forgot to cc the list
Bogdan-Andrei Iancu wrote:
> Hi Andreas,
>
> you are right - the problem was not the string itself, but the str
> itself. Anyhow, hopefully is fixed now.
>
> regards,
> bogdan
>
> Andreas Granig wrote:
>
>> Bogdan,
>>
>>> However, another problem accured.
>>>
>>> I see in my logs after an INVITE:
>>>
>>> set_dst_uri: Not enough memory
>>> lookup(): Failed to set dst_uri of Path
>>
>>
>>
>> In my original patch I made a copy of "route->nameaddr.uri" in
>> get_path_dst_uri() of modules/registrar/path.c and then free'd the
>> parsed route, but your optimization just points to that uri and also
>> free's that route, which invalidates the pointer to that uri, as far
>> as I understand.
>>
>> Can you please check that again, because if I comment out the
>> "free_rr(&route)", it does work (although it creates a mem-leak).
>>
>> Andy
>
>
>
>
Hi,
there is a way to accomplish the reverse function that group module do?
As i know the group module allow me to check if a user is inside a specified
group. How can i know in which group is a user without querying the system to
know if the user is in all the groups one by one?
Actually i'm doing it using avpops and setting an avp that tell me which is
the group of the user.
I was wandering if there is a more efficient (more performant) way.
Thanks,
bye,
Marcello
Hi All,
I found that PSTN GW usually put the "From: xxxx@ipaddress" as ANI (caller
ID) when it receive a SIP call. There is a function "rewritingURI" for
called URI, I am wandering if there was a function can do rewriting on the
the sorc "From" also.
For example:
user FROM:alice@abc.com make call to hugen(a)abf.com, can I rewrite the
alice(a)abc.com to 661234223(a)abc.com when I send INVITE to hugen(a)abf.com ?
Any help will be appreciated.
Steve
Hi,
before forwarding a call (INVITE) from the PSTN with flag "privacy=full"
in the Remote-Party-ID hf to a none-voice-licensed carrier I need to
make sure that the calling party number is completely wiped out from the
SIP requests.
I see this number at three places: the Remote-Party-ID hf itself, the
>From hf and the Contact hf.
The Remote-Party-ID hf can simply be dropped or replaced by something
anonymous.
The From hf can be rewritten using uac_replace_from.
But what about the Contact hf? This field is, as far as I can see,
required for routing purposes, to tell the called party side where to
send replies. So, if I would manage it to rewrite the field, I would
have to make sure, the replies sent to the rewritten address once again
will be rewritten to target for the meant destination.
Is there a best practice for this item?
Thanks,
Wolfgang
I've forgot to cc the list.
klaus
--------------------------------------------------------------------------
Hi Alan!
That's the correct behaviour. Please read RFC 3263. If the URI contains a
port, e.g. inside.acme.com:5060 there must not be an SRV lookup. Because
the SRV lookup leads to a port, which is already defined.
regards
klaus
On Tue, January 31, 2006 20:28, alan chiu said:
> Server: Sip EXpress router (0.8.12 (i86pc/solaris))
>
> when endpoint sends 18005551212(a)sip.acme.com, SER
> (without rewriting of R-URI, looks it up with SRV
> ***************
>
> 386.656875 192.168.1.246 -> 10.100.0.111 SIP/SDP
> Request: INVITE sip:18005551212@sip.acme.com, with
> session description
> 386.657307 10.100.0.111 -> 192.168.1.246 SIP Status:
> 100 trying -- your call is important to us
> 88.811658 10.100.0.111 -> 192.168.1.11 DNS Standard
> query SRV _sip._udp.sip.acme.com
> 98.825208 192.168.1.11 -> 10.100.0.111 DNS Standard
> query response, No such name
> 98.825496 10.100.0.111 -> 192.168.1.11 DNS Standard
> query A sip.acme.com
> 108.845136 192.168.1.11 -> 10.100.0.111 DNS Standard
> query response, No such name
> 108.845468 10.100.0.111 -> 192.168.1.11 DNS Standard
> query AAAA sip.acme.com
> 118.865215 192.168.1.11 -> 10.100.0.111 DNS Standard
> query response, No such name
> 416.711408 10.100.0.111 -> 192.168.1.246 SIP Status:
> 478 Unresolveable destination (478/TM)
> 407.921364 192.168.1.246 -> 10.100.0.111 SIP Request:
> ACK sip:18005551212@sip.acme.com
>
> ***********************************************************************************
>
>
> *************
> after rewritehostport("inside.acme.com:5060"), SER
> only does an A lookup
> ***************
>
> 424.851557 192.168.1.246 -> 10.100.0.111 SIP/SDP
> Request: INVITE sip:18005551212@nat.foo.com, with
> session descripti
> on
> 415.971953 10.100.0.111 -> 192.168.1.246 SIP Status:
> 100 trying -- your call is important to us
> 127.006508 10.100.0.111 -> 192.168.1.11 DNS Standard
> query A inside.foo.com
> 127.007557 192.168.1.11 -> 10.100.0.111 DNS Standard
> query response A 192.168.1.211
> 424.853464 10.100.0.111 -> 192.168.1.211 SIP/SDP
> Request: INVITE sip:18005551212@inside.foo.com:5060,
> with session descriptio
> n
> 415.973466 10.100.0.111 -> 192.168.1.211 SIP/SDP
> Request: INVITE sip:18005551212@inside.foo.com:5060,
> with session descriptio
> n
>
> ******************************************************************
>
> How can I get SER to consistently do SRV lookup after
> rewriting of R-URI?
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
Server: Sip EXpress router (0.8.12 (i86pc/solaris))
when endpoint sends 18005551212(a)sip.acme.com, SER
(without rewriting of R-URI, looks it up with SRV
***************
386.656875 192.168.1.246 -> 10.100.0.111 SIP/SDP
Request: INVITE sip:18005551212@sip.acme.com, with
session description
386.657307 10.100.0.111 -> 192.168.1.246 SIP Status:
100 trying -- your call is important to us
88.811658 10.100.0.111 -> 192.168.1.11 DNS Standard
query SRV _sip._udp.sip.acme.com
98.825208 192.168.1.11 -> 10.100.0.111 DNS Standard
query response, No such name
98.825496 10.100.0.111 -> 192.168.1.11 DNS Standard
query A sip.acme.com
108.845136 192.168.1.11 -> 10.100.0.111 DNS Standard
query response, No such name
108.845468 10.100.0.111 -> 192.168.1.11 DNS Standard
query AAAA sip.acme.com
118.865215 192.168.1.11 -> 10.100.0.111 DNS Standard
query response, No such name
416.711408 10.100.0.111 -> 192.168.1.246 SIP Status:
478 Unresolveable destination (478/TM)
407.921364 192.168.1.246 -> 10.100.0.111 SIP Request:
ACK sip:18005551212@sip.acme.com
***********************************************************************************
*************
after rewritehostport("inside.acme.com:5060"), SER
only does an A lookup
***************
424.851557 192.168.1.246 -> 10.100.0.111 SIP/SDP
Request: INVITE sip:18005551212@nat.foo.com, with
session descripti
on
415.971953 10.100.0.111 -> 192.168.1.246 SIP Status:
100 trying -- your call is important to us
127.006508 10.100.0.111 -> 192.168.1.11 DNS Standard
query A inside.foo.com
127.007557 192.168.1.11 -> 10.100.0.111 DNS Standard
query response A 192.168.1.211
424.853464 10.100.0.111 -> 192.168.1.211 SIP/SDP
Request: INVITE sip:18005551212@inside.foo.com:5060,
with session descriptio
n
415.973466 10.100.0.111 -> 192.168.1.211 SIP/SDP
Request: INVITE sip:18005551212@inside.foo.com:5060,
with session descriptio
n
******************************************************************
How can I get SER to consistently do SRV lookup after
rewriting of R-URI?
Hi Users, I should implement any RTP proxy solution for our SER
deployment to avoid NAT problem.
I have tried to look for in google some non subjective reasons to use
Mediaproxy or to use RTPproxy but I have not found anything.
I would like to know which solution is the best one, which one is most
common or witch one are you using, how it works and if its possible
some examples or documentation links.
Thxs in advance.
Dear all,
I am a student and am trying to setup a demonstration with TLS and DTLS
support between a SIP client and a Proxy.
Has anyone some information regarding adding DTLS support for a SIP
client/Proxy or a more generic one using OpenSSL. I just need it for
demonstration purposes and so error handling's if any need not be complete.
Just a simple support and a demonstration of both would do fine. If anyone
has any information regarding this, may be a how to start or any
introductory article would be of great help, kindly let me know. This would
be of great help. Any suggestions regarding the same are most welcome.
kindly let me know and thank you very much.
regards,
Pjothi
