Is there a way to restrict the number of simultaneous calls for a
particular sip client? For example, if I sell a service where each
line can have at most 4 simultaneous call voice paths and there are
two incoming calls and two separate outgoing calls, how can I make it
so that whenever a new calls comes in or a user tries to make another
call, they get fast busy or just busy?
Thanks,
Waldo
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi,
Is there anyone who uses fax over openser? Or can say something about
the reliability of fax over openser?
chris...
Hi,
only end devices need to support this feature. Nothing to be set on
openser side.
Ramona
unplug wrote:
>Hi,
> Does openser support fax? How?
>unplug
>
>_______________________________________________
>Users mailing list
>Users(a)openser.org
>http://openser.org/cgi-bin/mailman/listinfo/users
>
>
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.1 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
iD8DBQFEMiT6R0exH8dhr/YRAu8rAJ0cNvJRRqSHycU+dhMZvuJeVK+JmACg0Uox
rk/ECrupb92VR2+wBmkovmM=
=0pKc
-----END PGP SIGNATURE-----
We're looking into Asterisk-b2bua with SER to handle prepaid stuff, but we're
running into the snag that, well, there's pretty much absolutely NO
documentation for asterisk-b2bua. It's a few perl modules, a config file, and
an INSTALL file that consists of explaining how to download the source and
patch asterisk to use the program. That's it.
It mentions radius, and it LOOKS like radius needs to be running on the same
box as the asterisk-b2bua setup. I suppose that will work, as I can then point
asterisk to do its accounting/auth on the SER box where I have radius running
for auth. I've seen mention that asterisk-b2bua doesn't handle digest
authentication (although it has an auth server in its config file), but I
don't even know what it would be doing authentication FOR at that point and,
since SER uses digest, I don't know how the two would mesh.
Other than what I've been able to figure out from looking at the code (which
is, granted, not a whole lot), there's nothing even remotely resembling docs
(oh, my kingdom for the ability to force coders to hire technical writers). I
was kind of hoping someone might have a sample config for use with SER... or
be able to point me in the direction of one.
N.
I have been used SER and Asterisk ,I know asterisk have a function of meetme conference,but I don't know how to use it .Anyone know how to use and configure or administrat it
Thankful
Zhaomin
Hello,I have the following setup:
UA<->NAT<-->SER/MediaProxy<->Prepaid(B2BUA)<-->PSTNGW
|
IVR
Nat'd user calls a number that is forwarded (t_relay)
to Prepaid and the user is prompted for
PIN#/Destination Phone# and all works fine. Mediaproxy
is invoked and a mediaproxy port# (35774) is assigned
during the whole process.
Once the user enters the destination number, the
Prepaid forwards the call to the PSTNGW and the PSTN
phone rings (Call signalling working fine). When the
PSTN phone is answered, no media can be heard!!!
I did some investigation and here is what is
happening:
Once the user enters the destination# to dial, The
Prepaid/B2BUA does two things:
1) Sends a re-invite to put the UA on hold.
2) Sends an INVITE to the PSTN GW and retrieves the
SDP from the response from PSTN GW
3) Sends a second-reinvite to the UA via SER with the
SDP info of the PSTNGW
4) SER invokes the mediaproxy (since it is reinvite)
and assigns the SAME MEDIAPROXY PORT# as earlier on
when the media was flowing fine (SDP has audio port#
35774)
5) SER forwards the re-INVITE TO UA
6) UA responds with a 200 OK and sends 200 OK to SER.
SER agains invokes mediaproxy and assigns the SAME
MEDIAPROXY PORT# (audio port 35774) and forwards the
response to Prepaid (which sends it to PSTNGW).
I checked on Cisco PSTNGW that it is creating a
session with the mediaproxy (port:35774). BUT NO AUDIO
CAN BE HEARD IN ANY DIRECTION.
Can anyone please help? Is this a mediaproxy issue
that when reinvites are sent and mediaproxy is invoked
multiple times, issues arise?
I am running the latest mediaproxy
version.
Thanks
Dave
---------------------------------
Make Yahoo! Canada your Homepage Yahoo! Canada Homepage
Hello,I have the following setup:
UA<->NAT<-->SER/MediaProxy<->Prepaid(B2BUA)<-->PSTNGW
|
IVR
Nat'd user calls a number that is forwarded (t_relay)
to Prepaid and the user is prompted for
PIN#/Destination Phone# and all works fine. Mediaproxy
is invoked and a mediaproxy port# (35774) is assigned
during the whole process.
Once the user enters the destination number, the
Prepaid forwards the call to the PSTNGW and the PSTN
phone rings (Call signalling working fine). When the
PSTN phone is answered, no media can be heard!!!
I did some investigation and here is what is
happening:
Once the user enters the destination# to dial, The
Prepaid/B2BUA does two things:
1) Sends a re-invite to put the UA on hold.
2) Sends an INVITE to the PSTN GW and retrieves the
SDP from the response from PSTN GW
3) Sends a second-reinvite to the UA via SER with the
SDP info of the PSTNGW
4) SER invokes the mediaproxy (since it is reinvite)
and assigns the SAME MEDIAPROXY PORT# as earlier on
when the media was flowing fine (SDP has audio port#
35774)
5) SER forwards the re-INVITE TO UA
6) UA responds with a 200 OK and sends 200 OK to SER.
SER agains invokes mediaproxy and assigns the SAME
MEDIAPROXY PORT# (audio port 35774) and forwards the
response to Prepaid (which sends it to PSTNGW).
I checked on Cisco PSTNGW that it is creating a
session with the mediaproxy (port:35774). BUT NO AUDIO
CAN BE HEARD IN ANY DIRECTION.
Can anyone please help? Is this a mediaproxy issue
that when reinvites are sent and mediaproxy is invoked
multiple times, issues arise?
I am running ser-0.9.6 and the latest mediaproxy
version.
Thanks
Dave
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
Thanks.
Do you think is possible, instead to work with script, to work over source code and change it to let it do what I want?
Any thought?
-----Messaggio originale-----
Da: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro]
Inviato: giovedì 30 marzo 2006 15.56
A: D'Addelfio Davide
Cc: users(a)openser.org
Oggetto: Re: R: R: [Users] export SIP_Domain
On 03/29/06 12:50, D'Addelfio Davide wrote:
> Ok thanks, will take a look!
>
> Now that I've installed and running openser, I have to work over it. Do you know if is possibile to configure openser with some trigger that allows it to switch incoming message (INVITE, REQUEST, etc...) towards another element, like an application server? Do they exists? Where can I find it?
>
do you want to route certain request types to different hosts? Like if
request is INVITE, send it to an application server?
If so, then you can do:
if(is _method("INVITE")) {
t_relay("udp:app_srv_ip:port");
exit;
}
This example works with development version of openser, for versions
1.0.x below, check the documentation of tm module which is posted on
project's web site (http://openser.org). You have to use the textops
module as well.
Cheers,
Daniel
> Thanks for help
>
> -----Messaggio originale-----
> Da: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro]
> Inviato: mercoledì 29 marzo 2006 11.47
> A: D'Addelfio Davide
> Cc: users(a)openser.org
> Oggetto: Re: R: [Users] export SIP_Domain
>
> On 03/29/06 11:56, D'Addelfio Davide wrote:
>
>> Thanks for reply...it works very good!!!
>>
>> Still not understand which different between SER and OpenSER, could anyone explain me?
>>
>>
> openser started from the same code base as ser v0.9.x, and that was
> openser v0.9.x . Since then a lot of features has be added, new stable
> release was made as 1.0.0 in October last year. For more details see:
>
> http://openser.org
> http://openser.org/diffs-0.8.14.php
> http://openser.org/diffs-0.9.0.php
> http://openser.org/release-1.0.0.php
>
> Cheers,
> Daniel
>
>
>> Thanks
>>
>> -----Messaggio originale-----
>> Da: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro]
>> Inviato: martedì 28 marzo 2006 11.55
>> A: D'Addelfio Davide
>> Cc: users(a)openser.org
>> Oggetto: Re: [Users] export SIP_Domain
>>
>> Hello,
>>
>> On 03/28/06 11:59, D'Addelfio Davide wrote:
>>
>>
>>> Hi to all, i'm totally new in openser and i'd like to enjoy it...
>>>
>>> I've installed 1.0.1 version on my debian and seems it work fine with
>>> mysql database.
>>>
>>> Just a stupid question: I miss to export SIP_DOMAIN cause I can't
>>> understand what name I have to give...the default is openser.org...and
>>> I've seen on install file that I have to change it...could someone
>>> explain me what I have to do, please?
>>>
>>>
>>>
>> the SIP_DOMAIN has to be your domain or IP address:
>>
>> export SIP_DOMAIN="mydomain.com"
>>
>> Cheers,
>> Daniel
>>
>>
>>
>>> Thanks in advance
>>>
>>> Davide
>>>
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users(a)openser.org
>>> http://openser.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>
>>
>
>
Hi,
I know that t_relay() can use for stateful transaction relay.
What is a function that it can handle stateless relay?
Does it exist?
Thanks in advance.
Best regards,
Ren-Huang Liou
i havle little to none mysql skills so enless you can right a guide on how
to do it that road is imposible to me :P
sorry to say im on centos 4.3 server ed so no gui and no mysql administrator
befor i start installing a lamp on my box to get phpmyadmin running i was
wondering if there was a magic command that couold helkp me out like
instaead of serctl add someguy somepass some(a)thing.com
i could do serctl add someguy null some(a)thing.com
or some combo of " or ' or even - to trick mysql to not put anything in that
colum but keep ser happy :)
thanks for reading this far :)
regards
Alex Wood
