<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body dir="auto"><div>I downloaded latest repo from git whatever you mentioned. At compile time I got following error when it was compiling uac.c </div><div><br></div><div>Fatal error: auth.h file and directory not found. </div><div><br></div><div>Does it require any dependencies? <br><br>Sent from my iPhone</div><div><br>On Aug 25, 2014, at 8:10 AM, Vlad Paiu <<a href="mailto:vladpaiu@opensips.org">vladpaiu@opensips.org</a>> wrote:<br><br></div><blockquote type="cite"><div>
<meta content="text/html; charset=UTF-8" http-equiv="Content-Type">
<div class="moz-cite-prefix">Hello,<br>
<br>
Ok then, the patches should apply just fine - <br>
<a moz-do-not-send="true" class="moz-txt-link-freetext" href="https://github.com/OpenSIPS/opensips/commit/3316a2a518a2ac27401408369e4bd3adc70b4e48">https://github.com/OpenSIPS/opensips/commit/3316a2a518a2ac27401408369e4bd3adc70b4e48</a><br>
<br>
was already backported, so just apply <br>
<a moz-do-not-send="true" class="moz-txt-link-freetext" href="https://github.com/OpenSIPS/opensips/commit/7989c4fcf1825afccb8102b65d94d66105dbdf33">https://github.com/OpenSIPS/opensips/commit/7989c4fcf1825afccb8102b65d94d66105dbdf33</a><br>
<br>
and test it out.<br>
<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 13:36, Satish Patel wrote:<br>
</div>
<blockquote cite="mid:1F2837FB-C065-410A-B3E8-F90B1F75F635@gmail.com" type="cite">
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<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 moz-do-not-send="true" href="mailto:vladpaiu@opensips.org">vladpaiu@opensips.org</a>>
wrote:<br>
<br>
</div>
<blockquote type="cite">
<div>
<meta content="text/html; charset=UTF-8" http-equiv="Content-Type">
<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 moz-do-not-send="true" 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 moz-do-not-send="true" 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 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 13:31, Satish Patel wrote:<br>
</div>
<blockquote cite="mid:637B9FCF-B53C-4793-B34C-C1B4C23A82C6@gmail.com" type="cite">
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charset=UTF-8">
<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>
</div>
<blockquote type="cite">
<div>
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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>
</div>
<blockquote cite="mid:CAPgF-frYOOyynD-7fH1RWUYgLizSfU4yuuimPJE0+owTrLU6pA@mail.gmail.com" type="cite">
<div dir="ltr">
<div>
<div>
<div>
<div>
<div>
<div>
<div>
<div>
<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>
<blockquote class="gmail_quote" style="margin:0 0
0 .8ex;border-left:1px #ccc
solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
<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>
</div>
<blockquote type="cite">
<div>
<div class="h5">
<div dir="ltr">
<div>
<div>
<div>
<div><br>
</div>
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>
</div>
</div>
</div>
</div>
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