[asterisk-users] sipgate outgoing calls
Miguel Oyarzo
miguelaustro at gmail.com
Thu Sep 19 04:43:29 CDT 2013
It looks like the challenge response after INVITE is not been accepted.
Provide more detail.
$> sip set debug peer sipgate
--
==================================
Miguel Oyarzo
DevOps Engineer
http://www.linkedin.com/in/mikeaustralia
Linux User: # 483188 - counter.li.org
Melbourne, Australia
On 9/19/2013 7:10 PM, gpxctawjc5oh at irational.org wrote:
> On Thu, 19 Sep 2013, David Duffett wrote:
>
>
> i am getting these errors in asterisk cli
>
> -- Executing [01179553708 at default:1] Set("SIP/xxxx-0000015b",
> "CALLERID(num)=xxxxxx") in new stack
> -- Executing [01179553708 at default:2] Dial("SIP/xxxx-0000015b",
> "SIP/01179553708 at sipgate,30,trg") in new stack
> == Using SIP RTP CoS mark 5
> == Using SIP VRTP CoS mark 6
> -- Called 01179553708 at sipgate
> [Sep 19 10:08:03] NOTICE[28232]: chan_sip.c:17885
> handle_response_invite: Failed to authenticate on INVITE to '"xxxx"
> <sip:xxxxx at sipgate.co.uk>;tag=as055d9532'
> -- SIP/sipgate-0000015c is circuit-busy
> == Everyone is busy/congested at this time (1:0/1/0)
>
> any further ideas ?
>
> many thanks
>
>>
>> I believe registration is in place, otherwise inbound calls would not
>> work.
>>
>> Also, registration is not required for outbound calls to work.
>>
>> I would suggest cutting down your sip.conf profile to this minimal
>> configuration:
>>
>> host=sipgate.co.uk
>> username=xxxxxxx
>> fromuser=xxxxxxx
>> insecure=invite,port
>> secret=xxxxxxx
>> context=my-inbound-context
>> type=peer
>>
>> If outbound calls still do not with this, I would suggest that there
>> may be
>> an issue in the general section of your sip.conf
>>
>> Assuming calls do work, you can then add any other configuration
>> lines you
>> feel are necessary - but remember, as with all Asterisk configuration
>> files,
>> less is more :-)
>>
>> On 18 Sep 2013 22:06, "Administrator TOOTAI" <admin at tootai.net> wrote:
>> Le 18/09/2013 15:29, gpxctawjc5oh at irational.org a écrit :
>> Hello
>>
>>
>> Hi
>>
>>
>> i am trying to setup sipgate gateway
>>
>> i can get incoming calls fine, but when i dial in and
>> then try to dial
>> out i get this in asterisk command line
>>
>> -- Called 01179248615 at sipgate
>> [Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885
>> handle_response_invite: Failed to authenticate on
>> INVITE to
>> '"01179553708"
>> <sip:SIP-ID at sipgate.co.uk>;tag=as30eb9dd1'
>> -- SIP/sipgate-0000014d is circuit-busy
>> == Everyone is busy/congested at this time
>> (1:0/1/0)
>>
>>
>> here is my sip.conf file
>>
>>
>> [general]
>> port = 5060
>> bindaddr = 0.0.0.0
>> context=default
>> qualify=no
>> disallow=all
>> allow=alaw
>> allow=ulaw
>> allow=g729
>> allow=gsm
>> allow=slinear
>> srvlookup=yes
>> videosupport=yes
>> alwaysauthreject=yes
>>
>> register => SIP-ID:SIP-Password at sipgate.co.uk/SIP-ID
>>
>> [sipgate]
>> type=peer
>> secret=SIP_PASSWORD
>> insecure=invite
>> username=SIP-ID
>> defaultuser=SIP-ID
>> fromuser=SIP-ID
>> context=sipgate_in
>> fromdomain=sipgate.co.uk
>> host=sipgate.co.uk
>> outboundproxy=proxy.live.sipgate.co.uk
>> qualify=yes
>> disallow=all
>> allow=alaw
>> dtmfmode=rfc2833
>>
>>
>> SIP-ID:SIP-Password
>> obviously, i replace these with my login details
>>
>> but, are these the same thing ?
>> SIP-Password
>> SIP_PASSWORD
>>
>> the sipgate guides are contradictory
>>
>> http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk
>> http://www.live.sipgate.co.uk/faq/article/508/How_do_I_configure_Asteri
>> sk
>>
>>
>> any suggestions ?
>>
>> Many thanks
>>
>>
>> My setup with sipgate.de
>>
>> [sipgate]
>> type=peer
>> secret=MY-PASSWORD
>> defaultuser=SIP-ID
>> host=217.10.79.9
>> fromuser=SIP-ID
>> fromdomain=sipgate.de
>> context=incoming-sipgate
>> ;qualify=900
>> dtmfmode=info
>> directmedia=yes
>> insecure=port,invite
>> disallow=all
>> allow=ulaw,alaw
>> accountcode=MY-ACCOUNTCODE
>>
>> What you forget is to register with them:
>>
>> ; Sipgate
>> register => SIP-ID:MY-PASSWORD at sipgate.de/SIP-ID ;don't accept to
>> register without FQDN
>>
>> Hope that help
>>
>> --
>> Daniel
>>
>> --
>> _____________________________________________________________________
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>>
>>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
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