sr-users December 2016

sr-users@lists.kamailio.org

71 participants
104 discussions

why a register error?
by Olvera, Victor 09 Jan '17

09 Jan '17

As I attempt to use X-lite I get a 504 error. On the CSCFs I get the following: pcscf 0(2066) INFO: <script>: [REGISTER] from [sip:bob@epc.mnc001.mcc001.3gppnetwork.org] to [sip:bob@epc.mnc001.mcc001.3gppnetwork.org] 0(2066) ERROR: <script>: REGISTER (sip:bob@epc.mnc001.mcc001.3gppnetwork.org (192.168.56.1:55880) to sip:bob@epc.mnc001.mcc001.3gppnetwork.org, 81563ZDg1Y2MwZDExYWJiYWJjNDZmNzY0ZWEwYjE1ODNkYmM) icscf 11(2150) INFO: cdp [receiver.c:932]: peer_connect(): peer_connect(): Peer hss.epc.mnc001.mcc001.3gppnetwork.org<http://mcc001.3gppnetwork.org>:3868 connected 0(2138) INFO: <script>: [REGISTER] from [sip:bob@epc.mnc001.mcc001.3gppnetwork.org] to [sip:bob@epc.mnc001.mcc001.3gppnetwork.org] 0(2138) ERROR: <script>: REGISTER (sip:bob@epc.mnc001.mcc001.3gppnetwork.org (192.168.56.76:4060) to sip:bob@epc.mnc001.mcc001.3gppnetwork.org, 81563ZDg1Y2MwZDExYWJiYWJjNDZmNzY0ZWEwYjE1ODNkYmM) 0(2138) INFO: ims_icscf [cxdx_uar.c:71]: create_uaa_return_code(): created AVP successfully : [uaa_return_code] 6(2145) INFO: ims_icscf [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_server_name: Failed finding avp 6(2145) INFO: ims_icscf [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_result_code: Failed finding avp 6(2145) INFO: ims_icscf [cxdx_uar.c:71]: create_uaa_return_code(): created AVP successfully : [uaa_return_code] 0 scscf 11(2047) WARNING: cdp [receiver.c:864]: peer_connect(): peer_connect(): Trying to connect to 192.168.56.78 port 3868 11(2047) INFO: cdp [receiver.c:932]: peer_connect(): peer_connect(): Peer hss.epc.mnc001.mcc001.3gppnetwork.org<http://mcc001.3gppnetwork.org>:3868 connected 9(2045) ERROR: cdp [peerstatemachine.c:635]: I_Snd_CER(): I_Snd_CER(): Error on finding local host address > Socket operation on non-socket Great regards, Victor Olvera victor.olvera(a)verizonwireless.com<mailto:victor.olvera@verizonwireless.com> mobile # 001-214-923-9405

2 1

VERY IMPORTANT: deciding when to remove the MI code
by Daniel-Constantin Mierla 06 Jan '17

06 Jan '17

Hello, we started discussing about removing MI (so called management interface) for very long time, more or less since 2008. The RPC should remain the control interface, given its better structure for commands, parameters, etc ... MI is custom protocol using a line-oriented communication via fifo or socket file with kamailio (e.g., implemented mi_fifo and mi_datagram modules). RPC is the alternative, a more standardized concept, with better structured format. I think it's time to set a clear roadmap for doing the removal. Overall, it will be easier to maintain the code, right now being duplicated code for doing the same operation over MI or RPC, and MI shows its limitations (or complexity to deal with) for advanced needs (see the discussions about how to provide multi-line value parameters over MI). So, I want to know if there are many relying on MI directly and they still want to keep it, what would be the expected duration they need for upgrading their tools to work with RPC interface, other relevant aspects people have in favour of mi vs rpc. I am even willing to do the removal in time befire freezing the 5.0 branch. We will ensure a clean start of 5.x series. The main concern from my point of view is kamctl -- but I think we can preserve the compatibility for kamctl commands and parameters (so command line execution of kamctl will be the same), but the output might be different. That's because it should be easy to updated it to communicate with jsonrpc-s module, but then it will get json-formatted results. To summarize, two big questions to answer: a) Are you ok to remove the MI code/commands? b) If yes to a), are you ok to be done for v5.0? Not providing feedback will be considered as 'yes' for both questions, so **speak up if you want MI to be kept or delay it removal**. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

6 30

presentity tabla, faster update when drop TCP/TLS connection
by João Resende 05 Jan '17

05 Jan '17

I'm testing kamailio with a SIP client with presence support, when I get a connection drop from a client (i.e when the client lost the access to the Internet connection) I do not receive the change in the status to offline immediately, I just received when the timeout from the presentity database finishes. Exists a way to define this? or it is a bug? or should be a new feature? I know that I'm able to change the expiry parameter from the presentity table, but that increases my battery consumption and the data exchange, the same happens if I reduce the publish interval on the client side.

3 9

location
by Slava Bendersky 05 Jan '17

05 Jan '17

Hello Everyone, Why save location will ignore on pv var when trying save on reply 200 OK. if(is_method("REGISTER") && fnmatch("$rs", "200")) { xlog("L_INFO", "OnReply: Registration is $rs OK saving --> $avp(rcvcontact) in db\n"); $var(oar) = "sip:" + $tU + "@" + $(avp(rcvcontact){uri.host}) + ":" + $(avp(rcvcontact){uri.port}); if(!save("location", "0x04", "$var(oar)")) { xlog("L_INFO", "Error saving the location\n"); } Dec 26 11:43:20 cavprx00 kamailio: 4(5546) ERROR: *** cfgtrace:onreply_route=[LOCATION_REPLY] c=[/etc/kamailio/kamailio-prod.cfg] l=619 a=26 n=xlog Dec 26 11:43:20 cavprx00 kamailio: 4(5546) INFO: <script>: OnReply: Registration is 200 OK saving --> sip:4300@client_pub_ip:31376;transport=tls in db Dec 26 11:43:20 cavprx00 kamailio: 4(5546) ERROR: *** cfgtrace:onreply_route=[LOCATION_REPLY] c=[/etc/kamailio/kamailio-prod.cfg] l=620 a=63 n=assign Dec 26 11:43:20 cavprx00 kamailio: 4(5546) ERROR: *** cfgtrace:onreply_route=[LOCATION_REPLY] c=[/etc/kamailio/kamailio-prod.cfg] l=624 a=16 n=if Dec 26 11:43:20 cavprx00 kamailio: 4(5546) ERROR: *** cfgtrace:onreply_route=[LOCATION_REPLY] c=[/etc/kamailio/kamailio-prod.cfg] l=621 a=27 n=save Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: <core> [sruid.c:182]: sruid_next(): new sruid is [uloc-586147f5-15aa-1] (1 / 20) Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: registrar [save.c:410]: pack_ci(): generated ruid is: uloc-586147f5-15aa-1 Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: <core> [sruid.c:182]: sruid_next(): new sruid is [uloc-586147f5-15aa-2] (2 / 20) Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: registrar [save.c:410]: pack_ci(): generated ruid is: uloc-586147f5-15aa-2 Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: usrloc [ucontact.c:1004]: db_update_ucontact_ruid(): ruid:uloc-5860b463-10c5-1 Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: usrloc [ucontact.c:1133]: db_update_ucontact_ruid(): contact:sip:4300@10.18.130.27:5060;transport=udp ----> LAN side contact Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: db_postgres [km_val.c:163]: db_postgres_val2str(): PQescapeStringConn: in: 10 chars, out: 10 chars Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: db_postgres [km_val.c:163]: db_postgres_val2str(): PQescapeStringConn: in: 21 chars, out: 21 chars Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: db_postgres [km_val.c:163]: db_postgres_val2str(): PQescapeStringConn: in: 33 chars, out: 33 chars Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: db_postgres [km_val.c:163]: db_postgres_val2str(): PQescapeStringConn: in: 40 chars, out: 40 chars Dec 26 11:43:20 cavprx00 kamailio: 4(5546) DEBUG: db_postgres [km_val.c:163]: db_postgres_val2str(): PQescapeStringConn: in: 20 chars, out: 20 chars Slava.

2 7

Rewrite From domain
by Gonzalo Gasca Meza 03 Jan '17

03 Jan '17

Hi all, I'm using Kamailio to forward calls between 2 Service Providers and I need to rewrite the From header "domain" URI. Example: From: "+18888888888" <sip:+18888888888@sip.sp1.com> to From: "+18888888888" <sip:+18888888888@*sip.sp2.com <http://sip.sp2.com>*> *Call flow:* Phone A --- > SP1 ---> sip ----> (kamailio) SP2 --(LOCATION)-> Phone B When Phone A calls SP2 PhoneB, it contains original sip domain from sp1. ( sip.sp1.com) hence user in SP2 can see call comes from SP1. I would like to rewrite the From domain field in this conditions: a) Calls comes from "sip.sp1.com" AND b) Call is being routed to PhoneB. Right now Im using the following code to find user and send call to B. #!ifdef WITH_ALIASDB # search in DB-based aliases xlog("L_INFO","alias_db_lookup: Call received. $rU\n"); if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif I found this in documentation: $fd - From URI domain if($hdr(From)=~"sip.sp1\.com") { ...} But not sure where is the best place to overwrite the From URI domain header. Thanks

3 7

Freezing for release v5.0.0
by Daniel-Constantin Mierla 02 Jan '17

02 Jan '17

Hello, I propose to freeze the development of v5.0.0 on January 4, 2017. Hopefully it is just in time for getting the frozen version in next Debian stable. Meanwhile, I hope we can get rid of mi code to have a proper fresh start with cleaned up code in 5.0.0. The big code tree restructuring was already done, config files in embedded languages is there as well (it needs more exports, but that's only about available time to fill some structures, not new feature development). If anyone wants to make other proposals, reply to mailing lists and let's discuss. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

TOPOS contact issue on RE-INVITE
by Daren FERREIRA 30 Dec '16

30 Dec '16

Hello, I made lots of tests in order to make Kamailio act as a topology hiding proxy, but i always got problems when using topos to store needed Route and Record-Route informations. I thought that was because of possible error(s) playing with : - socket binding - destination tweaking that may have lead to transaction breaking, so i decided to use NAT mode (even if that only can be temporary). My problem is the same. I have not found any way to influence TOPOS on the domain it use on the « Contact » header. Everything is getting well, iPs on Contact Header are toggled between external and internal sides, until RE-INVITE. From there, Contact domain are no more toggled, and so: - Public IP appear on Private side : i can manage my divide to force it to reply to the sender of the request instead of the contact - Private IP appear on Public side : i can’t do anything to make the external side ignore contact’s domain I wonder if that is a bug, related to RE-INVITE, or if there is any variable i can change to say topos which domain to be advertised on Contact headers? PUBLIC NETWORK SIDE RFC6598 NETWORK SIDE ————————— | | PROVIDER Y.Y.Y.Y———————X.X.X.X—| KAMAILIO ]—100.64.10.1———————— 100.64.10.2 ASTERISK (or anything else) | | ————————— 20:58:46.215911 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 1004 INVITE sip:+0598@100.64.10.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 100.64.10.2:5060;branch=z9hG4bK38ea6539;rport Max-Forwards: 70 From: "FAX" <sip:+9997@100.64.10.2>;tag=as4cfa9905 To: <sip:+0598@100.64.10.1;user=phone> Contact: <sip:+9997@100.64.10.2:5060> Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.13.1 Date: Thu, 29 Dec 2016 19:59:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer P-Access-Network-Info: GSTN;operator-specific-GI=320615200;network-provided P-Asserted-Identity: <sip:+9997@voip-connect.fr;user=phone> Content-Type: application/sdp Content-Length: 246 v=0 o=root 79038203 79038203 IN IP4 100.64.10.2 s=Asterisk PBX 13.13.1 c=IN IP4 100.64.10.2 t=0 0 m=audio 13564 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv 20:58:46.216642 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 375 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 100.64.10.2:5060;branch=z9hG4bK38ea6539;rport=5060;received=100.64.10.2 From: "FAX" <sip:+9997@100.64.10.2>;tag=as4cfa9905 To: <sip:+0598@100.64.10.1;user=phone> Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 0 20:58:46.222391 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 1040 INVITE sip:+0598@Y.Y.Y.Y:5060 SIP/2.0 Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 Max-Forwards: 69 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone> Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.13.1 Date: Thu, 29 Dec 2016 19:59:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer P-Access-Network-Info: GSTN;operator-specific-GI=320615200;network-provided P-Asserted-Identity: <sip:+9997@voip-connect.fr;user=phone> Content-Type: application/sdp Content-Length: 264 Contact: <sip:btpsh-58656af1-769f-1@X.X.X.X> v=0 o=root 79038203 79038203 IN IP4 X.X.X.X s=Asterisk PBX 13.13.1 c=IN IP4 X.X.X.X t=0 0 m=audio 57810 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=nortpproxy:yes 20:58:46.234592 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 290 SIP/2.0 100 Trying Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone> Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE 20:58:46.548771 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 718 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 216 Contact: <sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp> Content-Disposition: session;handling=required Content-Type: application/sdp v=0 o=- 1510281183 168103904 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=audio 17818 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - 20:58:46.554903 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 719 SIP/2.0 183 Session Progress From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 232 Content-Disposition: session;handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 100.64.10.2:5060;received=100.64.10.2;branch=z9hG4bK38ea6539;rport=5060 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=- 1510281183 168103904 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=audio 62130 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - a=nortpproxy:yes 20:58:47.735919 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 709 SIP/2.0 180 Ringing Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 216 Contact: <sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp> Content-Disposition: session;handling=required Content-Type: application/sdp v=0 o=- 1510281183 168103904 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=audio 17818 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - 20:58:47.741229 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 710 SIP/2.0 180 Ringing From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 232 Content-Disposition: session;handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 100.64.10.2:5060;received=100.64.10.2;branch=z9hG4bK38ea6539;rport=5060 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=- 1510281183 168103904 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=audio 62130 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - a=nortpproxy:yes 20:58:48.050762 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 709 SIP/2.0 180 Ringing Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 216 Contact: <sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp> Content-Disposition: session;handling=required Content-Type: application/sdp v=0 o=- 1510281183 168103904 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=audio 17818 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - 20:58:48.056175 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 710 SIP/2.0 180 Ringing From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 232 Content-Disposition: session;handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 100.64.10.2:5060;received=100.64.10.2;branch=z9hG4bK38ea6539;rport=5060 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=- 1510281183 168103904 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=audio 62130 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - a=nortpproxy:yes 20:58:52.811066 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 774 SIP/2.0 200 OK Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 216 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO Contact: <sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp> Accept-Encoding: identity Content-Disposition: session;handling=required Content-Type: application/sdp v=0 o=- 1510281183 168103904 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=audio 17818 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - 20:58:52.818381 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 775 SIP/2.0 200 OK From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 232 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO Accept-Encoding: identity Content-Disposition: session;handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 100.64.10.2:5060;received=100.64.10.2;branch=z9hG4bK38ea6539;rport=5060 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=- 1510281183 168103904 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=audio 62130 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - a=nortpproxy:yes 20:58:52.819315 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 446 ACK sip:atpsh-58656af1-769f-2@100.64.10.1 SIP/2.0 Via: SIP/2.0/UDP 100.64.10.2:5060;branch=z9hG4bK18c31028;rport Max-Forwards: 70 From: "FAX" <sip:+9997@100.64.10.2>;tag=as4cfa9905 To: <sip:+0598@100.64.10.1;user=phone>;tag=SDc8s1299-016500890000adc2 Contact: <sip:+9997@100.64.10.2:5060> Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 13.13.1 Content-Length: 0 20:58:52.825894 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 491 ACK sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.d149220aa711a5230da51bfea457d8bc.0 Max-Forwards: 69 From: "FAX" <sip:+9997@100.64.10.2>;tag=as4cfa9905 To: <sip:+0598@100.64.10.1;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 13.13.1 Content-Length: 0 Contact: <sip:btpsh-58656af1-769f-1@X.X.X.X> 20:59:00.388481 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 910 INVITE sip:btpsh-58656af1-769f-1@X.X.X.X <sip:btpsh-58656af1-769f-1@X.X.X.X> SIP/2.0 Via: SIP/2.0/UDP Y.Y.Y.Y:5060;branch=z9hG4bKm3trae309oacrlkm13h0sb0000g00.1 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Content-Length: 273 Max-Forwards: 69 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO Contact: <sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp> Accept-Encoding: identity Content-Disposition: session;handling=required Content-Type: application/sdp v=0 o=- 1510281183 168103905 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=image 17818 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:1800 a=T38FaxMaxDatagram:260 a=T38FaxUdpEC:t38UDPRedundancy 20:59:00.390679 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 420 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP Y.Y.Y.Y:5060;branch=z9hG4bKm3trae309oacrlkm13h0sb0000g00.1;rport=5060 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Content-Length: 0 20:59:00.395723 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 890 INVITE sip:+9997@100.64.10.2:5060 SIP/2.0 Via: SIP/2.0/UDP 100.64.10.1;branch=z9hG4bK720b.88d9cf77da5d6f5fa4aac2c5623d8ac4.0 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Content-Length: 289 Max-Forwards: 68 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO Accept-Encoding: identity Content-Disposition: session;handling=required Content-Type: application/sdp Contact: <sip:btpsh-58656af1-769f-1@X.X.X.X> v=0 o=- 1510281183 168103905 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=image 62130 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:1800 a=T38FaxMaxDatagram:260 a=T38FaxUdpEC:t38UDPRedundancy a=nortpproxy:yes 20:59:00.397160 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 580 SIP/2.0 100 Trying Via: SIP/2.0/UDP 100.64.10.1;branch=z9hG4bK720b.88d9cf77da5d6f5fa4aac2c5623d8ac4.0;received=100.64.10.1;rport=5060 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:+9997@100.64.10.2:5060> Content-Length: 0 20:59:00.397344 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 875 SIP/2.0 200 OK Via: SIP/2.0/UDP 100.64.10.1;branch=z9hG4bK720b.88d9cf77da5d6f5fa4aac2c5623d8ac4.0;received=100.64.10.1;rport=5060 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:+9997@100.64.10.2:5060> Content-Type: application/sdp Content-Length: 266 v=0 o=root 79038203 79038204 IN IP4 100.64.10.2 s=Asterisk PBX 13.13.1 c=IN IP4 100.64.10.2 t=0 0 m=image 4882 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy 20:59:00.405004 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 874 SIP/2.0 200 OK From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 285 Via: SIP/2.0/UDP Y.Y.Y.Y:5060;rport=5060;branch=z9hG4bKm3trae309oacrlkm13h0sb0000g00.1 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=root 79038203 79038204 IN IP4 X.X.X.X s=Asterisk PBX 13.13.1 c=IN IP4 X.X.X.X t=0 0 m=image 57810 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy a=nortpproxy:yes 20:59:00.896864 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 875 SIP/2.0 200 OK Via: SIP/2.0/UDP 100.64.10.1;branch=z9hG4bK720b.88d9cf77da5d6f5fa4aac2c5623d8ac4.0;received=100.64.10.1;rport=5060 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:+9997@100.64.10.2:5060> Content-Type: application/sdp Content-Length: 266 v=0 o=root 79038203 79038204 IN IP4 100.64.10.2 s=Asterisk PBX 13.13.1 c=IN IP4 100.64.10.2 t=0 0 m=image 4882 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy 20:59:00.904049 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 874 TSIP/2.0 200 OK From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 285 Via: SIP/2.0/UDP Y.Y.Y.Y:5060;rport=5060;branch=z9hG4bKm3trae309oacrlkm13h0sb0000g00.1 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=root 79038203 79038204 IN IP4 X.X.X.X s=Asterisk PBX 13.13.1 c=IN IP4 X.X.X.X t=0 0 m=image 57810 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy a=nortpproxy:yes And so on until timeout and BYE

3 4

Astersik kamailio integration.
by Deep Pratap Singh Tomar 29 Dec '16

29 Dec '16

Hi I have modified the file so that if i send register request to kamailio it sud register on asterisk also. i have shared you the code but i cant see anyting happening in the asterisk server. please help me ASAP. L&T Technology Services Ltd www.LntTechservices.com<http://www.lnttechservices.com/> This Email may contain confidential or privileged information for the intended recipient (s). If you are not the intended recipient, please do not use or disseminate the information, notify the sender and delete it from your system.

2 1

Re: [SR-Users] Kamailio behind NAT, ACK to private IP not advertised public IP.
by Pranathi Venkatayogi 29 Dec '16

29 Dec '16

I implemented full NAT logic as per the sample config. Still unable to resolve the issue. How do I let Kamailio change record_route header to use public ip address? Please help!!! (attached are latest scripts) From: Pranathi Venkatayogi Sent: Wednesday, December 28, 2016 12:39 PM To: 'sr-users(a)lists.sip-router.org' <sr-users(a)lists.sip-router.org> Subject: Kamailio behind NAT, ACK to private IP not advertised public IP. Hi, I am encountering the same problem described in google groups<https://groups.google.com/forum/#!topic/2600hz-dev/-xvUZUrv4Y4>. However I dint not find any resolution hence writing again. 200 OK sent from the server has private Ip in its record route. As you see below, though the message is received on public IP (63.149.103.72) , the record route is set to private IP (172.31.211.31) I used listen with advertise of public IP, it did not work. Please find attached the config I am using. How do I change it send public ip only when talking to external world. Can someone point to me clear documentation how to configure Kamailio for NAT traversal. The following message is sent from Kamailio behind NAT to the public computer. 2016-12-27 17:19:24.526875 [blink.exe 5652]: RECEIVED: Packet 123, +0:08:42.690309 63.149.103.72:5061 -(SIP over TLS)-> 10.0.0.6:62912 SIP/2.0 200 OK Via: SIP/2.0/TLS 10.0.0.6:62912;rport=62912;received=50.175.10.190;branch=z9hG4bKPj2e381a96979945bd969989ffe9dca3a9;alias Record-Route: <sip:172.31.211.31:5061;transport=tls;lr;did=5.5e5> Call-ID: eb8670eec4354acdb69fd26f5625b75c From: "cust1" <sip:cust1@translation.sms-test.cyracom.com>;tag=2f25d2ae690747c48c874e0b415ca03c To: <sip:spanish@translation.sms-test.cyracom.com>;tag=1c33ad41f6f44cae8ae8e060f30fe119 CSeq: 4665 INVITE Server: Blink 3.0.0 (Windows) Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Contact: <sip:75329410@10.0.27.108:61381;transport=tls> Supported: 100rel, replaces, norefersub, gruu Content-Type: application/sdp Content-Length: 355 v=0 o=- 3691844303 3691844304 IN IP4 10.0.27.108 s=Blink 3.0.0 (Windows) t=0 0 m=message 2855 TCP/TLS/MSRP * c=IN IP4 10.0.27.108 a=path:msrps://10.0.27.108:2855/261d3f47be25612cc77c;tcp a=accept-types:message/cpim text/* image/* application/im-iscomposing+xml a=accept-wrapped-types:text/* image/* application/im-iscomposing+xml a=setup:active -- The following is the ACK sent by public computer in reply to the above message. Note this message never reaches the Kamailio server as it is sent to private IP. 2016-12-27 17:19:24.526875 [blink.exe 5652]: SENDING: Packet 124, +0:08:42.690309 10.0.0.6:62944 -(SIP over TLS)-> 172.31.211.31:5061 ACK sip:75329410@10.0.27.108:61381;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.0.0.6:62944;rport;branch=z9hG4bKPj7df757862e6546beba18a646cb965ba2;alias Max-Forwards: 70 From: "cust1" <sip:cust1@translation.sms-test.cyracom.com>;tag=2f25d2ae690747c48c874e0b415ca03c To: <sip:spanish@translation.sms-test.cyracom.com>;tag=1c33ad41f6f44cae8ae8e060f30fe119 Call-ID: eb8670eec4354acdb69fd26f5625b75c CSeq: 4665 ACK Route: <sip:172.31.211.31:5061;transport=tls;lr;did=5.5e5> User-Agent: Blink 3.0.0 (Windows) Content-Length: 0 Thanks, Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

3 4

String split function
by Grant Bagdasarian 29 Dec '16

29 Dec '16

Hi, Is there a string split function built into Kamailio for use in the cfg script? I'd like to split a string of key/value pairs inside the cfg script. For example: Key1=Value1;Key2=Value2; First split on ; Then on = I think I can use the String Transformation {s.select,index,separator}, but I was hoping there is a built in function. {param.valueat,index[, delimiter]} may also be an option. Regards, Grant

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sr-users December 2016