Hello,
Ok then, the patches should apply just fine -
https://github.com/OpenSIPS/opensips/commit/3316a2a518a2ac27401408369e4bd3adc70b4e48
was already backported, so just apply
https://github.com/OpenSIPS/opensips/commit/7989c4fcf1825afccb8102b65d94d66105dbdf33
and test it out.
Best Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com
On 25.08.2014 13:36, Satish Patel wrote:
I'm using 1.11 last week installed.
Sent from my iPhone
On Aug 25, 2014, at 6:34 AM, Vlad Paiu <[email protected]
<mailto:[email protected]>> wrote:
Hello,
What OpenSIPS version are you currently using ?
I've just committed a fix that implements a preliminary version of
this, see commits :
https://github.com/OpenSIPS/opensips/commit/3316a2a518a2ac27401408369e4bd3adc70b4e48
and
https://github.com/OpenSIPS/opensips/commit/7989c4fcf1825afccb8102b65d94d66105dbdf33
Please apply them to your sources and let me know how it oges
Best Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com
On 25.08.2014 13:31, Satish Patel wrote:
Great!! I can see light in tunnel now because last 1 week I tried
everything and now I was planing to go for B2B but I guess as you
said you guys working on so I'm holding my breath.
This is must needed solution because SIP service provide most of
time provide password to make outbound trunk call.
Sent from my iPhone
On Aug 24, 2014, at 11:13 PM, Bogdan-Andrei Iancu
<[email protected] <mailto:[email protected]>> wrote:
Hi Satish,
It is an known issue that OpenSIPS does not increases the cseq
number when performing UAC auth against another party. Asterisk
does not like that and consider the new branch INVITE with
credentials a simple retransmission (even if it has a different
VIA-branch :P) and discards them - this is why you get that timeout
from asterisk.
We have ongoing work (hopefully to be ready in 1-2 weeks) for
increasing the cseq number is a sip-wise manner. Just keep an eye
on the mailing list.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 25.08.2014 04:47, Satish Patel wrote:
I am seeing following and all transaction has CSeq: 2 INVITE, I
have notice one thing asterisk asking for 407 but opensips never
send any challenge response
Opensips ---> INVITE ---> Asterisk
Asterisk -----> 407 ------> Opensips (Proxy-Authenticate: Digest
algorithm=MD5, realm="asterisk", nonce="3a710e79".)
Opensips ----> ACK -----> Asterisk
Here opensips challenging SIP client and saying giving try to
asterisk and then following
Opensips ----> INVITE ---> Asterisk
Opensips ----> INVITE ----> Asterisk
Opensips ----> INVITE ----> Asterisk
After 3 tries opensips send SIP client 408 Request timeout..
On Sun, Aug 24, 2014 at 4:26 PM, Stefano Pisani
<[email protected] <mailto:[email protected]>>
wrote:
Check if the cseq was incremented by one in the second try.
Use ngrep.
Il 24/08/2014 22.24, Satish Patel ha scritto:
Hi,
my Opensips (UAC) registered to PSTN gateway and now i am
trying to call using my SIPphone which is register to opensip
but no success. I am getting 407 Proxy authentication
issue.. I am using following method but it didn't work. I
need solution badly..
PSTN gateway sending 407 Proxy auth and then my Opensip
sending 407 proxy auth to SIP phone.
Does anyone has any working example or some kind of document?
I haven;t see any single doc anywhere in Internet about
uac_auth issue
modparam("uac","credential","username:domain:password")
route {
....
t_on_failure("2");
t_relay( "udp:ip_addr:5060" );
...
}
failure_route[2] {
uac_auth();
t_relay("udp:ip_addr:5060");
}
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[email protected]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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