Hi Users,
I am Using SER-0.9.6 on a public ip and using another sip-proxy(not
SER) behind the NAT.
and when the users who registerd to Sip-proxy which is behind NAT,, calls
to the users who registered to public domain SER.. call is being
established .
but when hunging the phone the SER is not sending
"BYE" messages to the NAT device it is sending to the Sip-proxy (not SER)
that is for a private IP
. so even when i hung the call from the Nated Sip proxy ,SER is
getting messages through NAT but again SER is not sending "BYE" messages to
other party its sending back to the nated sip-proxy having private IP , so
the messages are going in a wrong direction and is being wasted .
It looks like this is a BUG in SER , i would take this
oppurtunity to report this.
If this is not a bug please send me the suggestions .....
To solve my problem. thank you.
regards,
Ravi.
Hi users ,
I am proceeding in this way i have a SER-0.9.6 running fine :-)
and I had given a username and password to a call-shop and this callshop
owner with his username and password he connects to another 6 phones,
Actually he bills to his 6
phones and I will bill him ."o.k overall Scenario is well and good".
I made a little changes in ser.cfg and when the call made from the call shop
up to call connecting is O.K but when we hung the phone the SER is not
generating "BYE" messages to other party , so the call is on.. and i am not
getting "Acct Stop" packet also
SO How I can solve my problem :-(
any suggestions will be appreciated .
below is the message i am getting from SER when I hung the phone on one side
-------------------<this message is came from callshop Nat
address>---------<"it sends bye to my SER
"------------------------------------------------------
U 82.102.69.105:32768 -> 81.21.33.35:5060
BYE sip:99106883@81.21.33.35:5060 SIP/2.0.
To: "99106883"<sip:99106883@81.21.33.35:5060>;tag=78F9ECC4-166C.
From: "12345"<sip:12345@81.21.33.35:5060>;tag=837e5b2ff0b4cf4a.
Via: SIP/2.0/UDP 192.168.1.100:5060
;branch=z9hG4bK-d87543-5983e94f152d226c82a4e76799fe58e5-1--d87543-;rport.
Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK9bf920205fe3bef9.
Call-ID: f1b0fe2b6cdf1456(a)192.168.1.102.
CSeq: 9533 BYE.
Route: <sip:99106883@81.21.33.35:5060;lr;nat=yes;ftag=837e5b2ff0b4cf4a>.
Route: <sip:192.168.1.100:5060>.
Record-Route: <sip:192.168.1.100:5060>.
Contact: <sip:192.168.1.100:5060>.
Max-Forwards: 69.
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE.
Proxy-Authorization: Digest username="12345", realm="81.21.33.35",
algorithm=MD5, uri="sip:192.168.1.100:5060",
nonce="44fd415a6754afa39f2668ce2ad11da6bfc65cde",
response="4001812db780010a557b40683efd2f9e".
Supported: replaces.
User-Agent: Grandstream BT110 1.0.8.23.
Content-Length: 0.
.
----------------<here it is as soon as SER recieve Bye message it has to
send Bye to other party >------<But it is sending to the hung up phone
itself>---------
#
U 81.21.33.35:5060 -> 192.168.1.100:5060
BYE sip:192.168.1.100:5060 SIP/2.0.
Record-Route: <sip:81.21.33.35;ftag=837e5b2ff0b4cf4a;lr=on>.
To: "99106883"<sip:99106883@81.21.33.35:5060>;tag=78F9ECC4-166C.
From: "12345"<sip:12345@81.21.33.35:5060>;tag=837e5b2ff0b4cf4a.
Via: SIP/2.0/UDP 81.21.33.35;branch=z9hG4bK936f.7ff7572.0.
Via: SIP/2.0/UDP 192.168.1.100:5060;received=82.102.69.105
;branch=z9hG4bK-d87543-5983e94f152d226c82a4e76799fe58e5-1--d87543-;rport=32768.
Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK9bf920205fe3bef9.
Call-ID: f1b0fe2b6cdf1456(a)192.168.1.102.
CSeq: 9533 BYE.
Record-Route: <sip:192.168.1.100:5060>.
Contact: <sip:192.168.1.100:5060>.
Max-Forwards: 16.
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE.
Proxy-Authorization: Digest username="12345", realm="81.21.33.35",
algorithm=MD5, uri="sip:192.168.1.100:5060",
nonce="44fd415a6754afa39f2668ce2ad11da6bfc65cde",
response="4001812db780010a557b40683efd2f9e".
Supported: replaces.
User-Agent: Grandstream BT110 1.0.8.23.
Content-Length: 0.
.
-----------------<And here I go iam getting this message and the call is not
being stopped >----------------------------------
#
U 81.21.33.35:5060 -> 82.102.69.105:32768
SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL).
To: "99106883"<sip:99106883@81.21.33.35:5060>;tag=78F9ECC4-166C.
From: "12345"<sip:12345@81.21.33.35:5060>;tag=837e5b2ff0b4cf4a.
Via: SIP/2.0/UDP 192.168.1.100:5060
;branch=z9hG4bK-d87543-5983e94f152d226c82a4e76799fe58e5-1--d87543-;rport=32768;received=
82.102.69.105.
Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK9bf920205fe3bef9.
Call-ID: f1b0fe2b6cdf1456(a)192.168.1.102.
CSeq: 9533 BYE.
Content-Length: 0.
Warning: 392 81.21.33.35:5060 "Noisy feedback tells: pid=27773 req_src_ip=
82.102.69.105 req_src_port=32768 in_uri=sip:99106883@81.21.33.35:5060
out_uri=sip:192.168.1.100:5060 via_cnt==2".
.
any suggestions will be appreciated:
Thank You.
Hi SER Users,
I am using SER-0.9.6 on public domain . and I am using
another Sip-proxy behind NAT (ip-address=192.168.1.100:5060)
i have some clients registered to that nated sip-proxy and here that nated
sip-proxy forwards PSTN calls to SER on public domain.
when call make from client (registered to nated sip-proxy)
the session establishing suscessfully , but when the client hung the phone
the SER on public domain is taking the BYE message and is not sending the
BYE message to the other end and it is sending directly to the Sip-proxy
which is a private-ip . So, here SER is not getting ACK and its keep on
sending the BYE requests to the privateip instead of sending to the NAT
address .
In some docs which jiri wrote on record route he
mentioned the same problem below is the link ; but he dint mention how to
solve that record route problem, If SER send messages back as it recieving
path there is no problem , but its changing the route to send BYE ,
For the request "CANCEL","ACK","INVITE","200 OK" every
thing is happening in the same Route but it is not happening only for the
"BYE" request .
Hope some body can help me.
Thank You.
here is the link which i told;-
http://216.239.59.104/search?q=cache:A32Unl9P0hkJ:voip.internet2.edu/meetin…
Regards,
Ravi.
I've got a dumb question.
Can I get t_relay() functionality (transactions and whatnot)
without getting the Via and potentially Record-Route tags in
the header???
I warned you it was dumb....
-g
--
Greg Fausak
greg(a)thursday.com
Dear all,
Thank you very much for your help.
A few days ago, I have sent a message that to ask about openser admin installation. Thank you very much for your help. Thanks
Now, I have tried to install it again. But, When I compile the rubygerms-0.9.0.tar.gz, I got error message. This is the task that I have done before:
1. Untar the rubygerms packet
# tar -zxvf rubygems-0.9.0.tar.gz
# cd rubygems-0.9.0
# ruby setup.rb
# gem install --include-dependencies
When I type this command, "gem install --include-dependencies", I got error message.
This is the error message:
Attempting local installation of 'rails'
Local gem file not found: rails*.gem
Attempting remote installation of 'rails'
Updating Gem source index for: http://gems.rubyforge.org
ERROR: While executing gem ... (Gem::RemoteSourceException)
Error fetching remote gem cache: getaddrinfo: Temporary failure in name esolution
What`s wrong? I do hope anybody can help me to solve this problem. Please help me...Please help because of my lack skills..Please
Note:
1. I do not have connection through the internet, so for rails packet, I have downloaded the rails-1.1.6.tar.gz.
I have untar it in root directory(same as rubygems) and type command
rails # script/server
But, Nothing happens. Why? Please...
Thank you very much
Regards,
Ferianto
---------------------------------
Do you Yahoo!?
Next-gen email? Have it all with the all-new Yahoo! Mail.
Hi,
Does anyone can help me to choose best NAT solution for SER/OpenSER
?NatHelper or Mediaproxy ?
And if they compare with STUN, TURN, ICE technology Nathelper or Mediaproxy
better than them ?
My target is I want my SIP Server can pass all NAT/Firewall just like SkyPE
which hard to be block by firewall and can pass all NAT system.
~Asep
hi,
I'm new to SER. I downloaded SER0.9.6 source to my local Fedora Core 5. I
run make all and make prefix=/ install, and then set SIPDOMAIN, and run ser.
However it didn't respond with incomging Registration(it showed incoming
Registration from Ethereal). (I used default ser.cfg). Can anybody tell me
what's the problem, or how to debug?
Thanks so much!
Henry
Hi Users,
Can any one tell about OSS ( Operation Support System) on openSER,
Why we need ?
But i think to reduce the load balance on openSER server,
Suppose is I use 3 OSS's on SIP server.
Sip users should also insert in to 3 OSS sip.
And other doubt is ?
Suppose if we using the redirect , that forwording into other openser
system,
When openSER redirect to other openSER, the user should also insert,
But i think that .... OSS and redirect on openSER are the same.
Please for me in English, I know its naughty questioin, please excuse me !
--
--
Thanks and Regards with cheers
Sunkara Ravi Prakash (Voip Developer)
Hyperion Technology
www.hyperion-tech.com
+91-9985077535
Hi All,
I am new to OpenSER/SER but have been receiving quite a bit of help and
would like to express my gratitude by letting you all know of the FREE DNS
Service that I am offering to the Internet Community. Go to
http://www.t4tm.net to find out more, signup and enjoy the benefits.
Tracy
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http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=…
Hi all!
I have just started using the SIP Express Router and I am looking for a
way to have a user preregistered when the router starts. Is there any
way to configure that user1 should be registered by default at an IP
address without needing to send the registration? I guess it will have
to do with messing a bit with the routing engine, but I don't dare to
touch it so far and I don't know how to do it. Any help would be very
much appreciated!
Thanks very much for any help!
/Nacho
---
,,,
(o o)
--ooO--(_)---Ooo-----------
Three may keep a secret, if two of them are dead.-
--oo0--------0oo-----------