Re: [Asterisk-Users] No sound when bridging two single FXO cards

2005-07-19 Thread Francois BERGERET

Ok, sorry for this lack of explanations.
I run Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k.

The log has recorded this during the last test of bridging two FXO lines :

"","","s","default","","Zap/2-1","Zap/3-1","Dial","Zap/3/ww0164417888","2005-07-15" 
12:34:25","2005-07-15 12:34:25","2005-07-15 
12:35:49",84,84,"ANSWERED","DOCUMENTATION"


The call progress normaly, but no sound.

And, yes, the incoming and outgoing calls are placed normaly with my cards 
(3 cards, in fact).

They are seen as X101P by Linux (at boot).

Zaptel tell that :

Zaptel Configuration
==

Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)

3 channels configured.

My zapata.conf :

cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=2000
rxgain=4
txgain=4
busydetect=yes
busycount=5
group = 1
context = pstn
signalling = fxs_ks
callerid=asreceived
amaflags = documentation
channel => 1-3

My zaptel.conf :

loadzone=fr
defaultzone=fr
fxsks=1-3

How to trace the bridge establishment from CLI ?
I know how for IAX or SIP protocol, not for zap interface...

TIA

Best Regards,
Francois BERGERET,
France.


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[Asterisk-Users] No sound when bridging two single FXO cards

2005-07-19 Thread Francois BERGERET

Wow ! No reply... May be I must talk about X100P instead of X101P ?
Is someone has yet encountered this kind of "no sound" problem when bridging 
two FXO lines like this (first as input, second as output) ?

Any idea ?
TIA.

Best Regards,
Francois BERGERET,
France.

- Original Message - 
From: "Francois BERGERET" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Saturday, July 16, 2005 7:41 PM
Subject: Bridging two FXO cards (X101P) problem



Hi the list  :-)

Wondering why I can't "bridge" two X101P FXO cards to forward an external 
call from a first X101P to another analog telephone outside my house 
throught a seconf X101P.


[VACATION]
exten => s,1,Answer
exten => s,2,Dial(Zap/3/ww0161417888),120
exten => s,3,Voicemail(u1001)
exten => s,4,Hangup
exten => s,104,Voicemail(b1001)
exten => s,105,Hangup

I temps to do that to avoid missing calls during my summer vacations.
Numbering is ok when receiving a call, but no sound is heard or only few 
peak distorsions in background if I speak loud in the phone.
Normal calls are running well, I use this Asterisk box every day and this 
FXO cards are ok.

What have I missed ?
Thanks for any help.

Best Regards,
Francois BERGERET,
France.




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Re: [Asterisk-Users] modprobe wcfxo fails.

2005-07-17 Thread Francois BERGERET



Hello Mike,
 
Is your Asterisk box running for a HAM's project 
that I am working on ?
 
73,
F6HQZ,
Francois BERGERET.

  - Original Message - 
  From: 
  Michael J. Tubby G8TIC 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Monday, July 18, 2005 12:30 
AM
  Subject: Re: [Asterisk-Users] modprobe 
  wcfxo fails.
  
  Tim,
   
  Yes, it was resolved but cannot remember how... 
  I'm using RedHat FC3, Asterisk 1.0.9 and it all just works assuming that 
  you:
   
  - RTFM with respect to udev setup
   
  - use "make linux26" to build the kernel module 
  for a 2.6 kernel
   
  Regards
   
   
  Mike
   
  
- Original Message - 
From: 
Tim King 

To: [EMAIL PROTECTED] 
Sent: Sunday, July 17, 2005 8:09 
PM
Subject: [Asterisk-Users] modprobe 
wcfxo fails.


I was reading a thread where you 
were helping someone out and noticed it ended without resolve. Was this 
issue ever taken care of?I seem to be having the exact same 
problem.
 
Thanks
 
 
Tim 
King
Network 
Engineer
Computer & 
Network Solutions LLC
1331 Plainfield 
Ave
Grand 
Rapids MI  49505
 
Phone: 
800-669-3290

 



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[Asterisk-Users] Bridging two FXO cards (X101P) problem

2005-07-16 Thread Francois BERGERET

Hi the list  :-)

Wondering why I can't "bridge" two X101P FXO cards to forward an external 
call from a first X101P to another analog telephone outside my house 
throught a seconf X101P.


[VACATION]
exten => s,1,Answer
exten => s,2,Dial(Zap/3/ww0161417888),120
exten => s,3,Voicemail(u1001)
exten => s,4,Hangup
exten => s,104,Voicemail(b1001)
exten => s,105,Hangup

I temps to do that to avoid missing calls during my summer vacations.
Numbering is ok when receiving a call, but no sound is heard or only few 
peak distorsions in background if I speak loud in the phone.
Normal calls are running well, I use this Asterisk box every day and this 
FXO cards are ok.

What have I missed ?
Thanks for any help.

Best Regards,
Francois BERGERET,
France.


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Re: [Asterisk-Users] VPN's

2005-07-16 Thread Francois BERGERET
Sure, I have more than 18 tunnels to manage here, and the only blocking 
effects are thuse that I have volontary encoded .

;-)
I believe that Peter has missed something in the VPN parmeters themselves or 
not correctly understood how are his IPtables onto this two IPSec secure 
gateway...

Peter, could you post us the content of your "/etc/ipsec.conf" file ?
We can take a look here and verify what is not good.

Best Regards,
Francois BERGERET,
France.

- Original Message - 
From: "Shamsul Arefin" <[EMAIL PROTECTED]>
To: "Francois BERGERET" <[EMAIL PROTECTED]>; "Asterisk Users Mailing 
List - Non-Commercial Discussion" 

Sent: Friday, July 15, 2005 11:48 PM
Subject: Re: [Asterisk-Users] VPN's


Hi,
We use firewall and VPN togather to connect around 5 remote sites, and
never encounter these problems. Make sure that port 10,000 and above
mentioned in ur rtp conf files are opened in ur vpn and firewall. also
when u connect from remote site don't use public ip use privte behind
firewall. If still have problem send me more detail and i will be more
then happy to sort this out .

Regards
Shamsul Arefin
Saktek
Broadband telephony experts

On 7/16/05, Francois BERGERET <[EMAIL PROTECTED]> wrote:

Hi men,

You have some IP ports blocked !
I use SuperFreeSwan and I encounter no problem with this kind of
configuration.
Do you have open all ports on your IPsec gateways ?
Think to have a look to your IPchains or any kind of firewall you are
running in your IPSec gateway.
I use shorewall and it is possible to miss some rules or to let pass few
ports only for protections between sites.
You must describe more your configurations to see what...

Good luck !

Francois BERGERET,
[EMAIL PROTECTED],
France.

- Original Message -
From: "Armin Schindler" <[EMAIL PROTECTED]>
To: "Peter Osborne" <[EMAIL PROTECTED]>
Cc: 
Sent: Friday, July 15, 2005 8:35 PM
Subject: Re: [Asterisk-Users] VPN's


> On Fri, 15 Jul 2005, Peter Osborne wrote:
>> Hi All,
>>
>> I'm using Asterisk for my PBX, I have a remote office that is connected
>> by a
>> VPN link. I am using Openswan on my side and a Linksys box on the 
>> remote
>> side. I have a Polycom IP300 on the remote side configured with a 
>> static

>> IP
>> address. When I call the phone on the remote side, it rings and
>> establishes
>> the call fine. The problem I am having is that the remote side can hear
>> the
>> call find but the local side hears nothing. Because of the VPN there 
>> are

>> no
>> firwalls in the way. Does anyone have some ideas or atleast how I can
>> track
>> down the problem.
>
> I had the same problem with VPN using 'netscreen' (or a similar name)
> boxes.
> When I switched from SIP to IAX protocol, it worked perfectly.
>
> I think the SIP voice UDP packets are blocked somehow, but I didn't
> investigated it further.
>
> Armin
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--
Best Regards
Shamsul Arefin
Saktek ,
Broadband Telephony experts 


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Re: [Asterisk-Users] VPN's

2005-07-15 Thread Francois BERGERET

Hi men,

You have some IP ports blocked !
I use SuperFreeSwan and I encounter no problem with this kind of 
configuration.

Do you have open all ports on your IPsec gateways ?
Think to have a look to your IPchains or any kind of firewall you are 
running in your IPSec gateway.
I use shorewall and it is possible to miss some rules or to let pass few 
ports only for protections between sites.

You must describe more your configurations to see what...

Good luck !

Francois BERGERET,
[EMAIL PROTECTED],
France.

- Original Message - 
From: "Armin Schindler" <[EMAIL PROTECTED]>

To: "Peter Osborne" <[EMAIL PROTECTED]>
Cc: 
Sent: Friday, July 15, 2005 8:35 PM
Subject: Re: [Asterisk-Users] VPN's



On Fri, 15 Jul 2005, Peter Osborne wrote:

Hi All,

I'm using Asterisk for my PBX, I have a remote office that is connected 
by a

VPN link. I am using Openswan on my side and a Linksys box on the remote
side. I have a Polycom IP300 on the remote side configured with a static 
IP
address. When I call the phone on the remote side, it rings and 
establishes
the call fine. The problem I am having is that the remote side can hear 
the
call find but the local side hears nothing. Because of the VPN there are 
no
firwalls in the way. Does anyone have some ideas or atleast how I can 
track

down the problem.


I had the same problem with VPN using 'netscreen' (or a similar name) 
boxes.

When I switched from SIP to IAX protocol, it worked perfectly.

I think the SIP voice UDP packets are blocked somehow, but I didn't
investigated it further.

Armin
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Re: [Asterisk-Users] Looking for a consutant in France about Asterisk.

2005-07-12 Thread Francois BERGERET



Hi all the list,
Bonsoir,
 
It could be nice to give a correct email adress to 
contact you, instead of an "over quota" one...
If you want a direct reply out of this list, 
please, give us one other.
 
Best Regards, 
a bientot,
Francois BERGERET,
France.
[EMAIL PROTECTED]

  - Original Message - 
  From: 
  Aref Cheikhrouhou 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Tuesday, July 12, 2005 1:08 
AM
  Subject: [Asterisk-Users] Looking for a 
  consutant in France about Asterisk. 
  
  
  Hi 
  All, 
  Looking 
  for a consultant in France to build a VoIP offer. 
  
   
  Thanks 
  for the reply
  Aref
  
  

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