, BERGANZ François
franc...@acropolistelecom.net wrote:
Hello,
I need to send a tiff via fax with my asterisk 1.6.1.0.
I tried in the dialplan
[default]
exten = _X.,1,SendFax(/root/test.tiff)
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Hello,
I need to send a tiff via fax with my asterisk 1.6.1.0.
I tried in the dialplan
[default]
exten = _X.,1,SendFax(/root/test.tiff)
but I have:
salledeconf1*CLI console dial 1...@default
[Dec 23 16:24:22] WARNING[31739]: chan_oss.c:492 setformat: Unable to
re-open DSP
was able to send.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of BERGANZ
François
Sent: Wednesday, December 23, 2009 10:50 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] fax problem
Hello,
I need to send
Try tcpdump to see where RTP go from asterisk.
Configure your x-lite
Use stun server ?
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De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de zehra yildiz
Envoyé :
It is a nat problem
François BERGANZ
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De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de zehra yildiz
Envoyé : mardi 22 décembre 2009 10:26
À :
Hello all,
I try to do a web interface for my conferences.
Is it possible to export in realtime all user in conferences as we could do
with a CLImeetme list all ?
François BERGANZ
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Hello,
I have conferences in my database.
I need at some moments, to access the database without asking pin access, or
with using cdr(accountcode).
Is it possible?
Thank you
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire
Sorry, there are some errors, here the right question:
Hello,
I have conferences in my database.
I need at some moments, to access the CONFEERENCE without asking pin access,
or with using cdr(accountcode).
Is it possible?
Thank you
Cordialement,
BERGANZ François
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de BERGANZ
François
Envoyé : jeudi 3 septembre 2009 11:42
À : 'Asterisk Users Mailing List - Non-Commercial Discussion'
Objet : Re: [asterisk-users] MeetMe unactive pin access
Sorry, there are some errors, here the right question
Hello,
How can Set(language()) in asterisk-1.6.x ?
Cordialement,
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AstriCon 2009 - October 13
Sorry, I just found the solution
Set(CHANNEL(language)=en)
Cordialement,
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De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de BERGANZ
François
Envoyé : mardi 1
Use the pattern matching P137 in Asterisk: the future of telephony
For example
Exten = 919X,n,
Cordialement,
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De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De
Use the pattern matching P137 in Asterisk: the future of telephony
For example
Exten = _919X,n,
Cordialement,
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
Kevin P. Fleming
Envoyé : vendredi 21 août 2009 17:11
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Asterisk 1.6 t 38 passthrough
BERGANZ François wrote:
I have that problem:
[Aug 21 15:57:35] WARNING[5198]: chan_sip.c:6737 get_ip_and_port_from_sdp
Zoa,
I could see that you could do T38 passthrough with asterisk ans Zoiper.
I have some problems to do it, can you help me?
Cordialement,
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,
BERGANZ François
cid:image001.gif@01C8F7CD.6BC1D2C0
http://www.acropolistelecom.net/ http://www.acropolistelecom.net
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De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de
The authentication is in the SER and asterisk trust the ser (insecure=invite)
Just do some iptables to be sure to don't receive sip from others...
Cordialement,
Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
-Message d'origine-
De :
In the sip.conf, just insert the host=
Cordialement,
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-Message d'origine-
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Ishfaq Malik
Envoyé : mardi 25 août
Try us...@lists.kamailio.org
http://www.acropolistelecom.net
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De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Ishfaq Malik
Envoyé : mardi
SER can't generate REGISTER
You can play to control if it is yours with adding headers if you want, we can
imagine that...
Cordialement,
Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
-Message d'origine-
De : asterisk-users-boun...@lists.digium.com
asterisk-addons-1.6.2.0-rc1.tar.gz
#cd asterisk-addons-1.6.2.0-rc1
#./configure make menuselect make make install make samples
thank you for help
Cordialement,
BERGANZ François
cid:image001.gif@01C8F7CD.6BC1D2C0
http://www.acropolistelecom.net/ http
Hello,
I need help to install a digium card TC400P.
I compiled the dahdi source, but dahdi dont find the card!
debian:/usr/src/dahdi-linux-complete-2.2.0.2+2.2.0# lspci
11:03.0 Ethernet controller: Digium, Inc. Wildcard TC400P transcoder base
card (rev 11)
Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] install the digium card TC400P howto?
BERGANZ François wrote:
debian:/usr/src/dahdi-linux-complete-2.2.0.2+2.2.0# /etc/init.d/dahdi
restart
Unloading DAHDI hardware modules: done
Loading DAHDI hardware modules
If I do #dmesg, I have it :
[ 101.994189] Unregistered codec translator 'DTE Decoder' with 92
transcoders (srcs=0101, dsts=000c)
[ 101.994189] Unregistered codec translator 'DTE Encoder' with 92
transcoders (srcs=000c, dsts=0101)
[ 101.999162] dahdi_transcode:
Hello,
How can I do t-38 passthrough with asterisk 1.6 ?
I know how to do with 1.4 but not with 1.6
Thank you
Cordialement,
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-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Kevin P. Fleming
Envoyé : vendredi 21 août 2009 15:05
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Asterisk 1.6 t 38 passthrough
BERGANZ François wrote:
How
media source on glared reinvite. Audio may not work
properly on this call.
Why?
It is at the second invite to do T38
Cordialement,
BERGANZ François
http://www.acropolistelecom.net
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-Message d'origine-
De : asterisk-users
hello
is it possible to lock a conference IF no admin are connected ?
or how to do to have a conference offline?
thank you
Cordialement,
BERGANZ François
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Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http
Hello,
In my dialplan, I do s,n,DIAL(
)
If my called phone response and after hangup, asterisk execute the h,1,
But, if I the caller hangup at ringing (cancel), it dont execute the h,1,
Know you why?
Thank you
Cordialement,
BERGANZ François
P Pensez à
I found a bug in asterisk 1.6:
http://lists.digium.com/pipermail/asterisk-dev/2009-April/037684.html
in fact, the h,1 in the macro don’t work with cancel!
Cordialement,
BERGANZ François
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-Message d'origine-
De : asterisk
/share/perl/5.8
/usr/local/lib/site_perl .) at Makefile.PL line 1.
BEGIN failed--compilation aborted at Makefile.PL line 1.
Why? What have I to do?
Thank you
Cordialement,
BERGANZ François
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Cordialement,
BERGANZ François
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asterisk-users mailing list
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http
Thank you!
I did understood that i twas THAT timeout :-)
I thought that it speak about my 'limit call'
Thank you
Cordialement,
BERGANZ François
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-Message d'origine-
De : asterisk-users-boun...@lists.digium.com
Dont work
I need that it suppr the
Thank you
Cordialement,
BERGANZ François
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Danny
Nicholas
Envoyé : mercredi 20 mai 2009 17:53
À : 'Asterisk Users Mailing List - Non
Hello,
I have a var like blabla with the
I need to suppr the
Is it possible with the ${var:x:y} ?
Thank you
Cordialement,
BERGANZ François
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I found !
exten = _X.,n,Set(var2=${CUT(var,',2)})
Cordialement,
BERGANZ François
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De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de BERGANZ
François
Envoyé
Hello,
I work on voicemail.conf and I need that ${VM_DATE} is in french!
How can I do it?
Thank you
Cordialement,
BERGANZ François
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(mnupprx1, 0170725000, 5) exited non-zero on
'SIP/33179977999-b6c18478'
I dont know why it dont exec it !
Cordialement,
BERGANZ François
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I have the same thing with AGI in the dialplan
And php is install
Cordialement,
BERGANZ François
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De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de
I have all permissioned
Cordialement,
BERGANZ François
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De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Danny
Nicholas
Envoyé : lundi 9 mars 2009 15:07
À
I am sorry it work !
In fact, I had mistakes in my config
Sorry
And thank you for answering
Cordialement,
BERGANZ François
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De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun
hello,
I will do a server to do a lots of conferences (MeetMe).
I want to know that if I dont use a digum card, the limit of simultaneous
calls is harder without a card than with a card ?if, yes, how harder is the
limit?
thank you
Cordialement,
BERGANZ François
P Pensez à
In the sip.conf
[2001]
...
Canreinvite=yes
[2002]
...
Canreinvite=no
Cordialement,
BERGANZ François
http://www.acropolistelecom.net
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-Message d'origine-
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk
Hello,
I am looking for a good prepaid solution.
What is the best ?
Cordialement,
BERGANZ François
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Happy birthday asterisk!
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de SIP
Envoyé : vendredi 5 décembre 2008 06:14
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Friday, Asterisk is 9 years old!
randulo
Hello,
I think, that I find a bug for a specific asterisk use.
My network test is:
Client 1
--asterisk0(client)---asterisk1(ss7)--Asterisk2(media)
Client 2
I need that a call from Client1 go through
asterisk0-asterisk1-asterisk2-asterisk1-asterisk0-Client2
When I
message at :23, :28, and :35 mins past the
hour? If you need immediate support you should contact Digium support
and pay for a service contract.
BERGANZ François wrote:
I need to test canreinvite=yes with 2softphones and 1 asterisk.
I want to have that :
http://www.panoramisk.com/wp-content
, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0
Have you an idea why ?
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de BERGANZ
François
Envoyé : jeudi 4 décembre
Hello,
I need help for that error message:
chan_sip.c:14889 handle_response_invite: Failed to authenticate on INVITE
to
My network is:
Client1--
---asterisk1--asterisk2
Client2--
· With client1, I do a call
· Asterisk1 forward the call to
Hello,
My network is:
Client_SS7_1--
---asterisk1--asterisk2
Client_SS7_2--
· I receive a fax from Client_SS7_1
· Asterisk1 forward the call to asterisk2
· Asterisk2 forward the call to asterisk1
· Then, asterisk2 forward the fax to
Hello,
I need to test canreinvite=yes with 2softphones and 1 asterisk.
I want to have that :
http://www.panoramisk.com/wp-content/uploads/2007/05/asterisk-call-flow-outb
ridge.png
But I have that http://www.zimagez.com/zimage/canreinvite.php
Canreinvite=yes work for all phones or
Hello,
I need to test canreinvite=yes with 2softphones and 1 asterisk..
I want to have that :
http://www.panoramisk.com/wp-content/uploads/2007/05/asterisk-call-flow-outb
ridge.png
But I have that http://www.zimagez.com/zimage/canreinvite.php
Canreinvite=yes work for all phones or
Someone have a solution for me ?
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de BERGANZ
François
Envoyé : mercredi 3 décembre 2008 18:24
À : asterisk-users@lists.digium.com
Objet : [asterisk-users] canreinvite=yes problem
Hello,
I need to test canreinvite=yes
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