Hi Dear
I have a problem, with my ser 0.9.6 server, as a sip
server.
the following results are with grandstream 487 or 287.
about 30% of customers, can install the sip phone very
easy with the first configuration.
about 25%, can register after changing adsl router, or
sip phone's configuration,
about 15%, can register with another sip phones, like
linksys, or. ...
and about 30% can not using sip services.
is it a natural event????
do u have any idea ????
best
Pezhman
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Hi,
With the recent development in PEAR Network, where XML_RPC isn't
available any longer, has anyone been able to install serweb recently?
I can't get XML_RPC package which is one of the requirements to get
SERWEB running.
Where I was able to get the raw files from PEAR but not ABLE to run
"pear install XML_RPC", i can't make install and the likes. Hence, i
doubt if XML_RPC will be successfully installed into my PEAR.
Any help!!
Mike.
FYI: I'm at the IETF meeting in the week of March/10th and at VoN
in the week of March/17th. Drop me a note if you would like to meet
whether for some SER/SIP related discussion or just a beer chat :-)
-jiri
--
Jiri Kuthan http://iptel.org/~jiri/
Hi, in order to avoid replying with early_media I'm trying to transform
a "183" with SDP into a "180".
Ok, my conclusion is that I cannot do it.
Textops functions seem to work just with SIP headers and body but never can
modify the SIP reply code, is true?
Also, I would need to delete the entire body in the 180|183 reply but
functions supposed to do it don't work as expected. For example:
* subst_body('/^.*//g')
does NOTHING, the only way it can do somehting is by:
subst_body('/^.*=.*//g')
any reason for this?
Thanks for any suggestion.
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
Hi,
I am working on project involving SER, SEMS. Things works OK as configured. I am curious to know how I can generate RTCP message towards caller.
Greatly appreciate the response.
Thanks,
Keyur Parikh
Hi,
I am working on a project that uses SER, SEMS and Media Proxy. I was wondering how and where to enable the RTCP reports. I could not find any useful information either in the product documentation or in the user groups.
Does any of these products support generating RTCP reports? IF not how do we add the feature and where would be best place to add it?
Thanks for the help.
--Srinivas Kotamarti
I've been looking at the possibility of using OpenSER as an
ingress/egress gateway, mediating access between the internet at large,
and a private network containing amongst other things SIP servers
through which a call may be routed to provide services such as IVR and
call archiving, but which should otherwise be hidden from the outside world.
I'm finding two interlinked problems:
(1) The internal layout of the network is revealed in Via headers - OK,
so this is somewhat intrinsic in SIP, and not really OpenSER's fault,
but....
(2) ... If an inbound SIP request has Route headers, loose_route()
pretty much sends it whereever the requester asks. There are admonitions
in the OpenSER docs about the need to secure loose_route(), but there's
no information I can find on how you should do this. In particular, a
simple authorization scheme is not good enough - just because someone
should be allowed to place calls through the gateway, doesn't mean it
should be allowed absolute control over the routing of the request, or
they could use information gleaned from Via headers of previous
transactions to add or bypass routing steps within the private network
at will.
It is possible to securely use OpenSER on a security boundary? If so, how?
Max.
Hi people...
With the Textops Module is possible search for the first match of re in the body of the message and appends txt after it, ex: search_append, search_append_body...
But I want append a TXT after the last occurrence of some thing...
Example: after the last occurrence of .*a=rtpmap:.*
Is there some way for my doubt?
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