Incoming anonymous calls are dropped

Been customer with Callcentric for a while but has always had the following problem. Whenever we receive an anonymous call it is dropped as soon as we try to answer it. On our Danish phone number through Telsome anonymous calls work just fine.

Calls to 777-xxx-xxxx and 864-xxx-xxxx all comes through the Callcentric account and it is of course the 864-number people use to call us mostly.

Can anyone give a hint if this a configuration error or maybe on Callcentric’s end? Already been in contact with Callcentric and they could give no clue on what the problem was.

If sip debug is needed I will post that as well. Asterisk is version 1.4.21.2 from Debian Lenny.

— sip.conf —
[general]
context=sip
registertimeout=15
registerattempts=0
;; #######
;; ACTIVE
;; #######
register => 49xxxxxx:xxxxxx@telsome/49xxxxxx
register => 1777xxxxxxx:xxxxxx@callcentric/1777xxxxxxx

realm=hjarding.dk ; Realm for digest authentication
useragent="Personal PBX"
port=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
Srvlookup=yes ; Enable DNS SRV lookups on outbound calls

disallow=all ; First disallow all codecs
allow=ulaw ; Allow codecs in order of preference
language=en ; Default language setting for all users/peers

dtmfmode = rfc2833

[telsome]
type=peer
host=voip9.telsome.com
dtmfmode=auto
canreinvite=no
username=49xxxxxx
secret=xxxxxx
context=incoming
fromuser=49xxxxxx
fromdomain=telsome.com
insecure=port,invite
nat=no

[callcentric]
type=peer
context=incoming
host=callcentric.com
username=1777xxxxxxx
secret=xxxxx
fromuser=1777xxxxxxx
fromdomain=callcentric.com
disallow=all
allow=ulaw,g729
insecure=port,invite

— extension.ael —
context local {
includes {
voicemail;
outgoing;
};
1000 => Dial(SIP/sipura3000,30);
1001 => Dial(SIP/snom320,30);
4711 => jump s@incoming;
};

context voicemail {
2000 => {
Ringing();
Wait(1);
VoiceMailMain(s4711);
};
};

macro std-exten-ael( ext ) {
Dial(SIP/snom320&SIP/sipura3000,30);
switch(${DIALSTATUS}) {
case BUSY:
Voicemail(b${ext});
PlayBack(vm-goodbye);
HangUp();
break;
default:
Voicemail(u${ext});
PlayBack(vm-goodbye);
HangUp();
};
catch a {
VoiceMailMain(${ext});
return;
};
};

context incoming {
s => &std-exten-ael(4711);

    1777xxxxxxx => jump s;
    1864xxxxxxx => jump s;
    49xxxxxx    => jump s;

};

context outgoing {
_8. => {
Dial(SIP/${EXTEN:1}@callcentric,120,tr);
Congestion();
Busy();
};

    _0. => {
            Dial(SIP/${EXTEN:1}@telsome,120,tr);
            Congestion();
            Busy();
    };

};

/Mark

Did you set this option in General section of sip.conf
srvlookup=no

Thanks,
Agni.

I have tried setting srvlookup to both yes and no and the result is the same.

When I try to pickup an anonymous call I get the following warnings on the CLI:
[Dec 26 09:19:05] WARNING[4280]: chan_sip.c:12351 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '1815809-3439268343-639122@msw2.telengy.net’. Giving up.
[Dec 26 09:19:05] WARNING[4280]: chan_sip.c:12886 handle_response: Remote host can’t match request BYE to call '1815809-3439268343-639122@msw2.telengy.net’. Giving up.

I don’t know if that is any help. I do have a sip debug log I can post if that is of any help.

Is there any specific need for you to go with nat=no in your configuration . Did you tried with the option canreinvite = no

place canreinvite = no in general section and make nat = yes for all your peer connection . Please update me the status.

Thanks,
Agni