Hi Iñaki,
I had the same issue with a Nortel CS1k switch, the easiest way to check this
problem is to use the Wireshark feature "Display raw text for SIP message"
which can be enabled under preferences / protocols / SIP
regards,
Andreas
Ovidiu Sas schrieb:
> Hello Inaki,
>
> Perform a raw captu
Hi all,
I made a reservation at 19:30 on "Granig". If the weather is ok, we've a
nice table outside.
See you there,
Andreas
Daniel-Constantin Mierla wrote:
> Hello,
>
> just to announce the next OPENSER Social Networking Event in Vienna,
> Austria, May 24, 2008, 19:30 at Plutzer Brau:
>
> h
Hi,
I have registered an UA 12345 , which is public.
In location table, I found NULL value for received.
I dial any UA , irrespective to their NAT status.
Call has been established but can't get audio both side.
It does not use mediaproxy for audio signaling.
How to resolve this problem?
Thanks
Hi,
Here is the scenario of the call.
12345 ---> 56565656 > PSTN
I have registered user 12345 using sjphone.
Dialing 56565656 which is registered to the same openser.
Now i let the call ring & after timeout call is transffered to PSTN number.
Which is terminated by PSTN gateway.
I am using me
hello all,
i am investigating the authentication on openSER. I search for a proper
explanations but unfortunately i did not find how it is exactly done so i
did some experiments. i assumed that the response is generated as the
following: note that i set the username and password with the same stri
Iñaki Baz Castillo wrote:
> Hi, if you get into:
> https://sourceforge.net/projects/openser/
> you won't read the word "OpenSER" but "Scalable SIP server".
>
>
> Also, in the "location" line we read:
>
> SF.net > Projects > Scalable SIP server > Summary
> Scalable SIP server
>
>
> I wonder: w
--- Begin Message ---
Hi Sergio Gutiérrez,
Thanks for quick reply of my mail
my problem is when user-A online from more then machine.
then user-B try to call user-A( Invite method ).
openser should send invite to all the machines where User-A is
online.
but it's not happen
Hi Inaki
Due to NAT issue and timer on a following proxy, the registration period of the
phones is 60 seconds. Thus, I have the possibility to keep track of the phone
status.
A proper behaviour for this case is assumed: Every phone, that sent a INVITE
and is alive (sends RE-REGISTER) must and
Hi mates,
I still need your pointers regarding my problem in this post, today i have
attached the routes suspected to be involved in this saga. From my config
file plz see below.
# -
# Unauthorized relay
# ---
El Thursday 22 May 2008 11:51:50 Schumann Sebastian escribió:
> Dear all
>
> I have a problem, which I thought I can solve with PUA_USRLOC. I have a
> OpenSER SIP Proxy and want him to create presence information about the
> messages that pass him.
>
> Some examples:
> - REGISTER: Publish as online
Hello,
Indeed pua_usrloc reacts only to Register message.
There are a lot of possibilities add that functionality for Invite and
Bye. In short, you need to:
- first parse the Invite or Bye message to extract the destination and
the source of the call
- then, get from database table 'presenti
22 maj 2008 kl. 11.48 skrev Iñaki Baz Castillo:
> El Thursday 22 May 2008 11:22:59 Johansson Olle E escribió:
>> 22 maj 2008 kl. 11.15 skrev Iñaki Baz Castillo:
>>> Hi, to be RFC3261 compliant a SIP proxy should accept hex encoded
>>> username in
>>> any URI, this is:
>>>
>>> sip:[EMAIL PROTECTED
Dear all
I have a problem, which I thought I can solve with PUA_USRLOC. I have a
OpenSER SIP Proxy and want him to create presence information about the
messages that pass him.
Some examples:
- REGISTER: Publish as online/available
- REGISTER+Expires=0: Publish as offline
- INVITE/200 OK: Publi
El Thursday 22 May 2008 11:22:59 Johansson Olle E escribió:
> 22 maj 2008 kl. 11.15 skrev Iñaki Baz Castillo:
> > Hi, to be RFC3261 compliant a SIP proxy should accept hex encoded
> > username in
> > any URI, this is:
> >
> > sip:[EMAIL PROTECTED] == sip:[EMAIL PROTECTED]
> >
> > For allowing this
22 maj 2008 kl. 11.15 skrev Iñaki Baz Castillo:
> Hi, to be RFC3261 compliant a SIP proxy should accept hex encoded
> username in
> any URI, this is:
>
> sip:[EMAIL PROTECTED] == sip:[EMAIL PROTECTED]
>
> For allowing this we must use, explicitely, the transformation
> $(rU{s.unescape.user}).
Alright, thanks.
Next time I'll use the docs mailing list, just thought it is directly module
related (as module documentation is build from the xml files)
Sebastian
-Original Message-
From: Anca Vamanu [mailto:[EMAIL PROTECTED]
Sent: Thursday, 22. May 2008 10:55 AM
To: Iñaki Baz Cas
Hi, to be RFC3261 compliant a SIP proxy should accept hex encoded username in
any URI, this is:
sip:[EMAIL PROTECTED] == sip:[EMAIL PROTECTED]
For allowing this we must use, explicitely, the transformation
$(rU{s.unescape.user}). Do OpenSer administrators allow this hex encoding?
--
Iñaki
Hi,
No need to do that anymore. Ifixed it.
Thanks,
Anca
Iñaki Baz Castillo wrote:
> El Thursday 22 May 2008 09:56:29 Schumann Sebastian escribió:
>
>> Dear all
>>
>> I noticed a small mistake in the documentation:
>>
>> I had some errors in config file using the pua_usrloc modparam
>> paramet
Hi, if you get into:
https://sourceforge.net/projects/openser/
you won't read the word "OpenSER" but "Scalable SIP server".
Also, in the "location" line we read:
SF.net > Projects > Scalable SIP server > Summary
Scalable SIP server
I wonder: why not name it as OpenSER?
--
Iñaki Baz C
El Thursday 22 May 2008 09:56:29 Schumann Sebastian escribió:
> Dear all
>
> I noticed a small mistake in the documentation:
>
> I had some errors in config file using the pua_usrloc modparam
> parameters with copy & paste.
>
> The problem were the semicolons ; behind the modparam definitions. As I
Dear all
I noticed a small mistake in the documentation:
I had some errors in config file using the pua_usrloc modparam
parameters with copy & paste.
The problem were the semicolons ; behind the modparam definitions. As I
had the error some time ago already I easily figured that out. Although
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