<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body dir="auto"><div>I'm using 1.11 last week installed. <br><br>Sent from my iPhone</div><div><br>On Aug 25, 2014, at 6:34 AM, Vlad Paiu <<a href="mailto:vladpaiu@opensips.org">vladpaiu@opensips.org</a>> wrote:<br><br></div><blockquote type="cite"><div>
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<div class="moz-cite-prefix">Hello,<br>
<br>
What OpenSIPS version are you currently using ?<br>
I've just committed a fix that implements a preliminary version of
this, see commits :<br>
<br>
<a class="moz-txt-link-freetext" href="https://github.com/OpenSIPS/opensips/commit/3316a2a518a2ac27401408369e4bd3adc70b4e48">https://github.com/OpenSIPS/opensips/commit/3316a2a518a2ac27401408369e4bd3adc70b4e48</a><br>
and<br>
<a class="moz-txt-link-freetext" href="https://github.com/OpenSIPS/opensips/commit/7989c4fcf1825afccb8102b65d94d66105dbdf33">https://github.com/OpenSIPS/opensips/commit/7989c4fcf1825afccb8102b65d94d66105dbdf33</a><br>
<br>
Please apply them to your sources and let me know how it oges<br>
Best Regards,<br>
<pre class="moz-signature" cols="72">Vlad Paiu
OpenSIPS Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a> </pre>
On 25.08.2014 13:31, Satish Patel wrote:<br>
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<div>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. </div>
<div><br>
</div>
<div>This is must needed solution because SIP service provide most
of time provide password to make outbound trunk call. <br>
<br>
Sent from my iPhone</div>
<div><br>
On Aug 24, 2014, at 11:13 PM, Bogdan-Andrei Iancu <<a moz-do-not-send="true" href="mailto:bogdan@opensips.org">bogdan@opensips.org</a>>
wrote:<br>
<br>
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<div class="moz-cite-prefix"><tt>Hi Satish,<br>
<br>
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.<br>
<br>
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.<br>
<br>
Regards,<br>
</tt>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a moz-do-not-send="true" class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a></pre>
On 25.08.2014 04:47, Satish Patel wrote:<br>
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<blockquote cite="mid:CAPgF-frYOOyynD-7fH1RWUYgLizSfU4yuuimPJE0+owTrLU6pA@mail.gmail.com" type="cite">
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<div>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 <br>
<br>
</div>
Opensips ---> INVITE ---> Asterisk<br>
</div>
Asterisk -----> 407 ------> Opensips
(Proxy-Authenticate: Digest algorithm=MD5,
realm="asterisk", nonce="3a710e79".)<br>
</div>
Opensips ----> ACK -----> Asterisk<br>
</div>
<br>
</div>
Here opensips challenging SIP client and saying
giving try to asterisk and then following<br>
<br>
</div>
Opensips ----> INVITE ---> Asterisk<br>
</div>
Opensips ----> INVITE ----> Asterisk <br>
</div>
Opensips ----> INVITE ----> Asterisk <br>
<br>
</div>
After 3 tries opensips send SIP client 408 Request
timeout.. <br>
</div>
<div class="gmail_extra"><br>
<br>
<div class="gmail_quote">On Sun, Aug 24, 2014 at 4:26 PM,
Stefano Pisani <span dir="ltr"><<a moz-do-not-send="true" href="mailto:stefano.pisani@omnianet.it" target="_blank">stefano.pisani@omnianet.it</a>></span>
wrote:<br>
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<div>Check if the cseq was incremented by one in the
second try.<br>
Use ngrep.<br>
<br>
<br>
<br>
Il 24/08/2014 22.24, Satish Patel ha scritto:<br>
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<div><br>
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Hi,<br>
<br>
</div>
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.. <br>
<br>
</div>
PSTN gateway sending 407 Proxy auth and
then my Opensip sending 407 proxy auth to
SIP phone. <br>
<br>
</div>
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<br>
<div>
<div>
<div><br>
<br>
<br>
<pre>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");
}
</pre>
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