Hi,
I'm using the latest openser version with presence supports. My User
Agent=kphoneSI.
My question is: How can i add my own events on the KphoneSI?
Kind regards,
Kc
Hi,
Did anyone successfully setup call accounting for call forward? I have
been searching for a working example for few days on google, I still
couldn't find any working example.
The config I have at the moment is
modparam("acc", "multi_leg_enabled", 1)
modparam("acc", "src_leg_avp_id", 110)
modparam("acc", "dst_leg_avp_id", 111)
In the main route,
if (!method=="REGISTER" && !method=="NOTIFY") {
avp_write("$from/username", "$avp(i:110)");
avp_write("$ruri/username", "$avp(i:111)");
};
Let say A call B and B forward call to C, the record in the acc table is all
A to C without B in any one of the record for billing purpose. The
Call forward was set on the UA.
Can anyone give me a hand on this please?
--
Howard Tang
Hello,
How to update call-info in openser (for an INVITE sip message)?
I want the call-info to contain the actual IP-address of called party.
(that we normally do for $ruri).
Thank You.
With warm regards,
Premnath, KN
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Hello All. After investigating quite a while about this issue I'm kind
of desperate now, I will appreciate your help on this.
I have a SER Server with multiple interfaces, I'm able to handle all the
SIP packets with the multi homed enabled. Now the part that I'm having
real trouble with is the RTP audio streams. After struggling with
MediaProxy I finally started using RTPProxy for this, since it has
bridge mode available. Now the problem is that I have more than one
network to connect audio from, so the internal and external
configuration is not enough for me. So my questions are:
1.-Can I have RTPProxy Bridge mode working for all the interfaces in my
server?
2.-Can I have more than one RTPProxy socket available for each of the
multiple interfaces I'm connecting in order to have the audio working,
so each route will use the proxy of their own network?
3.-If #2 is true, is this where the force_socket and force_rtp_proxy
functions play a role in my configuration?
Please help!!
Thanks you so much in advance
Gerardo Amaya
my configure in xlite:in system setting
in network I put these:
Auto detect IP:yes
Primary stun server:blank
in sip proxy I selected default and:
enabled:yes
display name:user
username:1234
Aotorisation user:1234
password:1234
Domain/Realm:10.10.10.138
sip proxy:10.10.10.138:5060
use outband proxy:default
send internal ip:Always
register:default
Is it any wrong?this xlite phone is on a pc that ser server
is.another one is on another PC and only in username and display and
password differs with this one.
I am in the back of a NAT but my two clients and server are
there.
what is so important in setting xlite?
thanks.
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Hi,
Is there some method to generate modules documentation from the cvs source??
Im running make install-modules-doc and nothing happens??
Bests
Tomasz
Has anyone ever tried routing calls based on codec in SER?
If so, do you have pointers for getting started?
Thanks,
Nathan
--
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www.nathanpralle.com
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