Folks,
I've just updated rtpproxy tarball available at
http://www.portaone.com/~sobomax/rtpproxy.tar to include ability to
alter location of the command socket (userful if there are multiple SERs
running on the same machine, appropriate patches for nathelper were
commited into trunk some time ago), as well as an ability to set
specific TOS for RTP packets (default is now 0xa0), which can be used to
do QoS.
Regards,
Maxim
I am in the process of establishing a SIP-based system using SER which will allow my end users to connect to each other and to dial out to the pstn using a SIP/pstn gateway run by a provider downstream.
Some of the users on my system are consultants who work on several projects at once. They would like to prefix the numbers they dial with their own arbitrary 4-digit project codes for different calls and have the accounting I send them show the project code they used for each call. Since my accounting is done downstream at the pstn gateway, and it shows the username for each call, I would like to change the username on each of the calls going out of my SER to include the project code. Then no changes would need to be made in the billing software and each call will be easy to identify as to user and project.
Of course the project code would get stripped off so that just the correct uri is sent, and I have figured out how to do that using the strip function. But I am new to SER and just figuring my way through this, so I need a bit of help with the rest of it.
Here is an example of what I want to set up.
If user Mike wishes to call 1-111-1111 and wants to code it to project 1234, he would dial:
*123411111111(a)wherever.com
and the outgoing uri would be just:
11111111(a)wherever.com (this is done using strip)
and the outgoing username would be:
mike1234
Note: the asterisk just indicates that a project code follows. Not all users will want this feature and calls without an asterisk will be handled differently.
If I can do the last part (change the outgoing username to mike1234) I'll have it cracked.
It would be best to do this within SER, but I don't see a way to do it. If there is a way, please let me know.
So then it seems to me that the best thing might be to use the exec_dst command to pass a set of commands to the shell would cause characters 2 through 5 of $SIP_ORURI to be appended to the end of $SIP_USER. So something like:
EXPORT $SIP_USER=(characters 2-5 of $SIP_ORURI) ...... Obviously my bash-knowledge needs a bit of work here ;-) If this would work, can someone tell me the shell commands to use?
Alternatively, what about calling a small executable that reads the appropriate environment variables and changes the value of $SIP_USER before returning to SER? I had at first thought this might be the simplest way to do it, but now I understand that when a program modifies an environment variable the variable returns to its original value when the child process ends. If that is correct, then SER would never see the change.
Note that I'm aware that it would be easier to do this accounting on my own SER system, where I have access to the original user name. However, since the pstn gateway at the next hop does the official accounting and billing, I want the billing it produces to include the project codes, and I think the easiest way to do this is for me to modify the outgoing username.
I have been working away at this, starting from zero knowledge of Linux, but I've come about as far as I can get. I would appreciate any advice.
Thanks,
Mike
That's caused by new Messenger's _proprietary_ session model -- messaging
session are initiated and expected to be initiated with INVITE.
There is a hint how to avoid the 487 in our FAQ
http://iptel.org/~faqomatic/fom-serve/cache/64.html
Nevertheless, it only softens some scenarios, as said the core problem is
the whole session model is violently proprietary. From standard compliance
(and related capability to talk to anything else then Messenger itself),
4.7 was much better.
-jiri
At 10:05 PM 10/17/2003, David Quenzler wrote:
>I've seen similar postings before, but could not find the resolution.
>
>SER 0.8.11
>Windows Messenger 5.0.0381
>
>Connecting two clients to the SER machine... both show online in their
>messenger windows.
>
>Testcase
>SER: 'ngrep -n 5060 user1'
>USER2: attempts IM 'hello world' to USER1
>
>Flow control as seen with ngrep
>
>User2 SER User1
>1> invite
> 100 <2
> 3> invite
> bye <4
> bye <5
>6> 481
> 7> 481
> 200 <8
> 200 <9
>10> ack
> 11> ack
>12> message
> 13> message
> 481 <14
> 481 <15
>
>Has anyone seen this before?
>
>- Dave
>
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
I've seen similar postings before, but could not find the resolution.
SER 0.8.11
Windows Messenger 5.0.0381
Connecting two clients to the SER machine... both show online in their
messenger windows.
Testcase
SER: 'ngrep -n 5060 user1'
USER2: attempts IM 'hello world' to USER1
Flow control as seen with ngrep
User2 SER User1
1> invite
100 <2
3> invite
bye <4
bye <5
6> 481
7> 481
200 <8
200 <9
10> ack
11> ack
12> message
13> message
481 <14
481 <15
Has anyone seen this before?
- Dave
Hello All,
Can anyone tell me whether the following setup can be achieved with SER?
[UA] <-------> [Gateway/NAT/SER Proxy] <--------> [SER Registrar]
1. The UA is behind a NAT gateway.
2. Whenever the SER Proxy on the NAT Gateway receives a registration request from a UA ( tester@gateway )on the private network, it sends a registration request ( tester@ser-registrar ) to the "SER Registrar" on behalf of the UA. This means the SER Proxy should be able to translate a SIP URI on the public side ( tester@ser-registrar ) to one on the private side ( tester@gateway ) and forward it to the right machine.
Any feedback will be greatly appreciated.
Regards,
Dhiraj Bhuyan
Security Research Engineer
BTexact
Tel: +44 1473 643932
Mob: +44 7753231984
Email: dhiraj.2.bhuyan(a)bt.com
Hi,
I have installed a ser on my one linux-machine in a LAN which also has an
other Linux-machine running DNS server.
SIP server has a domain like, sip.ur_net.lab and its ip address is
10.3.0.1
i can resolve the sip.ur_net.lab to 10.3.0.1 on the this sip machine by
running command
dig sip.ur_net.lab at the command line.
Now i have connected two windows hosts to same network and installed SIP
user agents on these windows hosts.
now in client when i use my sip domain and sip proxy as sip.ur_net.lab,
the server does not register me.
Also on windows hosts i can resolve sip.ur_net.lab to 10.3.0.1 by using
ping sip.ur_net.lab.
But in windows host when i use 10.3.0.1 as my sip domain and sip proxy
then all things go ok and we can talk through sip user agents.
Do i have to add some special entry in DNS server for
username(a)sip.ur_net.lab to resolve to username(a)10.3.0.1 or any other
suggestion.
Thanks.
-ajeet.
--
Ajeet
ajeetnk(a)fastmail.fm
--
http://www.fastmail.fm - Accessible with your email software
or over the web
Thanks for the reply.
A quick google search and I found one B2BUA from http://www.vovida.org/applications/downloads/b2bua/
Is there any other open source B2BUA for Linux/Unix platform?
Thanks,
Dhiraj
-----Original Message-----
From: Jan Janak [mailto:jan@iptel.org]
Sent: 17 October 2003 15:25
To: Bhuyan,D,Dhiraj,XVR3 R
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] Outbound Proxy
Hello,
no, it is not possible because you can't change To and From header
fields in the message and also the gateway ser can't respond to digest
challenge from the registrar because it doesn't know user's credentials.
You will need a B2BUA for this kind of job.
Jan.
On 17-10 14:43, dhiraj.2.bhuyan(a)bt.com wrote:
> Hello All,
>
> Can anyone tell me whether the following setup can be achieved with SER?
>
>
> [UA] <-------> [Gateway/NAT/SER Proxy] <--------> [SER Registrar]
>
>
> 1. The UA is behind a NAT gateway.
> 2. Whenever the SER Proxy on the NAT Gateway receives a registration request from a UA ( tester@gateway )on the private network, it sends a registration request ( tester@ser-registrar ) to the "SER Registrar" on behalf of the UA. This means the SER Proxy should be able to translate a SIP URI on the public side ( tester@ser-registrar ) to one on the private side ( tester@gateway ) and forward it to the right machine.
>
>
> Any feedback will be greatly appreciated.
>
> Regards,
>
> Dhiraj Bhuyan
> Security Research Engineer
> BTexact
>
> Tel: +44 1473 643932
> Mob: +44 7753231984
> Email: dhiraj.2.bhuyan(a)bt.com
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
Dear Pablo,
we are also setting up VM/SER, so pls. do send the configs if not to the list at least to me.
Regards,
Harry
Dr. Harry Behrens
Projektleitung BIB3R
DAI Labor - Technische Universität Berlin
Sekretariat GOR 1-1, Franklinstrasse 28/29, 10587 Berlin
Fon: +49 30 314 23383
Fax: +49 30 314 21799
Email: harry.behrens(a)dai-labor.de
http://www.dai-labor.de
> -----Ursprüngliche Nachricht-----
> Von: Pablo Murillo [mailto:ser@rednet.com.ar]
> Gesendet: Freitag, 17. Oktober 2003 16:57
> An: serusers(a)lists.iptel.org
> Betreff: [Serusers] Finally, ser & vm working ok
>
>
> Hi
>
> Finally I made ser w/vm works ok
> I did it with two instances of ser running
> - One to process all connections
> - Other only for VM w/ans_machine
>
> Works on not on-line or not answer
>
> If it's ok, I can send all the cfgs to the list
>
>
> Pablo Murillo
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
>
>
Thanks for that, Jiri.
Jaime
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Hi all,
We're looking for informations regarding an INVITE action we would like to
plan for SER server;
We're manufacturing a new SIP phone, and in case of DSL-modem router, we're
developing a special REGISTER with 2 Contacts for the same SIP phone:
A public IP address, and a private IP address. In each case, SER server is
placed on external part.
Here is a scheme of what is done:
@IP1
SIP phone1-----| @IPWan
|---[modem-routeur]-----------(......INTERNET.......)-----------------[Exter
nal SER server]
SIP phone2-----|
Location Table : SIP phone-1 <=> {@IP1 ; @IPWan}
@IP2
SIP phone-2 <=> {@IP2 ; @IPWan}
What we want to do is following:
When SIP phone1 is looking for a call-session with SIP phone2, we want SER
server to compare WAN contacts in mysql location table, and in case of same
Response (=@IPWan),
it will redirect INVITE by sending a "302 Moved Temporarly" to SIP-phone1,
with a special new contact as SIP-phone2 private IP address (@IP2).
Could anyone ever done that before?
Any ideas are welcome.
Really thanks for your support.
Best Regards.
Marc