[Serusers] SER/PSTN Gateway conf

[Greger V. Teigre]

Your record routing is OK.
g-)

flavio wrote:
> 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 at 10.28.19.202 SIP/2.0.
> To: <sip:06605XXXXX at 10.28.19.202>.
> From: "06720XXXXX" 
> <sip:06720XXXXX at 10.28.52.105>;tag=DXsf22387576811Yaz079427.
> Via: SIP/2.0/UDP 
> 10.28.52.105:5060;branch=z9hG4bK2387576811429488787179425.
> Remote-Party-ID: "06720XXXXX"
> <sip:06720XXXXX at 10.28.52.105>;party=calling;screen=yes;privacy=off.
> Contact: <sip:10.28.52.105:5060>.
> Call-ID: 238757681179426 at 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 at 10.28.19.202>.
> From: "06720XXXXX" 
> <sip:06720XXXXX at 10.28.52.105>;tag=DXsf22387576811Yaz079427.
> Via: SIP/2.0/UDP 
> 10.28.52.105:5060;branch=z9hG4bK2387576811429488787179425.
> Call-ID: 238757681179426 at 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 at 10.28.19.202
> out_uri=sip:06605XXXXX at 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 at 10.28.19.124;user=phone SIP/2.0.
> Record-Route: <sip:10.28.19.202;ftag=DXsf22387576811Yaz079427;lr=on>.
> To: <sip:0660522014 at 10.28.19.202>.
> From: "06720XXXXX" 
> <sip:06720XXXXX at 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 at 10.28.52.105>;party=calling;screen=yes;privacy=off.
> Contact: <sip:10.28.52.105:5060>.
> Call-ID: 238757681179426 at 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 <K.Marneris at otenet.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 at 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
>>
>>
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