New subject: [Serusers] SER/PSTN Gateway conf

15 May 2007

My Record_Route and Loose section are the follow:

# -----------------------------------------------------------------
# Record Route Section
# -----------------------------------------------------------------
if (method!="REGISTER") {
record_route();
};
# -----------------------------------------------------------------
# Loose Route Section
# -----------------------------------------------------------------
if (loose_route()) {
    if (method=="INVITE") {
      if (!allow_trusted()) {
        if (!proxy_authorize("","subscriber")) {
          proxy_challenge("","0");
          break;
        } else if (!check_from()) {
          sl_send_reply("403", "Use From=ID");
          break;
        };
        consume_credentials();
      };
    };
route(1);
break;
};

As follow reported, SER Record Route INVITE original INVITE message, that lack
of 'Route:' header, and forward it to IP Phone. Why not for BYE? =_="

Thanks for support and big patience. ^_^

U 2007/05/15 16:14:20.789400 10.28.52.105:5060 -> 10.28.19.202:5060
INVITE sip:06605XXXXX@10.28.19.202 SIP/2.0.
To: <sip:06605XXXXX@10.28.19.202>.
From: "06720XXXXX"
<sip:06720XXXXX@10.28.52.105>;tag=DXsf22387576811Yaz079427.
Via: SIP/2.0/UDP 10.28.52.105:5060;branch=z9hG4bK2387576811429488787179425.
Remote-Party-ID: "06720XXXXX"
<sip:06720XXXXX@10.28.52.105>;party=calling;screen=yes;privacy=off.
Contact: <sip:10.28.52.105:5060>.
Call-ID: 238757681179426(a)10.28.52.105.
Max-Forwards: 70.
User-Agent: Netsynt-Gateway/SIP_AC_SRVlkup_3.1.12_4.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, REGISTER, NOTIFY.
CSeq: 23392 INVITE.
Content-Length: 274.
Content-Type: application/sdp.
.
v=0.
o=NetsyntSIP-GW-UserAgent 34373 1 IN IP4 10.28.52.105.
s=SIP Call.
c=IN IP4 10.28.52.105.
t=0 0.
m=audio 16572 RTP/AVP 18 8 0.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:8 PCMA/8000.
a=silenceSupp:off - - - -.
a=rtpmap:0 PCMU/8000.
a=silenceSupp:off - - - -.

#
U 2007/05/15 16:14:20.790166 10.28.19.202:5060 -> 10.28.52.105:5060
SIP/2.0 100 trying -- your call is important to us.
To: <sip:06605XXXXX@10.28.19.202>.
From: "06720XXXXX"
<sip:06720XXXXX@10.28.52.105>;tag=DXsf22387576811Yaz079427.
Via: SIP/2.0/UDP 10.28.52.105:5060;branch=z9hG4bK2387576811429488787179425.
Call-ID: 238757681179426(a)10.28.52.105.
CSeq: 23392 INVITE.
Server: Sip EXpress router (0.9.6 (i386/linux)).
Content-Length: 0.
Warning: 392 10.28.19.202:5060 "Noisy feedback tells:  pid=3803
req_src_ip=10.28.52.105 req_src_port=5060
in_uri=sip:0660522014@10.28.19.202
out_uri=sip:06605XXXXX@10.28.19.124;user=phone via_cnt==1".
.

#
U 2007/05/15 16:14:20.790257 10.28.19.202:5060 -> 10.28.19.124:5060
INVITE sip:06605XXXXX@10.28.19.124;user=phone SIP/2.0.
Record-Route: <sip:10.28.19.202;ftag=DXsf22387576811Yaz079427;lr=on>.
To: <sip:0660522014@10.28.19.202>.
From: "06720XXXXX"
<sip:06720XXXXX@10.28.52.105>;tag=DXsf22387576811Yaz079427.
Via: SIP/2.0/UDP 10.28.19.202;branch=z9hG4bK4639.ea946db1.0.
Via: SIP/2.0/UDP 10.28.52.105:5060;branch=z9hG4bK2387576811429488787179425.
Remote-Party-ID: "06720XXXXX"
<sip:06720XXXXX@10.28.52.105>;party=calling;screen=yes;privacy=off.
Contact: <sip:10.28.52.105:5060>.
Call-ID: 238757681179426(a)10.28.52.105.
Max-Forwards: 16.
User-Agent: Netsynt-Gateway/SIP_AC_SRVlkup_3.1.12_4.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, REGISTER, NOTIFY.
CSeq: 23392 INVITE.
Content-Length: 274.
Content-Type: application/sdp.
.
v=0.
o=NetsyntSIP-GW-UserAgent 34373 1 IN IP4 10.28.52.105.
s=SIP Call.
c=IN IP4 10.28.52.105.
t=0 0.
m=audio 16572 RTP/AVP 18 8 0.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:8 PCMA/8000.
a=silenceSupp:off - - - -.
a=rtpmap:0 PCMU/8000.
a=silenceSupp:off - - - -.

2007/5/15, Kostas Marneris &lt;K.Marneris(a)otenet.gr&gt;gr>:
- Nascondi testo tra virgolette -
...
  As far as I know and understand it seems that the
 R-URI of the BYE mesg (1st line) which GW sends to SER is wrong.

 I expected to see :
         BYE sip:user@SIP_Phone_IP_Address SIP/2.0

 or the problem is that the 'Route:' header is missing from the BYE mesg
 so it's not loose_routed.

 Do you Record_Route the original INVITE ?

 Check also : rfc3261 / 16.12.1

 Kostas

Re: [Serusers] SER/PSTN Gateway conf