[asterisk-users] asterisk trunk

[olivier taylor]

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...

Any idea someone of what I did wrong?

Olivier




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Re: [asterisk-users] Brazilian.

[olivier taylor]

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 a question?
See you.

Ronaldo.


Jozeph Brasil wrote:
  
  
Some brazilian here on list?



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RE : [Asterisk-Users] Codec Selection

[Olivier Taylor]

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 PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Abdul Lateef
Envoyé : dimanche 5 février 2006 20:00
À : asterisk-users@lists.digium.com
Objet : Re: [Asterisk-Users] Codec Selection



Hi,

I am using Perl AGI to dial the carrier (Gateway), i
am little experiencing how to do TRUN in Perl AGI.

this is my script how i am dialing the number to
Gateways, So before dialing the number i want to
select the codecs according to our Gateway.


my $discr = $AGI-get_variable(DIALSTATUS);
if ($discr == CONGESTION || $discr == NOANSWER ||
$discr == CHANUNAVAIL)
{
my $dialstr = $gwtype/$gwip/ . $dialednum .
|30|tTL( . ($crdeit*1000) .:7000:5000);
$AGI-exec('Dial', $dialstr);
$discr = ;
}

Any idea?




Yours,
Abdul Lateef
Computer Programmer
HATIF COM
Mob: +974 - 5405022
ICQ: 276994704
MSN: [EMAIL PROTECTED]
GoogleTalk: [EMAIL PROTECTED]
YM!: abdul_zu
Doha Qatar
http://www.hatif.com

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RE : RE : [Asterisk-Users] Codec Selection

[Olivier Taylor]

You will need  to buy licences for the codec you want, I don't know for
g723, but g729 costs 10$ by channel.

Cheers

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Abdul Lateef
Envoyé : dimanche 5 février 2006 21:50
À : asterisk-users@lists.digium.com
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 :(

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 PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Abdul Lateef
Envoyé : dimanche 5 février 2006 20:00
À : asterisk-users@lists.digium.com
Objet : Re: [Asterisk-Users] Codec Selection



Hi,

I am using Perl AGI to dial the carrier (Gateway), i
am little experiencing how to do TRUN in Perl AGI.

this is my script how i am dialing the number to
Gateways, So before dialing the number i want to
select the codecs according to our Gateway.


my $discr = $AGI-get_variable(DIALSTATUS);
if ($discr == CONGESTION || $discr == NOANSWER ||
$discr == CHANUNAVAIL)
{
my $dialstr = $gwtype/$gwip/ . $dialednum . |30|tTL( .
($crdeit*1000) .:7000:5000);
$AGI-exec('Dial', $dialstr);
$discr = ;
}

Any idea?






Yours,
Abdul Lateef
Computer Programmer
HATIF COM
Mob: +974 - 5405022
ICQ: 276994704
MSN: [EMAIL PROTECTED]
GoogleTalk: [EMAIL PROTECTED]
YM!: abdul_zu
Doha Qatar
http://www.hatif.com

__
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http://mail.yahoo.com 
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RE : [Asterisk-Users] G729 Commercial Licenses.

[Olivier Taylor]

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 PROTECTED] De la part de 
  [EMAIL PROTECTED]Envoyé: samedi 28 janvier 2006 
  14:36À: Asterisk Users Mailing List - Non-Commercial 
  DiscussionObjet: Re: [Asterisk-Users] G729 Commercial 
  Licenses.Thanks. Doug for the precise 
  clarifications..Dan
  On 28/01/06, Doug 
  Lytle [EMAIL PROTECTED] 
  wrote: 
  [EMAIL PROTECTED] wrote: 
Rob,  Thanks, well i had gone through it before but i had 
some different comments from a couple of friends on the same topic 
but let me clarify. currently i have 2 commercial licenses 
and suppose i a have backups of  the licenses and once a i do a full 
revamp and i place my .lic files back at the respective folders... 
im gonna have a sure go on the same PC? am i 
right..? Please correct me if i am wrong. 
According to the license, it's based on MAC address, as long as 
thatdoesn't change you should be all set:A G.729 key must be 
re-registered if any of the ethernet devices in your 
Asteriskserver are changed, added, or 
removed.The unique G.729 license file which 
islocated in your /var/lib/asterisk/licenses directory is 
tied to the MAC address ofall the ethernet devices installed 
in your system.Doug--Ben Franklin quote: 
"Those who would give up Essential Liberty to purchase a little 
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RE : [Asterisk-Users] Voip Provider

[Olivier Taylor]

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: 
  asterisk-users@lists.digium.comObjet: [Asterisk-Users] Voip 
  Provider
  
  Hi 
  Everyone, 
  I know 
  this may be off subject but I am not sure who to ask. I am currently looking 
  for voip termination that is closest to replicating U.S. 
  pots service. I run I.V.R. systems and I want to point Sipura 2100s to a voip 
  terminator and have the DTMF tones properly detected. All that I need is 
  outbound service and the problem I run into now is that when the called party 
  presses a key on the phone it does not play it back properly to my system. I 
  have tried to dial through voxee and plain voip and they both have the same 
  problem. Im not sure if this is an asterisk issue or what. When I dial through 
  packet 8, aptella or vonage everything works fine. I think my problems are 
  because I am going through their asterisk servers. If anyone can help I would 
  appreciate it, there is a potential for me using thousands of minutes per day 
  if I could only find compatible service. 
  I use 
  the generic term U.S. Pots service because my dialers work perfectly on normal 
  analog phone lines. Ive been looking for service for 2 months and I havent 
  had any luck.
  P.S. I 
  do not need any special services, just proper DTMF tone handling. 
  
  
  Mark 
  AdamsInfinity Marketing 1-800-430-1478 Main 530-579-8856 Fax 
  
  
  
  


  


  

  
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RE : [Asterisk-Users] X-web Lite

[Olivier Taylor]

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: [Asterisk-Users] X-web 
  Lite
  
  Hello,
  
  Im using X-web lite in a webpage 
  to connect to one of our asterisk server.
  But now I have a problem, when you 
  are connected to a voice script the voice will not be heard after a couple of 
  seconds.
  When you press or say something 
  that the voice will come back for a couple of 
  seconds.
  
  When I thy X-Lite (stand-alone 
  version) I had the same problem, but when I turned off the silence suppression 
  in X-lite ("Transmit Silence"=YES) the problem in X-lite was 
  over.
  
  Does anybody have the same problem 
  with X-web lite and does anybody have a solution for this 
  problem.
  
  Or does anybody know an other 
  embedded web based SIP client?
  
  Kind 
  Regards,
  
  Arjan 
  KroonMobillion B.V. Copernicuslaan 
  30 Postbus 554 / PO Box 554 6710 BN Ede tel: +31 (0)318-648920 
  fax: +31 (0)318-648839 mobile: +31 (0)6-55871460 email: 
  [EMAIL PROTECTED] internet: www.mobillion.nl 
  
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RE : RE : [Asterisk-Users] X-web Lite

[Olivier Taylor]

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: RE : [Asterisk-Users] X-web Lite
  
  Hi,
  
  It is not distributed 
  by X-ten.
  I found a copy on 
  another forum.
  I can send it to you, 
  if you want it.
  
  Kind 
  regards,
  
  
  Arjan 
  Kroon
  
  
  
  
  From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Olivier TaylorSent: vrijdag 13 januari 2006 
  9:40To: 'Asterisk Users 
  Mailing List - Non-Commercial Discussion'Subject: RE : [Asterisk-Users] X-web 
  Lite
  
  
  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: [Asterisk-Users] X-web 
Lite
Hello,

Im using X-web lite in a 
webpage to connect to one of our asterisk 
server.
But now I have a problem, when 
you are connected to a voice script the voice will not be heard after a 
couple of seconds.
When you press or say something 
that the voice will come back for a couple of 
seconds.

When I thy X-Lite (stand-alone 
version) I had the same problem, but when I turned off the silence 
suppression in X-lite ("Transmit Silence"=YES) the problem in X-lite was 
over.

Does anybody have the same 
problem with X-web lite and does anybody have a solution for this 
problem.

Or does anybody know an other 
embedded web based SIP client?

Kind 
Regards,

Arjan 
KroonMobillion B.V. 
Copernicuslaan 30 Postbus 554 / PO Box 554 6710 BN Ede tel: +31 
(0)318-648920 fax: +31 (0)318-648839 mobile: +31 (0)6-55871460 
email: [EMAIL PROTECTED] internet: www.mobillion.nl 


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RE : [Asterisk-Users] Re: RE : Re: RE : codecs order and so on

[Olivier Taylor]

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 answer, that's just a fact

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Tomislav
Parcina
Envoyé : jeudi 12 janvier 2006 10:31
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] Re: RE : Re: RE : codecs order and so on


In article [EMAIL PROTECTED], 
[EMAIL PROTECTED] says...
 Calling zap = no problem, Ulaw is choosen
 Calling pstn provider =fail (I need g729 but Ulaw is choosen) Call 
 from zap = no problem Ulaw is choosen Call from pstn = no problem 
 g729 used...

When you call out * establishes two channels. One is between Ua and *, 
and another between * and Zap (or provider).

If you call out, asterisk first negotiate codec for that channel. Then 
it tries to nagotiate codec for second channel. When you call your 
provider it can't nagotiate because he doesn't have g729 codec.

This is reason why you have problem, and I have explain how to solw it. 
There is nothing else I can say to help you.


-- 

Tomislav Parcina
[EMAIL PROTECTED]

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RE : [Asterisk-Users] codecs order and so on

[Olivier Taylor]

Just have a lok at this config :

[general]
Disallow=all
Allow=g729
Allow=ulaw

[pstn]
Disallow=all
Allow=g729

[zap]
Disallow=all
Allow=ulaw

In extensions.conf, I change the context for each call, Asterisk doesn't
care of codecs in contexts, it uses the order of general...
Could be good to have Ssterisk making a match between codecs in General and
the context used to make a call
But definitiely, Asterisk choose g729 even if I am in the zap context

Any idea, help is welcome.

Olivier








-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Moises Silva
Envoyé : mardi 10 janvier 2006 22:51
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] codecs order and so on


Doing in the console show translation i can see that it seems not be
possible to translate from any to g729 codec, or from g729 to any. So, let
me try to find a reason for this.

 When you have first allow=g729  (preferred codec)
all the calls to pstn providers work because the phones and asterisk agree
to use g729, so no codec translation is done. all the calls to and from fxo
fails because no translation can be made from ULAW to g729, and from g729
(phones) to ulaw. then asterisk is not smart enough to realize that can ask
the phones to use ulaw (i assume the phones support ulaw) to not use
translation to call the fxo???

 When you have first allow=ulaw (prefered codec)
all the calls to and from fxo works because the prefered codec is ulaw, then
from fxo to phones using ulaw, no codec translation is made all the calls to
pstn providers fails, again, because it seems asterisk gives preference to
ulaw codec (the first list codec) so, the phones use ulaw, and is not
possible to translate ulaw to g729 and viceversa??

im 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 to a pstn provider accepting only g729 Internal 
 calls doesn't care of codecs All Uas have g729 (g729 is then pass-thru 
 when needed) All Uas have ulaw(of course)
 If I have in [general]
 disallow=all
 allow=g729
 allow=ulaw

 In this case:

 all calls to pstn providers works
 all calls to and from fxo fails because of : No translator path exists 
 for 

 If I have in [general]
 disallow= all
 allow= ulaw
 allow= g729

 In this case:

 all calls to and from fxo works
 all calls to pstn providers fails because of : No translator path 
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--
Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org;
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RE : [Asterisk-Users] Re: RE : codecs order and so on

[Olivier Taylor]

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 is choosen
Calling pstn provider =fail (I need g729 but Ulaw is choosen)
Call from zap = no problem Ulaw is choosen
Call from pstn = no problem g729 used...

What does it mean?
Strange isn't it?

In fact Asterisk let the Uas negociates the codec for incoming calls and
doesn't care for outgoing calls.
In a context for incoming, no problems
In a context for outgoing(I use goto context,extension,priority)Asterisk
doesn't take care of the context codecs priority.

It's then false to say that asterisk uses the prefered codec of Uas, I have
here a Ua wich uses differents codecs for incoming calls.
Question is : Why Asterisk doesn't care of codecs in an outgoing context?

Any good idea is welcome.

Ps: the solution is to have a g729 codec form Digium, ok, I have it and it
works, but it takes a lot of cpu (50% of my Soekris box).














-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Tomislav
Parcina
Envoyé : mercredi 11 janvier 2006 12:28
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] Re: RE : codecs order and so on


In article [EMAIL PROTECTED], 
[EMAIL PROTECTED] says...
 In extensions.conf, I change the context for each call, Asterisk 
 doesn't care of codecs in contexts, it uses the order of general... 
 Could be good to have Ssterisk making a match between codecs in 
 General and the context used to make a call But definitiely, Asterisk 
 choose g729 even if I am in the zap context
 
 Any idea, help is welcome.

Phones usualy use only one prefered codec. So, if your phone supports 
ulaw and g729, it will use only one of those two to communicate with *. 

Once the phone is authenticated with * he allways use the same codec. So 
you have to get use that on that side is that specific codec. What is on 
another side (SIP, Zap, IAX2...) and what codec other side uses, 
determinates do you need codec translation in * box. If you need codec 
translation then you need to have licence (for g729).

I hope I have make it clear for you.

Solution:
Count do you get more outside ulaw or g729 calls (at the same time). If 
you get more ulaw calls then use ulaw codec on SIP phones. Buy the same 
number of g729 licences as you need simultanius phone calls to that 
provider.


-- 

Tomislav Parcina
[EMAIL PROTECTED]

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RE : [Asterisk-Users] Outbound routing

[Olivier Taylor]

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 : [Asterisk-Users] Outbound routing


Hi all, I've 3 providers (A, B, and C) the A is giving me freecalls to USA,
the B is giving my freecalls to Europe, and C is to call the otre
destinations. My question is, how can I configure the outboud routing to
select the right trunk for every destination?

All the providers uses the dialing form 00 1 123 4567890 when 00 is the
number dialed to call, 1 the country code, 123 the area code and 4567890 the
phone number.

I've the following outbound routing with AMP, but the calls are been started
by the first provider in the trunk sequence list:

Route Name: International
Dial Patterns : 00.
Trunk Sequence: A
B
C

I want to make that the USA calls going with A, Europe calls with B and rest
of the world with C.

Is this possible ? Can you gime a little of help with this... 

Than you in advance. :)


-- 
Guillermo Salas M.
Telconet S.A. Manta
Calle 15 y Av. 24 Esq.
Phone : 593 5 262 8071
Mobile: 593 9 985 5138
SIP   : [EMAIL PROTECTED]
e-mail: [EMAIL PROTECTED]
www   : http://www.telconet.net
http://www.telcocarrier.net

Linux User: 255902
Soporte en Linea en http://www.manta.telconet.net

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

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[Asterisk-Users] codecs order and so on

[Olivier Taylor]

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 codecs
  All Uas have g729 
  (g729 is then pass-thru when needed)
  All Uas have 
  ulaw(of course)
If I have in 
[general]
disallow=all
allow=g729 
allow=ulaw

In this 
case:

all calls to 
pstn providers works
all calls to 
and from fxo fails because of :  No translator path exists for 


If I have in [general]
disallow=all
allow=ulaw 

allow=g729

In this 
case:

all calls to 
and from fxo works
all calls to 
pstn providers fails because of : No translator path exists for 

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[Asterisk-Users] Zap hangup issue

[Olivier Taylor]

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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[Asterisk-Users] agi variables list

[Olivier Taylor]

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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RE : [Asterisk-Users] agi variables list

[Olivier Taylor]

Not really, I am looking for a list of available headers in agi.

I know the way to read them, but for example, I need to read the contact
header, but I don't know the variable name in agi.

Olivier



-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Steve 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:[EMAIL PROTECTED]
 Sent: Saturday, December 10, 2005 12:44 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: [Asterisk-Users] agi variables list
 
 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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RE : [Asterisk-Users] How to exit from Asterisk console.

[Olivier Taylor]

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 d'origine-De: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] De la part de 
  gcEnvoyé: mercredi 30 novembre 2005 22:15À: 
  asterisk-users@lists.digium.comObjet: [Asterisk-Users] How to 
  exit from Asterisk console.
  I am new to Asterisk.
  Asterisk 1.2
  
  I started * like this: asterisk 
  -vgc
  now I am in CLI mode: *CLI
  
  How do I get out this CLI mode to linux shell 
  without kill asterisk process?
  
  I tried EXIT, QUIT, exit and quit. None of them 
  work.
  
  If I use ^c, this also kill asterisk 
  process.
  
  GC
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RE : RE : RE : [Asterisk-Users] What does it mean?

[Olivier Taylor]

Logger.conf seems to have nothing to see with my problem

Olivier

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de harry gaillac
Envoyé : jeudi 24 novembre 2005 23:25
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : RE: RE : RE : [Asterisk-Users] What does it mean?


Je ne donne pas de réponse !
Il me semble t'avoir suggèrer asterisk comme système
de messagerie vocale au lieu d'SEMS, avoir fourni
quelques fichiers de configuration, ce n'étaient pas
des devinettes.

Conbien de fois on ma répondu personne n'est obligé
de faire ton 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] a écrit
:

 Cela veut simplement dire que tu te plains de ne pas
 avoir de réponses, mais
 qu'en fait tu n'en donnes pas non plus, sauf sous
 forme de devinette.
 Auquel cas, il est plus simple de ne pas répondre,
 
 merci
 
 -Message d'origine-
 De : [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] De
 la part de 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 sybillines que tes
 questions
  :)
  
  Olivier
  
  -Message d'origine-
  De : [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED]
 De
  la part de harry gaillac
  Envoyé : jeudi 24 novembre 2005 16:45
  À : Asterisk Users Mailing List - Non-Commercial
  Discussion
  Objet : RE: [Asterisk-Users] 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.1:
   /usr/lib/asterisk/modules/res_config_mysql.so:
   Undefined symbol ast_config_load
   
   What's wrong?
   
   Olivier
   
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RE : [Asterisk-Users] Asterisk doesn't start

[Olivier Taylor]

Yes, beta2 works perfectly, but 1.2 released version gives me this error.

Olivier

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de harry gaillac
Envoyé : vendredi 25 novembre 2005 11:24
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : 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/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
 
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[Asterisk-Users] What does it mean?

[Olivier Taylor]

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

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RE : [Asterisk-Users] What does it mean?

[Olivier Taylor]

Tes réponses sont aussi sybillines que tes questions :)

Olivier

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de harry gaillac
Envoyé : jeudi 24 novembre 2005 16:45
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : RE: [Asterisk-Users] 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.1:
 /usr/lib/asterisk/modules/res_config_mysql.so:
 Undefined symbol ast_config_load
 
 What's wrong?
 
 Olivier
 
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RE : RE : [Asterisk-Users] What does it mean?

[Olivier Taylor]

SIBYLLIN, INE. adj. Qui appartient aux sibylles. Il n'est guère usité au
sens propre que dans ces locutions : Les oracles, les livres, les vers
sibyllins, Les oracles, les livres, les vers des sibylles. 
Il signifie au figuré Qui est mystérieux obscur, dont le sens est difficile
à saisir. Il m'a répondu en termes sibyllins. Des paroles sibyllines. Un
langage sibyllin.


-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de 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 sybillines que tes questions
 :)
 
 Olivier
 
 -Message d'origine-
 De : [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] De
 la part de harry gaillac
 Envoyé : jeudi 24 novembre 2005 16:45
 À : Asterisk Users Mailing List - Non-Commercial
 Discussion
 Objet : RE: [Asterisk-Users] 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.1:
  /usr/lib/asterisk/modules/res_config_mysql.so:
  Undefined symbol ast_config_load
  
  What's wrong?
  
  Olivier
  
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RE : RE : [Asterisk-Users] What does it mean?

[Olivier Taylor]

Cela veut simplement dire que tu te plains de ne pas avoir de réponses, mais
qu'en fait tu n'en donnes pas non plus, sauf sous forme de devinette.
Auquel cas, il est plus simple de ne pas répondre,

merci

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de 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 sybillines que tes questions
 :)
 
 Olivier
 
 -Message d'origine-
 De : [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] De
 la part de harry gaillac
 Envoyé : jeudi 24 novembre 2005 16:45
 À : Asterisk Users Mailing List - Non-Commercial
 Discussion
 Objet : RE: [Asterisk-Users] 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.1:
  /usr/lib/asterisk/modules/res_config_mysql.so:
  Undefined symbol ast_config_load
  
  What's wrong?
  
  Olivier
  
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RE : RE : RE : [Asterisk-Users] What does it mean?

[Olivier Taylor]

Merci

Olivier

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de harry gaillac
Envoyé : jeudi 24 novembre 2005 23:25
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : RE: RE : RE : [Asterisk-Users] What does it mean?


Je ne donne pas de réponse !
Il me semble t'avoir suggèrer asterisk comme système
de messagerie vocale au lieu d'SEMS, avoir fourni
quelques fichiers de configuration, ce n'étaient pas
des devinettes.

Conbien de fois on ma répondu personne n'est obligé
de faire ton 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] a écrit
:

 Cela veut simplement dire que tu te plains de ne pas
 avoir de réponses, mais
 qu'en fait tu n'en donnes pas non plus, sauf sous
 forme de devinette.
 Auquel cas, il est plus simple de ne pas répondre,
 
 merci
 
 -Message d'origine-
 De : [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] De
 la part de 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 sybillines que tes
 questions
  :)
  
  Olivier
  
  -Message d'origine-
  De : [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED]
 De
  la part de harry gaillac
  Envoyé : jeudi 24 novembre 2005 16:45
  À : Asterisk Users Mailing List - Non-Commercial
  Discussion
  Objet : RE: [Asterisk-Users] 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.1:
   /usr/lib/asterisk/modules/res_config_mysql.so:
   Undefined symbol ast_config_load
   
   What's wrong?
   
   Olivier
   
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[Asterisk-Users] Asterisk doesn't start

[Olivier Taylor]

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

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[Asterisk-Users] RE : [Serusers] Re: [Users] open letter

[Olivier Taylor]

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 PROTECTED] [mailto:[EMAIL PROTECTED] De la
part de Iqbal
Envoyé : mardi 22 novembre 2005 16:52
À : harry gaillac
Cc : [EMAIL PROTECTED]; asterisk-users@lists.digium.com; users@openser.org
Objet : [Serusers] Re: [Users] open letter


Okay almost there :-)

So UA --- asterisk --- SER --- UA

is that it

harry gaillac wrote:

  

okay, so ALL your users are registering to
asterisk...is that correct.



Correct via ser as outbound sip proxy
  

If so the problem is howto accept users from behind
a NAT into asterisk,
or am I confusing things further.



the problem is in contact field.
when user agents send register we have in sip hf
Contact sip:[EMAIL PROTECTED]
So asterisk store this AOR and try to contact agent
via nat box instead of SER

  

If the above are true, where is SER in this, or are
users hitting SER
and you are sending the REGISTER from ser into
asterisk.



SER is an outbound sip proxy which handle IM presence
nat

Harry

  

One box
   ---
   |     |
   |  | asterisk pbx |   | 
   |     |
   |||   |
   |  ----
   |  |   SER  ||NAT box | private
   |  ---- 
   |--
  




   

   
   
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.

  


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RE : [Asterisk-Users] Can not build zaptel with kernel-2.6.12

[Olivier Taylor]

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 kernel-2.6.12


Hello,

I try to compile zaptel .
I installed kernel-sources but when i run :
make linux26
/
serveur1:/usr/local/src/ASTERISK/zaptel-1.2.0# make
linux26
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   
-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\
/etc/zaptel.conf\   -c -o gendigits.o gendigits.c
cc -o gendigits gendigits.o -lm
./gendigits
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   
-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\
/etc/zaptel.conf\makefw.c   -o makefw
./makefw tormenta2.rbt tor2fw  tor2fw.h
Loaded 69900 bytes from file
./makefw pciradio.rbt radfw  radfw.h
Loaded 42096 bytes from file
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   
-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\
/etc/zaptel.conf\   -c -o ztcfg.o ztcfg.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE   
-DSTANDALONE_ZAPATA -DZAPTEL
_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o
zonedata.lo zonedata.c
cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE   
-DSTANDALONE_ZAPATA -DZAPTEL
_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o
tonezone.lo tonezone.c
ar rcs libtonezone.a zonedata.lo tonezone.lo
cc -o ztcfg ztcfg.o libtonezone.a -lm
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   
-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\
/etc/zaptel.conf\   -c -o torisatool.o torisatool.c
cc -o torisatool torisatool.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   
-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\
/etc/zaptel.conf\   -c -o ztmonitor.o ztmonitor.c
cc -o ztmonitor ztmonitor.o
cc -o ztspeed.o -c ztspeed.c
cc -o ztspeed ztspeed.o
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   
-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\
/etc/zaptel.conf\zttest.c   -o zttest
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   
-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\
/etc/zaptel.conf\   -c -o fxotune.o fxotune.c
cc -o fxotune fxotune.o -lm
ir
You do not appear to have the sources for the
2.6.12-1-386 kernel installed.
make: *** [linux26] Error 1
//


Something don't match in makefile with debian sarge
3.1 here
linux26: prereq $(BINS)
@echo $(KSRC)
@if [ -z $(KSRC) -o ! -d $(KSRC) ]; then
echo You do not appear to have the sources for the
$(KVERS) kernel installed.; exit 1 ; fi
$(MAKE) -C $(KSRC) SUBDIRS=$(PWD) modules


Harry






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RE : RE : [Asterisk-Users] Can not build zaptel with kernel-2.6.12

[Olivier Taylor]

Ça y est, on utilise le b2b asterisk avec nos modifications java et jradius,
ça marche impeccable en attendant nos modifs définities sur vovida.
Astersik pour ce qui est pbx et ser pour le reste, je suis assez content du
résultat.
Pour le fax, notre provider l'accepte en h323, asterisk transforme sip en
h323 et go :)

Olivier

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de harry gaillac
Envoyé : lundi 21 novembre 2005 15:09
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : RE: RE : [Asterisk-Users] Can not build zaptel with kernel-2.6.12


Hello Olivier,

Non je ne suis pas fâché !
Alors ce *b2bua ?
En fait je cherche une solution pour intègrer
SER+Asterisk sur la même machine.

Ser est un bon proxy asterisk un bon ipbx.
Je souhaite utilisé ser pour le routage sip avec
asterisk et pour fournir les service de téléponie
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 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 kernel-2.6.12
 
 
 Hello,
 
 I try to compile zaptel .
 I installed kernel-sources but when i run :
 make linux26

/
 serveur1:/usr/local/src/ASTERISK/zaptel-1.2.0# make
 linux26
 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   
 -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\
 /etc/zaptel.conf\   -c -o gendigits.o gendigits.c
 cc -o gendigits gendigits.o -lm
 ./gendigits
 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   
 -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\
 /etc/zaptel.conf\makefw.c   -o makefw
 ./makefw tormenta2.rbt tor2fw  tor2fw.h
 Loaded 69900 bytes from file
 ./makefw pciradio.rbt radfw  radfw.h
 Loaded 42096 bytes from file
 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   
 -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\
 /etc/zaptel.conf\   -c -o ztcfg.o ztcfg.c
 cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE   
 -DSTANDALONE_ZAPATA -DZAPTEL
 _CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo 
 zonedata.c
 cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE   
 -DSTANDALONE_ZAPATA -DZAPTEL
 _CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo 
 tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo
 cc -o ztcfg ztcfg.o libtonezone.a -lm
 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   
 -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\
 /etc/zaptel.conf\   -c -o torisatool.o
 torisatool.c
 cc -o torisatool torisatool.o
 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   
 -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\
 /etc/zaptel.conf\   -c -o ztmonitor.o ztmonitor.c
 cc -o ztmonitor ztmonitor.o
 cc -o ztspeed.o -c ztspeed.c
 cc -o ztspeed ztspeed.o
 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   
 -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\
 /etc/zaptel.conf\zttest.c   -o zttest
 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE   
 -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\
 /etc/zaptel.conf\   -c -o fxotune.o fxotune.c
 cc -o fxotune fxotune.o -lm
 ir
 You do not appear to have the sources for the
 2.6.12-1-386 kernel installed.
 make: *** [linux26] Error 1

//
 
 
 Something don't match in makefile with debian sarge
 3.1 here
 linux26: prereq $(BINS)
 @echo $(KSRC)
 @if [ -z $(KSRC) -o ! -d $(KSRC) ]; then
 echo You do not appear to have the sources for the
 $(KVERS) kernel installed.; exit 1 ; fi
 $(MAKE) -C $(KSRC) SUBDIRS=$(PWD) modules
 
 
 Harry
 
 
   
 
   
   

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[Asterisk-Users] G729 trancoder

[Olivier Taylor]

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] codecs

[Olivier Taylor]

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 example?

Any idea is welcome.

Olivier

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RE : [Asterisk-Users] codecs

[Olivier Taylor]

Right,

I must suppose I need gsm codec to hear gsm files, I miss?

olivier

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Angelito
Manansala
Envoyé : mercredi 9 novembre 2005 12:28
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [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 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 example?

 Any idea is welcome.

 Olivier

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--
Best Regards,
Angelito Manansala
www.voicefidelity.net
Mobile: +639175425807
DID: (+63) 44 7906770
msn: [EMAIL PROTECTED]
skype: bulcrack
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RE : [Asterisk-Users] MP3 or OGG

[Olivier Taylor]

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 : asterisk-users@lists.digium.com
Objet : Re: [Asterisk-Users] MP3 or OGG


On Wed, Nov 09, 2005 at 11:43:34AM +1100, Mark Edwards wrote:
 Hey Waldo.
 
 AFAIK there is quite a lot of scope for tuning the compression of 
 speex
 - just as there is for mp3. I have no doubt that if you tune complexity,
 quality and bitrate parameters you will be able to get that filesize
 down even further. Can't see any reason at all why you shouldn't be able
 to whack mp3 for filesize. 

One thing off the top of my head: streams from Asterisk as 8mhz. Is there
any use in recording in a higher bit-rate? (any anti-aliasing-like
effect?)

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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RE : RE : [Asterisk-Users] codecs

[Olivier Taylor]

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 Sahil Gupta
Envoyé : mercredi 9 novembre 2005 12:33
À : Asterisk 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 gsm files, I miss?

 olivier

 -Message d'origine-
 De : [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] De la part de 
 Angelito Manansala Envoyé : mercredi 9 novembre 2005 12:28
 À : Asterisk Users Mailing List - Non-Commercial Discussion
 Objet : Re: [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 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 example?

 Any idea is welcome.

 Olivier

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 --
 Best Regards,
 Angelito Manansala
 www.voicefidelity.net
 Mobile: +639175425807
 DID: (+63) 44 7906770
 msn: [EMAIL PROTECTED]
 skype: bulcrack ___
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RE : [Asterisk-Users] asterisk-1.2-bêta2 | presence/ subscription support in the SIP channel driver

[Olivier Taylor]

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] asterisk-1.2-bêta2 | presence/subscription
support in the SIP channel driver


I'm not a developper !
What do you mean   Some parts of it, yes.

harry
--- BJ Weschke [EMAIL PROTECTED] a écrit :

 Some parts of it, yes.
 
 On 11/9/05, harry gaillac [EMAIL PROTECTED]
 wrote:
  Does asterisk support RFC3265 ?
 
  Harry
  --- Matt Riddell [EMAIL PROTECTED] a
 écrit :
 
   harry gaillac wrote:
nobody has an answer here!
  
   Actually someone asked for you config details.
  
   --
   Cheers,
  
   Matt Riddell ___
  
   http://www.sineapps.com/news.php (Daily Asterisk
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[Asterisk-Users] Codecs problem

[Olivier Taylor]

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 description format G729
Found description format telephone-event
Capabilities: us - 0x70f (g723|gsm|ulaw|alaw|g729|speex|ilbc), peer -
audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1
(g723)
Nov  9 16:46:13 NOTICE[402]: channel.c:1763 ast_set_read_format: Unable to
find a path from g729 to gsm
Nov  9 16:46:13 NOTICE[402]: channel.c:1730 ast_set_write_format: Unable to
find a path from ilbc to g729

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RE : [Asterisk-Users] Codecs problem

[Olivier Taylor]

Unfortunately, we are on sip :(

Olivier

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de William Lloyd
Envoyé : mercredi 9 novembre 2005 18:12
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] Codecs problem 


I've 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 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 description format G729
 Found description format telephone-event
 Capabilities: us - 0x70f (g723|gsm|ulaw|alaw|g729|speex|ilbc), peer - 
 audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) 
 Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723),
 combined - 0x1
 (g723)
 Nov  9 16:46:13 NOTICE[402]: channel.c:1763 ast_set_read_format:  
 Unable to
 find a path from g729 to gsm
 Nov  9 16:46:13 NOTICE[402]: channel.c:1730 ast_set_write_format:  
 Unable to
 find a path from ilbc to g729

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[Asterisk-Users] Asterisk as an internal pbs for a samall company

[Olivier Taylor]

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 provider and be
able to ceceive calls.

Any advice is welcome.

Sorry for the noob question,

Olivier

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RE : [Asterisk-Users] Asterisk as an internal pbs for a samall company

[Olivier Taylor]

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 Asterisk, no Ivr at this time, at
least an answering machine in case of busy or not available users.
For the rest, we need to be able to have external calls to pstn, or even to
other sip phones form other providers.
Is that enough?

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de trixter aka
Bret McDanel
Envoyé : mercredi 2 novembre 2005 13:48
À : Asterisk Users Mailing List - Non-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 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 provider 
 and be able to ceceive calls.
 
 Any advice is welcome.
 
 Sorry for the noob question,
 
 Olivier

What you want to do depends largely on what you want to do.  While that
seems like a cylic statement I will try to explain.  You have said that you
want to route calls between your asterisk box and the PSTN via a VoIP
provider that you have.  So far that seems simple, but how are those calls
going to go bewteen the office workers and asterisk?  You will need
configurations for that.  How are the inbound calls going to be routed?  Via
an IVR?  Well you will have to configure that.  There is a lot of
information that is missing from this setup.  

www.voip-info.org has a lot of asterisk examples including configuration
files.  You may find something there that does what you want.

I cant easily help you solve this problem (and suspect that no one else can
either) until you provide more information on exactly what you want.  

If you wish to discuss this offl ist feel free to email me directly.

-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378

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RE : RE : [Asterisk-Users] Asterisk as an internal pbs for a samallcompany

[Olivier Taylor]

It seems to be what I needed
Thanks for help.

Best regards,

Olivier

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de trixter aka
Bret McDanel
Envoyé : mercredi 2 novembre 2005 14:09
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: 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 Asterisk, no Ivr 
 at this time, at least an answering machine in case of busy or not 
 available users. For the rest, we need to be able to have external 
 calls to pstn, or even to other sip phones form other providers. Is 
 that enough?

Not for 100% setup, but enoughto at least get you started. From what I
understand this is what it appears you want (I may be wrong, if I am let me
know).

You will want voicemail for each user.  This is configured in voicemail.conf
http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf

You will need to edit sip.conf for the voip provider (register and
context) and if the office workers use sip to asterisk one for each of them
as well. http://www.voip-info.org/wiki-Asterisk+config+sip.conf

Lastly you will want to create a dialplan so that when a call comes in from
the DID it will then dial the appropriate user and if busy/no answer goto
voicemail.  This is done from extensions.conf.
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf

You may want a macro like:
[macro-dialvmb]
exten = s,1,Dial(${ARG1},20,t)
exten = s,2,Voicemail(u${ARG2})
exten = s,3,Hangup
exten = s,102,Voicemail(b${ARG2})
exten = s,103,Hangup 

Then for each inbound DID something like:
exten = 18005551212,1,Macro(dialvmb,SIP/user1,1234)

where user1 is the user defined in sip.conf, 1234 is the voicemail extension
defined in voicemail.conf and 18005551212 is the extension that a given did
goes to (ie last part of the register line).  

Hope this helps

-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378

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RE : [Asterisk-Users] Re: www.openpbx.org

[Olivier Taylor]

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 PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Jean-Michel
Hiver
Envoyé : dimanche 9 octobre 2005 15:35
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] Re: www.openpbx.org


Matt Riddell a écrit :

*PLONK*
  

I was only stating the obvious... sorry you don't like it.

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RE : [Asterisk-Users] Asterisk on windows

[Olivier Taylor]

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 windows


Why on earth would you want to run it on Windows?  First off, your
performance is going to go down because of the GUI... oh your call quality
just went down the toilet?  Yeah sorry the screen saver just
kicked in.   Having issues making calls?  Oh sorry we had to reboot
for a critical update.   Yeah I know audio isn't working right, the
swap file is a little large right now, we need to reboot.

Are you on crack?!?!   Asterisk runs well on Linux because of the lack
of a GUI... sleek simple interface (text) to it.   Linux is free,
windows adds a license cost.   Since you shouldn't be running any
other applications on the server anyway, why not just install Linux? 
Trying to run it on windows seems like a bad idea to me.

On 9/28/05, Kanishka Somaratne [EMAIL PROTECTED] wrote:
 why can't we compile the asterisk coading in windows, it's done in c++ 
 so it should work in windows as well

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RE : [Asterisk-Users] Asterisk realtime beta

[Olivier Taylor]

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: [Asterisk-Users] Asterisk realtime beta


So, you admit that you can do what you want using RealTime Static, but you
are just unwilling to do so. So, how is that a limitation if you 'can' do
it?

-Matthew

 From: Urban [EMAIL PROTECTED]
 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Date: Mon, 19 Sep 2005 11:03:09 +0200
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Subject: Re: [Asterisk-Users] Asterisk realtime beta
 
 Matthew Boehm wrote:
 
 I currently not use it due to some limitations in * realtime .

 
 
Such as?
 
 -Matthew
 
 
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 My current configuration uses a lot of include statements to split up 
 the context's such as security contexts included per extension (allow 
 national, internation calls etc). Since realtime does not have this 
 type of feature (if you not using static) I decided it was to much 
 work to redeisgn the dialplan at the moment.
 
 /urban
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RE : [Asterisk-Users] Asterisk realtime beta

[Olivier Taylor]

Well, in fact I have compilation error, Unixodbc is installed.
But I get :

gcc -shared -Xlinker -x -o res_odbc.so res_odbc.o -lodbc
/usr/bin/ld: cannot find -lodbc
gmake[1]: *** [res_odbc.so] Error 1
gmake[1]: Leaving directory `/usr/local/src/asterisk-1.2.0-beta1/res'
gmake: *** [subdirs] Error 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 wrote:

Does anybody has intalled it on freebsd with unixodbc or libiodbc and 
have it working?

Regards,

Olivier

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I have installed unixodbc for realtime on freebsd 5.4 and it works fine. 
I currently not use it due to some limitations in * realtime .

cheers
urban
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RE : 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 Discussion'
Objet : RE : [Asterisk-Users] Asterisk realtime beta


Well, in fact I have compilation error, Unixodbc is installed. But I get :

gcc -shared -Xlinker -x -o res_odbc.so res_odbc.o -lodbc
/usr/bin/ld: cannot find -lodbc
gmake[1]: *** [res_odbc.so] Error 1
gmake[1]: Leaving directory `/usr/local/src/asterisk-1.2.0-beta1/res'
gmake: *** [subdirs] Error 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 wrote:

Does anybody has intalled it on freebsd with unixodbc or libiodbc and
have it working?

Regards,

Olivier

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I have installed unixodbc for realtime on freebsd 5.4 and it works fine. 
I currently not use it due to some limitations in * realtime .

cheers
urban
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[Asterisk-Users] Asterisk realtime beta

[Olivier Taylor]

Does anybody has intalled it on freebsd with unixodbc or libiodbc and have
it working?

Regards,

Olivier

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