Re: [asterisk-users] sipgate outgoing calls

Miguel Oyarzo Thu, 19 Sep 2013 02:44:45 -0700


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, gpxctawjc...@irational.org wrote:

On Thu, 19 Sep 2013, David Duffett wrote:


i am getting these errors in asterisk cli

-- Executing [01179553708@default:1] Set("SIP/xxxx-0000015b","CALLERID(num)=xxxxxx") in new stack-- Executing [01179553708@default:2] Dial("SIP/xxxx-0000015b","SIP/01179553708@sipgate,30,trg") in new stack

  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
    -- Called 01179553708@sipgate

[Sep 19 10:08:03] NOTICE[28232]: chan_sip.c:17885handle_response_invite: Failed to authenticate on INVITE to '"xxxx"<sip:xx...@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 notwork.


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 theremay be

an issue in the general section of your sip.conf

Assuming calls do work, you can then add any other configurationlines youfeel are necessary - but remember, as with all Asterisk configurationfiles,

less is more :-)

On 18 Sep 2013 22:06, "Administrator TOOTAI" <ad...@tootai.net> wrote:
      Le 18/09/2013 15:29, gpxctawjc...@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@sipgate
            [Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885
            handle_response_invite: Failed to authenticate on
            INVITE to
            '"01179553708"
            <sip:sip...@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-passw...@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-passw...@sipgate.de/SIP-ID ;don't accept to
      register without FQDN

      Hope that help

      --
      Daniel

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Re: [asterisk-users] sipgate outgoing calls

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