[Users] CALL FORWARD / PSTN GW
Kjell Arild Frigstad
kaf at fastcom.no
Wed Dec 7 18:32:34 CET 2005
Hi,
confirms that it was the RPID who was wrong, removed it, and it works OK!
Thanks!
Best regards
K.A.Frigstad
Bogdan-Andrei Iancu wrote:
> 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
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at openser.org
>> http://openser.org/cgi-bin/mailman/listinfo/users
>>
More information about the sr-users
mailing list