sr-users April 2015

sr-users@lists.kamailio.org

89 participants
144 discussions

Timeout session Kamailio w/ rtpproxy v2
by Igor Potjevlesch 13 Apr '15

13 Apr '15

Hi everyone, I've sent my request on the rtpproxy google group without answer so try here. I'm using Kamailio 4.2.3 (and few older versions) with the new release of rtpproxy. I can't get the info of a session timeout (-n or -i options) from rtpproxy to my Kamailio proxy, using -n and the timeout socket. I tried: - Kamailio side: *modparam("rtpproxy", "timeout_socket", "unix:/var/run/rtpproxy_timeout.sock")* - rtpproxy side: *-n **unix:/var/run/rtpproxy_timeout.sock* and other configuration like tcp, or udp socket, but nothing is happening on the Kamailio part. The Kamailio rtpproxy man page give the example: *modparam("rtpproxy", "timeout_socket", "xmlrpc:http://127.0.0.1:8000/RPC2 <http://127.0.0.1:8000/RPC2>")* Does it mean Kamailio only handles XMLRPC socket ? In this case, is rtpproxy able to handle XMLRPC (I remember a patch in an older rtpproxy version) In the case Kamailio could handle other socket types, which one could allow me to get a session timeout signal from rtpproxy in order to generate BYE message at the signalling side ? I could add that rtpproxy stops well RTP sessions that timed out. Regards, Igor Potjevlesc

2 1

Re: [SR-Users] Avoiding "script writer didn't release transaction" warnings
by Mickael Marrache 13 Apr '15

13 Apr '15

Any idea? From: Mickael Marrache [mailto:mickaelmarrache@gmail.com] Sent: Thursday, April 09, 2015 2:08 PM To: sr-users(a)lists.sip-router.org Subject: RE: Avoiding "script writer didn't release transaction" warnings I think understand in which case it happens. When I receive a REGISTER, I create a transaction since I need to use the t_continue feature to wake up a suspended transaction. However, the save() command of the REGISTRAR module doesn't seem to call t_reply for replying the 200 (it looks like the SL module is used instead). Therefore, I get the warning every time it happens since the transaction is not released explicitly. Should I call t_release just before exiting the block handling the REGISTER? From: Mickael Marrache [mailto:mickaelmarrache@gmail.com] Sent: Thursday, April 09, 2015 1:43 PM To: sr-users(a)lists.sip-router.org Subject: Avoiding "script writer didn't release transaction" warnings Hi, In my logs, I get a lot of the following warnings: WARNING: tm [t_lookup.c:1476]: t_unref(): WARNING: script writer didn't release transaction I force transaction creation from my script to absorb retransmissions as soon as possible. I understand that I need to explicitly call t_release in order to avoid this warning. What is the most appropriate place in the script to call t_release? Thanks, Mickael

2 6

Maximum feasible size of htable?
by Brooks Bridges 13 Apr '15

13 Apr '15

I have a small-ish project that could benefit from using an htable for storage of very basic data that doesn't change much, and I'd like to get an idea of what the group's experience is with storing larger amounts of data? Something in the 15-20GB range. Yes I could use Redis, but the idea here is to minimize the amount of dependencies and processing required. Thoughts? Brooks Bridges | Sr. Voice Services Engineer O1 Communications 5190 Golden Foothill Pkwy El Dorado Hills, CA 95762 office: 916.235.2097 | main: 888.444.1111, Option 2 email: bbridges(a)o1.com<mailto:bbridges@o1.com> | web: www.o1.com<http://www.o1.com/>

2 2

CDR fields are all strings
by Mickael Marrache 13 Apr '15

13 Apr '15

Hi, Why are all the CDR fields inserted as strings? For extra fields, I can understand it is easier to implement. However, for the built in fields (e.g. duration), we need to perform queries based on duration and time and it is not really possible using strings. I see that the time field for failed transactions is inserted as a date. Thanks, Mickael

2 2

Sipml5-Kamailio -RtpEngine -ThridpartySipServers- Not all sip servers working
by Karthik Kondapaneni 13 Apr '15

13 Apr '15

Dear experts , I am using this setup . Sipml5 webrtc client(chrome) -----> Kamailio with RTP engine ------> ThridpartySip Server kamalio will act as mediator between webRTC client and NonWebRTC sip server . User will be registered on sip server and calls will be terminated to PSTN . This setup is working fine with one thirdparty sip server but not working with other SIP server . First server ( working ) is happily replying INVITE request and reponding with "TRYING " . But Second SIP server never reply to kamailio for INVITE requests and kamailio keep sending SDP and finally on SipML5 it shows "request timed out " Ports are open on Firewall . SIP Server 1: 192.168.102.237 SIP Server 2: 192.168.87.101 Kamailio IP : 172.17.185.46 Client PUblic ip : 79.207.241.94 1. Working Server Trace ---- INVITE sip:917638900890@192.168.102.237 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bK3929.368fd6fd45db4710e7acd52df40d3903.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKJHBbZaGh9noFlUgS1VVr8DCImVbITu0q;rport=54273 From: "2016"<sip:testcard@192.168.102.237>;tag=6mGSo8IPI4pGsvh6To1A To: <sip:917638900890@192.168.102.237> Contact: "2016"<sip:testcard@df7jal23ls0d.invalid ;alias=79.207.241.94~54273~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 89be25db-d9ed-616e-118e-c6b3a863d8fd CSeq: 62565 INVITE Content-Type: application/sdp Content-Length: 819 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8650276303411124000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS tosDjBW9pSViLfckZBR4gh6AolFzau3iBjgG m=audio 30246 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:2398380477 cname:8WcQLQ8hOhk5MhWZ a=ssrc:2398380477 msid:tosDjBW9pSViLfckZBR4gh6AolFzau3iBjgG 63b396a3-5126-405f-a389-75548cf0a634 a=ssrc:2398380477 mslabel:tosDjBW9pSViLfckZBR4gh6AolFzau3iBjgG a=ssrc:2398380477 label:63b396a3-5126-405f-a389-75548cf0a634 a=sendrecv a=rtcp:30247 SIP/2.0 407 Proxy Authentication Required Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bK3929.368fd6fd45db4710e7acd52df40d3903.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKJHBbZaGh9noFlUgS1VVr8DCImVbITu0q;rport=54273 From: "2016" <sip:testcard@192.168.102.237>;tag=6mGSo8IPI4pGsvh6To1A To: <sip:917638900890@192.168.102.237> Contact: "2016" <sip:testcard@192.168.102.237:5060> Call-Id: 89be25db-d9ed-616e-118e-c6b3a863d8fd CSeq: 62565 INVITE Proxy-Authenticate: Digest realm="sip.dingdongbell.net", nonce="c2133e5e1e40805244c5f201cfbaee4d", algorithm=MD5 Content-Length: 0 ACK sip:917638900890@192.168.102.237 SIP/2.0 Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bK3929.368fd6fd45db4710e7acd52df40d3903.0 From: "2016"<sip:testcard@192.168.102.237>;tag=6mGSo8IPI4pGsvh6To1A To: <sip:917638900890@192.168.102.237> Call-ID: 89be25db-d9ed-616e-118e-c6b3a863d8fd CSeq: 62565 ACK Content-Length: 0 Max-Forwards: 69 INVITE sip:917638900890@192.168.102.237 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bK0929.61a39db61290521db7dc657d252c4182.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKnswxFdcreaCt0SMSRCqs2HPPOSN9adR6;rport=54273 From: "2016"<sip:testcard@192.168.102.237>;tag=6mGSo8IPI4pGsvh6To1A To: <sip:917638900890@192.168.102.237> Contact: "2016"<sip:testcard@df7jal23ls0d.invalid ;alias=79.207.241.94~54273~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 89be25db-d9ed-616e-118e-c6b3a863d8fd CSeq: 62566 INVITE Content-Type: application/sdp Content-Length: 819 Max-Forwards: 69 Proxy-Authorization: Digest username="testcard",realm="sip.dingdongbell.net ",nonce="c2133e5e1e40805244c5f201cfbaee4d",uri=" sip:917638900890@192.168.102.237 ",response="fb9c97e380c4231725d1119256f7f7d9",algorithm=MD5 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8650276303411124000 2 IN IP4 172.17.185.46 s=DD- chrome t=0 0 a=msid-semantic: WMS tosDjBW9pSViLfckZBR4gh6AolFzau3iBjgG m=audio 30246 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:2398380477 cname:8WcQLQ8hOhk5MhWZ a=ssrc:2398380477 msid:tosDjBW9pSViLfckZBR4gh6AolFzau3iBjgG 63b396a3-5126-405f-a389-75548cf0a634 a=ssrc:2398380477 mslabel:tosDjBW9pSViLfckZBR4gh6AolFzau3iBjgG a=ssrc:2398380477 label:63b396a3-5126-405f-a389-75548cf0a634 a=sendrecv a=rtcp:30247 SIP/2.0 100 Trying Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bK0929.61a39db61290521db7dc657d252c4182.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKnswxFdcreaCt0SMSRCqs2HPPOSN9adR6;rport=54273 From: "2016" <sip:testcard@192.168.102.237>;tag=6mGSo8IPI4pGsvh6To1A To: <sip:917638900890@192.168.102.237> Contact: <sip:917638900890@192.168.102.237:5060> Call-Id: 89be25db-d9ed-616e-118e-c6b3a863d8fd CSeq: 62566 INVITE Content-Length: 0 2. Not working Server trace ( second server)-------------------- INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 INVITE sip:917638900890@192.168.87.101 SIP/2.0 Record-Route: <sip:172.17.185.46:5090;nat=yes;lr=on> Via: SIP/2.0/UDP 172.17.185.46:5090 ;branch=z9hG4bKc8b9.d6c920a694694e5d3104f492c48d2582.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;received=79.207.241.94;branch=z9hG4bKBXNMor2TC7eWrTb87WqF4qQN34s6MtQ0;rport=50933 From: "2017"<sip:testcard2@192.168.87.101>;tag=axqOflPwynQ898Qu2Tue To: <sip:917638900890@192.168.87.101> Contact: "2017"<sip:testcard2@df7jal23ls0d.invalid ;alias=79.207.241.94~50933~5;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 6b04e418-dbdf-21aa-39b7-1c2fb1d72f6b CSeq: 8601 INVITE Content-Type: application/sdp Content-Length: 811 Max-Forwards: 69 User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18 Organization: DD v=0 o=- 8501880612358765000 2 IN IP4 172.17.185.46 s=DD - chrome t=0 0 a=msid-semantic: WMS 2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 m=audio 30038 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 172.17.185.46 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:68402186 cname:W2sB0/Nke1Gs8u89 a=ssrc:68402186 msid:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 62dff51b-f07b-4788-98cb-72c25ff2ab1b a=ssrc:68402186 mslabel:2pj4LaFD0X7B4Sa0IkGLFLZhZAoDcQm5nhk8 a=ssrc:68402186 label:62dff51b-f07b-4788-98cb-72c25ff2ab1b a=sendrecv a=rtcp:30039 Kamailio Config :----------------------------------- request_route { if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS))) { xlog("L_WARN", "SIP request received on $Rp\n"); sl_send_reply("403", "Forbidden"); exit; } # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); if (!is_method("REGISTER|INVITE|ACK|BYE|CANCEL|PRACK|MESSAGE|INFO|UPDATE")) { sl_send_reply("405", "Method not allowed"); exit; } # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route_preset("172.17.185.46:5090;nat=yes"); # handle registrations route(REGISTRAR); # if ($rU==$null) { # # request with no Username in RURI # sl_send_reply("484","Address Incomplete"); # exit; # } # if (!is_method("INVITE")) { # route(RELAY); # exit; # } route(RELAY); } route[SETUP_BY_TRANSPORT] { if ($ru =~ "transport=ws") { xlog("L_INFO", "Request going to WS"); if(sdp_with_transport("RTP/SAVPF")) { rtpengine_manage("force trust-address replace-origin replace-session-connection ICE=force"); t_on_reply("REPLY_WS_TO_WS"); return; } # rtpengine_manage("froc+SP"); rtpengine_manage("force trust-address replace-origin replace-session-connection ICE=force RTP/SAVPF"); t_on_reply("REPLY_FROM_WS"); } else if ($proto =~ "ws") { xlog("L_INFO", "Request coming from WS"); # rtpengine_manage("froc-sp"); rtpengine_manage("force rtcp-mux-demux symmetric trust-address replace-origin replace-session-connection ICE=remove RTP/AVP"); t_on_reply("REPLY_TO_WS"); } else { xlog("L_INFO", "This is a classic phone call"); # rtpengine_manage("co"); rtpengine_manage("replace-origin replace-session-connection"); t_on_reply("MANAGE_CLASSIC_REPLY"); } } event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT #!ifdef WITH_TLS && $Rp != MY_WSS_PORT #!endif ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } xlog("L_INFO", "HTTP Request Received\n"); if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm=~"GET") { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("404", "Not Found", "", ""); } route[RELAY] { if (is_method("INVITE")) { route(SETUP_BY_TRANSPORT); if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; }

1 0

End time in CDR
by Mickael Marrache 13 Apr '15

13 Apr '15

Hi, Very short question. Why the time at which the dialog ended is also included in the CDRs? There already are the start time and duration. Is it there to avoid end time computation? Mickael

1 0

Dialog based CDRs and failed dialogs
by Mickael Marrache 13 Apr '15

13 Apr '15

Hi, I'm using the ACC module to generate dialog based CDRs and I see that CDRs are generated even for failed calls. For example, a CDR is generated for cancelled calls. At first, I thought that dialog based CDRs are generated only for successfully established dialogs but it doesn't seem to be the case. Thanks, Mickael

2 4

Re: [SR-Users] MULTIDOMAIN Environment
by Bruno Emer 13 Apr '15

13 Apr '15

Hello Daniel! Thanks for your help again!! I placed the instruction just after the "authentication" and now it is working as I want! Bruno Emer 2015-04-09 15:33 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > Hello all! > > I have a scenario here where I need to put more than two companies into > the same Kamailio server and to accomplish this I'm using a multi-domain > configuration. > > For now, I am able to create users in different domains and register all > of them. The users are able to place and receive internal calls (as I am > using this just to get internal communications working), so I am ok with > this part of my configuration. > > The problem that I'm facing now is that users from domain "A" are able to > place calls to users from domain "B". I need to deny calls between > different domains, and I don't know the best way to achieve this. > > Here are my domains create (output from "kamctl domain show"): > > domain:: enterprise.com > did:: enterprise.com > domain:: enterprise2.com > did:: enterprise2.com > > > And here are my users created: > > username: bob > domain: enterprise.com > > username: alice > domain: enterprise2.com > > > So, can you please help me to get this? > > > Here is my .cfg file: > > > #!KAMAILIO > #!define WITH_PGSQL > #!define WITH_AUTH > #!define WITH_USRLOCDB > ##!define WITH_NAT > ##!define WITH_DEBUG > #!define WITH_MULTIDOMAIN > ##!define WITH_SIPTRACE > #!define WITH_SQLOPS > #!define WITH_XMLRPC > ##!define WITH_603 > #!define WITH_TLS > # > > > #!substdef "!MY_WS_PORT!80!g" > #!substdef "!MY_WSS_PORT!81!g" > #!substdef "!MY_WS_ADDR!eth1:MY_WS_PORT!g" > #!substdef "!MY_WSS_ADDR!tls:eth1:MY_WSS_PORT!g" > > #!define WITH_WEBSOCKETS > > > #!ifdef ACCDB_COMMENT > ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; > ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; > ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; > ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; > ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; > ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; > ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL > DEFAULT ''; > ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL > DEFAULT ''; > ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default > ''; > ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL > DEFAULT ''; > ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL > DEFAULT ''; > ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL > DEFAULT ''; > #!endif > > ####### Include Local Config If Exists ######### > import_file "kamailio-local.cfg" > > ####### Defined Values ######### > > # *** Value defines - IDs used later in config > #!ifdef WITH_PGSQL > # - database URL - used to connect to database server by modules such > # as: auth_db, acc, usrloc, a.s.o. > #!define DBURL "postgres://kamailio:kamailiorw@localhost/kamailio" > #!endif > #!ifdef WITH_MULTIDOMAIN > # - the value for 'use_domain' parameters > #!define MULTIDOMAIN 1 > #!else > #!define MULTIDOMAIN 0 > #!endif > > # - flags > # FLT_ - per transaction (message) flags > # FLB_ - per branch flags > #!define FLT_ACC 1 > #!define FLT_ACCMISSED 2 > #!define FLT_ACCFAILED 3 > #!define FLT_NATS 5 > > #!define FLB_NATB 6 > #!define FLB_NATSIPPING 7 > > ####### Global Parameters ######### > > ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR > #!ifdef WITH_DEBUG > debug=4 > log_stderror=yes > #!else > debug=2 > log_stderror=no > #!endif > > > memdbg=5 > memlog=5 > > log_facility=LOG_LOCAL0 > > fork=yes > children=4 > > /* uncomment the next line to disable TCP (default on) */ > disable_tcp=no > tcp_accept_no_cl=yes > > /* uncomment the next line to disable the auto discovery of local aliases > based on reverse DNS on IPs (default on) */ > #auto_aliases=no > > /* add local domain aliases */ > #alias="sip.mydomain.com" > > /* uncomment and configure the following line if you want Kamailio to > bind on a specific interface/port/proto (default bind on all available) > */ > #listen=udp:10.0.0.10:5060 > > /* port to listen to > * - can be specified more than once if needed to listen on many ports */ > port=5060 > > #!ifdef WITH_TLS > enable_tls=yes > #!endif > > # life time of TCP connection when there is no traffic > # - a bit higher than registration expires to cope with UA behind NAT > tcp_connection_lifetime=3605 > tcp_rd_buf_size=6144 > > listen=eth1 > #!ifdef WITH_WEBSOCKETS > listen=MY_WS_ADDR > #!ifdef WITH_TLS > listen=eth1 > #!endif > #!endif > > ####### Custom Parameters ######### > > # These parameters can be modified runtime via RPC interface > # - see the documentation of 'cfg_rpc' module. > # > # Format: group.id = value 'desc' description > # Access: $sel(cfg_get.group.id) or @cfg_get.group.id > # > > #!ifdef WITH_PSTN > # PSTN GW Routing > # > # - pstn.gw_ip: valid IP or hostname as string value, example: > # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" > # > # - by default is empty to avoid misrouting > pstn.gw_ip = "" desc "PSTN GW Address" > pstn.gw_port = "" desc "PSTN GW Port" > #!endif > > #!ifdef WITH_VOICEMAIL > # VoiceMail Routing on offline, busy or no answer > # > # - by default Voicemail server IP is empty to avoid misrouting > voicemail.srv_ip = "" desc "VoiceMail IP Address" > voicemail.srv_port = "5060" desc "VoiceMail Port" > #!endif > > ####### Modules Section ######## > > # set paths to location of modules (to sources or installation folders) > #!ifdef WITH_SRCPATH > mpath="modules/" > #!else > mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" > #!endif > > #!ifdef WITH_PGSQL > loadmodule "db_postgres.so" > #!endif > > loadmodule "mi_fifo.so" > loadmodule "kex.so" > loadmodule "corex.so" > loadmodule "tm.so" > loadmodule "tmx.so" > loadmodule "sl.so" > loadmodule "rr.so" > loadmodule "pv.so" > loadmodule "maxfwd.so" > loadmodule "usrloc.so" > loadmodule "registrar.so" > loadmodule "textops.so" > loadmodule "siputils.so" > loadmodule "xlog.so" > loadmodule "sanity.so" > loadmodule "ctl.so" > loadmodule "cfg_rpc.so" > loadmodule "mi_rpc.so" > loadmodule "acc.so" > > #!ifdef WITH_AUTH > loadmodule "auth.so" > loadmodule "auth_db.so" > #!ifdef WITH_IPAUTH > loadmodule "permissions.so" > #!endif > #!endif > > #!ifdef WITH_ALIASDB > loadmodule "alias_db.so" > #!endif > > #!ifdef WITH_SPEEDDIAL > loadmodule "speeddial.so" > #!endif > > #!ifdef WITH_MULTIDOMAIN > loadmodule "domain.so" > #!endif > > #!ifdef WITH_PRESENCE > loadmodule "presence.so" > loadmodule "presence_xml.so" > #!endif > > #!ifdef WITH_NAT > loadmodule "nathelper.so" > loadmodule "rtpproxy.so" > #!endif > > #!ifdef WITH_TLS > loadmodule "tls.so" > #!endif > > #!ifdef WITH_ANTIFLOOD > loadmodule "htable.so" > loadmodule "pike.so" > #!endif > > #!ifdef WITH_XMLRPC > loadmodule "xmlrpc.so" > #!endif > > #!ifdef WITH_DEBUG > loadmodule "debugger.so" > #!endif > > #!ifdef WITH_SIPTRACE > loadmodule "siptrace.so" > #!endif > > #!ifdef WITH_SQLOPS > loadmodule "sqlops.so" > #!endif > > #!ifdef WITH_WEBSOCKETS > loadmodule "xhttp.so" > loadmodule "websocket.so" > loadmodule "nathelper.so" > #!endif > > # ----------------- setting module-specific parameters --------------- > > # ---- sip_trace params ---- > #!ifdef WITH_SIPTRACE > modparam("siptrace", "db_url", "postgres://kamailio:kamailiorw@localhost > /kamailio") > modparam("siptrace", "trace_on", 1) > modparam("siptrace", "trace_flag", 22) > modparam("siptrace", "trace_sl_acks", 0) > #!endif > > # ----- mi_fifo params ----- > modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") > > > # ----- tm params ----- > # auto-discard branches from previous serial forking leg > modparam("tm", "failure_reply_mode", 3) > # default retransmission timeout: 30sec > modparam("tm", "fr_timer", 30000) > # default invite retransmission timeout after 1xx: 120sec > modparam("tm", "fr_inv_timer", 120000) > > # add value to ;lr param to cope with most of the UAs > modparam("rr", "enable_full_lr", 1) > # do not append from tag to the RR (no need for this script) > modparam("rr", "append_fromtag", 0) > > > # ----- registrar params ----- > modparam("registrar", "method_filtering", 1) > /* uncomment the next line to disable parallel forking via location */ > # modparam("registrar", "append_branches", 0) > /* uncomment the next line not to allow more than 10 contacts per AOR */ > #modparam("registrar", "max_contacts", 10) > # max value for expires of registrations > modparam("registrar", "max_expires", 300) > modparam("registrar", "min_expires", 30) > # set it to 1 to enable GRUU > modparam("registrar", "gruu_enabled", 0) > > > # ----- acc params ----- > /* what special events should be accounted ? */ > modparam("acc", "early_media", 0) > modparam("acc", "report_ack", 0) > modparam("acc", "report_cancels", 0) > /* by default ww do not adjust the direct of the sequential requests. > if you enable this parameter, be sure the enable "append_fromtag" > in "rr" module */ > modparam("acc", "detect_direction", 0) > /* account triggers (flags) */ > modparam("acc", "log_flag", FLT_ACC) > modparam("acc", "log_missed_flag", FLT_ACCMISSED) > modparam("acc", "log_extra", > "src_user=$fU;src_domain=$fd;src_ip=$si;" > "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") > modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) > /* enhanced DB accounting */ > #!ifdef WITH_ACCDB > modparam("acc", "db_flag", FLT_ACC) > modparam("acc", "db_missed_flag", FLT_ACCMISSED) > modparam("acc", "db_url", DBURL) > modparam("acc", "db_extra", > "src_user=$fU;src_domain=$fd;src_ip=$si;" > "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") > #!endif > > > # ----- usrloc params ----- > /* enable DB persistency for location entries */ > #!ifdef WITH_USRLOCDB > modparam("usrloc", "db_url", DBURL) > modparam("usrloc", "db_mode", 2) > modparam("usrloc", "use_domain", MULTIDOMAIN) > #!endif > > > # ----- auth_db params ----- > #!ifdef WITH_AUTH > modparam("auth_db", "db_url", DBURL) > modparam("auth_db", "calculate_ha1", 0) > #modparam("auth_db", "password_column", "password") > modparam("auth_db", "load_credentials", "") > modparam("auth_db", "use_domain", MULTIDOMAIN) > #modparam("auth_db", "use_domain", 1) > > # ----- permissions params ----- > #!ifdef WITH_IPAUTH > modparam("permissions", "db_url", DBURL) > modparam("permissions", "db_mode", 1) > #!endif > > #!endif > > > # ----- alias_db params ----- > #!ifdef WITH_ALIASDB > modparam("alias_db", "db_url", DBURL) > modparam("alias_db", "use_domain", MULTIDOMAIN) > #!endif > > > # ----- speeddial params ----- > #!ifdef WITH_SPEEDDIAL > modparam("speeddial", "db_url", DBURL) > modparam("speeddial", "use_domain", MULTIDOMAIN) > #!endif > > > # ----- domain params ----- > #!ifdef WITH_MULTIDOMAIN > modparam("domain", "db_url", DBURL) > # register callback to match myself condition with domains list > modparam("domain", "register_myself", 1) > #!endif > > > #!ifdef WITH_PRESENCE > # ----- presence params ----- > modparam("presence", "db_url", DBURL) > > # ----- presence_xml params ----- > modparam("presence_xml", "db_url", DBURL) > modparam("presence_xml", "force_active", 1) > #!endif > > > #!ifdef WITH_NAT > # ----- rtpproxy params ----- > modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") > # ----- nathelper params ----- > modparam("nathelper", "natping_interval", 30) > modparam("nathelper", "ping_nated_only", 1) > modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) > modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") > > # params needed for NAT traversal in other modules > modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") > modparam("usrloc", "nat_bflag", FLB_NATB) > #!endif > > > #!ifdef WITH_TLS > # ----- tls params ----- > modparam("tls", "connection_timeout", 60) > modparam("tls", "tls_debug", 10) > modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") > #modparam("tls", "verify_certificate", 1) > #modparam("tls", "require_certificate", 0) > #!endif > > #!ifdef WITH_ANTIFLOOD > # ----- pike params ----- > modparam("pike", "sampling_time_unit", 2) > modparam("pike", "reqs_density_per_unit", 16) > modparam("pike", "remove_latency", 4) > > # ----- htable params ----- > # ip ban htable with autoexpire after 5 minutes > modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") > #!endif > > #!ifdef WITH_XMLRPC > # ----- xmlrpc params ----- > modparam("xmlrpc", "route", "XMLRPC"); > modparam("xmlrpc", "url_match", "^/RPC") > #!endif > > #!ifdef WITH_DEBUG > # ----- debugger params ----- > modparam("debugger", "cfgtrace", 1) > #!endif > > > #!ifdef WITH_WEBSOCKETS > # ----- nathelper params ----- > modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") > # Note: leaving NAT pings turned off here as nathelper is _only_ being > used for > # WebSocket connections. NAT pings are not needed as WebSockets have > # their own keep-alives. > #!endif > > > > ####### Routing Logic ######## > > # Main SIP request routing logic > # - processing of any incoming SIP request starts with this route > # - note: this is the same as route { ... } > request_route { > > route(REQINIT); > > > #!ifdef WITH_SIPTRACE > if($hdr(X-Omnz-SipTrace) !=$null) > { > sip_trace(); > setflag(22); > } > #!endif > > > #!ifdef WITH_WEBSOCKETS > if (nat_uac_test(64)) { > # Do NAT traversal stuff for requests from a WebSocket > # connection - even if it is not behind a NAT! > # This won't be needed in the future if Kamailio and the > # WebSocket client support Outbound and Path. > force_rport(); > if (is_method("REGISTER")) > fix_nated_register(); > else { > if (!add_contact_alias()) { > xlog("L_ERR", "Error aliasing contact > <$ct>\n"); > sl_send_reply("400", "Bad Request"); > exit; > } > } > } > #!endif > > > # NAT detection > route(NATDETECT); > > # CANCEL processing > if (is_method("CANCEL")) > { > if (t_check_trans()) { > route(RELAY); > } > exit; > } > > # handle requests within SIP dialogs > route(WITHINDLG); > > ### only initial requests (no To tag) > > t_check_trans(); > > # authentication > route(AUTH); > > # record routing for dialog forming requests (in case they are > routed) > # - remove preloaded route headers > remove_hf("Route"); > if (is_method("INVITE|SUBSCRIBE")) > record_route(); > > # account only INVITEs > if (is_method("INVITE")) > { > setflag(FLT_ACC); # do accounting > } > > # dispatch requests to foreign domains > route(SIPOUT); > > ### requests for my local domains > > # handle presence related requests > route(PRESENCE); > > > # handle registrations > route(REGISTRAR); > > if ($rU==$null) > { > # request with no Username in RURI > sl_send_reply("484","Address Incomplete"); > exit; > } > > # dispatch destinations to PSTN > route(PSTN); > > # user location service > route(LOCATION); > } > > > route[RELAY] { > > # enable additional event routes for forwarded requests > # - serial forking, RTP relaying handling, a.s.o. > > > #!ifdef WITH_603 > if (is_method("INVITE")) { > if($hdr(X-Omz-int-Id) ==$null) > { > # request with no Username in RURI > sl_send_reply("603","Declined"); > exit; > } > } > #!endif > > > > if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { > if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); > } > if (is_method("INVITE|SUBSCRIBE|UPDATE")) { > if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); > } > if (is_method("INVITE")) { > if(!t_is_set("failure_route")) > t_on_failure("MANAGE_FAILURE"); > } > > if (!t_relay()) { > sl_reply_error(); > } > exit; > } > > > # Per SIP request initial checks > route[REQINIT] { > #!ifdef WITH_ANTIFLOOD > # flood dection from same IP and traffic ban for a while > # be sure you exclude checking trusted peers, such as pstn gateways > # - local host excluded (e.g., loop to self) > if(src_ip!=myself) > { > if($sht(ipban=>$si)!=$null) > { > # ip is already blocked > xdbg("request from blocked IP - $rm from $fu > (IP:$si:$sp)\n"); > exit; > } > if (!pike_check_req()) > { > xlog("L_ALERT","ALERT: pike blocking $rm from $fu > (IP:$si:$sp)\n"); > $sht(ipban=>$si) = 1; > exit; > } > } > #!endif > > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483","Too Many Hops"); > exit; > } > > if(!sanity_check("1511", "7")) > { > xlog("Malformed SIP message from $si:$sp\n"); > exit; > } > } > > > # 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()) { > > > #!ifdef WITH_WEBSOCKETS > if ($du == "") { > if (!handle_ruri_alias()) { > xlog("L_ERR", "Bad alias <$ru>\n"); > sl_send_reply("400", "Bad > Request"); > exit; > } > } > #!endif > > > > route(DLGURI); > if (is_method("BYE")) { > setflag(FLT_ACC); # do accounting ... > setflag(FLT_ACCFAILED); # ... even if the > transaction fails > } > else if ( is_method("ACK") ) { > # ACK is forwarded statelessy > route(NATMANAGE); > } > else if ( is_method("NOTIFY") ) { > # Add Record-Route for in-dialog NOTIFY as > per RFC 6665. > record_route(); > } > route(RELAY); > } else { > if (is_method("SUBSCRIBE") && uri == myself) { > # in-dialog subscribe requests > route(PRESENCE); > exit; > } > if ( is_method("ACK") ) { > if ( t_check_trans() ) { > # no loose-route, but stateful ACK; > # must be an ACK after a 487 > # or e.g. 404 from upstream server > route(RELAY); > exit; > } else { > # ACK without matching transaction > ... ignore and discard > exit; > } > } > sl_send_reply("404","Not here"); > } > exit; > } > } > > # Handle SIP registrations > route[REGISTRAR] { > if (is_method("REGISTER")) > { > if(isflagset(FLT_NATS)) > { > setbflag(FLB_NATB); > # uncomment next line to do SIP NAT pinging > ## setbflag(FLB_NATSIPPING); > } > if (!save("location")) > sl_reply_error(); > > exit; > } > } > > # USER location service > route[LOCATION] { > > #!ifdef WITH_SPEEDDIAL > # search for short dialing - 2-digit extension > if($rU=~"^[0-9][0-9]$") > if(sd_lookup("speed_dial")) > route(SIPOUT); > #!endif > > #!ifdef WITH_ALIASDB > # search in DB-based aliases > if(alias_db_lookup("dbaliases")) > route(SIPOUT); > #!endif > > $avp(oexten) = $rU; > if (!lookup("location")) { > $var(rc) = $rc; > route(TOVOICEMAIL); > t_newtran(); > switch ($var(rc)) { > case -1: > case -3: > send_reply("404", "Not Found"); > exit; > case -2: > send_reply("405", "Method Not Allowed"); > exit; > } > } > > # when routing via usrloc, log the missed calls also > if (is_method("INVITE")) > { > setflag(FLT_ACCMISSED); > } > > route(RELAY); > exit; > } > > # Presence server route > route[PRESENCE] { > if(!is_method("PUBLISH|SUBSCRIBE")) > return; > > #!ifdef WITH_PRESENCE > if (!t_newtran()) > { > sl_reply_error(); > exit; > }; > > if(is_method("PUBLISH")) > { > handle_publish(); > t_release(); > } > else > if( is_method("SUBSCRIBE")) > { > handle_subscribe(); > t_release(); > } > exit; > #!endif > > # if presence enabled, this part will not be executed > if (is_method("PUBLISH") || $rU==$null) > { > sl_send_reply("404", "Not here"); > exit; > } > return; > } > > # Authentication route > route[AUTH] { > #!ifdef WITH_AUTH > > #!ifdef WITH_IPAUTH > if((!is_method("REGISTER")) && allow_source_address()) > { > # source IP allowed > return; > } > #!endif > > if (is_method("REGISTER") || from_uri==myself) > { > # authenticate requests > if (!auth_check("$fd", "subscriber", "1")) { > auth_challenge("$fd", "0"); > exit; > } > # user authenticated - remove auth header > if(!is_method("REGISTER|PUBLISH")) > consume_credentials(); > } > # if caller is not local subscriber, then check if it calls > # a local destination, otherwise deny, not an open relay here > if (from_uri!=myself && uri!=myself) > { > sl_send_reply("403","Not relaying"); > exit; > } > > #!endif > return; > } > > # Caller NAT detection route > route[NATDETECT] { > #!ifdef WITH_NAT > force_rport(); > if (nat_uac_test("19")) { > if (is_method("REGISTER")) { > fix_nated_register(); > } else { > add_contact_alias(); > } > setflag(FLT_NATS); > } > #!endif > return; > } > > # RTPProxy control > route[NATMANAGE] { > #!ifdef WITH_NAT > if (is_request()) { > if(has_totag()) { > if(check_route_param("nat=yes")) { > setbflag(FLB_NATB); > } > } > } > if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) > return; > > rtpproxy_manage("co"); > > if (is_request()) { > if (!has_totag()) { > if(t_is_branch_route()) { > add_rr_param(";nat=yes"); > } > } > } > if (is_reply()) { > if(isbflagset(FLB_NATB)) { > add_contact_alias(); > } > } > #!endif > return; > } > > # URI update for dialog requests > route[DLGURI] { > #!ifdef WITH_NAT > if(!isdsturiset()) { > handle_ruri_alias(); > } > #!endif > return; > } > > # Routing to foreign domains > route[SIPOUT] { > if (!uri==myself) > { > append_hf("P-hint: outbound\r\n"); > route(RELAY); > } > } > > # PSTN GW routing > route[PSTN] { > #!ifdef WITH_PSTN > # check if PSTN GW IP is defined > if (strempty($sel(cfg_get.pstn.gw_ip))) { > xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not > defined\n"); > return; > } > > # route to PSTN dialed numbers starting with '+' or '00' > # (international format) > # - update the condition to match your dialing rules for PSTN > routing > if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) > return; > > # only local users allowed to call > if(from_uri!=myself) { > sl_send_reply("403", "Not Allowed"); > exit; > } > > if (strempty($sel(cfg_get.pstn.gw_port))) { > $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); > } else { > $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" > + $sel(cfg_get.pstn.gw_port); > } > > route(RELAY); > exit; > #!endif > > return; > } > > # XMLRPC routing > #!ifdef WITH_XMLRPC > route[XMLRPC] { > # allow XMLRPC from localhost > if ((method=="POST" || method=="GET") > && (src_ip==127.0.0.1)) { > # close connection only for xmlrpclib user agents (there > is a bug in > # xmlrpclib: it waits for EOF before interpreting the > response). > if ($hdr(User-Agent) =~ "xmlrpclib") > set_reply_close(); > set_reply_no_connect(); > dispatch_rpc(); > exit; > } > send_reply("403", "Forbidden"); > exit; > } > #!endif > > # route to voicemail server > route[TOVOICEMAIL] { > #!ifdef WITH_VOICEMAIL > if(!is_method("INVITE")) > return; > > # check if VoiceMail server IP is defined > if (strempty($sel(cfg_get.voicemail.srv_ip))) { > xlog("SCRIPT: VoiceMail rotuing enabled but IP not > defined\n"); > return; > } > if($avp(oexten)==$null) > return; > > $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) > + ":" + $sel(cfg_get.voicemail.srv_port); > route(RELAY); > exit; > #!endif > > return; > } > > # manage outgoing branches > branch_route[MANAGE_BRANCH] { > xdbg("new branch [$T_branch_idx] to $ru\n"); > route(NATMANAGE); > } > > # manage incoming replies > onreply_route[MANAGE_REPLY] { > xdbg("incoming reply\n"); > if(status=~"[12][0-9][0-9]") > route(NATMANAGE); > } > > # manage failure routing cases > failure_route[MANAGE_FAILURE] { > route(NATMANAGE); > > if (t_is_canceled()) { > exit; > } > > #!ifdef WITH_BLOCK3XX > # block call redirect based on 3xx replies. > if (t_check_status("3[0-9][0-9]")) { > t_reply("404","Not found"); > exit; > } > #!endif > > #!ifdef WITH_VOICEMAIL > # serial forking > # - route to voicemail on busy or no answer (timeout) > if (t_check_status("486|408")) { > $du = $null; > route(TOVOICEMAIL); > exit; > } > #!endif > } > > > #!ifdef WITH_WEBSOCKETS > onreply_route { > if (nat_uac_test(64)) { > # Do NAT traversal stuff for replies to a WebSocket > connection > # - even if it is not behind a NAT! > # This won't be needed in the future if Kamailio and the > # WebSocket client support Outbound and Path. > add_contact_alias(); > } > } > > event_route[xhttp:request] { > set_reply_close(); > set_reply_no_connect(); > > if ($Rp != MY_WS_PORT > > #!ifdef WITH_TLS > && $Rp != MY_WSS_PORT > #!endif > > ) { > xlog("L_WARN", "HTTP request received on $Rp\n"); > xhttp_reply("403", "Forbidden", "", ""); > exit; > } > > xlog("L_DBG", "HTTP Request Received\n"); > > if ($hdr(Upgrade)=~"websocket" > && $hdr(Connection)=~"Upgrade" > && $rm=~"GET") { > xlog("L_DBG", "WebSocket\n"); > xlog("L_DBG", " Host: $hdr(Host)\n"); > xlog("L_DBG", " Origin: $hdr(Origin)\n"); > > if ($hdr(Host) == $null ) { > xlog("L_WARN", "Bad host $hdr(Host)\n"); > xhttp_reply("403", "Forbidden", "", ""); > exit; > } > > # Optional... validate Origin > # Optional... perform HTTP authentication > > # ws_handle_handshake() exits (no further configuration > file > # processing of the request) when complete. > if (ws_handle_handshake()) > { > # Optional... cache some information abou the > # successful connection > exit; > } > } > > xhttp_reply("404", "Not found", "", ""); > } > > event_route[websocket:closed] { > xlog("L_INFO", "WebSocket connection from $si:$sp has closed\n"); > } > #!endif > > > >

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Re: [SR-Users] Welcome to the "Users" mailing list
by John Jhj 13 Apr '15

13 Apr '15

Hello, We have setup Kamailio SIP Server v4.1.7 at latest our Rackspace server (ubuntu 14.10) Chat works fine on all cases and calls are working for Local network only( I think its when it doesn't uses rtpproxy). Over Internet( on real ips rather than local one), we are able to make calls(audio/video) and we can see that Our packets are being transmitted properly to the server, but we are unable to get those packets to client side back from the server. So, Client-to-Server : works but, Server-to-Client : it doesn't works FYI, We have confirmed that firewalll is inactive on our server and required ports are open. We have tried with and without tls (self signed certificates only). The SIP client we used to test the environment is JITSI. We are getting following error : Apr 10 10:46:39 telezang-app /usr/local/sbin/kamailio[25670]: ERROR: rtpproxy [rtpproxy.c:1756]: select_rtpp_node(): script error -no valid set selected Apr 10 10:46:39 telezang-app /usr/local/sbin/kamailio[25670]: ERROR: rtpproxy [rtpproxy.c:2636]: force_rtp_proxy(): no available proxies Apr 10 10:46:44 telezang-app /usr/local/sbin/kamailio[25641]: ERROR: rtpproxy [rtpproxy.c:1756]: select_rtpp_node(): script error -no valid set selected Apr 10 10:46:44 telezang-app /usr/local/sbin/kamailio[25641]: ERROR: rtpproxy [rtpproxy.c:2636]: force_rtp_proxy(): no available proxies Apr 10 10:46:44 telezang-app /usr/local/sbin/kamailio[25641]: NOTICE: acc [acc.c:315]: acc_log_request(): ACC: transaction answered: timestamp=1428662804;method=INVITE;from_tag=978a791a;to_tag=3458055147;call_id=6c08683633dd586d0e634f5e2bbfa99c@0 :0:0:0:0:0:0:0;code=200;reason=OK;src_user=user1;src_domain= mobileappplatform.com ;src_ip=202.157.76.21;dst_ouser=user4;dst_user=user4;dst_domain=192.168.5.115 Apr 10 10:46:44 telezang-app /usr/local/sbin/kamailio[25643]: ERROR: rtpproxy [rtpproxy.c:1756]: select_rtpp_node(): script error -no valid set selected Apr 10 10:46:44 telezang-app /usr/local/sbin/kamailio[25643]: ERROR: rtpproxy [rtpproxy.c:2636]: force_rtp_proxy(): no available proxies Apr 10 10:46:48 telezang-app /usr/local/sbin/kamailio[25670]: ERROR: rtpproxy [rtpproxy.c:1756]: select_rtpp_node(): script error -no valid set selected Apr 10 10:46:48 telezang-app /usr/local/sbin/kamailio[25670]: ERROR: rtpproxy [rtpproxy.c:1960]: unforce_rtp_proxy(): no available proxies Apr 10 10:46:49 telezang-app /usr/local/sbin/kamailio[25644]: NOTICE: acc [acc.c:315]: acc_log_request(): ACC: transaction answered: timestamp=1428662809;method=BYE;from_tag=978a791a;to_tag=3458055147;call_id=6c08683633dd586d0e634f5e2bbfa99c@0 :0:0:0:0:0:0:0;code=200;reason=OK;src_user=user1;src_domain= mobileappplatform.com ;src_ip=202.157.76.21;dst_ouser=user4;dst_user=user4;dst_domain=202.157.76.21 Thanks On Fri, Apr 10, 2015 at 4:38 PM, <users-request(a)lists.opensips.org> wrote: > Welcome to the Users(a)lists.opensips.org mailing list! > > To post to this list, send your email to: > > users(a)lists.opensips.org > > General information about the mailing list is at: > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > If you ever want to unsubscribe or change your options (eg, switch to > or from digest mode, change your password, etc.), visit your > subscription page at: > > > http://lists.opensips.org/cgi-bin/mailman/options/users/followtest111%40gma… > > > You can also make such adjustments via email by sending a message to: > > Users-request(a)lists.opensips.org > > with the word `help' in the subject or body (don't include the > quotes), and you will get back a message with instructions. > > You must know your password to change your options (including changing > the password, itself) or to unsubscribe. It is: > > 123456 > > Normally, Mailman will remind you of your lists.opensips.org mailing > list passwords once every month, although you can disable this if you > prefer. This reminder will also include instructions on how to > unsubscribe or change your account options. There is also a button on > your options page that will email your current password to you. >

2 1

TLS & log files
by Tributh 13 Apr '15

13 Apr '15

Hi, is there any way to configure the tls curve like secp384r1 in the config? I use actually version 4.2.4 On the other side i can't see any connection Information in the logs. I would like to see something like: "Connected with TLSv1.1 using cipher ECDHE-RSA-AES256-GCM-SHA384 and curve secp384r1" With this logs the variety of the offered ciphers can be shortened to the most secure and only needed by the clients. Regard T.Tributh

2 1

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sr-users April 2015