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<br>
It looks like the challenge response after INVITE is not been
accepted.<br>
<br>
Provide more detail.<br>
<br>
$> sip set debug peer sipgate<br>
<br>
<br>
<pre class="moz-signature" cols="72">--
==================================
Miguel Oyarzo
DevOps Engineer
<a class="moz-txt-link-freetext" href="http://www.linkedin.com/in/mikeaustralia">http://www.linkedin.com/in/mikeaustralia</a>
Linux User: # 483188 - counter.li.org
Melbourne, Australia
</pre>
<br>
<br>
<div class="moz-cite-prefix">On 9/19/2013 7:10 PM,
<a class="moz-txt-link-abbreviated" href="mailto:gpxctawjc5oh@irational.org">gpxctawjc5oh@irational.org</a> wrote:<br>
</div>
<blockquote
cite="mid:alpine.DEB.2.00.1309191009180.14092@server.irational.org"
type="cite">On Thu, 19 Sep 2013, David Duffett wrote:
<br>
<br>
<br>
i am getting these errors in asterisk cli
<br>
<br>
-- Executing [01179553708@default:1] Set("SIP/xxxx-0000015b",
"CALLERID(num)=xxxxxx") in new stack
<br>
-- Executing [01179553708@default:2] Dial("SIP/xxxx-0000015b",
"SIP/01179553708@sipgate,30,trg") in new stack
<br>
== Using SIP RTP CoS mark 5
<br>
== Using SIP VRTP CoS mark 6
<br>
-- Called 01179553708@sipgate
<br>
[Sep 19 10:08:03] NOTICE[28232]: chan_sip.c:17885
handle_response_invite: Failed to authenticate on INVITE to
'"xxxx" <a class="moz-txt-link-rfc2396E" href="sip:xxxxx@sipgate.co.uk"><sip:xxxxx@sipgate.co.uk></a>;tag=as055d9532'
<br>
-- SIP/sipgate-0000015c is circuit-busy
<br>
== Everyone is busy/congested at this time (1:0/1/0)
<br>
<br>
any further ideas ?
<br>
<br>
many thanks
<br>
<br>
<blockquote type="cite">
<br>
I believe registration is in place, otherwise inbound calls
would not work.
<br>
<br>
Also, registration is not required for outbound calls to work.
<br>
<br>
I would suggest cutting down your sip.conf profile to this
minimal
<br>
configuration:
<br>
<br>
host=sipgate.co.uk
<br>
username=xxxxxxx
<br>
fromuser=xxxxxxx
<br>
insecure=invite,port
<br>
secret=xxxxxxx
<br>
context=my-inbound-context
<br>
type=peer
<br>
<br>
If outbound calls still do not with this, I would suggest that
there may be
<br>
an issue in the general section of your sip.conf
<br>
<br>
Assuming calls do work, you can then add any other configuration
lines you
<br>
feel are necessary - but remember, as with all Asterisk
configuration files,
<br>
less is more :-)
<br>
<br>
On 18 Sep 2013 22:06, "Administrator TOOTAI"
<a class="moz-txt-link-rfc2396E" href="mailto:admin@tootai.net"><admin@tootai.net></a> wrote:
<br>
Le 18/09/2013 15:29, <a class="moz-txt-link-abbreviated" href="mailto:gpxctawjc5oh@irational.org">gpxctawjc5oh@irational.org</a> a écrit :
<br>
Hello
<br>
<br>
<br>
Hi
<br>
<br>
<br>
i am trying to setup sipgate gateway
<br>
<br>
i can get incoming calls fine, but when i dial in
and
<br>
then try to dial
<br>
out i get this in asterisk command line
<br>
<br>
-- Called 01179248615@sipgate
<br>
[Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885
<br>
handle_response_invite: Failed to authenticate on
<br>
INVITE to
<br>
'"01179553708"
<br>
<a class="moz-txt-link-rfc2396E" href="sip:SIP-ID@sipgate.co.uk"><sip:SIP-ID@sipgate.co.uk></a>;tag=as30eb9dd1'
<br>
-- SIP/sipgate-0000014d is circuit-busy
<br>
== Everyone is busy/congested at this time
<br>
(1:0/1/0)
<br>
<br>
<br>
here is my sip.conf file
<br>
<br>
<br>
[general]
<br>
port = 5060
<br>
bindaddr = 0.0.0.0
<br>
context=default
<br>
qualify=no
<br>
disallow=all
<br>
allow=alaw
<br>
allow=ulaw
<br>
allow=g729
<br>
allow=gsm
<br>
allow=slinear
<br>
srvlookup=yes
<br>
videosupport=yes
<br>
alwaysauthreject=yes
<br>
<br>
register =>
<a class="moz-txt-link-abbreviated" href="mailto:SIP-ID:SIP-Password@sipgate.co.uk/SIP-ID">SIP-ID:SIP-Password@sipgate.co.uk/SIP-ID</a>
<br>
<br>
[sipgate]
<br>
type=peer
<br>
secret=SIP_PASSWORD
<br>
insecure=invite
<br>
username=SIP-ID
<br>
defaultuser=SIP-ID
<br>
fromuser=SIP-ID
<br>
context=sipgate_in
<br>
fromdomain=sipgate.co.uk
<br>
host=sipgate.co.uk
<br>
outboundproxy=proxy.live.sipgate.co.uk
<br>
qualify=yes
<br>
disallow=all
<br>
allow=alaw
<br>
dtmfmode=rfc2833
<br>
<br>
<br>
SIP-ID:SIP-Password
<br>
obviously, i replace these with my login details
<br>
<br>
but, are these the same thing ?
<br>
SIP-Password
<br>
SIP_PASSWORD
<br>
<br>
the sipgate guides are contradictory
<br>
<br>
<a class="moz-txt-link-freetext" href="http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk">http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk</a>
<br>
<a class="moz-txt-link-freetext" href="http://www.live.sipgate.co.uk/faq/article/508/How_do_I_configure_Asteri">http://www.live.sipgate.co.uk/faq/article/508/How_do_I_configure_Asteri</a>
<br>
sk
<br>
<br>
<br>
any suggestions ?
<br>
<br>
Many thanks
<br>
<br>
<br>
My setup with sipgate.de
<br>
<br>
[sipgate]
<br>
type=peer
<br>
secret=MY-PASSWORD
<br>
defaultuser=SIP-ID
<br>
host=217.10.79.9
<br>
fromuser=SIP-ID
<br>
fromdomain=sipgate.de
<br>
context=incoming-sipgate
<br>
;qualify=900
<br>
dtmfmode=info
<br>
directmedia=yes
<br>
insecure=port,invite
<br>
disallow=all
<br>
allow=ulaw,alaw
<br>
accountcode=MY-ACCOUNTCODE
<br>
<br>
What you forget is to register with them:
<br>
<br>
; Sipgate
<br>
register => <a class="moz-txt-link-abbreviated" href="mailto:SIP-ID:MY-PASSWORD@sipgate.de/SIP-ID">SIP-ID:MY-PASSWORD@sipgate.de/SIP-ID</a> ;don't
accept to
<br>
register without FQDN
<br>
<br>
Hope that help
<br>
<br>
--
<br>
Daniel
<br>
<br>
--
<br>
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<br>
</blockquote>
<br>
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