Hello,
How can I use textops module to rewrite all ip private
addresses in sip contact hf with a public ip
I can't write the right regexp.
if search("^Contact sip:user@private ip$)) {
replace_all ("private ip", "public ip");
};
Harry
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Hi all!
I want to know if it is possible to configure openser to communicate
with another sip server? if it's possible , how can i do it?
Best regards!
Serge!
Hi all,
Does openser offer any support for call queues?
Thanks,
Giordanna
---------------------------------
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Friends,
I am using 2 SJPhone UAs (B, C) on two different networks, that register
with my SER server
using the same (username=XXXXXXX, password).
>SJPhone - B (username:XXXXXXX)
/
INVITE
/
SJPhone - A (username: YYYYYYY)
\
INVITE
\
>SJPhone - C (username: XXXXXXX)
I dial the phone number XXXXXXX from a third SJPhone (A).
When only one of the destination SJPhones is registered with SER,
let's say B, then B rings. Then while B is ringing, if "ignore" button
is pushed, a 486 is generated by B and sent back to SER. Then
SER receives the 486 in failure_route and processes it.
But when both B and C are online, both B and C ring.
when "ignore" button is pressed on B, 486 is generated by B and
received by SER but SER does not enter the failure_route.
When ignore is pressed on the first destination, I expect SER to
send a CANCEL to the second destination and receive the 486
generated by the first destination in the failure_route so that I
can forward the first destination to voicemail!
Can anyone share any insight as to why this is happening?
Thank you for your help in advance
ramin
HI all
*I'm doing a project on comparison of SER an OSIP*
*can anybody plz assist me in telling that *
*is it possible to isolate SIP stack and install only that*
*if yes then how*
**
*thanks*
**
*Nitin*
Which Gateways is better.
I want to know which gateways are better.
I am comparing Audiocodes and Mediatrix.
Anyone have any opinions on either one good or bad?
Thanks
Hi List
I am using rtpproxy from long now without actually knowing its proper
internals. As far as i know, it is used to modify the SDP coming from Natted
clients. Am i correct?
Second, can i use it to develop Prepaid Application?
Best Regards
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I try to rewrite contact field in sip hf because of
fix_nat_contact() fail
with this diagram
-----ppp0[ ser/nathelper ] eth0----private network
debian 3.1
when agents send INVITE or REGISTER to ser which
listen on ppp0 the ip address in contact sip hf stay a
private ip address, why ?
I try siproxd which rewrite it .
how can i build a ser.cfg in order to contact in sip
hf will be rewritte
Harry
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Hi all,
I am using Quintum tenor platform (radius + pstn gw) to send my
calls outwards to pstn. I am using radius authorization. The radius server
is provided by Quintum itself as a packaged billing+gw solution.
The radius auth packets seem to go from my ser and they even land
on the radius server, but they are not visible in the "incoming requests"
monitor of the radius server or its log, although they are visible in the
Ethereal on the radius machine. What I doubt is whether the Quintum radius
is supports digest requests or not, because my radius packets are working
very fine with freeradius, although on a different machine.
Correct me or ask for clarification when and where needed.
Any help will be highly appreciated.
PS: - As evident, I am not having any problem in my radiusclient.conf.
- The radius server is running on standard ports.
-There is no firewall policy on the Quintum radius server that
blocks/denies/rejects any packets from ser.
Regards,
Ashutosh Kumar
Hello!
Can someone explain me how i can correctly check if 2 UA's are behind same NAT ?
for example: A and B UA's have private 192.168.1.1 and 192.168.1.2, SER have a public ip
A calling B
For A all is clear for me - SRCIP have an A external ip and i can use it
But i dont see any B external ip, RURI have a private ip part (192.168.1.2)
Pls show me the way!
