May I ask something else karoo related?
Basic calls are working. Now I tested attended/unattended transfer through
karoo. In my current configuration the Refer message is forwarded through
karoo which is not desired. The documentation of karoo says “*Bridging the
call will make Karoo Bridge handle the REFER locally so that it does not
propagate to the ITSP which may not support it.*”. So I added
profile.bridge to my routing.
function is_my_pbx_routable(profile, ifaceAddr, ifacePort, toUser, toHost)
{
if ( ifaceAddr != sip_interface_address[0] ||
ifacePort != sip_interface_port[0])
return false;
if(toHost == "mypublicdomain")
{
profile.setInterfaceAddress(sip_interface_address[1],
sip_interface_port[1]);
profile.setTargetAddress("udp", "mylocaldomain", "5060");
profile.sipMessage.setRequestUriUser("404");
profile.setTargetDomain("mylocaldomain");
profile.bridge("no","authentication");
return true;
}
else
{
return false;
}
}
Now Invite messages become routed to the freeswitch behind karoo but it
just answers 404 not found.
[CID=583608e3] SIP/2.0 404 Not Found
[CID=583608e3] Via: SIP/2.0/UDP
xxx.xxx.xxx.xxx:5080;branch=z9hG4bK14885936779861960987;rport=5080
[CID=583608e3] From: "IANT" <sip:IANT@mylocaldomain
>;tag=714294677
[CID=583608e3] To: <sip:xxx@mylocaldomain>;tag=S7K0253KmcDmQ
[CID=583608e3] Call-ID:
[email protected]
[CID=583608e3] CSeq: 26 INVITE
[CID=583608e3] User-Agent: OSS Karoo FS
[CID=583608e3] Accept: application/sdp
[CID=583608e3] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
[CID=583608e3] Supported: timer, precondition, path,
replaces
[CID=583608e3] Allow-Events: talk, hold, refer
[CID=583608e3] Content-Length: 0
[CID=583608e3] P-Asserted-Identity: "xxx"
<sip:xxx@mylocaldomain>
Could the reason be that I cannot start karoo by /etc/init.d/karoo-init
start because for some reason the sahara.cfg is empty? I always need to
start karoo manually with karoo –D. ps does not show a freeswitch process.
But if it is not there: Where does the 404 come from?
Sincerely
Jan
*Von:* Joegen Baclor [mailto:[email protected]]
*Gesendet:* Freitag, 30. März 2012 13:33
*An:* Discussion list for users of sipXecs software
*Cc:* Jan Fricke
*Betreff:* Re: [sipx-users] Karoo B2BUA?
By the way, did you restart karoo after you made the rtp port range
config? It requries a restart. not just a reload-config
On 03/30/2012 06:32 PM, Jan Fricke wrote:
OK found /var/log/karoo/sbc.log
Karoo seems to act as B2BUA.
Log shows:
10:16:52.303: [CID=22b5a92d] JS: Setting transaction property
require-rtp-proxy=1
Incoming SDP Part:
[CID=22b5a92d] s=Polycom IP Phone
[CID=22b5a92d] c=IN IP4 10.99.148.60
[CID=22b5a92d] t=0 0
[CID=22b5a92d] a=sendrecv
[CID=22b5a92d] m=audio 2258 RTP/AVP 9 0 8 18 101
Outgoing SDP Part:
[CID=22b5a92d] s=Polycom IP Phone
[CID=22b5a92d] c=IN IP4 141.41.40.10
[CID=22b5a92d] t=0 0
[CID=22b5a92d] a=sendrecv
[CID=22b5a92d] m=audio 30002 RTP/AVP 9 0 8 18 101
But why does it ignore the settings of sip.conf?
rtp-proxy-port-base = 10000;
rtp-proxy-port-max = 10200;
Sip.conf attached.
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