- "Oguzhan Kayhan" wrote:
> Hi, as far as i know 's' is wildcard for "all calls" because as i see on
> asteriskgui it is written as 's' (CatchAll) which means redirect all
> calls to that extension.
That is not correct. The 's' extension only matches analog calls (because they
have no dia
> 2009/3/19 Oguzhan Kayhan
>
>> Hi, i managed to connect to Ericsson MD110 with PRI at last.
>> And made a successful call thru asterisk to ericsson.
>>
>> But when i try to call from ericsson to asterisk i got an error on
>> asterisk side.
>> And i couldnt figure out why.
>>
>> Here's my extensio
2009/3/19 Oguzhan Kayhan
> Hi, i managed to connect to Ericsson MD110 with PRI at last.
> And made a successful call thru asterisk to ericsson.
>
> But when i try to call from ericsson to asterisk i got an error on
> asterisk side.
> And i couldnt figure out why.
>
> Here's my extensions.conf abo
sorry i cant help you :-(
i only can sugest add another peer in sip.conf in one use only audio and in
the other one use only T38.
you should post it again whit a subject like "T38 problem" or "please
help!! t38 problem".
David
Sorry again.
2009/2/27 michel freiha
> Dear David,
>
> Please
Dear David,
Please find on http://pastebin.com/m69b8559d my sip.conf file
Thanks a lot
On Fri, Feb 27, 2009 at 1:05 PM, David fire wrote:
> paste your sip.conf.
> David
>
> 2009/2/26 michel freiha
>
>> Dear All,
>> I have created an inbound context in SIP .conf that forward incoming call
>> t
paste your sip.conf.
David
2009/2/26 michel freiha
> Dear All,
> I have created an inbound context in SIP .conf that forward incoming call
> to opensips server...The problem appears as soon as I enable t38pt_udptl =
> yes...The Asterisk negotiate the SIP session with OpenSIPS without adding
> vo
Von: Crazy Boy [mailto:[EMAIL PROTECTED]
Gesendet: Donnerstag, 7. September 2006 14:25
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] Incoming call problem-calling part is busy(IPKall)
Hi,
I have registered with IPKall ang got the number i.e., 206XXX. When I
call to this numbe
Crazy Boy wrote:
I have given my total configuration. Please tell me the solution.
Looking forward to your response. Thank you.
You need to also include the output from the console.
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary Safe
I have a wrt54g as well and I did not have to
open any ports. I also have internal ipaddresses.
Let me see what your sip.conf looks like first and
then if that is ok let's see your extension.conf
- Original Message -
From:
Matt
Schwartz
To: asterisk-users@lists.digi
There is no way for asterisk to know which extension you want to call when
you have a single analog line. You can either send the call to an extension
or group, or you could create a menu system that allow the caller to select
which extension to call (press 1 for Steve, 2 for Dave etc.).
/Anders
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