-extracting
everything from the tar
recompile everything and now jabber is working or almost. When I
On 2/10/07, Patrick [EMAIL PROTECTED] wrote:
Where can I find that patch?
Thanks,
Patrick
Hi Patrick,
I downloaded the patch from here
http://bugs.digium.com/view.php?id=7764
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Hi,
I download the last svn and I also look around but I cannot find the source,
I only found the patch
http://bugs.digium.com/print_bug_page.php?bug_id=8919
any one can help me out.
thx
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The point is to use more than one port, I think the only way is to use the
redirect from iptables
On 2/6/07, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Ciao,
just change port value in sip.conf.
Giorgio
Il Neofita wrote:
Hi,
I was wondering if it is possible to set asterisk in order
I start the patch and automatically created the file. But now on the menu I
cannot select chan_cellphone
I launched ./bootstrap.sh
and after ./configure
in my /usr/include/bluetooth I have the header
but I cannot select the option
any idea?
On 2/9/07, Il Neofita [EMAIL PROTECTED] wrote:
Hi,
I
JABBER: gtalk_account OUTGOING: ?xml version='1.0'?stream:stream
xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to='
gmail.com' version='1.0'
localhost*CLI jabber show tes
JABBER: gtalk_account INCOMING: ?xml version=1.0
encoding=UTF-8?stream:stream from=gmail.com
Hi,
I was wondering if it is possible to set asterisk in order to listen to
different ports for the sip or I need to do this operation with iptables?
All of this since some time the port 5060 is blocked.
Thank you
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Hi,
I was wondering if someone had problems with chanskype.
Since I am wondering if they are a credible company or not.
See you
On 1/20/07, Moises Silva [EMAIL PROTECTED] wrote:
Hi,
I tried the try version of chanskype, however, everytime that I make a
call
asterisk generate an error
So
Hi,
I tried the try version of chanskype, however, everytime that I make a call
asterisk generate an error
Anyone has experience with this? Since I tried to contact the support but
they never replied.
Thank you
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Someone know why my asterisk gives me the following msgs?
Thank you
- Got SIP response 603 Declined (no dialog) back from
X.X.X.Xhttp://82.51.224.34/
-- Got SIP response 603 Declined (no dialog) back from
X.X.X.Yhttp://82.51.224.34/http://82.104.4.192/http://82.104.4.192/
-- Got SIP
I do not think that there are some company that offer a toll free number
(Numero verde in italian)
But contact on of these three providers
http://www.eutelia.it/tlc/
http://www.unidata.it/
http://messagenet.it/
If they have one of these should be able to give to you
See you
On 1/8/07, CM
On 7/3/06, Jean-Denis Girard [EMAIL PROTECTED] wrote:
But I'm not sure that MozPhone is what the original poster asked.No, however, I like to read all these different point of view.
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Hi,someone know a good webphone, possibily a free oneThx
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on.
pFrom: Il Neofita
[EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.comDate: Sun, 18 Jun 2006 05:01:20 -0400Subject: Re: [Asterisk-Users] Canreinvite This is the dial in extensionsexten = _40001,1,Dial(SIP/40001,30)
exten =
_40002,1,Dial(SIP
This is the dial in extensionsexten = _40001,1,Dial(SIP/40001,30) exten = _40002,1,Dial(SIP/40002,30)
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cosa vedo a console -- Executing Dial(SIP/40001-3760, SIP/40002|30) in new stack -- Called 40002 -- SIP/40002-4753 is ringing -- SIP/40002-4753 answered SIP/40001-3760
-- Attempting native bridge of SIP/40001-3760 and SIP/40002-4753srvlinux*CLI sip show channelsPeer User/ANR Call ID Seq (Tx/Rx)
I put canreinvite=yes in my sip, for a sipura 3000 and a xlite, however, if I call the traffic still go throw the asterisk. How come?
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Hi,when I am going to compile the zaptel I receive this messageake -C /lib/modules/2.6.12-12mdk/build SUBDIRS=/usr/src/asterisk/zaptel-1.2.5 XPPMOD= modulesmake[1]: Entering directory `/usr/src/linux-2.6.12-12mdk
' WARNING: Symbol version dump /usr/src/linux-2.6.12-12mdk/Module.symvers is missing;
Hi,one of my WiFI phone has problem with the notify asterisk signal to me the following Apr 29 06:49:16 WARNING[6455] chan_sip.c: Host '192.168.100.124' does not implement 'NOTIFY'
In theory the phone support this function.Any idea?
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I agree with you, but I would like to find a way to use the notification. I tough that there was a work around.On 4/29/06, tom
[EMAIL PROTECTED] wrote:Il Neofita wrote: Hi, one of my WiFI phone has problem with the notify asterisk signal to me
the following Apr 29 06:49:16 WARNING[6455]
Hi,I try to use my phone that has a SMS capability with asterisk.I am not able to receive SMS, someone can help me out?This what I am able to have but nothing more-- Executing SMS(SIP/503-7d2e, 508|sa) in new stack
-- SMS TX 93 00 6D
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Hi,I am pretty sure that you already answer to this question, but I was not able to find the solutionon the console I have over and over the following msgs-- Remote UNIX connection disconnected -- Remote UNIX connection disconnected
-- Remote UNIX connection disconnected -- Remote UNIX connection
Hi,Has anyone proved the chan_gsm_bt ??Any impression?
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Someone know the meaing of this error?chan_sip.c:3853 copy_via_headers: No header field 'Via' present to copy
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This is my debugtried with netmeeting I can still hear but when I talk nothing happenmygw*CLI h.323 debugH323 debug enabledmygw== New H.323 Connection created. -- Setting up Call -- oCall token: [ip$192.168.1.219:1057/8226]
-- oCalling party name: [myPersonal] -- oCalling party number: [] --
I have the license for G729, however I need to use a different codec for the prepaid service, but when the call is started I have this errorAsked to transmit frame type 256, while native formats is 4 (read/write = 4/4)
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I never so this error.I am using H323 with Asterisk 1.2.6 Any idea what can be the problem?
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Hi,I installed H323, however when I make a call from SIP Phone - Asterisk H323 - Provider H323 the provider can hear me, but I cannot hear nothing.The asterisk is 1.2.6 with G729 license, and the asterisk is connect direct to internet with a public IP.
Any thoughts?
There is a proble to put an H323 Asterisk server behind an iptables firewall?
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Hi,I compiled for my asterisk 1.2.4 the openh323but when I give this commandh.323 show codecsI receive thisAllowed Codecs: Table:Set:I cannot test with msn if everything is working since I am outside and I cannot access to the firewall.
Someone can tell me if I need to install the oh323 since I do
Hi,I am wondering if it is possible with astcc to make a second call without hangup and be oblige to re-enter all the codes.Any idea how to do?Thank you
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Hi,I have a TDM400P with 4 FXO The TDM after it receive a call do not hangup properly, it takes the line occupied.
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Yes, I changed.Thank youOn 3/27/06, Infobox Peru [EMAIL PROTECTED] wrote:
Maybe your lines use polarity reversals for hangup detection.On 3/27/06,
Il Neofita [EMAIL PROTECTED]
wrote:
Hi,I have a TDM400P with 4 FXO The TDM after it receive a call do not hangup properly, it takes the line occupied
Sorry if I am always here asking for MWI, but I do not know how to
solve this issue, I have my ATAs (Azatel 200 and Fritz!Box) that they
think that I have a message waiting.
Anyone knows how to solve this issue?
Thank you
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Hi,
I was not able to find any indication for this problem that I have right now.
My phones connected to an Azatel 200 they always indicate that I have a message waiting to be listen.
However, I do not have any message.
I also checked using the console show voicemail user for context but I have 0
I am not able to register an external ATA on my asterisk 2.0 Beta
This is the debug
Any idea?
server01*CLI
-- SIP read from CLIENTIP:5060:
REGISTER sip:SIPSERVERIP SIP/2.0
Via: SIP/2.0/UDP PRIVATEIP;rport;branch=z9hG4bK455E4AEA9A9954FB135D6D788DA2
From: sip:[EMAIL PROTECTED];tag=1564789518
To:
I installed astersik 1.2beta and from that point the led that indicate a new call flash.
The ATA installed is an AZATEL.
Any idea what can I check?
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Hi,
do you know what it means the following:
Call failed to go through, reason 3
I received it when I try to send a FAX and no one answer it.
Thank you
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HI Chris
I am interested, I would like to know how I can have the opportunity to test your program.
On 9/9/05, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Sep 08, 2005 at 06:42:49PM -0500, Chris Shipman wrote: I started working on a program using Ghostscript and Redmon to generate the tif in
This is the script I do not remeber the name of the author
For me is working
On 9/9/05, Chris Shipman [EMAIL PROTECTED] wrote:
Thanks Luki,Maybe it was the wrong place to mention it.Back to the other matter: The program will create a PDF or tif. Itcan submit the fax by FTP or by AstFax.If someone
Hi,
I would like to use the * when I am in the asnwer machine, but I received a message asking for the temporary pass code.
Where I need to put this pass?
I am using asterisk 1.2.0 beta 1
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Is theresome way to know if the fax was received correctly or not?
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To
The site is in italian, and you need to register your self in order to download the script.
The script is in perl and you need to start in order to simulate the hylafax daemon after that you can use WHFC
On 9/8/05, Matthew Gibson [EMAIL PROTECTED] wrote:
Sorry to interrupt :)But I believe what you
Hi,
I found on a forum a script that emulate a hylafax this is the linkhttp://www.vocesuip.com/viewtopic.php?p=2423
You can use the WHFC in order to send a fax to asterisk.
On 9/5/05, Harald Klein [EMAIL PROTECTED] wrote:
Hi Chris, Hi Arne,Am 5.9.2005 schrieb Chris Shipman [EMAIL
>From my point of view I do not see any issue with that scenario.
On 9/5/05, housi mueller [EMAIL PROTECTED] wrote:
I am new with asterisk and hope somebody can help me.
Is a configuration like shown on the picture with asterisk
correct? Some phone calls arriving in Branch 1 should be
redirected
The call generate from branch2 can be send to the asterisk in Branch1
with a trunk the same think the call received from branch1 the only
thing that is not cleat how you want transfer automatically the
call received from the pstn. What rule you want use?On 9/5/05, housi mueller [EMAIL PROTECTED]
Hi,
I put on sip.conf the following line
#include sip.d/*.conf
inside I have files like that
provider1.conf
provider2.conf
But asterisk does not want to load it
This is the error
Sep 1 13:18:35 VERBOSE[8756]: == Parsing
'/etc/asterisk/sip.d/*.conf': Sep 1 13:18:35
VERBOSE[8756]: == Parsing
Sorry, I'm using the 1.0.9On 9/1/05, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Il Neofita wrote: Hi, I put on sip.conf the following line #include sip.d/*.confYou neglected to include the most important piece of information: whatversion of Asterisk you are using
Hi,
I'm using a script in order to send out my faxes with the application
txfax, therefore, I do not know how to see if the faxes are sent.
Any idea?
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Thank you was that the problem.
On 8/25/05, Darren Wiebe [EMAIL PROTECTED] wrote:
When did you install it? Try running the update database function
from the configure menu.
Darren Wiebe
[EMAIL PROTECTED]
Il Neofita wrote:
My installation of ASTCC does not update the cdrs tables
I am looking for a calling card application which is able to advise me
during a call when the credit is almost finish. For examples 1 minute
before the end of the credit.
Thank you
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My installation of ASTCC does not update the cdrs tables .
It is a problem of ASTCC or it is a configuration problem?
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Why my X100P detect the ring after 3 o 4 seconds?
The funny thing that when I have an incoming call asterisk receive a
signal but the commands start after 3 or 4 seconds. Moreover, when the
call end the hungup has the same delay.
any ideas?
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Hi,
I installed this program but I am not able to configure, it does not
want to work.
Someone can help me?
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Hi,
I checked but I did not find any info regarding the config files. I
tough that I configured everything in the right way but I am not able
to see anything on the web page.
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