- sr-users - kamailio.org

pua_dialoginfo avps and entity
by Dmitri Savolainen 22 May '17

22 May '17

hi all! i'm trying to use pua_dialoginfo module (current master) with modparam("pua_dialoginfo", "use_pubruri_avps", 1) so for $avp(puburis_caller) = "sip:111111@mydomain"; i have PUBLISH with xml "entity" from "From:" field, not from my avp. PUBLISH sip:111111@mydomain SIP/2.0 Via: SIP/2.0/UDP 213.170.66.150:5071 ;branch=z9hG4bK9a08.f20b7367000000000000000000000000.0 To: <sip:111111@mydomain> From: <sip:111111@mydomain>;tag=b5f1e2ebdfeaa37cea72edd75fe37385-0dfc CSeq: 10 PUBLISH Call-ID: 6d71159d7ab5b9ff-23010@mydomain Content-Length: 260 User-Agent: Dev_Kamailio_UA Max-Forwards: 70 Event: dialog Expires: 41 Content-Type: application/dialog-info+xml <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:003@mydomain"> <dialog id="yciKcJRNdO1HRjFg0qOoPUe4V3IhbqIW" direction="initiator"> <state>Trying</state> </dialog> </dialog-info> is it ok? according to doc: > If use_pubruri_avps is enabled, PUBLISH-requests reporting > dialog-information about the caller (entity=caller) are sent using the > value of the specified avp as R-Uri. > -- Savolainen Dmitri

1 0

Lookup using the last N digits
by Emanuele Gambaro 22 May '17

22 May '17

Hello everyone I need to do a lookup for an existing user using the last N digit of its phone number. For example, if a subscriber has a phone number equal to "+391234567890" I would like to call him using the full phone number or also the last 8 digits "34567890". There is a simple way to do this using kamailio 5.0.1? -- Emanuele Gambaro --- email: emanuele.gambaro(a)pynlab.com skype: sarbyn_work OpenPGP Key: https://goo.gl/fdeVnI

2 1

Siremis upgrade and features
by javier falbo 21 May '17

21 May '17

Hi, I notice Siremis is not usable for me as it is very limited in the user management interface. I will suggest the developpers to include a USER management GUI visual interfase, as nowadays i use command line to do the following by putty tool: a) "Create Users" kamctl add <user> <password> b) "Delete Users" kamctl rm <user> c) "Show users in database" kamctl db show subscriber d) "Show online users" kamctl online Adding this on Siremis will help a lot for daily use as a Manager Account. Someone knows the Siremis author to add this suggestion? or someone has already coded a VISUAL GUI interfase for the above and could share it? Regards, Javier

4 7

Siremis searching from htable
by Olli Attila 21 May '17

21 May '17

Hello, Is there any way to make partial search from all htable entries with for example: %partofkeyname%? I tried to put that on search box but I get no results. If I use the exact full name of the key in the search i get the result. The search is here: Sip admin services - > server services -> htable list -> search I would like that the search function could return all of the keys that match the searched pattern but I have not succeed in making that kind of search. Cheers, Olli

2 2

How to debug Kamailio
by Remsik,Robert 19 May '17

19 May '17

Hello! I've got a base install of Kamailio and I'm trying to have it forward packets generated by sipsak to a different proxy (a migration of an existing ruleset by opensips). I can get the Kamailio process started but I can't figure out why Kamailio isn't forwarding the packet and I haven't been able to get debugging working to determine what's going on inside Kamailio to fix it. Can anyone please help me figure out how to debug this? Thank you in advance, Robert Robert Remsik ACNS Desk Phone: 970 491 7120 Robert.Remsik(a)colostate.edu

3 4

Repeat of packets
by Jean Cérien 18 May '17

18 May '17

Hello I am getting started with Kamailio (or restarted, used it briefly years ago), with the final objective to do load balancing. For the time being, I am just trying to have one asterisk and one kamailio, on the same box. I have setup a box with an asterisk 11.3, and kamailio 4.4. I've taken the config file from http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . My idea is that asterisk runs on port 5080, while kamailio on port 5060. Client interacts with Kamailio on port 5060. It almost works... Registration is fine, but when I send an invite, it is properly acknowledged (by asterisk 100 trying then 200 OK) - but the OK message gets repeated multiple times and asterisk issues its infamous 'Retransmission timeout reached ...' - as if Kamailio wasnt processing it. See below ngrep traces between asterisk and kamailio Any ideas where to look ? Thanks J. # U +18.289105 192.168.2.228:5060 -> 192.168.2.228:5080 INVITE sip:102@192.168.2.228 SIP/2.0..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b9922 4198ef593.0..Via: SIP/2.0/UDP 192.168.2.200:5085;received=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip:102@ 192.168.2.228>..Call-ID: 1571382735..CSeq: 21 INVITE..Contact: <sip:iper@(null)>..Content-Type: application/sdp..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY , MESSAGE, SUBSCRIBE, INFO..Max-Forwards: 69..User-Agent: Linphone/3.6.1 (eXosip2/4.1.0)..Subject: Phone call..Content-Length: 437....v=0..o=199 2799 2990 IN IP4 192. 168.2.200..s=Talk..c=IN IP4 192.168.2.200..t=0 0..m=audio 7078 RTP/AVP 124 111 110 0 8 101..a=rtpmap:124 opus/48000..a=fmtp:124 useinbandfec=1; usedtx=1..a=rtpmap:111 s peex/16000..a=fmtp:111 vbr=on..a=rtpmap:110 speex/8000..a=fmtp:110 vbr=on..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-11..m=video 9078 RTP/AVP 103 99..a=rtpmap:103 VP8/90000..a=rtpmap:99 MP4V-ES/90000..a=fmtp:99 profile-level-id=3.. # U +0.000683 192.168.2.228:5080 -> 192.168.2.228:5060 OPTIONS sip:199@192.168.2.228:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.2.228:5080;branch=z9hG4bK7c2d71fb..Max-Forwards: 70..From: "asterisk" <sip:199@192.168.2.228:5080>; tag=as57c98c4b..To: <sip:199@192.168.2.228:5060>..Contact: < sip:199@192.168.2.228:5080>..Call-ID: 52fa034372ce18ca2b93fc1817ad38a5@192.168.2.228:5080..CSeq: 102 OPTIONS ..User-Agent: Asterisk PBX 11.3.0..Date: Tue, 16 May 2017 21:08:29 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: re places, timer..Content-Length: 0.... # U +0.001035 192.168.2.228:5080 -> 192.168.2.228:5060 OPTIONS sip:199@192.168.2.228:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.2.228:5080;branch=z9hG4bK4248158e..Max-Forwards: 70..From: "asterisk" <sip:199@192.168.2.228:5080>; tag=as30d9cfb4..To: <sip:199@192.168.2.228:5060>..Contact: < sip:199@192.168.2.228:5080>..Call-ID: 4331da391bca02965b2af65254717a18@192.168.2.228:5080..CSeq: 102 OPTIONS ..User-Agent: Asterisk PBX 11.3.0..Date: Tue, 16 May 2017 21:08:29 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: re places, timer..Content-Length: 0.... # U +0.000407 192.168.2.228:5080 -> 192.168.2.228:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;rec eived=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip :102@192.168.2.228>..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H..Supported: replaces, timer..Contact: <sip:102@192.168.2.228:5080>..Content-Length: 0.... # U +0.003961 192.168.2.228:5080 -> 192.168.2.228:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;receive d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip:102 @192.168.2.228>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, I NFO, PUBLISH..Supported: replaces, timer..Contact: < sip:102@192.168.2.228:5080>..Content-Type: application/sdp..Content-Length: 312....v=0..o=root 350189084 350189084 I N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4 192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telep hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99.. # U +0.087859 192.168.2.228:5060 -> 192.168.2.228:5080 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228:5080;branch=z9hG4bK7c2d71fb..From: "asterisk" <sip:199@192.168.2.228:5080>;tag=as57c98c4b..To: < sip:199@192.168.2.228:506 0>;tag=524348182..Call-ID: 52fa034372ce18ca2b93fc1817ad38a5@192.168.2.228:5080..CSeq: 102 OPTIONS..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, SUBSCRIBE, NOTIFY, INFO..Accept: application/sdp..User-Agent: Linphone/3.6.1 (eXosip2/4.1.0)..Content-Length: 0.... # U +0.000213 192.168.2.228:5060 -> 192.168.2.228:5080 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228:5080;branch=z9hG4bK4248158e..From: "asterisk" <sip:199@192.168.2.228:5080>;tag=as30d9cfb4..To: < sip:199@192.168.2.228:506 0>;tag=939659485..Call-ID: 4331da391bca02965b2af65254717a18@192.168.2.228:5080..CSeq: 102 OPTIONS..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, SUBSCRIBE, NOTIFY, INFO..Accept: application/sdp..User-Agent: Linphone/3.6.1 (eXosip2/4.1.0)..Content-Length: 0.... # U +0.011138 192.168.2.228:5080 -> 192.168.2.228:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;receive d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip:102 @192.168.2.228>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, I NFO, PUBLISH..Supported: replaces, timer..Contact: < sip:102@192.168.2.228:5080>..Content-Type: application/sdp..Content-Length: 312....v=0..o=root 350189084 350189084 I N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4 192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telep hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99.. # U +0.200291 192.168.2.228:5080 -> 192.168.2.228:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;receive d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip:102 @192.168.2.228>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, I NFO, PUBLISH..Supported: replaces, timer..Contact: < sip:102@192.168.2.228:5080>..Content-Type: application/sdp..Content-Length: 312....v=0..o=root 350189084 350189084 I N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4 192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telep hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99.. # U +0.400478 192.168.2.228:5080 -> 192.168.2.228:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via: SIP/2.0/UDP 192.168.2.200:5085;receive d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route: <sip:192.168.2.228;lr=on;ftag=1034946464>..From: <sip:199@192.168.2.228>;tag=1034946464..To: <sip:102 @192.168.2.228>;tag=as497f35c0..Call-ID: 1571382735..CSeq: 21 INVITE..Server: Asterisk PBX 11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, I NFO, PUBLISH..Supported: replaces, timer..Contact: < sip:102@192.168.2.228:5080>..Content-Type: application/sdp..Content-Length: 312....v=0..o=root 350189084 350189084 I N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4 192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telep hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99.. #

3 5

IMS: Including contents in the body of third-party REGIISTER request.
by Haggerty, Paul (phaggert) 18 May '17

18 May '17

It looks like the Kamailio IMS modules don't support including the original REGISTER request in third party registrations as defined in 3GPP TS 24.229. Can someone confirm this? Should this be a tracked defect or enhancement? In the meantime, is there any workaround available? Our application server needs to get the contents from the original REGISTER message from the UE. Thanks, Paul

2 4

Publish/notify for group kamailio
by Nguyen Tran Nhan 18 May '17

18 May '17

Hi, Is there any way to send Publish/Subscribe to group and Notify for all group (conference)? I have tried to send Subscribe to conference URI (SEMS) but I got method not implemented error. Any idea? Thanks, Nhan

1 0

kamcmd cli autocomplete fetaure stopped working after update from 4.3.4 to 4, 4.4 and 4.4.5
by Nuno Reis 18 May '17

18 May '17

Hi guys. I've just noticed that I can't use autocomplete under kamcmd cli anymore, after upgrading from 4.3.4 to latest versions. Should I open a github issue for this? -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* <http://www.wavecom.pt/> <http://www.wavecom.pt/pt/>

2 5

no audio with kamailio & rtpproxy bebind NAT
by 赵国杰 18 May '17

18 May '17

Hi everyone, I'm new to kamailio and rtpproxy and the problem's been bothering me for days. Any help would highly appreciated. My network topology is as follows: kamailio5.0 and rtpproxy2.0 are running on the same server. The server is behind NAT, its private ip is 10.240.0.62, public ip is <PUBLICIP> the server has 5060 tcp&udp, 35000-65000 udp port enabled. There're 2 sip clients(test && test1) running in another LAN(175.100.202.xxx) . Situation: Both clients registered successfully and they can call each other. But neither of them has audio.(If i kill rtpproxy, audio is ok). My kamailio cfg(only posted the part that i change), #!KAMAILIO #!define WITH_MYSQL #!define WITH_DEBUG #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ANTIFLOOD #!define WITH_NAT #!define WITH_PSTN listen=udp:10.240.0.62:5060 advertise <PUBLICIP>:5060 modparam("rtpproxy", "rtpproxy_sock", "udp:10.240.0.62:64000") rtpproxy cmd: rtpproxy -A <PUBLICIP> -F -l 10.240.0.62 -m 35000 -M 65000 -s udp:*:64000 -d INFO I used ngrep -d any -qt -W byline port 5060 to capture the sip msg, they are as follows: U 2017/05/18 05:56:29.320152 175.100.202.33:10870 -> 10.240.0.62:5060 INVITE sip:test1@<PUBLICIP> SIP/2.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport;branch=z9hG4bK-524287-1---9e44840a531df765. Max-Forwards: 70. Contact: <sip:test@175.100.202.33:10870;rinstance=6238e8023060ce02>. To: <sip:test1@<PUBLICIP>>. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 1 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.8 stamp 84253. Content-Length: 331. . v=0. o=- 1500767650 1 IN IP4 175.100.202.33. s=X-Lite release 4.9.8 stamp 84253. c=IN IP4 175.100.202.33. t=0 0. m=audio 63960 RTP/AVP 9 8 120 0 84 101. a=rtpmap:120 opus/48000/2. a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000. a=rtpmap:84 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2017/05/18 05:56:29.321103 10.240.0.62:5060 -> 175.100.202.33:10870 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;branch=z9hG4bK-524287-1---9e44840a531df765;received=175.100.202.33. To: <sip:test1@<PUBLICIP>>;tag=24b9bbecdb24b798805f15f8519a2604.ea26. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 1 INVITE. Proxy-Authenticate: Digest realm="<PUBLICIP>", nonce="WR04uVkdN40MqPSv+qEh76S0Jk2d2uq8". Server: kamailio (5.0.1 (x86_64/linux)). Content-Length: 0. . U 2017/05/18 05:56:29.384015 175.100.202.33:10870 -> 10.240.0.62:5060 ACK sip:test1@<PUBLICIP> SIP/2.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport;branch=z9hG4bK-524287-1---9e44840a531df765. Max-Forwards: 70. To: <sip:test1@<PUBLICIP>>;tag=24b9bbecdb24b798805f15f8519a2604.ea26. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 1 ACK. Content-Length: 0. . U 2017/05/18 05:56:29.388008 175.100.202.33:10870 -> 10.240.0.62:5060 INVITE sip:test1@<PUBLICIP> SIP/2.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Max-Forwards: 70. Contact: <sip:test@175.100.202.33:10870;rinstance=6238e8023060ce02>. To: <sip:test1@<PUBLICIP>>. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 2 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Proxy-Authorization: Digest username="test",realm="<PUBLICIP>",nonce="WR04uVkdN40MqPSv+qEh76S0Jk2d2uq8",uri="sip:test1@<PUBLICIP>",response="e7c42929b9a62de980651327b3a6999c",algorithm=MD5. Supported: replaces. User-Agent: X-Lite release 4.9.8 stamp 84253. Content-Length: 331. . v=0. o=- 1500767650 1 IN IP4 175.100.202.33. s=X-Lite release 4.9.8 stamp 84253. c=IN IP4 175.100.202.33. t=0 0. m=audio 63960 RTP/AVP 9 8 120 0 84 101. a=rtpmap:120 opus/48000/2. a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000. a=rtpmap:84 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2017/05/18 05:56:29.389271 10.240.0.62:5060 -> 175.100.202.33:10870 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;branch=z9hG4bK-524287-1---7a361a514cf88b6b;received=175.100.202.33. To: <sip:test1@<PUBLICIP>>. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 2 INVITE. Server: kamailio (5.0.1 (x86_64/linux)). Content-Length: 0. . U 2017/05/18 05:56:29.389935 10.240.0.62:5060 -> 175.100.202.254:53006 INVITE sip:test1@175.100.202.254:53006;ob SIP/2.0. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Via: SIP/2.0/UDP <PUBLICIP>:5060;branch=z9hG4bKc9f1.223c15db97157715ffa93dfeb61a6d8e.0. Via: SIP/2.0/UDP 175.100.202.33:62434;received=175.100.202.33;rport=10870;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Max-Forwards: 69. Contact: <sip:test@175.100.202.33:10870;rinstance=6238e8023060ce02;alias=175.100.202.33~10870~1>. To: <sip:test1@<PUBLICIP>>. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 2 INVITE. Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite release 4.9.8 stamp 84253. Content-Length: 349. . v=0. o=- 1500767650 1 IN IP4 <PUBLICIP>. s=X-Lite release 4.9.8 stamp 84253. c=IN IP4 <PUBLICIP>. t=0 0. m=audio 46862 RTP/AVP 9 8 120 0 84 101. a=rtpmap:120 opus/48000/2. a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000. a=rtpmap:84 speex/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=nortpproxy:yes. U 2017/05/18 05:56:29.782323 175.100.202.254:53006 -> 10.240.0.62:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP <PUBLICIP>:5060;received=<PUBLICIP>;branch=z9hG4bKc9f1.223c15db97157715ffa93dfeb61a6d8e.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;received=175.100.202.33;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. From: <sip:test@<PUBLICIP>>;tag=106c953a. To: <sip:test1@<PUBLICIP>>. CSeq: 2 INVITE. Content-Length: 0. . U 2017/05/18 05:56:29.846192 175.100.202.254:53006 -> 10.240.0.62:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP <PUBLICIP>:5060;received=<PUBLICIP>;branch=z9hG4bKc9f1.223c15db97157715ffa93dfeb61a6d8e.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;received=175.100.202.33;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. From: <sip:test@<PUBLICIP>>;tag=106c953a. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. CSeq: 2 INVITE. Contact: <sip:test1@175.100.202.254:53006;ob>. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Content-Length: 0. . U 2017/05/18 05:56:29.846709 10.240.0.62:5060 -> 175.100.202.33:10870 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;received=175.100.202.33;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. From: <sip:test@<PUBLICIP>>;tag=106c953a. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. CSeq: 2 INVITE. Contact: <sip:test1@175.100.202.254:53006;ob;alias=175.100.202.254~53006~1>. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Content-Length: 0. . U 2017/05/18 05:56:32.768386 175.100.202.254:53006 -> 10.240.0.62:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP <PUBLICIP>:5060;received=<PUBLICIP>;branch=z9hG4bKc9f1.223c15db97157715ffa93dfeb61a6d8e.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;received=175.100.202.33;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. From: <sip:test@<PUBLICIP>>;tag=106c953a. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. CSeq: 2 INVITE. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Contact: <sip:test1@175.100.202.254:53006;ob>. Supported: replaces, 100rel, timer, norefersub. Content-Type: application/sdp. Content-Length: 233. . v=0. o=- 3704075788 3704075789 IN IP4 10.60.208.171. s=pjmedia. c=IN IP4 10.60.208.171. t=0 0. m=audio 4008 RTP/AVP 0 101. c=IN IP4 10.60.208.171. a=sendrecv. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. U 2017/05/18 05:56:32.769284 10.240.0.62:5060 -> 175.100.202.33:10870 SIP/2.0 200 OK. Via: SIP/2.0/UDP 175.100.202.33:62434;rport=10870;received=175.100.202.33;branch=z9hG4bK-524287-1---7a361a514cf88b6b. Record-Route: <sip:<PUBLICIP>;lr;nat=yes>. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. From: <sip:test@<PUBLICIP>>;tag=106c953a. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. CSeq: 2 INVITE. Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS. Contact: <sip:test1@175.100.202.254:53006;ob;alias=175.100.202.254~53006~1>. Supported: replaces, 100rel, timer, norefersub. Content-Type: application/sdp. Content-Length: 255. . v=0. o=- 3704075788 3704075789 IN IP4 <PUBLICIP>. s=pjmedia. c=IN IP4 <PUBLICIP>. t=0 0. m=audio 42584 RTP/AVP 0 101. c=IN IP4 <PUBLICIP>. a=sendrecv. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=nortpproxy:yes. U 2017/05/18 05:56:32.836731 175.100.202.33:10870 -> 10.240.0.62:5060 ACK sip:test1@175.100.202.254:53006;ob;alias=175.100.202.254~53006~1 SIP/2.0. Via: SIP/2.0/UDP 175.100.202.33:62434;rport;branch=z9hG4bK-524287-1---ee5d0f26a850b671. Max-Forwards: 70. Route: <sip:<PUBLICIP>;lr;nat=yes>. Contact: <sip:test@175.100.202.33:10870;rinstance=6238e8023060ce02>. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 2 ACK. User-Agent: X-Lite release 4.9.8 stamp 84253. Content-Length: 0. . U 2017/05/18 05:56:32.837266 10.240.0.62:5060 -> 175.100.202.254:53006 ACK sip:test1@175.100.202.254:53006;ob SIP/2.0. Via: SIP/2.0/UDP <PUBLICIP>:5060;branch=z9hG4bKc9f1.c283a6093e5f4143adec82663a0ff1d1.0. Via: SIP/2.0/UDP 175.100.202.33:62434;received=175.100.202.33;rport=10870;branch=z9hG4bK-524287-1---ee5d0f26a850b671. Max-Forwards: 69. Contact: <sip:test@175.100.202.33:10870;rinstance=6238e8023060ce02;alias=175.100.202.33~10870~1>. To: <sip:test1@<PUBLICIP>>;tag=iGrC0-.CMQN.LvSA2guPmwBlvTWEsUOu. From: <sip:test@<PUBLICIP>>;tag=106c953a. Call-ID: 84253NGRlOTIzM2M0NTMyNDY3MWRkNzBmODlmNDIzYWUxOTQ. CSeq: 2 ACK. User-Agent: X-Lite release 4.9.8 stamp 84253. Content-Length: 0. . U 2017/05/18 05:56:33.025120 10.240.0.62:5060 -> 175.100.202.254:53006 .... U 2017/05/18 05:56:38.025707 10.240.0.62:5060 -> 175.100.202.33:1490 .... U 2017/05/18 05:56:38.025731 10.240.0.62:5060 -> 175.100.202.33:10870 .... -- cheers, Jesse

1 0

Kamailio 4.4.X bug fixes
by José Seabra 17 May '17

17 May '17

Hello there, I apology for the following question but i didn't find this information on the kamailio site. When the development cycle in terms of bug fixes ends on version 4.4.x? Thank you Regards José

2 2

PV Setting Question
by Duarte Rocha 17 May '17

17 May '17

Good Afternoon, I'm having trouble with creation and value setting of some pseudo-variables. In the first cases, I used the pvapic.c function pv_get_sintval() like this : pv_get_sintval(msg, param, res, number_int_value); Using pv_get_sintval() i can assign the value to the PV and print it in the configuration file without any problem. However, if i need to assign a char* to the PV and use pv_get_strzval(msg, param, res, number_char_value);, it doesn't reach the configuration file. I made some debugs inside pv_get_strzval() in order to see the values of "res->rs.s" and res->rs.len and both values are fine. I just can't access them in the configuration file. Thanks in advance for your help! Cheers, Duarte Rocha

2 1

dmq_t_replicate
by Jay Patel 17 May '17

17 May '17

Hello Guys, Has anyone used dmq_t_replicate function to replicate INVITE? Documenation says it is useful for REGISTER , but does not mention anything about INVITE. I am trying to do something like this Box 1 ) Runs kamailio stateless ( no dialog module ) and calls dmq_t_replicate to box 2. Box 2 ) Runs kamailio with dialog module , This box generates CDR and tracks call count Goal is to restart kamailio on Box 1 without loosing CDR. Any input is appreciated.!! Regards, Jay

3 2

How to gateway MSRP messages into XMPP
by Pranathi Venkatayogi 16 May '17

16 May '17

I read about XMPP module in Kamailio, but it looks like it works with "MESSAGE" packets (SIMPLE). How do I gateway MSRP packets to XMPP in Kamailio. Do we have module support? Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

1 0

Failing in dropping duplicated 200 OK on onreply_route()
by Stefano Bertuola 16 May '17

16 May '17

Hi. I am trying to drop duplicated 200 OK on onreply_route(), but Kamailio still delivering them: if (is_method("INVITE") && t_check_status("200") && sdp_content()) { xlog("L_INFO", "200 OK Received\n"); if (!isflagset(20)) { xlog("L_INFO", "200 OK Relayed\n"); setflag(20); } else { xlog("L_INFO", "200 OK Dropped\n"); drop; } } >From logs, I can see the script is executed, but still 200 OK is relayed: INFO: <script>: 200 OK Received INFO: <script>: 200 OK Dropped DEBUG: tm [t_reply.c:1234]: t_should_relay_response(): ->>>>>>>>> T_code=200, new_code=200 DEBUG: tm [t_reply.c:1239]: t_should_relay_response(): DBG: t_should_relay_response: 200 INV after final sent DEBUG: tm [t_reply.c:1752]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 [...] How can I proceed to get the desired behavior? Br. Stefano

2 2

Why is the To URI the default in save()?
by Daniel Tryba 16 May '17

16 May '17

The save function from the registrar module uses the To header to disect and store the username for the location table according to observations and documentation http://www.kamailio.org/docs/modules/stable/modules/registrar.html#registra… After troubleshooting a ticket from an enduser unable to receive calls where all looked fine but the username used for authentication wasn't showing up in the location database. Finally I found the REGISTER was added to the location database, but not with the user its username, instead it was using the username (phonenumber) specified in the To header. Till now I always assumed that the username in the location table would be the username used during authentication(*). This opens the door to hijacking incoming calls to other users on the same kamailio registrar if one knows/guesses other usernames and use those in the To header. This realisation is kind of shocking to me. The solution is simple (if authentication is required): save("location", "0x00", "sip:$au@$rd"); *: which kind of answers my question in the subject, what else can be used if there is no authentication required?

3 3

✔I have finally found that stuff!
by poker poker 16 May '17

16 May '17

Greetings! You are going to be surprised a lot! I've finally found the stuff I told you about some time ago, just take a look http://www.realhopgar.com/wp-includes/pomo/rest.php?4c4d Thx, poker poker From: sr-users [mailto:sr-users@kamailio.org] Sent: Tuesday, May 16, 2017 2:49 AM To: pokerino76(a)libero.it Subject: Are you kidding? "Planets?" I squint my eyes and frown. "Who are you?" "I am Prince Eri. You have proven an excellent tactician and master manipulator of public perception." The (wo)man? looked at me with what I could only describe as hunger. Still rather confused I ask the only thing I could. "How do you figure that? I work a dead end job looking after two kids that aren't even mine." I scoff off what they said. "Simple. Remember when three months ago you received an email for a game in beta testing?" Wait, the game where I had to enact policies of keeping the public opinion of my army positive whilst simultaneously attempting to take over the known universe. "Wait don't tell me...." "Yes, it was real. Did you not think it odd how you were only able to take over two planets in three months? You were silently giving us our orders." The alien smiles? I don't know. "Why?" A sinking feeling enters the pit of his stomach. "Boredom mostly, we were mostly tired of steam rolling every planet we saw. So I devised the limitation of following some random person from Earth's orders. Surprisingly, the galactic opinion of us improved. We relied less and less on genocide. They now view us as a benevolent empire seeking to spread freedom and liberty. Thank you for that." I stood there dumbstruck. This isn't real, this is a dream. "Either join us and gain riches your entire world could never conceive of, or when we inevitably come after this planet, watch it all burn down." The choice already made for me I accept. But if what they say about me is true, then I will teach them something that never occurred to their alien minds. Humanity. Sent from Mail for Windows 10

1 0

Kamailio 5.0.1 install fails on Xenial
by Justin Clacherty 15 May '17

15 May '17

Hi, I’ve just set up a fresh install of Ubuntu 16.04.2 and tried to install Kamailio 5.0.1 using the apt repository at deb.kamailio.org <http://deb.kamailio.org/> but it fails to install during the configuration stage. Has anyone else seen this issue and are there any workarounds? Cheers, Justin. Setting up kamailio (5.0.1+xenial) ... Job for kamailio.service failed because the control process exited with error code. See "systemctl status kamailio.service" and "journalctl -xe" for details. invoke-rc.d: initscript kamailio, action "start" failed. ● kamailio.service - Kamailio (OpenSER) - the Open Source SIP Server Loaded: loaded (/lib/systemd/system/kamailio.service; enabled; vendor preset: enabled) Active: failed (Result: exit-code) since Sun 2017-05-14 23:54:15 AEST; 6ms ago Process: 1881 ExecStart=/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f $CFGFILE -m $SHM_MEMORY -M $PKG_MEMORY -u $USER -g $GROUP (code=exited, status=255) May 14 23:54:15 km01 kamailio[1881]: CRITICAL: <core> [core/cfg.y:3400]: yyerror_at(): parse error in config file /etc/kamaili... module May 14 23:54:15 km01 kamailio[1881]: ERROR: <core> [core/sr_module.c:570]: load_module(): could not find module <maxfwd> in </...odules> May 14 23:54:15 km01 kamailio[1881]: CRITICAL: <core> [core/cfg.y:3400]: yyerror_at(): parse error in config file /etc/kamaili... module May 14 23:54:15 km01 kamailio[1881]: ERROR: <core> [core/sr_module.c:570]: load_module(): could not find module <usrloc> in </...odules> May 14 23:54:15 km01 kamailio[1881]: CRITICAL: <core> [core/cfg.y:3400]: yyerror_at(): parse error in config file /etc/kamaili... module May 14 23:54:15 km01 kamailio[1881]: ERROR: bad config file (46 errors) May 14 23:54:15 km01 systemd[1]: kamailio.service: Control process exited, code=exited status=255 May 14 23:54:15 km01 systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. May 14 23:54:15 km01 systemd[1]: kamailio.service: Unit entered failed state. May 14 23:54:15 km01 systemd[1]: kamailio.service: Failed with result 'exit-code'. Hint: Some lines were ellipsized, use -l to show in full. dpkg: error processing package kamailio (--configure): subprocess installed post-installation script returned error exit status 1

2 2

SCTP Support for Rx/Dx interfaces
by Fred Schulz 15 May '17

15 May '17

Hi everybody, as I can read in the docs is the support for SCTP possible via Module just for SIP. Is there a possibility to add SCTP support to Rx/Dx interfaces at ICSCF? As read in the specs SCTP is mandatory for that interface… Thanks Mit freundlichen Grüßen fred schulz e-mail: fschulz(a)blackned.de<mailto:fschulz@blackned.de> tel.: +49 8331 9959-6-00 fax: +49 8331 9959-6-09 [cid:image001.png@01D2CD6A.382B1260] blackned gmbh · www.blackned.de<http://www.blackned.de/> niederlassung memmingen: zugspitzstr. 1 · 87751 heimertingen niederlassung münchen :friedrichshafener strasse 3 · 82205 gilching zentrale: dorfstrasse 3 · 88416 erlenmoos geschäftsführer: timo haas · josef stadler · hrb ulm 724147 Diese E-Mail enthält vertrauliche und/oder rechtlich geschützte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und löschen Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and delete this e-mail. Any unauthorized copying, disclosure or distribution of the contents in this e-mail is strictly forbidden.

2 2

forced tls over tcp
by David Villasmil 15 May '17

15 May '17

Hello guys, I'm listening on port 443 for tls and 5111 for tcp. Client registers on 443 and makes a call over tls which kamailio forwards over tcp. Because freeswitch doesn't support "sips" yet, when replies come back from freeswitch (which contact set as "sip:" i manually change that tp "sips:" for the client to not die with "SIPS Required". Call is established and the client responds with ACK. At this point i need to change back the contact "sips:" to "sip:". This is the route that changes it: route[FIXCONTACT] { # This is freeswitch sending a message, so we change "sip:" to "sips:" xlog("L_ERR","[FIXCONTACT]: User-Agent is: $ua\n"); if( $ua =~ "^FreeSWITCH" ) { if( subst('/^(Contact:.*)sip:/\1sips:/') ) { xlog("L_ERR","[FIXCONTACT]: Method $rm Status $rs Changed contact coming from freeSWITCH from sip to sips [$ct]!\n"); } # Else it is the client, when change it back from "sips:" to "sip:" } else { if( subst("/^(Contact:.*)sips:/\1sip:/") ) { xlog("L_ERR","[FIXCONTACT]: Method $rm Status $rs Changed contact coming from CLIENT from sips to sip [$ct]!\n"); } } return; } When kamailio is preparing to send out the ACK via TCP I'm seeing: [forward.c:268]: get_send_socket2(): protocol/port mismatch And I really don't know why it's doing this... Help is appreciated! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 3

kamcmd domain.dump ERROR: reply too big
by Denys Pozniak 15 May '17

15 May '17

Hello! I have updated Kamailio to version 5.1.0-dev2 and started to obtain next error: *kamcmd> domain.dump* *ERROR: reply too big* Tried to increase ctl buffer size, but result is the same. *loadmodule "ctl.so"* *modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl")* *modparam("ctl", "binrpc_max_body_size", 32000)* *modparam("ctl", "binrpc_struct_max_body_size", 3200)* Please point me where to find issue. *# uname -a* *Linux 4.4.0-64-generic #85-Ubuntu SMP Mon Feb 20 11:50:30 UTC 2017 x86_64 x86_64 x86_64 GNU/Linux* *# cat /etc/lsb-release* *DISTRIB_ID=Ubuntu* *DISTRIB_RELEASE=16.04* *DISTRIB_CODENAME=xenial* *DISTRIB_DESCRIPTION="Ubuntu 16.04.2 LTS"* BR, Denys

2 2

Kamailio + Sems with TURN server
by Nguyen Tran Nhan 15 May '17

15 May '17

Hi, I am trying to enable TURN with Kamailio + SEMS. Is there any link for guide? Thanks, Nhan

2 2

MSILO DUMPING SAME MESSAGE MULTIPLE TIMES
by sagar malam 15 May '17

15 May '17

Hello, I have been using Kamailio 4.4.In there i have used MSILO module to store messages when B party is not available and dump message when it gets registered. But when user gets registered,MSILO is dumping same messages multiple times. On further observation i found that MSILO stops dumping message when 200 OK for any one of the multiple message requests arrives to server.Hence i tried configuring "check_time" parameter to higher value but it did not help. I am am using same example script as provided in below link : http://kamailio.org/docs/modules/4.4.x/modules/msilo.html Except tm module parameters. Please help me on this issue. Thanks in advance.

2 1

IP Address used for outgoing Packets?
by Keith Whyte 15 May '17

15 May '17

Hi All, tl;dr - Kamailio is sending traffic to the Voip server with a source IP of the private (vpn) interface. For some weeks, I've been experimenting with Kamailio v 4.4.4, trying different options, including running (or not) a vpn interface on the kamailio server with and without NAT/rtpproxy. I have not as yet fully "installed" kamailo with database etc, I have just been playing with the config file, mostly running with -D For most of the configurations I tried, I was routing SIP to/from the VPN via another box on the network, and I tried to get RTP streams running without using rtpproxy, sometimes it worked, but I have two many double/triple NAT problems with some of the freeswitch nodes. Now I have decided to run a VPN interface directly on the kamailio server, for simplicity. Essentially what I want to do is have kamailio/rtpproxy bridge between the VPN and the VOIP provider, so it should be a pretty standard NAT config. My setup looks like this: VPN CLUSTER of Freeswitch nodes <-10.23.0.1-> KAMAILIO Server <-192.168.1.1 - gateway - PUBLIC_IP-> Voip Provider The Kamailio Server has two interfaces, one VPN interface and one facing the Public Internet. It is behind a firewall, and so has a private IP address assigned on the interface, but the is dedicated public IP that forwards all relevant ports, and traffic seen on the public side originates from this address. I have two listen directives, listen=udp:192.168.1.1:5060 advertise PUBLIC_IP:5060 listen=udp:10.23.0.1:5060 I expected Kamailio to receive SIP INVITE on the 10.23.0.1 address and forward it to the voip provider via the default route (the 192.168.1.1 address), but it is sending the packet out to the voip provider with a source address of 10.23.0.1 !! Can anyone tell me how I can configure this? I cannot find anything obvious in the manual. Thanks!

3 2

SIREMIS 4.3.0 and Kamilio 5.1.0-dev1
by Emil Morganti 12 May '17

12 May '17

Hi everybody, I am installing siremis 4.3.0 with kamilio 5.1.0-dev1 but when i step to the second step i get an error: ERROR: SQLSTATE[23000]: Integrity constraint violation: 1062 Duplicate entry '127.0.0.1' for key 'domain_idx' Some solution?

2 1

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