Well,
Installed asterisk, libpri, zaptel,... trunk
Parameters seems ok for asterisk and ss7, linkset is ok
Problem is astersik doesn't matter about the sip messages sent to him,
Ngrep see the messages on port 5060 but astersik doesn't react...
Even sip set debug on doesn't give me any infos...
Chineese now in asterisk mailing list?
Ary Junior a crit:
Isso nao vai parar?
On 7/30/07, Josu Conti [EMAIL PROTECTED] wrote:
Yep! From So Paulo - SP
Where we can help?
Regards
Josu
2007/7/30, Ronaldo [EMAIL PROTECTED]:
Hi,
I'm brazilian. By the way, Why such
Hi,
Just forget to choose the Codec on asterisk :(
Only solution is :
Disallow=all
Allow=YourCodec
If client doesn't have that codec you will need to transcode on asterisk.
If client has that codec,asterisk will do pass-thru and it will work.
Olivier
-Message d'origine-
De : [EMAIL
Objet : RE : [Asterisk-Users] Codec Selection
Hi,
Is there any special configuration for transcoding on
asterisk? Or Asterisk will do it automatically?
---
Olivier Taylor
Sun, 05 Feb 2006 11:51:51 -0800
Hi,
Just forget to choose the Codec on asterisk
Title: Message
I have
25 licences here, u will have the possibility to Re-register once in case of
failure, even if your mac-addresses are different.
After
that, they will ask you some explanations.
Olivier
-Message d'origine-De:
[EMAIL PROTECTED]
[mailto:[EMAIL
Title: Message
Hi,
feel free to contact me off-list, we can have a test if you
want.
[EMAIL PROTECTED]
-Message d'origine-De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Mark
AdamsEnvoyé: samedi 28 janvier 2006 15:50À:
Title: Message
just a
question, where did u get xweb lite, is it still
distributed?
Olivier
-Message d'origine-De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Arjan
KroonEnvoyé: vendredi 13 janvier 2006 9:20À:
asterisk-users@lists.digium.comObjet:
Title: Message
could
be nice :)
Is it
allowed to use it?
Olivier
-Message d'origine-De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Arjan
KroonEnvoyé: vendredi 13 janvier 2006 9:55À:
Asterisk Users Mailing List - Non-Commercial DiscussionObjet:
RE:
Thanks for all,
But Asterisk is able to use g729 pass-tru and both ends have g729, then the
question is:
Why asterisk doesn't use g729 pass-thru when both ends have it?
For incoming calls from Voip, G729 is not a problem, problems appears when I
make a call to Voip...
Olivier
Ps: No need to
interested in knowing the reason too, any guidelines?
regards
On 1/10/06, Olivier Taylor [EMAIL PROTECTED] wrote:
The problem :
an asterisk box with 2 fxo
First fxo just receive calls from pstn (ulaw)
Second fxo receive and send call to mobile network thru a sipbox(ulaw)
Calls to pstn are sent
Well, another try
[general]
Disallow=all
Allow=ulaw
Allow=g729
For the Uas, they are sets to have g729 first
Calls to/from pstn needs g729
Calls to/from zap needs Ulaw
ALL incoming calls works OK even if the caller is G729(I have made a caller
using g729 only)...
Calling zap = no problem, Ulaw
Give me your providers and I give you the agi script to do that :)
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Guillermo
Salas M
Envoyé : mercredi 11 janvier 2006 17:17
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet :
Title: Message
The problem
:
an asterisk box
with 2 fxo
First fxo just
receive calls from pstn (ulaw)
Second fxo receive
and send call to mobile network thru a sipbox(ulaw)
Calls to pstn are
sent to a pstn provider accepting only g729
Internal calls
doesn't care of
Hi all,
We are located in Belgium and using an asterisk as internal Pbx.
We have many problems with Zap lines, in fact, very often, Zap doesn't
release the line after a call or an unanswered call.
Any idea is welcome,
Olivier
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Title: Message
hello
all,
where
can I find a list of agi variables that can be read by a external
script?
Thanks,
Olivier
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To UNSUBSCRIBE or update options
Totaro
Envoyé : samedi 10 décembre 2005 18:52
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : RE: [Asterisk-Users] agi variables list
Is this what you mean?
http://www.voip-info.org/tiki-index.php?page=Asterisk+AGI
-Original Message-
From: Olivier Taylor [mailto
Title: Message
just
press Ctrl-C or type exit
You
will kill asterisk, of course...
Start
asterisk by typing asterisk
and
then go toCLI by typing asterisk -r
then,
when u will quit, asterisk will not be killed
U will
be then in CLI mode
have
fun
-Message
tavail, tu n'as qu'a payé pour ce que tu
demandes.
IL me semble même me souvenir avoir lu un développeur
te faire la remarque les utilisateurs de nos projets
vous ne profitez que de notre travail !.
Pour répondre à ton problème configure logger.conf .
Harry
--- Olivier Taylor [EMAIL PROTECTED
: RE: [Asterisk-Users] Asterisk doesn't start
Hello,
You built asterisk on freebsd ?
Harry
--- Olivier Taylor [EMAIL PROTECTED] a écrit
:
Hello
Whan starting astersik(1.2) (asterisk -vvc), I
get this message :
[res_config_mysql.so] = (MySQL RealTime
Configuration Driver)
/libexec
Hello,
I have compiled asterisk cvs under freebsd, no problems.
When starting asterisk, I get :
[res_config_mysql.so] = (MySQL RealTime Configuration Driver)
/libexec/ld-elf.so.1: /usr/lib/asterisk/modules/res_config_mysql.so:
Undefined symbol ast_config_load
What's wrong?
Olivier
] What does it mean?
Hello,
Read the Makefile in apps.
Harry
--- Olivier Taylor [EMAIL PROTECTED] a écrit
:
Hello,
I have compiled asterisk cvs under freebsd, no
problems.
When starting asterisk, I get :
[res_config_mysql.so] = (MySQL RealTime
Configuration Driver)
/libexec/ld-elf.so
: [Asterisk-Users] What does it mean?
Je ne connais pas la signification de sybillines.
Harry
--- Olivier Taylor [EMAIL PROTECTED] a écrit
:
Tes réponses sont aussi sybillines que tes questions
:)
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De
la part
harry gaillac
Envoyé : jeudi 24 novembre 2005 17:54
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : RE: RE : [Asterisk-Users] What does it mean?
Je ne connais pas la signification de sybillines.
Harry
--- Olivier Taylor [EMAIL PROTECTED] a écrit
:
Tes réponses sont aussi
.
IL me semble même me souvenir avoir lu un développeur
te faire la remarque les utilisateurs de nos projets
vous ne profitez que de notre travail !.
Pour répondre à ton problème configure logger.conf .
Harry
--- Olivier Taylor [EMAIL PROTECTED] a écrit
:
Cela veut simplement dire que tu te
Hello
Whan starting astersik(1.2) (asterisk -vvc), I get this message :
[res_config_mysql.so] = (MySQL RealTime Configuration Driver)
/libexec/ld-elf.so.1: /usr/lib/asterisk/modules/res_config_mysql.so:
Undefined s
ymbol ast_config_load
What did I forgot to do?
Olivier
Just one thing,
Register the Uas to asterisk also as outbound proxy.
Asterisk will register to SER all the Uas.
We use this design:
Ua --Asterisk(NAT)-- Ser(public Ip)-- where do you want to go
It works perfectly.
Maybe I miss something?
Olivier
-Message d'origine-
De : [EMAIL
Salut Harry, plus de nouvelles de toi :(
Serais tu faché?
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de harry gaillac
Envoyé : lundi 21 novembre 2005 13:34
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] Can not build zaptel with
d'entreprise plus l'IM et presence via SIMPLE
qu'asterisk ne propose pas !
Mon problème est le champ contact dans le Sip HF avec
des clients natés
Une idée ?
Harry
--- Olivier Taylor [EMAIL PROTECTED] a écrit
:
Salut Harry, plus de nouvelles de toi :(
Serais tu faché?
Olivier
-Message
Hi asterisk lovers,
Does anyone know a good trancoder to produce g729 files from gsm or wav.
Regards,
Olivier
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Asterisk-Users@lists.digium.com
Hi all,
We use asterisk as a local pbx and we connect to a pstn/sip provider for
calls to pstn.
Since the messages on asterisk are on gsm format, we need gsm, but to call
pstn, we need g729 or g723.
How can we enable both codecs to be able to call pstn and hearing voicemail
messages for
: [Asterisk-Users] codecs
i think gsm you mention is gsm sound files not gsm codecs.
On 11/9/05, Olivier Taylor [EMAIL PROTECTED] wrote:
Hi all,
We use asterisk as a local pbx and we connect to a pstn/sip provider
for calls to pstn.
Since the messages on asterisk are on gsm format, we need
User-agents have g729, g723.1 and gsm, isn't it possible to force user-agent
to use gsm for voicemail and g729 for outbound calls?
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Tzafrir Cohen
Envoyé : mercredi 9 novembre 2005 12:35
À :
Users Mailing List - Non-Commercial Discussion
Objet : Re: RE : [Asterisk-Users] codecs
You simply need to have g729/g723 codecs. Asterisk comes with gsm by
default.
Regards,
Sahil Gupta
VoiceValley
On Wed, 9 Nov 2005, Olivier Taylor wrote:
Right,
I must suppose I need gsm codec to hear
Salut Harry,
Tu quittes Ser pour asterisk?
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de harry gaillac
Envoyé : mercredi 9 novembre 2005 13:22
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users]
That's a call to pstn
Callee and caller have 9729 but asterisk (astlinux and soekris) tell me that
there is no match and give me an error :(
Any idea?
Kind regards,
Olivier
9 headers, 11 lines
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 82.146.123.246:38098
found that happens when one version of asterisk is 1.2 and the
other end is running 1.0.9 and you are connecting over IAX2.
If you bridge the two servers with SIP it will be fine.
-bill
On 9-Nov-05, at 11:52 AM, Olivier Taylor wrote:
That's a call to pstn
Callee and caller have 9729
Hello all,
We'd like to use asteriek as an internal pbx connected to an external sip
provider to make outbound/inbound calls to pstn.
We have the provider and have installed an asterisk at the office.
Does anyone have a sample config?
We need 25 telephone numbers(dids), to be registerd to the
-Commercial Discussion
Objet : Re: [Asterisk-Users] Asterisk as an internal pbs for a samall
company
On Wed, 2005-11-02 at 13:36 +0100, Olivier Taylor wrote:
Hello all,
We'd like to use asteriek as an internal pbx connected to an external
sip provider to make outbound/inbound calls to pstn. We have
: RE : [Asterisk-Users] Asterisk as an internal pbs for a
samallcompany
On Wed, 2005-11-02 at 13:58 +0100, Olivier Taylor wrote:
Well,
U right, many missing informations.
The case is quite simple(I guess), we have dids, and each call to
these dids has to be routed to the right handset thru
Very funny discussion :)
Same thing arrives with a lot of Gpl softwares,
A few months ago, it was another Voip software wich forked (Sip Express
Router aka Ser), there were pro and cons, now the discussion is finished and
they all cooperate.
Olivier
-Message d'origine-
De : [EMAIL
Just press Ctrl-Alt-Del
Usual on windows ;)
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Matt
Envoyé : mercredi 28 septembre 2005 15:22
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] Asterisk on
The limitation is that it doesn't work on freebsd, probably due to
libiodbc...
That's a limitation, isn't it?
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Matthew Boehm
Envoyé : lundi 19 septembre 2005 16:42
À : Asterisk Users
Objet : Re:
1
Any help is welcome
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Urban
Envoyé : samedi 17 septembre 2005 19:28
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] Asterisk realtime beta
Olivier Taylor
Ok I use Libiobc instead of unixodbc.
But Now asterisk doesn't find the database...
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Olivier
Taylor
Envoyé : dimanche 18 septembre 2005 12:58
À : 'Asterisk Users Mailing List - Non-Commercial
Does anybody has intalled it on freebsd with unixodbc or libiodbc and have
it working?
Regards,
Olivier
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