Ciao Daniel, here the data you request..
Of course Kamailio2 use the color 9990 to send the call to CISCO Gw because its a required.. so CISCO send back the call with 9990 to Kamailio2 and it to Kamailio1 and after that to Customer.. What I espect was that CISCO replied 9990 to Kamailio2, Kama2 replied 9053 to Kamailio1 and Kamailio1 replied 9999 to Customer1. Here the sip trace you request Kamailio1 --> Kamailio2 U 2016/08/10 10:54:29.269917 2.2.2.2:5060 -> 3.3.3.3:5060 INVITE sip:90534912345678@3.3.3.3 SIP/2.0. Record-Route: <sip:2.2.2.2;lr;did=4f3.8501;nat=yes>. Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK5c29.8288fcc6bf463fb8f82d3609b0c4893c.0. Via: SIP/2.0/UDP 1.1.1.1:5060;received=1.1.1.1;branch=z9hG4bK06b62a40;rport=5060. Max-Forwards: 69. From: "151512345678" <sip:151512345678@1.1.1.1>;tag=as7f0dee78. To: <sip:90534912345678@3.3.3.3>. Contact: <sip:151512345678@1.1.1.1:5060>. Call-ID: 406a307158b1016b3a9936cd476b3c89@1.1.1.1:5060. CSeq: 102 INVITE. Date: Wed, 10 Aug 2016 08:54:27 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 308. User-Agent: Fagians VOIP 2.4. . v=0. o=root 869935480 869935480 IN IP4 1.1.1.1. s=Asterisk PBX 1.8.32.3. c=IN IP4 x.x.x.x.x. t=0 0. m=audio 36398 RTP/AVP 3 18 8 101. a=rtpmap:3 GSM/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. Kamailio2 --> CISCO.. LCR need to use 9990 to send call to CISCO.. U 2016/08/10 10:54:29.301085 3.3.3.3:5060 -> 4.4.4.4:5060 INVITE sip:99904912345678@4.4.4.4 SIP/2.0. Record-Route: <sip:3.3.3.3;lr;did=4f3.9ce2;nat=yes>. Record-Route: <sip:2.2.2.2;lr;did=4f3.8501;nat=yes>. Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK5c29.c1bcba318da7a0a1ae45efbe2ee682c5.0. Via: SIP/2.0/UDP 2.2.2.2:5060;rport=5060;branch=z9hG4bK5c29.8288fcc6bf463fb8f82d3609b0c4893c.0. Via: SIP/2.0/UDP 1.1.1.1:5060;received=1.1.1.1;branch=z9hG4bK06b62a40;rport=5060. Max-Forwards: 68. From: "151512345678" <sip:151512345678@1.1.1.1>;tag=as7f0dee78. To: <sip:99904912345678@4.4.4.4>. Contact: <sip:151512345678@1.1.1.1:5060>. Call-ID: 406a307158b1016b3a9936cd476b3c89@1.1.1.1:5060. CSeq: 102 INVITE. Date: Wed, 10 Aug 2016 08:54:27 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 309. User-Agent: Fagians VOIP 2.4. . v=0. o=root 869935480 869935480 IN IP4 1.1.1.1. s=Asterisk PBX 1.8.32.3. c=IN IP4 x.x.x.x.. t=0 0. m=audio 58242 RTP/AVP 3 18 8 101. a=rtpmap:3 GSM/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. CISCO ---> Kamailio2 180/200 messages U 2016/08/10 10:54:29.361634 4.4.4.4:5060 -> 3.3.3.3:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK5c29.c1bcba318da7a0a1ae45efbe2ee682c5.0,SIP/2.0/UDP 2.2.2.2:5060;rport=5060;branch=z9hG4bK5c29.8288fcc6bf463fb8f82d3609b0c4893c.0,SIP/2.0/UDP 1.1.1.1:5060;received=1.1.1.1;branch=z9hG4bK06b62a40;rport=5060. From: "151512345678" <sip:151512345678@1.1.1.1>;tag=as7f0dee78. To: <sip:99904912345678@4.4.4.4>;tag=5F0E7DF4-172F. Date: Wed, 10 Aug 2016 08:54:29 GMT. Call-ID: 406a307158b1016b3a9936cd476b3c89@1.1.1.1:5060. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 102 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. Allow-Events: telephone-event. Contact: <sip:99904912345678@4.4.4.4:5060>. Record-Route: <sip:3.3.3.3;lr;did=4f3.9ce2;nat=yes>,<sip:2.2.2.2;lr;did=4f3.8501;nat=yes>. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 249. . v=0. o=CiscoSystemsSIP-GW-UserAgent 9803 1571 IN IP4 4.4.4.4. s=SIP Call. c=IN IP4 x.x.x.x. t=0 0. m=audio 18838 RTP/AVP 3 101. c=IN IP4 83.147.65.249. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:10. U 2016/08/10 10:54:39.486505 4.4.4.4:5060 -> 3.3.3.3:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK5c29.c1bcba318da7a0a1ae45efbe2ee682c5.0,SIP/2.0/UDP 2.2.2.2:5060;rport=5060;branch=z9hG4bK5c29.8288fcc6bf463fb8f82d3609b0c4893c.0,SIP/2.0/UDP 185.24.22 0.141:5060;received=1.1.1.1;branch=z9hG4bK06b62a40;rport=5060. From: "151512345678" <sip:151512345678@1.1.1.1>;tag=as7f0dee78. To: <sip:99904912345678@4.4.4.4>;tag=5F0E7DF4-172F. Date: Wed, 10 Aug 2016 08:54:29 GMT. Call-ID: 406a307158b1016b3a9936cd476b3c89@1.1.1.1:5060. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 102 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. Supported: replaces. Allow-Events: telephone-event. Contact: <sip:99904912345678@4.4.4.4:5060>. Record-Route: <sip:3.3.3.3;lr;did=4f3.9ce2;nat=yes>,<sip:2.2.2.2;lr;did=4f3.8501;nat=yes>. Content-Type: application/sdp. Content-Length: 249. . v=0. o=CiscoSystemsSIP-GW-UserAgent 9803 1571 IN IP4 4.4.4.4. s=SIP Call. c=IN IP4 x.x.x.x. t=0 0. m=audio 18838 RTP/AVP 3 101. c=IN IP4 83.147.65.249. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:10. Il 10/08/16 13:33, Daniel Grotti ha scritto: > Ciao Laura, > would be interesting to see the INVITE from kamailo2-->Cisco and see > the headers there, as well as the 180/200 from Cisco->kamailio2. > As Carsten said, probably Cisco is messing up From/To headers. The > 9990 color is not present in any of the INVITEs you provided, so would > be nice to understand where is come from. > > > Cheers, > Daniel > > > > On 08/10/2016 12:27 PM, Laura wrote: >> >> Sorry for delay on my reply.. >> >> >> I need to expalin better the situazione.. >> >> Customer1 Ip : 1.1.1.1 >> Kamailio1 ip : 2.2.2.2 >> Kamailio2 ip: 3.3.3.3 >> CiscoGW ip: 4.4.4.4 >> >> Kamailio1 is on USA for example >> Kamailio2 is on Germany for example >> >> Customer1 --> Kamailio platform1 --> Kamailio Platform2 --> CISCO GW >> SIP/TDM for PTSN termination >> >> Customer1 is sending a call using his specific color 9999 to number >> 4912345678 and from sender 151512345678 >> >> U 2016/08/10 09:54:29.250974 1.1.1.1:5060 ->2.2.2.2:5060 >> INVITE sip:*9999*4912345678@2.2.2.2 SIP/2.0. >> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK06b62a40;rport. >> Max-Forwards: 70. >> From: "151512345678" <sip:151512345678@1.1.1.1>;tag=as7f0dee78. >> To: <sip:*9999*4912345678@2.2.2.2>. >> Contact: <sip:151512345678@1.1.1.1:5060>. >> Call-ID: 406a307158b1016b3a9936cd476b3c89@1.1.1.1:5060. >> CSeq: 102 INVITE. >> User-Agent: Asterisk PBX 1.8.32.3. >> Date: Wed, 10 Aug 2016 08:54:27 GMT. >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, >> INFO, PUBLISH, MESSAGE. >> Supported: replaces, timer. >> Content-Type: application/sdp. >> Content-Length: 309. >> . >> v=0. >> o=root 869935480 869935480 IN IP4 1.1.1.1. >> s=Asterisk PBX 1.8.32.3. >> c=IN IP4 1.1.1.1. >> t=0 0. >> m=audio 15710 RTP/AVP 3 18 8 101. >> a=rtpmap:3 GSM/8000. >> a=rtpmap:18 G729/8000. >> a=fmtp:18 annexb=no. >> a=rtpmap:8 PCMA/8000. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-16. >> a=ptime:20. >> a=sendrecv. >> >> >> After that the Kamailio1 platform is checking the LCR and route it >> with the color of its supplier (9053) to Kamailio2. Kamailio2 is a >> supplier of Kamailio1 >> >> U 2016/08/10 09:54:29.2525272.2.2.2:5060 -> 3.3.3.3:5060 >> INVITE sip:*9053*4912345678@3.3.3.3 SIP/2.0. >> Record-Route: <sip:2.2.2.2;lr;did=4f3.8501;nat=yes>. >> Via: >> SIP/2.0/UDP2.2.2.2:5060;branch=z9hG4bK5c29.8288fcc6bf463fb8f82d3609b0c4893c.0. >> Via: SIP/2.0/UDP >> 1.1.1.1:5060;received=1.1.1.1;branch=z9hG4bK06b62a40;rport=5060. >> Max-Forwards: 69. >> From: "151512345678" <sip:151512345678@1.1.1.1>;tag=as7f0dee78. >> To: <sip:*9053*4912345678@3.3.3.3>. >> Contact: <sip:151512345678@1.1.1.1:5060>. >> Call-ID: 406a307158b1016b3a9936cd476b3c89@1.1.1.1:5060. >> CSeq: 102 INVITE. >> Date: Wed, 10 Aug 2016 08:54:27 GMT. >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, >> INFO, PUBLISH, MESSAGE. >> Supported: replaces, timer. >> Content-Type: application/sdp. >> Content-Length: 308. >> User-Agent: Fagians VOIP 2.4. >> . >> v=0. >> o=root 869935480 869935480 IN IP4 1.1.1.1. >> s=Asterisk PBX 1.8.32.3. >> c=IN IP4 51.254.158.37. >> t=0 0. >> m=audio 36398 RTP/AVP 3 18 8 101. >> a=rtpmap:3 GSM/8000. >> a=rtpmap:18 G729/8000. >> a=fmtp:18 annexb=no. >> a=rtpmap:8 PCMA/8000. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-16. >> a=ptime:20. >> a=sendrecv. >> >> Kamailio2 use its LCR and send the call to Cisco Gateway that use its >> color and send the call on termination to TDM Switch. >> Naturally Kamailio2 receive the replies from Cisco and send it back >> to Kamailio1. >> >> >> Here is the Session progress Kamailio1 receive from Kamailio2 that it >> got from Cisco. >> >> U 2016/08/10 09:54:29.375669 3.3.3.3:5060 ->2.2.2.2:5060 >> SIP/2.0 183 Session Progress. >> Via: >> SIP/2.0/UDP2.2.2.2:5060;rport=5060;branch=z9hG4bK5c29.8288fcc6bf463fb8f82d3609b0c4893c.0,SIP/2.0/UDP >> 1.1.1.1:5060;received=1.1.1.1;branch=z9hG4bK06b62a40;rport=5060. >> From: "151512345678" <sip:151512345678@1.1.1.1>;tag=as7f0dee78. >> To: <sip:*9990*4912345678@4.4.4.4>;tag=5F0E7DF4-172F. >> Date: Wed, 10 Aug 2016 08:54:29 GMT. >> Call-ID: 406a307158b1016b3a9936cd476b3c89@1.1.1.1:5060. >> CSeq: 102 INVITE. >> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, >> SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. >> Allow-Events: telephone-event. >> Contact: <sip:99904912345678@4.4.4.4:5060>. >> Record-Route: >> <sip:3.3.3.3;lr;did=4f3.9ce2;nat=yes>,<sip:2.2.2.2;lr;did=4f3.8501;nat=yes>. >> Content-Disposition: session;handling=required. >> Content-Type: application/sdp. >> Content-Length: 251. >> User-Agent: Fagians VOIP 2.4. >> . >> v=0. >> o=CiscoSystemsSIP-GW-UserAgent 9803 1571 IN IP4 4.4.4.4. >> s=SIP Call. >> c=IN IP4 83.147.127.247. >> t=0 0. >> m=audio 58240 RTP/AVP 3 101. >> c=IN IP4 83.147.127.247. >> a=rtpmap:3 GSM/8000. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-16. >> a=ptime:10. >> >> To: <sip:99904912345678@4.4.4.4>;tag=5F0E7DF4-172F. ->> 9990 is the >> color that use CISCO to terminate the call on TDM Switch >> >> After some other messages Kamailio1 receive the 200 OK and send it >> back to Customer1 >> >> >> Kamailio2 --> Kamailio1 >> >> U 2016/08/10 09:54:39.507885 3.3.3.3:5060 ->2.2.2.2:5060 >> SIP/2.0 200 OK. >> Via: >> SIP/2.0/UDP2.2.2.2:5060;rport=5060;branch=z9hG4bK5c29.8288fcc6bf463fb8f82d3609b0c4893c.0,SIP/2.0/UDP >> 1.1.1.1:5060;received=1.1.1.1;branch=z9hG4bK06b62a40;rport=5060. >> From: "151512345678" <sip:151512345678@1.1.1.1>;tag=as7f0dee78. >> To: <sip:*9990*4912345678@4.4.4.4>;tag=5F0E7DF4-172F. >> Date: Wed, 10 Aug 2016 08:54:29 GMT. >> Call-ID: 406a307158b1016b3a9936cd476b3c89@1.1.1.1:5060. >> CSeq: 102 INVITE. >> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, >> SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. >> Supported: replaces. >> Allow-Events: telephone-event. >> Contact: <sip:99904912345678@4.4.4.4:5060>. >> Record-Route: >> <sip:3.3.3.3;lr;did=4f3.9ce2;nat=yes>,<sip:2.2.2.2;lr;did=4f3.8501;nat=yes>. >> Content-Type: application/sdp. >> Content-Length: 251. >> User-Agent: Fagians VOIP 2.4. >> . >> v=0. >> o=CiscoSystemsSIP-GW-UserAgent 9803 1571 IN IP4 4.4.4.4. >> s=SIP Call. >> c=IN IP4 83.147.127.247. >> t=0 0. >> m=audio 58240 RTP/AVP 3 101. >> c=IN IP4 83.147.127.247. >> a=rtpmap:3 GSM/8000. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-16. >> a=ptime:10. >> >> Kamailio1 --> Customer1 >> >> U 2016/08/10 09:54:39.5120362.2.2.2:5060 -> 1.1.1.1:5060 >> SIP/2.0 200 OK. >> Via: SIP/2.0/UDP >> 1.1.1.1:5060;received=1.1.1.1;branch=z9hG4bK06b62a40;rport=5060. >> From: "151512345678" <sip:151512345678@1.1.1.1>;tag=as7f0dee78. >> To: <sip:*9990*4912345678@4.4.4.4>;tag=5F0E7DF4-172F. >> Date: Wed, 10 Aug 2016 08:54:29 GMT. >> Call-ID: 406a307158b1016b3a9936cd476b3c89@1.1.1.1:5060. >> CSeq: 102 INVITE. >> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, >> SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. >> Supported: replaces. >> Allow-Events: telephone-event. >> Contact: <sip:99904912345678@4.4.4.4:5060>. >> Record-Route: >> <sip:3.3.3.3;lr;did=4f3.9ce2;nat=yes>,<sip:2.2.2.2;lr;did=4f3.8501;nat=yes>. >> Content-Type: application/sdp. >> Content-Length: 249. >> User-Agent: Fagians VOIP 2.4. >> . >> v=0. >> o=CiscoSystemsSIP-GW-UserAgent 9803 1571 IN IP4 4.4.4.4. >> s=SIP Call. >> c=IN IP4 51.254.158.37. >> t=0 0. >> m=audio 56710 RTP/AVP 3 101. >> c=IN IP4 51.254.158.37. >> a=rtpmap:3 GSM/8000. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-16. >> a=ptime:10. >> >> So the real question is how to fix that on Kamailio ?.. >> >> We need to use always the original messages and data into sdp header >> when we talk with other parts.. >> >> On our configuration we permit to transit that modified messages.. >> like you can see Customer1 is getting back datas modified from CiscoGW. >> >> >> Hope that will be more clear to you all.. >> >> >> Anyone can suggest us a way ? >> >> >> Regards >> >> Laura >> >> >> Il 01/08/16 14:25, Carsten Bock ha scritto: >>> Hi, >>> >>> do you use "uac_replace_from" or "uac_replace_to" in your logic? >>> >>> If not, it seems to me, that your supplier is messing around with >>> the SIP-Replies. >>> >>> Thanks, >>> Carsten >>> >>> 2016-08-01 14:10 GMT+02:00 Laura <red_...@plugit.net >>> <mailto:red_...@plugit.net>>: >>> >>> Dear list, >>> >>> i'm asking here a question about Kamailio config. >>> >>> We are testing a wide area configuration of Kamailio over separates >>> countries and we are still facing with an issue. >>> >>> We configured Kamailio 4.3.5 with dialog support over the TM >>> modules and >>> we use LCR module for menage ours LCRs rule set profiles. >>> >>> For some technicals reasons we use tech prefix for our customer >>> so for >>> exaples customer1 send traffic to us with 1111 prefix, customer2 >>> send >>> traffic to us with 2222 and something similar.. >>> >>> Our supplier, of course, are using tech prefix too so for >>> examples if i >>> want to send the call to supplier1 i need to use tech prefix 1789 or >>> something similar.. >>> >>> The point is.. >>> >>> >>> When customer1 is sending an invite to us.. it send us something >>> like >>> (Bangladesh mobile 8801xxx) >>> >>> INVITE sip:11118801xxxx...@aaa.bbb.ccc.ddd >>> >>> Our Kamailio will reply with the Trying and then it goes to LCR >>> module >>> and match our supplier1 so it make a new invite like this >>> >>> INVITE sip:17898801xxx...@supplier.ip >>> >>> The problem come when supplier1 reply to us and we replies back to >>> customer1.. >>> >>> Customer1 view the From: field with the 17898801xxxxxx numbers.. and >>> some of our customers don't like it. >>> >>> We don't use anymore the topoh module becuase we found some troubles >>> using it.. so.. >>> >>> Is there a way that we can use for fix this situation ? >>> >>> >>> Best regards. >>> >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >>> mailing list >>> sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >>> >>> >>> -- >>> Carsten Bock >>> CEO (Geschäftsführer) >>> >>> ng-voice GmbH >>> Millerntorplatz 1 >>> 20359 Hamburg / Germany >>> >>> http://www.ng-voice.com >>> mailto:cars...@ng-voice.com <mailto:cars...@ng-voice.com> >>> >>> Office +49 40 5247593-40 >>> Fax +49 40 5247593-99 >>> >>> Sitz der Gesellschaft: Hamburg >>> Registergericht: Amtsgericht Hamburg, HRB 120189 >>> Geschäftsführer: Carsten Bock >>> Ust-ID: DE279344284 >>> >>> Hier finden Sie unsere handelsrechtlichen Pflichtangaben: >>> http://www.ng-voice.com/imprint/ >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users@lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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