[Users] CALL FORWARD / PSTN GW
Bogdan-Andrei Iancu
bogdan at voice-system.ro
Wed Dec 7 16:55:45 CET 2005
Hi,
I would say the problem is the RPID header . when a call from PSTN gets
back to PSTN, it will carry the original RPID inserted by the GW.
AFAIK, the CISCO gw looks first to RPID and than to FROM when
determining the caller. If you look into the logs you send, the GW says
"Calling Number=38092382" which is the RPID and not the FROM 38707130.
So, either change the RPID for calls to PSTN (recommended) or if use
FROM, remove the RPID hdr before fwd to GW.
regards,
bogdan
Kjell Arild Frigstad wrote:
> Hello!
> I am trying to implement Call forwarding.
> Server runs openser 1.0.0 and I have a Cisco PSTN GW where only defined
> calling numbers are permitted (because of accounting on the "upstream"
> provider). My problem is when a call from PSTN is beeing forwarded and
> I have to "turn it back" to the PSTN gatway.
>
> Calls from :
> 1) Sip server ---> SIP-CALL-FW ---> PSTN GW OK, the calling number is
> also changed to the number which has turned call forwarding on. The
> display on the phone also showing the number which has turned Call
> forwarding on..
>
> 2) PSTN ---> SIP-CALL-FW ---> SIP SERVER Works OK
>
> 3) PSTN ----> SIP-CALL-FW ---> PSTN is not working. Dialpeer matching
> in Cisco not OK because calling number not OK. I can define a way so
> that all calls can go through the GW. Then I sometimes get a "482 loop
> detected". The disply on the phone showing the orginal calling number
> (not the number wich hat turnd Call forwarding on).
>
> Is this the right way to "turn" a call back to where it came from? Have
> a missed a "rebuild" of the message?
>
>
>
> if (avp_db_load("$ruri","$fw_to") & method=="INVITE") {
> # CALL Forward
> log(1, "INFO: AVP_db_load fw_to true \n");
> #save called number
> avp_write("$ruri/username","s:org_to");
> #Insert call forward number
> avp_pushto("$ruri/username","$fw_to");
> #is call forward number local or to PSTN
> if (does_uri_exist()) {
> log(1, "INFO:Call FW to Local users\n");
> if (!lookup("location")) {
> log(1, "INFO: call FW to local user, not
> online\n");
> route(4);
> return;
> };
>
> log(1, "INFO:Call FW to local users online\n");
>
> uac_replace_from("$avp(s:org_to)","sip:$avp(s:org_to)@212.125.213.151:5060");
>
> if (!t_relay()) {
> sl_reply_error();
> return;
> };
> return;
>
> } else {
> log(1, "INFO: Call FW to PSTN\n");
>
> #here is somting wrong. What???
> #move Called number as From number and send to PSTN GW
>
> uac_replace_from("$avp(s:org_to)","sip:$avp(s:org_to)@212.125.213.151:5060");
>
> rewritehostport("80.239.101.34:5060");
> append_hf("P-hint: call fwd to PSTN\r\n");
> append_branch();
> record_route();
> if (!t_relay()) {
> sl_reply_error();
> return;
> };
>
> };
> return;
> }
>
> Cisco trace : calling from 38092382 (PSTN) to 38707130 (on the SIP
> server) witch is forwarded to 95066833 (PSTN)
>
> Call comming in from PSTN to SIP-server :
> Sent: (from Cisco)
> INVITE sip:38707130 at 212.125.213.151:5060 SIP/2.0
> Via: SIP/2.0/UDP 80.239.101.34:5060;branch=z9hG4bK69972
> From: <sip:38092382 at 80.239.101.34>;tag=2A699224-CC6
> To: <sip:38707130 at 212.125.213.151>
> Date: Fri, 02 Dec 2005 18:41:38 GMT
> Call-ID: 1B42584B-629A11DA-AFCCC9A9-E3CBCB60 at 80.239.101.34
> Supported: 100rel,timer
> Min-SE: 1800
> Cisco-Guid: 457212883-1654264282-2767847443-3293903248
> User-Agent: Cisco-SIPGateway/IOS-12.x
> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
> SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
> CSeq: 101 INVITE
> Max-Forwards: 70
> Remote-Party-ID:
> <sip:38092382 at 80.239.101.34>;party=calling;screen=yes;privacy=off
> Timestamp: 1133548898
> Contact: <sip:38092382 at 80.239.101.34:5060>
> Expires: 180
> Allow-Events: telephone-event
> Content-Type: application/sdp
> Content-Length: 253
>
> v=0
> o=CiscoSystemsSIP-GW-UserAgent 6274 5968 IN IP4 80.239.101.34
> s=SIP Call
> c=IN IP4 80.239.101.34
> t=0 0
> m=audio 16788 RTP/AVP 18 8 0
> c=IN IP4 80.239.101.34
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
>
> *Dec 2 18:41:38.278: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> Received:
> SIP/2.0 100 trying -- your call is important to us
> Via: SIP/2.0/UDP 80.239.101.34:5060;branch=z9hG4bK69972
> From: <sip:38092382 at 80.239.101.34>;tag=2A699224-CC6
> To: <sip:38707130 at 212.125.213.151>
> Call-ID: 1B42584B-629A11DA-AFCCC9A9-E3CBCB60 at 80.239.101.34
> CSeq: 101 INVITE
> Server: OpenSer (1.0.0 (i386/linux))
> Content-Length: 0
> Warning: 392 212.125.213.151:5060 "Noisy feedback tells: pid=29232
> req_src_ip=80.239.101.34 req_src_port=55717
> in_uri=sip:38707130 at 212.125.213.151:5060
> out_uri=sip:95066833 at 80.239.101.34:5060 via_cnt==1"
>
>
> *Dec 2 18:41:38.282: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> Received:
> INVITE sip:95066833 at 80.239.101.34:5060 SIP/2.0
> Record-Route: <sip:212.125.213.151;ftag=2A699224-CC6;lr=on>
> Via: SIP/2.0/UDP 212.125.213.151;branch=z9hG4bK1e4.11a1a9d3.0
> Via: SIP/2.0/UDP 80.239.101.34:5060;branch=z9hG4bK69972
> From: <sip:38707130 at 212.125.213.151:5060>;tag=2A699224-CC6
> To: <sip:38707130 at 212.125.213.151>
> Date: Fri, 02 Dec 2005 18:41:38 GMT
> Call-ID: 1B42584B-629A11DA-AFCCC9A9-E3CBCB60 at 80.239.101.34
> Supported: 100rel,timer
> Min-SE: 1800
> Cisco-Guid: 457212883-1654264282-2767847443-3293903248
> User-Agent: Cisco-SIPGateway/IOS-12.x
> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
> SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
> CSeq: 101 INVITE
> Max-Forwards: 69
> Remote-Party-ID:
> <sip:38092382 at 80.239.101.34>;party=calling;screen=yes;privacy=off
> Timestamp: 1133548898
> Contact: <sip:38092382 at 80.239.101.34:5060>
> Expires: 180
> Allow-Events: telephone-event
> Content-Type: application/sdp
> Content-Length: 253
> P-hint: call fwd to PSTN
>
> v=0
> o=CiscoSystemsSIP-GW-UserAgent 6274 5968 IN IP4 80.239.101.34
> s=SIP Call
> c=IN IP4 80.239.101.34
> t=0 0
> m=audio 16788 RTP/AVP 18 8 0
> c=IN IP4 80.239.101.34
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
>
>
>
> Dial peer maching (call forward from SIP server). Calling number is
> wrong, should be 38707130 (number who has put forwarding on)
> Calling Number=38092382, Called Number=95066833, Voice-Interface=0x0,
> Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
> Type=PEER_TYPE_VOICE,
> Peer Info Type=DIALPEER_INFO_SPEECH
> *Dec 2 19:44:08.330: //-1/D6732AD9A532/DPM/dpAssociateIncomingPeerCore:
> Result=NO_MATCH(-1) After All Match Rules Attempt
> *Dec 2 19:44:08.334: //-1/D6732AD9A532/DPM/dpMatchPeersCore:
> Calling Number=, Called Number=95066833, Peer Info
> Type=DIALPEER_INFO_SPEECH
> *Dec 2 19:44:08.334: //-1/D6732AD9A532/DPM/dpMatchPeersCore:
> Match Rule=DP_MATCH_DEST; Called Number=95066833
>
>
>
> Best regards
> Kjell Arild
>
>
>
>
>
>
>
>
>
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