Hello,
I've ever posted this question many times.
I try again.
method.get_missed_calls.php script from serweb require
domain column in aliases tables and username/domain
columns in missed_calls table .
What are the ways to make serweb missed calls useable?
* Make acc module set values into username/domain
columns in missed_calls table when setflag() is
called.
How can we set values in domain column for aliases
tables ?
* Rewrite method.get_missed_calls.php script.
* Using avpops !?
If you really don't understand my english let me know
it .
Regards
Harry
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Hello,
I am wondering whether the SER process for dealing with SIP REGISTER
messages supports instance identifiers? i.e. an instance media feature
tag defined in the SIP REGISTER contact field. It is defined according
to the following internet draft:
http://www.ietf.org/internet-drafts/draft-jennings-sipping-instance-id-0
1.txt
Could someone let me know if this is supported?
BR,
Aisling.
_______________________________________
Aisling O' Driscoll
Research Student
Centre for Adaptive Wireless Systems
Cork Institute of Technology.
Email: ashling.odriscoll(a)cit.ie
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Hi all,
I'm working for a nat traversal scenerio.
The case is both endpoints are behind NAT (ADSL clients) and SER is on the
real network. In this project I dont want to implement a rtp proxy because I
have limited bandwidth on real network. I want clients to pass signaling to
me and rtp directly between eachother.
I have tried some configurations with module mangler but i could not manage
to create rtp session properly.
What do you think about the solution and if you can mail a sample
configuration I will be very happy.
Regards,
Omer
hello,
after googling for some information about nathelper it seems that
nathelper should be able to traverse any type of NAT (even symmetric
ones) as long as both user agents support symmetric RTP.
is this right?
are there any situations where just using nathelper may not be
sufficient? (such as transfers between an internal phone to an external
phone or transfers between two internal phones).
bye,
/gst
Hi all,
I wrote a perl script to register many sip uri in one time using the FIFO of
SER and the ul_add method.
I'd like to know if there's other concurrent process than my script writing to
the FIFO (for example SER itself for registration?)and is there a way to
protect the fifo to be sure that the information sent to the FIFO will be
parsed correctly (ie prevent information to be interlaced in the fifo).
Thanks very much,
Rodrigue
BTW, I am running httpd version 2.0.46
Thx
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Hi,
I am facing the following problem: I force a maximum
of one registration per sip user (
modparam("registrar", "max_contacts", 1). I also
authenticate INVITES (proxy_authorize()).
When a second UA tries to register, it is denied as
expected and registration fails. But when this second
client places a call, it suceeds even though it is
not registered (The proxy_authorize for INVITE suceeds
since the credentials are good).
I do not want this second client to place a call since
it is not registered. I have looked at all the various
functions and there seems to be no way this. Checking
whether the "from" uri is registered will not help
since the first client is already registered.
Is there any way to handle this?
Thanks
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