Hi,
I have the following SIP scenario:
UA_A -> SER_A -> other SIP Proxy -> SER_A -> UA_B
I want to force SER_A instance to forward the received request to other SIP proxy element and ensure that that request will return to SER_A afterwards.
To do that I use an insert_hf() function from textops module inside ser.cfg to insert a double Route header field - first value the sip uri of the 'other SIP proxy' and second one the SER_A sip uri to assure that the forwarded request will return to SER_A.
Whenever request enters the ser.cfg script logic my SER code performs the loose_route() check in general. In my situation when request initialy enters the logic it doesn't have any Route headers. When request enters a particular route block responsible for handling this request there is an lookup_user() invocation to check if the user should be serviced by this SER. If lookup_user() returns true and this particular user has a flag set which indicates routing through the extra SIP proxy I add the Route headers and pass the request to the 'FORWARD' route block where I do the t_relay().
My problem is that when I do t_relay() after inserting the Route headers into the request it is not forwarded according to the topmost Route header field value??
Is it bug in ser??
Please point me what do I missed.
Thank you in advance.
Tomasz
Thought I would let people know that (in case no one realised) SER -
Getting Started Guide (html and pdf versions) seems not to be accessible
any more. Seems all the links that point to onsip.org on
iptel.org/ser/doc/gettingstarted page can't load, yet onsip.org main
page can be loaded.
Use google cache in the mean time but not sure how up to date it is:
http://www.google.com/search?q=ser+gettingstarted&ie=utf-8&oe=utf-8&aq=t&rl…
--
Shaun Hofer
Email: shaun.hofer(a)voxpak.com
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Hello,
Is it possible to choose the protocol, without specifying the destination as we do with the t_relay()?
With the t_relay_to_[tcp|udp|tls] I can choose the protocol but I must specify a destination address in the config file.
Thanks
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Hi,
Since yesterday (I think) I always run into timeouts when trying to
access the SER bug tracker at iptel.org. Once I also got an error
response (after a long delay) which indicated that the server's
JVM was running out of memory.
Just wanted to report this in case nobody noticed it so far. Or am I the
only one having problems accessing the tracker?
Regards,
Jan
--
Jan Andres <jan.andres(a)freenet.ag>
Hi,
I am new to SER and have been having this problem for a while
which I'm not sure how to resolve. Any pointers would be greatly
appreciated
On the openser.cfg when I use the "forward" or the "send" calls
with an IP inside I see the following upon running openser ( Note I have
masked the real IP's)
./openser
Listening on
udp: 208.xxx.xx.xx
tcp: 208.xxx.xx.xx
WARNING: no fork mode
stateless - initializing
0(0) Maxfwd module- initializing
textops - initializing
0(0) AUTH module - initializing
0(0) AUTH_DB module - initializing
0(0) ERROR: mk_proxy: could not resolve hostname:
"sip:208.1xx.44.yy:5060"
ERROR: error -478 while trying to fix configuration
Anybody have any ideas? I have tried putting a DNS name to the IP
address as well but it comes back with the same error.
Regards
Piyush
Hello,
I have the following line in my script which causes OpenSER (1.2) to
crash when the execution gets there for an ACK.
if ($du=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)")
It gets at this line when an ACK is received and after loose_route()
returns true. So I am guessing the destination uri is null at that point
and the check causes OpenSER to crash.
If I add the following check then everything works fine.
if (isdsturiset() &&
$du=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)")
thank you
George
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Hi all,
I find a way to execute an external script when I get a trigger. The
trigger could be BYE or INVITE for example.
I read the documentation and some examples but I don't find tips to help
me.
Is it possible to have a "return false" in the script wich is executed
and use it in "if (exec())" in SER ? If it's possible I can do IF ELSE
in ser.cfg according to the return of the script.
The best should be that I return a value in the script and I will able
to test it in SER.
I run SER 0.9.6.
Thanks for your support and ideas.
Adrien
Hi, I'm starting with OpenSer 1.2.1-1 and have a strange problem. This occurs
always with users behind NAT with a mapped NAT port different of his device
SIP port. For example:
* user 807:
device: sip:192.186.0.204:5061
is mapped in the NAT router as:
sip:212.121.235.18:1055
So in "location" table I see:
---------------------------------
username contact received
807 sip:807@192.168.1.204:5061 sip:212.121.235.18:1055
800 sip:800@212.121.235.18:5080 NULL
---------------------------------
800 is a user with STUN.
I have some services in Asterisk accesible via public and private numbers in
OpenSer. Public numbers (500) don't need auth in INVITE.
When 807 call to any private number (or internal users as 800) OpenSer ask him
for auth, but OpenSer replies this messaje to a wrong port:
---------------------------------
# 807 INVITE 800
U 212.121.235.18:1055 -> 82.94.0.210:5060
INVITE sip:800@openser.domain.org SIP/2.0
# OpenSer 407
U 82.94.0.210:5060 -> 212.121.235.18:5061 <--- WRONG DST PORT
SIP/2.0 407 Proxy Authentication Required
---------------------------------
As you see, OpenSer send the 407 message to the NAT IP but to the device SIP
port instead of the NAT port. So 807 sends again and again the INVITE.
But if 807 calls to a public number (500) it get the call but when 807 does
BYE, the response to that BYE is again wrong:
---------------------------------
# 807 INVITE 500
U 212.121.235.18:1055 -> 82.94.0.210:5060
INVITE sip:500@openser.domain.org SIP/2.0
# OpenSer OK
U 82.94.0.210:5060 -> 212.121.235.18:1055
SIP/2.0 200 OK
# 807 BYE 500
U 212.121.235.18:1055 -> 82.94.0.210:5060
BYE sip:1800@82.94.0.210:5070 SIP/2.0
# OpenSer OK
U 82.94.0.210:5060 -> 212.121.235.18:5061 <--- WRONG DST PORT
SIP/2.0 200 OK
---------------------------------
I hope this is a bug in my OpenSer configuration. Please, could anyone tell me
with funcions related to this issue should I verify?
Thanks a lot.
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
Hi I am trying st set up media proxy on my sip express router setup.
this is the problem that I am running into.
I have placed Media proxy on a seperate box and I have configured the INI
file.
I am able to start the media proxy. I then go to the sip express router and
I add the proxydispatcher to the box.
the documentation that is on onsip.org does not tell me what directory this
should go into.
this is what I decided to do. I placed the file in /usr/share. I then
tried to start the proxydispatcher.py by typing
python proxydispatcher.py and it failed.
This is the error that I got.
# python proxydispatcher.py
Traceback (most recent call last):
File "proxydispatcher.py", line 31, in ?
import daemon
ImportError: No module named daemon
I am a little confused as to how to set up the whole media proxy nat
traversal.
Can any one let me know if I did this correctluy up to this point.
Part 2. I cant get any external phones to register onto the server. The
getting started config works fine.
I am now using a nathelper config and I cant get any external phones to
register.
Help help
Hi,
What is the use of the config item children in the openser.cfg file. How is it related to command line options
(-n process-no / -N tcp process-no).
Rgds,
Sreekanth