Hello!
Just curious: I do remove_hf("Contact") right after fix_nated_contact() (for
whatever reason).
Then the remove_hf() removes anything but the new SIP URI from the message,
leaving e.g.
sip:user@received_ip:port
in front of the headerfield following the Contact. e.g.
Before:
Some-Header: bla
Contact: Name <sip:user@privateip:port>
Authorization: bla
Then:
Some-Header: bla
sip:user@public_ip:portAuthorization: bla
How to remove that Contact: after having it nat_helped and then saved?
br
Walter
Hi friends,
I am new in OpenSER, I want to use it only for SIP Proxying with freeradius.
I made plan to install openser with freeradius on Virtual Server to get only
100 cuncurent calls.
1- Is it possible to install on Virtual Server?
2- Which Codecs or used, because i want to calculate the bandhwidth
according to the codec?
3- What RAM should be used to handel 100 cuncurent calls?
4- It can accept h323-credit-time from radius to control max credit call
time?
I will appriciate for your kind of suggestion.
Regards,
www.Go4Calls.Com
VoIP Forums
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Hi,
I am looking for a solution to analyze sip and rtp packets on a Ser + Mediaproxy server.
SIPAlanyzerV6 seems to receive remote capturing, but there is no information on its manuals for remote capture.
What is the best solution to analyze sip and rtp packets remotely from a SER server ?
I also would like to have some statistical output, if possible.
Thanks for your comments.
ilker
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Hi all,
how can i remove/add and reload the dispatcher module when it is in
service?
I've to servers with openser + dm which are dispatching the traffic
to two different OpenSERs which handle the registrations etc.
If one of this two servers dies, i want to have this one remove from
the dispatch Servers. How can i do that?
TIA
--
Mit freundlichen Grüßen / Kind regards
Dominik Bay
Cablesurf Technik
Hi,
I ve added /etc/openser/dictionary.radius to
/etc/radiusclient-ng/dictionary and
to BSDradius dictionary file too
Largo wrote:
> Hi
>
> Try to use the file provided in the folder "/usr/local/etc/openser"
> called "dictionary.radius" as a dictionary.
>
> Bye
>
> Cseke Tamas schrieb:
>
>> Hello,
>>
>> We 'd like to use openser with radius authentication, accounting
>>
>> We have installed openser from debian package on an ubuntu system:
>> deb http://www.openser.org/debian sarge main
>>
>> But radius authentication failed, and we need avp_radius module too,
>> wich is not included in this package i guess
>>
>> error message in the log:
>> ERROR:auth_radius:radius_authorize_sterman: rc_auth failed
>>
>> on http://openser.org/docs/openser-radius-1.0.x.html i ve 'read that
>> openser should be compiled (with some macros defined in makefile) to
>> work with
>> radius.
>> Is there any possibility to install some deb packages instead of
>> compilation?
>> Which package should we install?
>>
>> Thanks,
>> Tamas
>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users(a)openser.org
>> http://openser.org/cgi-bin/mailman/listinfo/users
>
>
>
Hello,erery one,
I want to use mysql's regexp operator in ser,but with db-api ,I
can't do this.Because the patten is stored in the mysql table.The db-api can
only do key op value.I want to do the value REGEXP key operation,
Any one has advices to me?
Thanks very much.
>Hi Greger. SER and Asterisk can be configured to use TCP for SIP/SDP
>messages?
>
Try this before forwarding the INVITE to Asterisk:
if(!uri_param("transport")) {
add_uri_param("transport=tcp"); }
>The reason I said that the caller wouldn't receive audio is because the
>callee's RTP stream would be directed to SER, not the caller.
>-- Nick
How come? I don't see rtpproxy or mediaproxy in your setup. So, unless you force an RTP proxy in your SER, I would expect Asterisk and caller to communicate directly on RTP.
g-)
Nick Hoffman wrote:
>> Nick Hoffman wrote:
>>
>>> Hi guys. Say you have this setup, with an account for the caller on
>>> both Asterisk and SER:
>>> Caller -> SER -> Asterisk -> VoIP Provider -> Callee
>>>
>>> If the caller were to spoof SER's IP address and place a call directly
>>> to Asterisk (thus circumventing SER), what would happen?
>>>
>>> If the call was in fact setup, obviously the caller would not receive
>>> any audio from the callee. However, would the call be setup? When
>>> Asterisk responds to the caller's request and sends SIP packets back
>>> (to SER), would SER say "I don't know anything about this call!
>>> Asterisk, kill this call please."?
>>>
>>> Thanks for your input!
>>> -- Nick
>>> e: nick.hoffman(a)altcall.com
>>> p: +61 7 5591 3588
>>> f: +61 7 5591 6588
>>>
>
>
> On Wed July 5 2006 17:58, "Greger V. Teigre" <greger(a)teigre.com> wrote:
>
>> Depends on the config and what type of message the caller managed to
>> make asterisk create... You could (and probably should) put asterisk on
>> a private routable network (i.e. NATed behind a firewall). The best
>> would be to put ser and asterisk on the same network and only allow
>> outside world to contact ser and let ser contact asterisk using the
>> private address of asterisk. Alternatively you could use tcp to
>> asterisk and stop udp traffic.
>> Why the caller wouldn't receive audio, I don't understand...
>> g-)
>>
>
>
> Hi Greger. SER and Asterisk can be configured to use TCP for SIP/SDP
> messages?
>
> The reason I said that the caller wouldn't receive audio is because the
> callee's RTP stream would be directed to SER, not the caller.
> -- Nick
> e: nick.hoffman(a)altcall.com
> p: +61 7 5591 3588
> f: +61 7 5591 6588
>
> If you receive this email by mistake, please notify us and do not make any
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>
Dear All,
I use openser 1.0.1 and use rtp proxy for some cases other than nat option. At voice calls I do not have any problem. But When I try to send fax,it is unsuccesful. I investigated the logs and saw that, after the reinvites ( to take the session from g729 to G711u bypass mode ), the last OK message's SDP that is sending from openser to user agent is not correct.Etheral logs are attached.
Is it a bug of rtpproxy ? Any suggestion.
Best Regards,
Hakan.
Hi users,
i have the task to implement "IP authentication for OPENSER using the database"IP are trusted IP's, I have the idea that it would be done through AVPOPS module but for this i need your help .
if any one implment AVPOPS module please send me OPENSER.cnf file (if possible) so that i will take idea about implementing the module
thanks a lot for your help,
TANZEEL
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