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37522 discussions

rtpproxy and CANCEL in serial forked scenarios
by Andreas Granig 05 Apr '12

05 Apr '12

Hi, I ran into a scenario with couple of serial forks where kamailio loops to itself and, due to the looping, the INVITE to a new branch happens before the CANCEL to an old branch. What I do at the moment is force rtpproxy in branch route, and stop rtpproxy in the failure route. The problem with this scenario is that in rtpproxy_offer and unforce_rtpproxy, the rtpproxy module only passes the call-id and from-tag to rtpproxy (because there is no to-tag yet), which then in unforce_rtpproxy for a CANCEL deletes all calls related to it (because it can only match on from-tag and call-id, obviously). This means that when I do a subsequent rtpproxy_answer for my new branch, rtpproxy doesn't find the session anymore, since it has been removed with the CANCEL. A fix I can think of is to also pass the branch of the top-most via to rtpproxy in order to perform a more fine-grained matching. Are there solutions to that out there somewhere, or is it something I should just introduce, eg. as a new param to offer/answer/unforce functions? Objections? Other approaches? Andreas

3 2

in doubt about in-dialog INVITE Route params processing
by Andrew Pogrebennyk 05 Apr '12

05 Apr '12

I've been pondering an issue with Route header parameters being not mirrored by kamailio proxy into Record-Route field on in-dialog requests over the last few hours so I thought I'd just whether I'm missing something obvious. The call scenario in UA -> lb -> proxy -> sems. Everything is fine with initial INVITE transaction where we add rr param ;rtpprx=yes. (We also use dialog module.) The UA does re-INVITE where it mirrors RR headers to Route set just fine. However, when forwarding re-INVITE kamailio proxy removes own Route header and inserts RR header without rtpproxy, did etc params. The kamailio default config from branch 3.1 routes in-dialog requests in the same fashion and I can't tell really if this is how it is supposed to work? I can workaround this by add_rr_param() but shouldn't missing did param cause a problem with dialog matching? Here is the debug with log level 3: > Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: rr [loose.c:792]: Topmost route URI: 'sip:127.0.0.1:5062;lr;ftag=8pTM6oKWVtC4H03478xjIR34SSsYuTU5;did=8a3.aab32831;mpd=ii;rtpprx=yes;vsf=aDNlNDEjPSkBJB9/YykkNnYYdTZ0KGUsfGZxVndVe1M-' is me > Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: <core> [parser/msg_parser.c:103]: found end of header > Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: rr [loose.c:257]: No next Route HF found > Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: rr [loose.c:811]: No next URI found > Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: rr [rr_cb.c:97]: callback id 1 entered with <lr;ftag=8pTM6oKWVtC4H03478xjIR34SSsYuTU5;did=8a3.aab32831;mpd=ii;rtpprx=yes;vsf=aDNlNDEjPSkBJB9/YykkNnYYdTZ0KGUsfGZxVndVe1M-> > Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: dialog [dlg_handlers.c:910]: route param is '8a3.aab32831' (len=12) > Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: dialog [dlg_hash.c:442]: ref dlg 0x7f0a0be918b0 with 1 -> 3 > Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: dialog [dlg_hash.c:444]: dialog id=327302058 found on entry 936 > Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: dialog [dlg_hash.c:757]: dialog 0x7f0a0be918b0 changed from state 4 to state 4, due event 8 > Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: dialog [dlg_handlers.c:1042]: sequential request successfully processed > Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: dialog [dlg_timer.c:117]: inserting 0x7f0a0be91900 for 127601858 > Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: dialog [dlg_hash.c:411]: cseq is 10760 > Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: rr [rr_cb.c:97]: callback id 0 entered with <lr;ftag=8pTM6oKWVtC4H03478xjIR34SSsYuTU5;did=8a3.aab32831;mpd=ii;rtpprx=yes;vsf=aDNlNDEjPSkBJB9/YykkNnYYdTZ0KGUsfGZxVndVe1M-> > Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: uac [from.c:421]: getting 'vsf' Route param > Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: uac [from.c:429]: route param is 'aDNlNDEjPSkBJB9/YykkNnYYdTZ0KGUsfGZxVndVe1M-' (len=44) > Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: uac [from.c:479]: decoded uris are: new=[sip:43991001@192.168.51.210] old=[sip:sipwise-user1@192.168.51.210] > Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: rr [loose.c:983]: params are <;lr;ftag=8pTM6oKWVtC4H03478xjIR34SSsYuTU5;did=8a3.aab32831;mpd=ii;rtpprx=yes;vsf=aDNlNDEjPSkBJB9/YykkNnYYdTZ0KGUsfGZxVndVe1M-> > Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: siputils [checks.c:76]: totag found > Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: textops [textops.c:1789]: content type is 196611 > Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: rr [loose.c:983]: params are <;lr;ftag=8pTM6oKWVtC4H03478xjIR34SSsYuTU5;did=8a3.aab32831;mpd=ii;rtpprx=yes;vsf=aDNlNDEjPSkBJB9/YykkNnYYdTZ0KGUsfGZxVndVe1M-> > Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: rr [loose.c:983]: params are <;lr;ftag=8pTM6oKWVtC4H03478xjIR34SSsYuTU5;did=8a3.aab32831;mpd=ii;rtpprx=yes;vsf=aDNlNDEjPSkBJB9/YykkNnYYdTZ0KGUsfGZxVndVe1M-> > Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: INFO: <script>: Use mediaproxy for forward direction for IPv4/IPv4 - M=INVITE R=sip:127.0.0.1:5080 F=sip:sipwise-user1@192.168.51.210 T=sip:43991002@192.168.51.210 IP=192.168.51.1:35189 (127.0.0.1:5060) ID=M6obxlxpUOPPR3CQYE-d2t9-fgybDo6y > Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type <application/sdp> found valid > Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: DEBUG: rtpproxy [rtpproxy.c:2260]: proxy reply: 30080 192.168.51.210 4#012 > Apr 4 13:23:21 sp1 /usr/sbin/kamailio[7520]: INFO: <script>: Relaying request - M=INVITE R=sip:127.0.0.1:5080 F=sip:sipwise-user1@192.168.51.210 T=sip:43991002@192.168.51.210 IP=192.168.51.1:35189 (127.0.0.1:5060) ID=M6obxlxpUOPPR3CQYE-d2t9-fgybDo6y

2 2

Problem with xpath updates of XCAP documents
by Peter Dunkley 05 Apr '12

05 Apr '12

Hi, I have been experimenting with uploading, and modifying existing, XCAP documents. I have been trying out the example from RFC 4827 and I can't get it to work with Kamailio. To upload the document (this works) I use the command: curl --digest -u 111:111 -T pidf.xml -X PUT http://192.168.0.130:5060/xcap-root/pidf-manipulation/users/sip:111@mydomai… To modify the document (this doesn't work) I use the command: curl --digest -u 111:111 -T pidf_modify.xml -H "Content-Type: application/xcap-el+xml" -X PUT http://192.168.0.130:5060/xcap-root/pidf-manipulation/users/sip:111@mydomai…'x8eg92n'%5d/note When I try to modify the document I get the error: "Apr 4 17:12:08 pd-laptop-linux kamailio[16862]: ERROR: xcap_server [xcap_server.c:596]: no new content" I went through the XCAP server code and added some debug and found that the check on xcap_misc.c:491 is succeeding (which means "no selection for xpath parent expression") and the document is not being changed. Also, from my limited understanding of XPath and libxml2, I would have expected the code path which should have been followed to go through the else clause of the conditional on xcap_misc.c:468 - which means I don't think that the xcaps_xpath_set() function is doing the right thing at all. Is there anyone here that is more familiar with libxml2 that can give me any pointers here. As far as I can tell the XCAP server code looks OK to me, and I believe the documents (attached) and my XCAP queries are correct as well. Thanks, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd

2 3

Acc Module - logging forked calls.
by Stephen Dodge (Bistech) 05 Apr '12

05 Apr '12

Hello, We currently use the acc module to capture information to generate CDR missed call info. I have now introduced serial forking and I would like to see if I can improve how accounting is working for missed calls. I am using the following configuration. # turn accounting on for all new calls if (is_method("INVITE")){ setflag(1); # do accouting setflag(2); # missed calls record_route(); } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")){ setflag(1); # do accounting setflag(3); # even if the transaction fails xdbg("## BISNET ## - RTPPROXY CHECK \n"); # clear any rtpproxy sessions in progress unforce_rtp_proxy(); } I've noticed that accounting for missed calls is generated for the main branch (1st destination) only, whether this is a 486 response or a 408 on expiry of the 'fr_inv_timer'. After a receiving a failure another destination is created/tried by the script, we are allowing up to 4 serial forks. I do see the 183 message for each fork and if the call is answered then the acc table is updated correctly. How can I log the missed call response for each fork, or only log the final failure message for the call? Logging failed transactions seems to put them into the main acc table and not the missed call table which I would have expected. Thanks, Steve. ________________________________ Information in this message, including any attachments, is confidential to the person to whom it is addressed and may be legally privileged. If you are not the intended recipient please notify the sender and delete the message from your system. Please note that Bistech Group plc, Bistech plc, Bisnet Limited and the sender do not accept any responsibility for viruses. It is your responsibility to check the e-mail and any attachments for viruses. Calls may be monitored and recorded.

2 2

KAMAILIO no audio establishment
by Ryan Gholam 05 Apr '12

05 Apr '12

Hello , i am facing an issue concerning kamailio where i am trying to connect the kamailio to the asterisk using a private ip and the kamailio is connected using a public ip for the clients .I am trying to create a call from the client the phone rings but there is no audio conversation happening , what is the best method to use ? N.B : i have tried RTPproxy but it didnt work i think because their is no proper NATING . Thank you

2 1

Questions on multihomed setup
by Karsten Horsmann 05 Apr '12

05 Apr '12

Hi all! I try to setup an kamailio 3.2.2 that is multihomed (mhomed=1) with public and private ip addresses. It should *always* use rtpproxy for all rtp traffic. In wireshark i see that an ACK is missing, the log shows me an error in rr module: rr [loose.c:103]: failed to parse Route HF So my questions are: 1.) Is my NATMANAGE route logic correct? 2.) Why is the ACK dropped by kamailio (ends without transactions)? 3.) Should the SDP/contact field to the 3rd party sipserver have the public ip??? Now it becomes the private IVR ip address. Thanks! Cheers Karsten ########################################## Kamailio public ip 222.222.222.222 Kamailio private ip 172.20.100.74 ########################################## # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT xlog("NATMANAGE M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (isbflagset(FLB_NATB)) return; if (dst_ip == 222.222.222.222) { xlog("rtpproxy_manage ext to int\n"); rtpproxy_manage("FAEI","172.20.10.74"); } else { xlog("rtpproxy_manage int to ext\n"); rtpproxy_manage("FAIE","222.222.222.222"); } if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } ########################################## Kamailio public ip 222.222.222.222 Kamailio private ip 172.20.100.74 IVR ip 172.20.100.61 ########################################## U 2012/04/04 14:05:59.918472 217.10.79.9:5060 -> 222.222.222.222:5060 INVITE sip:kalkbrenner@222.222.222.222:5060 SIP/2.0. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Record-Route: <sip:172.20.40.2;lr=on>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK6021.0ca5d221.1. Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK6021.0ca5d221.1. Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.226;branch=z9hG4bK703f9495. Via: SIP/2.0/UDP 217.10.67.13:5060;received=217.10.67.13;branch=z9hG4bK703f9495;rport=5060. Max-Forwards: 67. From: "alice" <sip:alice@sipgate.de>;tag=as0ebb1d67. To: <sip:bob@sipgate.de>. Contact: <sip:alice@217.10.67.13>. Call-ID: 2d9296f200e0212711db47c036e68917(a)sipgate.de. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 428. . v=0. o=root 1567926566 1567926566 IN IP4 217.10.67.13. s=sipgate VoIP GW. c=IN IP4 217.10.67.13. t=0 0. m=audio 17430 RTP/AVP 8 0 3 97 18 112 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:3 GSM/8000. a=rtpmap:97 iLBC/8000. a=fmtp:97 mode=30. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:112 G726-32/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 2012/04/04 14:05:59.919629 222.222.222.222:5060 -> 217.10.79.9:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK6021.0ca5d221.1;rport=5060. Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK6021.0ca5d221.1. Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.226;branch=z9hG4bK703f9495. Via: SIP/2.0/UDP 217.10.67.13:5060;received=217.10.67.13;branch=z9hG4bK703f9495;rport=5060. From: "alice" <sip:alice@sipgate.de>;tag=as0ebb1d67. To: <sip:bob@sipgate.de>. Call-ID: 2d9296f200e0212711db47c036e68917(a)sipgate.de. CSeq: 102 INVITE. Server: Kamailio. Content-Length: 0. Warning: 392 222.222.222.222:5060 "Noisy feedback tells: pid=12262 req_src_ip=217.10.79.9 req_src_port=5060 in_uri=sip:kalkbrenner@222.222.222.222:5060 out_uri=sip:kalkbrenner@172.20.100.61 via_cnt==4". . U 2012/04/04 14:05:59.921070 172.20.100.74:5060 -> 172.20.100.61:5060 INVITE sip:kalkbrenner@172.20.100.61 SIP/2.0. Record-Route: <sip:172.20.100.74;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Record-Route: <sip:222.222.222.222;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Record-Route: <sip:172.20.40.2;lr=on>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Via: SIP/2.0/UDP 172.20.100.74;branch=z9hG4bK6021.34ce1643.0. Via: SIP/2.0/UDP 217.10.79.9:5060;rport=5060;branch=z9hG4bK6021.0ca5d221.1. Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK6021.0ca5d221.1. Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.226;branch=z9hG4bK703f9495. Via: SIP/2.0/UDP 217.10.67.13:5060;received=217.10.67.13;branch=z9hG4bK703f9495;rport=5060. Max-Forwards: 66. From: "alice" <sip:alice@sipgate.de>;tag=as0ebb1d67. To: <sip:bob@sipgate.de>. Contact: <sip:alice@217.10.67.13>. Call-ID: 2d9296f200e0212711db47c036e68917(a)sipgate.de. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 448. . v=0. o=root 1567926566 1567926566 IN IP4 222.222.222.222. s=sipgate VoIP GW. c=IN IP4 222.222.222.222. t=0 0. m=audio 47798 RTP/AVP 8 0 3 97 18 112 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:3 GSM/8000. a=rtpmap:97 iLBC/8000. a=fmtp:97 mode=30. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:112 G726-32/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 U 2012/04/04 14:05:59.925396 172.20.100.61:5060 -> 172.20.100.74:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 172.20.100.74;branch=z9hG4bK6021.34ce1643.0. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK6021.0ca5d221.1;rport=5060. Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK6021.0ca5d221.1. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK703f9495;received=217.10.68.226. Via: SIP/2.0/UDP 217.10.67.13:5060;rport=5060;branch=z9hG4bK703f9495;received=217.10.67.13. From: "alice" <sip:alice@sipgate.de>;tag=as0ebb1d67. To: <sip:bob@sipgate.de>;tag=ds-46dd3aff-40a69134. Call-ID: 2d9296f200e0212711db47c036e68917(a)sipgate.de. CSeq: 102 INVITE. Content-Length: 0. . U 2012/04/04 14:05:59.966415 172.20.100.61:5060 -> 172.20.100.74:5060 SIP/2.0 200 Ok. Via: SIP/2.0/UDP 172.20.100.74;branch=z9hG4bK6021.34ce1643.0. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK6021.0ca5d221.1;rport=5060. Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK6021.0ca5d221.1. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK703f9495;received=217.10.68.226. Via: SIP/2.0/UDP 217.10.67.13:5060;rport=5060;branch=z9hG4bK703f9495;received=217.10.67.13. From: "alice" <sip:alice@sipgate.de>;tag=as0ebb1d67. To: <sip:bob@sipgate.de>;tag=ds-46dd3aff-40a69134. Call-ID: 2d9296f200e0212711db47c036e68917(a)sipgate.de. CSeq: 102 INVITE. Content-Length: 218. Content-Type: application/sdp. Record-Route: <sip:172.20.100.74;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Record-Route: <sip:222.222.222.222;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Record-Route: <sip:172.20.40.2;lr=on>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Supported: replaces. Supported: 100rel. Allow: INVITE. Allow: ACK. Allow: BYE. Allow: CANCEL. Allow: OPTIONS. Allow: NOTIFY. Allow: REFER. Allow: PRACK. Allow: INFO. Allow: UPDATE. Allow: MESSAGE. Contact: <sip:172.20.100.61>. . v=0. o=aculab-0204CC01 1431055130 1431055130 IN IP4 172.20.100.61. s=-. c=IN IP4 172.20.100.71. t=0 0. m=audio 19512 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2012/04/04 14:05:59.966775 222.222.222.222:5060 -> 217.10.79.9:5060 SIP/2.0 200 Ok. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK6021.0ca5d221.1;rport=5060. Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK6021.0ca5d221.1. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK703f9495;received=217.10.68.226. Via: SIP/2.0/UDP 217.10.67.13:5060;rport=5060;branch=z9hG4bK703f9495;received=217.10.67.13. From: "alice" <sip:alice@sipgate.de>;tag=as0ebb1d67. To: <sip:bob@sipgate.de>;tag=ds-46dd3aff-40a69134. Call-ID: 2d9296f200e0212711db47c036e68917(a)sipgate.de. CSeq: 102 INVITE. Content-Length: 236. Content-Type: application/sdp. Record-Route: <sip:172.20.100.74;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Record-Route: <sip:222.222.222.222;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Record-Route: <sip:172.20.40.2;lr=on>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Supported: replaces. Supported: 100rel. Allow: INVITE. Allow: ACK. Allow: BYE. Allow: CANCEL. Allow: OPTIONS. Allow: NOTIFY. Allow: REFER. Allow: PRACK. Allow: INFO. Allow: UPDATE. Allow: MESSAGE. Contact: <sip:172.20.100.61>. . v=0. o=aculab-0204CC01 1431055130 1431055130 IN IP4 222.222.222.222. s=-. c=IN IP4 222.222.222.222. t=0 0. m=audio 52564 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=nortpproxy:yes. U 2012/04/04 14:06:00.009715 217.10.79.9:5060 -> 222.222.222.222:5060 ACK sip:222.222.222.222:5060 SIP/2.0. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK6021.0ca5d221.3. Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK6021.0ca5d221.3. Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.226;branch=z9hG4bK55e1bf7a. Via: SIP/2.0/UDP 217.10.67.13:5060;received=217.10.67.13;branch=z9hG4bK55e1bf7a;rport=5060. Route: <sip:222.222.222.222;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>,<sip:172.20.100.74;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Max-Forwards: 67. From: "alice" <sip:alice@sipgate.de>;tag=as0ebb1d67. To: <sip:bob@sipgate.de>;tag=ds-46dd3aff-40a69134. Contact: <sip:alice@217.10.67.13>. Call-ID: 2d9296f200e0212711db47c036e68917(a)sipgate.de. CSeq: 102 ACK. Content-Length: 0. X-hint: rr-enforced. . U 2012/04/04 14:06:00.467283 172.20.100.61:5060 -> 172.20.100.74:5060 SIP/2.0 200 Ok. Via: SIP/2.0/UDP 172.20.100.74;branch=z9hG4bK6021.34ce1643.0. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK6021.0ca5d221.1;rport=5060. Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK6021.0ca5d221.1. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK703f9495;received=217.10.68.226. Via: SIP/2.0/UDP 217.10.67.13:5060;rport=5060;branch=z9hG4bK703f9495;received=217.10.67.13. From: "alice" <sip:alice@sipgate.de>;tag=as0ebb1d67. To: <sip:bob@sipgate.de>;tag=ds-46dd3aff-40a69134. Call-ID: 2d9296f200e0212711db47c036e68917(a)sipgate.de. CSeq: 102 INVITE. Content-Length: 218. Content-Type: application/sdp. Record-Route: <sip:172.20.100.74;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Record-Route: <sip:222.222.222.222;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Record-Route: <sip:172.20.40.2;lr=on>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Supported: replaces. Supported: 100rel. Allow: INVITE. Allow: ACK. Allow: BYE. Allow: CANCEL. Allow: OPTIONS. Allow: NOTIFY. Allow: REFER. Allow: PRACK. Allow: INFO. Allow: UPDATE. Allow: MESSAGE. Contact: <sip:172.20.100.61>. . v=0. o=aculab-0204CC01 1431055130 1431055130 IN IP4 172.20.100.61. s=-. c=IN IP4 172.20.100.71. t=0 0. m=audio 19512 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2012/04/04 14:06:00.467915 222.222.222.222:5060 -> 217.10.79.9:5060 SIP/2.0 200 Ok. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK6021.0ca5d221.1;rport=5060. Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK6021.0ca5d221.1. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK703f9495;received=217.10.68.226. Via: SIP/2.0/UDP 217.10.67.13:5060;rport=5060;branch=z9hG4bK703f9495;received=217.10.67.13. From: "alice" <sip:alice@sipgate.de>;tag=as0ebb1d67. To: <sip:bob@sipgate.de>;tag=ds-46dd3aff-40a69134. Call-ID: 2d9296f200e0212711db47c036e68917(a)sipgate.de. CSeq: 102 INVITE. Content-Length: 236. Content-Type: application/sdp. Record-Route: <sip:172.20.100.74;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Record-Route: <sip:222.222.222.222;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Record-Route: <sip:172.20.40.2;lr=on>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Supported: replaces. Supported: 100rel. Allow: INVITE. Allow: ACK. Allow: BYE. Allow: CANCEL. Allow: OPTIONS. Allow: NOTIFY. Allow: REFER. Allow: PRACK. Allow: INFO. Allow: UPDATE. Allow: MESSAGE. Contact: <sip:172.20.100.61>. . v=0. o=aculab-0204CC01 1431055130 1431055130 IN IP4 222.222.222.222. s=-. c=IN IP4 222.222.222.222. t=0 0. m=audio 52564 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=nortpproxy:yes. U 2012/04/04 14:06:01.468092 172.20.100.61:5060 -> 172.20.100.74:5060 SIP/2.0 200 Ok. Via: SIP/2.0/UDP 172.20.100.74;branch=z9hG4bK6021.34ce1643.0. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK6021.0ca5d221.1;rport=5060. Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK6021.0ca5d221.1. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK703f9495;received=217.10.68.226. Via: SIP/2.0/UDP 217.10.67.13:5060;rport=5060;branch=z9hG4bK703f9495;received=217.10.67.13. From: "alice" <sip:alice@sipgate.de>;tag=as0ebb1d67. To: <sip:bob@sipgate.de>;tag=ds-46dd3aff-40a69134. Call-ID: 2d9296f200e0212711db47c036e68917(a)sipgate.de. CSeq: 102 INVITE. Content-Length: 218. Content-Type: application/sdp. Record-Route: <sip:172.20.100.74;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Record-Route: <sip:222.222.222.222;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Record-Route: <sip:172.20.40.2;lr=on>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Supported: replaces. Supported: 100rel. Allow: INVITE. Allow: ACK. Allow: BYE. Allow: CANCEL. Allow: OPTIONS. Allow: NOTIFY. Allow: REFER. Allow: PRACK. Allow: INFO. Allow: UPDATE. Allow: MESSAGE. Contact: <sip:172.20.100.61>. . v=0. o=aculab-0204CC01 1431055130 1431055130 IN IP4 172.20.100.61. s=-. c=IN IP4 172.20.100.71. t=0 0. m=audio 19512 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2012/04/04 14:06:01.469460 222.222.222.222:5060 -> 217.10.79.9:5060 SIP/2.0 200 Ok. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK6021.0ca5d221.1;rport=5060. Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK6021.0ca5d221.1. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK703f9495;received=217.10.68.226. Via: SIP/2.0/UDP 217.10.67.13:5060;rport=5060;branch=z9hG4bK703f9495;received=217.10.67.13. From: "alice" <sip:alice@sipgate.de>;tag=as0ebb1d67. To: <sip:bob@sipgate.de>;tag=ds-46dd3aff-40a69134. Call-ID: 2d9296f200e0212711db47c036e68917(a)sipgate.de. CSeq: 102 INVITE. Content-Length: 236. Content-Type: application/sdp. Record-Route: <sip:172.20.100.74;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Record-Route: <sip:222.222.222.222;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Record-Route: <sip:172.20.40.2;lr=on>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Supported: replaces. Supported: 100rel. Allow: INVITE. Allow: ACK. Allow: BYE. Allow: CANCEL. Allow: OPTIONS. Allow: NOTIFY. Allow: REFER. Allow: PRACK. Allow: INFO. Allow: UPDATE. Allow: MESSAGE. Contact: <sip:172.20.100.61>. . v=0. o=aculab-0204CC01 1431055130 1431055130 IN IP4 222.222.222.222. s=-. c=IN IP4 222.222.222.222. t=0 0. m=audio 52564 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=nortpproxy:yes. U 2012/04/04 14:06:03.468631 172.20.100.61:5060 -> 172.20.100.74:5060 SIP/2.0 200 Ok. Via: SIP/2.0/UDP 172.20.100.74;branch=z9hG4bK6021.34ce1643.0. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK6021.0ca5d221.1;rport=5060. Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK6021.0ca5d221.1. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK703f9495;received=217.10.68.226. Via: SIP/2.0/UDP 217.10.67.13:5060;rport=5060;branch=z9hG4bK703f9495;received=217.10.67.13. From: "alice" <sip:alice@sipgate.de>;tag=as0ebb1d67. To: <sip:bob@sipgate.de>;tag=ds-46dd3aff-40a69134. Call-ID: 2d9296f200e0212711db47c036e68917(a)sipgate.de. CSeq: 102 INVITE. Content-Length: 218. Content-Type: application/sdp. Record-Route: <sip:172.20.100.74;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Record-Route: <sip:222.222.222.222;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Record-Route: <sip:172.20.40.2;lr=on>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Supported: replaces. Supported: 100rel. Allow: INVITE. Allow: ACK. Allow: BYE. Allow: CANCEL. Allow: OPTIONS. Allow: NOTIFY. Allow: REFER. Allow: PRACK. Allow: INFO. Allow: UPDATE. Allow: MESSAGE. Contact: <sip:172.20.100.61>. . v=0. o=aculab-0204CC01 1431055130 1431055130 IN IP4 172.20.100.61. s=-. c=IN IP4 172.20.100.71. t=0 0. m=audio 19512 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2012/04/04 14:06:03.472282 222.222.222.222:5060 -> 217.10.79.9:5060 SIP/2.0 200 Ok. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK6021.0ca5d221.1;rport=5060. Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK6021.0ca5d221.1. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK703f9495;received=217.10.68.226. Via: SIP/2.0/UDP 217.10.67.13:5060;rport=5060;branch=z9hG4bK703f9495;received=217.10.67.13. From: "alice" <sip:alice@sipgate.de>;tag=as0ebb1d67. To: <sip:bob@sipgate.de>;tag=ds-46dd3aff-40a69134. Call-ID: 2d9296f200e0212711db47c036e68917(a)sipgate.de. CSeq: 102 INVITE. Content-Length: 236. Content-Type: application/sdp. Record-Route: <sip:172.20.100.74;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Record-Route: <sip:222.222.222.222;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Record-Route: <sip:172.20.40.2;lr=on>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Supported: replaces. Supported: 100rel. Allow: INVITE. Allow: ACK. Allow: BYE. Allow: CANCEL. Allow: OPTIONS. Allow: NOTIFY. Allow: REFER. Allow: PRACK. Allow: INFO. Allow: UPDATE. Allow: MESSAGE. Contact: <sip:172.20.100.61>. . v=0. o=aculab-0204CC01 1431055130 1431055130 IN IP4 222.222.222.222. s=-. c=IN IP4 222.222.222.222. t=0 0. m=audio 52564 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=nortpproxy:yes. U 2012/04/04 14:06:07.469772 172.20.100.61:5060 -> 172.20.100.74:5060 SIP/2.0 200 Ok. Via: SIP/2.0/UDP 172.20.100.74;branch=z9hG4bK6021.34ce1643.0. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK6021.0ca5d221.1;rport=5060. Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK6021.0ca5d221.1. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK703f9495;received=217.10.68.226. Via: SIP/2.0/UDP 217.10.67.13:5060;rport=5060;branch=z9hG4bK703f9495;received=217.10.67.13. From: "alice" <sip:alice@sipgate.de>;tag=as0ebb1d67. To: <sip:bob@sipgate.de>;tag=ds-46dd3aff-40a69134. Call-ID: 2d9296f200e0212711db47c036e68917(a)sipgate.de. CSeq: 102 INVITE. Content-Length: 218. Content-Type: application/sdp. Record-Route: <sip:172.20.100.74;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Record-Route: <sip:222.222.222.222;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Record-Route: <sip:172.20.40.2;lr=on>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Supported: replaces. Supported: 100rel. Allow: INVITE. Allow: ACK. Allow: BYE. Allow: CANCEL. Allow: OPTIONS. Allow: NOTIFY. Allow: REFER. Allow: PRACK. Allow: INFO. Allow: UPDATE. Allow: MESSAGE. Contact: <sip:172.20.100.61>. . v=0. o=aculab-0204CC01 1431055130 1431055130 IN IP4 172.20.100.61. s=-. c=IN IP4 172.20.100.71. t=0 0. m=audio 19512 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2012/04/04 14:06:07.470020 172.20.100.74:5060 -> 217.10.79.9:5060 SIP/2.0 200 Ok. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK6021.0ca5d221.1;rport=5060. Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK6021.0ca5d221.1. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK703f9495;received=217.10.68.226. Via: SIP/2.0/UDP 217.10.67.13:5060;rport=5060;branch=z9hG4bK703f9495;received=217.10.67.13. From: "alice" <sip:alice@sipgate.de>;tag=as0ebb1d67. To: <sip:bob@sipgate.de>;tag=ds-46dd3aff-40a69134. Call-ID: 2d9296f200e0212711db47c036e68917(a)sipgate.de. CSeq: 102 INVITE. Content-Length: 218. Content-Type: application/sdp. Record-Route: <sip:172.20.100.74;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Record-Route: <sip:222.222.222.222;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Record-Route: <sip:172.20.40.2;lr=on>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Supported: replaces. Supported: 100rel. Allow: INVITE. Allow: ACK. Allow: BYE. Allow: CANCEL. Allow: OPTIONS. Allow: NOTIFY. Allow: REFER. Allow: PRACK. Allow: INFO. Allow: UPDATE. Allow: MESSAGE. Contact: <sip:172.20.100.61>. . v=0. o=aculab-0204CC01 1431055130 1431055130 IN IP4 172.20.100.61. s=-. c=IN IP4 172.20.100.71. t=0 0. m=audio 19512 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2012/04/04 14:06:11.470798 172.20.100.61:5060 -> 172.20.100.74:5060 SIP/2.0 200 Ok. Via: SIP/2.0/UDP 172.20.100.74;branch=z9hG4bK6021.34ce1643.0. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK6021.0ca5d221.1;rport=5060. Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK6021.0ca5d221.1. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK703f9495;received=217.10.68.226. Via: SIP/2.0/UDP 217.10.67.13:5060;rport=5060;branch=z9hG4bK703f9495;received=217.10.67.13. From: "alice" <sip:alice@sipgate.de>;tag=as0ebb1d67. To: <sip:bob@sipgate.de>;tag=ds-46dd3aff-40a69134. Call-ID: 2d9296f200e0212711db47c036e68917(a)sipgate.de. CSeq: 102 INVITE. Content-Length: 218. Content-Type: application/sdp. Record-Route: <sip:172.20.100.74;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Record-Route: <sip:222.222.222.222;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Record-Route: <sip:172.20.40.2;lr=on>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Supported: replaces. Supported: 100rel. Allow: INVITE. Allow: ACK. Allow: BYE. Allow: CANCEL. Allow: OPTIONS. Allow: NOTIFY. Allow: REFER. Allow: PRACK. Allow: INFO. Allow: UPDATE. Allow: MESSAGE. Contact: <sip:172.20.100.61>. . v=0. o=aculab-0204CC01 1431055130 1431055130 IN IP4 172.20.100.61. s=-. c=IN IP4 172.20.100.71. t=0 0. m=audio 19512 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2012/04/04 14:06:11.471041 172.20.100.74:5060 -> 217.10.79.9:5060 SIP/2.0 200 Ok. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK6021.0ca5d221.1;rport=5060. Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK6021.0ca5d221.1. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK703f9495;received=217.10.68.226. Via: SIP/2.0/UDP 217.10.67.13:5060;rport=5060;branch=z9hG4bK703f9495;received=217.10.67.13. From: "alice" <sip:alice@sipgate.de>;tag=as0ebb1d67. To: <sip:bob@sipgate.de>;tag=ds-46dd3aff-40a69134. Call-ID: 2d9296f200e0212711db47c036e68917(a)sipgate.de. CSeq: 102 INVITE. Content-Length: 218. Content-Type: application/sdp. Record-Route: <sip:172.20.100.74;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Record-Route: <sip:222.222.222.222;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Record-Route: <sip:172.20.40.2;lr=on>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Supported: replaces. Supported: 100rel. Allow: INVITE. Allow: ACK. Allow: BYE. Allow: CANCEL. Allow: OPTIONS. Allow: NOTIFY. Allow: REFER. Allow: PRACK. Allow: INFO. Allow: UPDATE. Allow: MESSAGE. Contact: <sip:172.20.100.61>. . v=0. o=aculab-0204CC01 1431055130 1431055130 IN IP4 172.20.100.61. s=-. c=IN IP4 172.20.100.71. t=0 0. m=audio 19512 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2012/04/04 14:06:13.282635 217.247.216.82:12780 -> 222.222.222.222:5060 . . .............. U 2012/04/04 14:06:15.468037 172.20.100.61:5060 -> 172.20.100.74:5060 SIP/2.0 200 Ok. Via: SIP/2.0/UDP 172.20.100.74;branch=z9hG4bK6021.34ce1643.0. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK6021.0ca5d221.1;rport=5060. Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK6021.0ca5d221.1. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK703f9495;received=217.10.68.226. Via: SIP/2.0/UDP 217.10.67.13:5060;rport=5060;branch=z9hG4bK703f9495;received=217.10.67.13. From: "alice" <sip:alice@sipgate.de>;tag=as0ebb1d67. To: <sip:bob@sipgate.de>;tag=ds-46dd3aff-40a69134. Call-ID: 2d9296f200e0212711db47c036e68917(a)sipgate.de. CSeq: 102 INVITE. Content-Length: 218. Content-Type: application/sdp. Record-Route: <sip:172.20.100.74;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Record-Route: <sip:222.222.222.222;r2=on;lr=on;ftag=as0ebb1d67;nat=yes>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Record-Route: <sip:172.20.40.2;lr=on>. Record-Route: <sip:217.10.79.9;lr;ftag=as0ebb1d67>. Supported: replaces. Supported: 100rel. Allow: INVITE. Allow: ACK. Allow: BYE. Allow: CANCEL. Allow: OPTIONS. Allow: NOTIFY. Allow: REFER. Allow: PRACK. Allow: INFO. Allow: UPDATE. Allow: MESSAGE. Contact: <sip:172.20.100.61>. . v=0. o=aculab-0204CC01 1431055130 1431055130 IN IP4 172.20.100.61. s=-. c=IN IP4 172.20.100.71. t=0 0. m=audio 19512 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2012/04/04 14:06:15.468272 172.20.100.74:5060 -> 217.10.79.9:5060 SIP/2.0 200 Ok. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK6021.0ca5d221.1;rport=5060. Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK6021.0ca5d221.1. Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK703f9495;received=217.10.68.226. Via: SIP/2.0/UDP 217.10.67.13:5060;rport=5060;branch=z9hG4bK703f9495;received=217.10.67.13. From: "alice" <sip:alice@sipgate.de>;tag=as0ebb1d67. To: <sip:bob@sipgate.de>;tag=ds-46dd -- Mit freundlichen Grüßen *Karsten Horsmann*

2 2

LCR
by Jeff Russell 05 Apr '12

05 Apr '12

As a rule, I have managed to work my way through any issues however, I would really appreciate and help in getting the LCR module working (with postgres), what I really need is working examples of working config and database tables, Many thanks, Jeff

3 8

Re: [SR-Users] Kamailio and billing services .
by Daniel-Constantin Mierla 05 Apr '12

05 Apr '12

Hello, do not write private emails started from mailing list -- they will be ignored. Also, if it is a different topic, create a new email with appropriate subject. Then provide details about what you tried (config file parts) and results (log messages and sip traces). Otherwise nobody can guess what happens and cannot give hints. Cheers, Daniel On 4/4/12 4:10 PM, Ryan Gholam wrote: > Hello Daniel , > > I thank you for the last email you've sent me it helped me alot , but > now i am facing an issue concerning kamailio where i am trying to > connect the kamailio to the asterisk using a private ip and the > kamailio is connected using a public ip for the clients . > > I am trying to create a call from the client the phone rings but there > is no audio conversation happening , what is the best method to use ? > > N.B : i have tried RTPproxy but it didnt work i think because their > is no proper NATING . > > Thank you -- Daniel-Constantin Mierla Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany http://www.asipto.com/index.php/kamailio-advanced-training/

2 1

kamailio rpm build
by Min Wang 05 Apr '12

05 Apr '12

HI I tried to build Kamailio 3.2.2 rpm pkg on Centos 6.2 following instruction from: http://kb.asipto.com/kamailio:install:3.1.x-rpms-centos5x I got the following error ( at the end of the email) also tried: pkg-config --cflags glib-2.0 export C_INCLUDE_PATH="/usr/include/glib-2.0 /usr/lib64/glib-2.0/include" export CPLUS_INCLUDE_PATH="/usr/include/glib-2.0 /usr/lib64/glib-2.0/include" the try to build again, got the same error. anything could be wrong? thanks. min ////////////////////////////////////////////////////////// + make every-module group_include=kpurple CC (gcc) [M purple.so] clientaccount.o In file included from /usr/include/glib-2.0/glib/galloca.h:34, from /usr/include/glib-2.0/glib.h:32, from /usr/include/libpurple/account.h:30, from clientaccount.c:21: /usr/include/glib-2.0/glib/gtypes.h:410:2: error: #error unknown ENDIAN type In file included from /usr/include/glib-2.0/glib.h:33, from /usr/include/libpurple/account.h:30, from clientaccount.c:21: /usr/include/glib-2.0/glib/garray.h:50: error: expected specifier-qualifier-list before 'guint8' /usr/include/glib-2.0/glib/garray.h:152: error: expected '=', ',', ';', 'asm' or '__attribute__' before '*' token /usr/include/glib-2.0/glib/garray.h:157: error: expected ';', ',' or ')' before '*' token /usr/include/glib-2.0/glib/garray.h:160: error: expected ';', ',' or ')' before '*' token In file included from /usr/include/glib-2.0/glib/gerror.h:30, from /usr/include/glib-2.0/glib/gthread.h:34, from /usr/include/glib-2.0/glib/gasyncqueue.h:34, from /usr/include/glib-2.0/glib.h:34, from /usr/include/libpurple/account.h:30, from clientaccount.c:21: /usr/include/glib-2.0/glib/gquark.h:38: error: expected '=', ',', ';', 'asm' or '__attribute__' before 'GQuark' /usr/include/glib-2.0/glib/gquark.h:42: error: expected '=', ',', ';', 'asm' or '__attribute__' before 'g_quark_try_string' /usr/include/glib-2.0/glib/gquark.h:43: error: expected '=', ',', ';', 'asm' or '__attribute__' before 'g_quark_from_static_string' /usr/include/glib-2.0/glib/gquark.h:44: error: expected '=', ',', ';', 'asm' or '__attribute__' before 'g_quark_from_string' /usr/include/glib-2.0/glib/gquark.h:45: error: expected ')' before 'quark' In file included from /usr/include/glib-2.0/glib/gthread.h:34, ....

2 5

t_relay is failing after 45 seconds...
by Krishna Kurapati 04 Apr '12

04 Apr '12

Hi, I have # Time in seconds after a TCP connection will be closed if it is not available for writing in this interval tcp_send_timeout=2 And I am using save("location", "0x04")) to save only one contact per user. However, when I tried to relay message to a user, sometimes it is taking over 45 seconds for t_relay to timeout and return 477 error. How can I reduce the t_relay timeout? Thanks Krish Kura

2 2

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sr-users