sr-users June 2015

sr-users@lists.kamailio.org

82 participants
136 discussions

several uac_replace_from in one dialog
by Никитенко Виталий 10 Jun '15

10 Jun '15

Hi! Making calls via different trunks through kamailio for simply so uac_replace_from ("sip:regname@ipaddress"); if (! t_relay ()) { uac_replace_from ("sip:regname2@ipaddress"); t_relay () } exit and run tcpdump i see in fisrt INVITE From: "testname" <sip: regname @ ipaddress>; tag = as21aea6e0 in second INVITE From: "testname" <sip: regname @ ipaddress sipregname2 @ ipaddress>; tag = as21aea6e0 and in the log file EROOR: can not parse From URI, and then can not find a BYE to terminate the call I try do remove_hf insert_hf but then i do second remove_hf it realy not delete and i have two field From. Who can tell me how few times to change the From field in a one dialog?

2 1

kamcmd permissions problem
by Uri Shacked 10 Jun '15

10 Jun '15

Hi, I am using exec_avp() command from kamailio. The command execute a shell script that runs kamctl and kamcmd commands. kamailio runs with user 'kamailio' and group 'kamailio' (defaults). There is not problem with the kamctl commands. But, the kamcmd commands fails due to permissions issues. Any ideas? Cheers, Uri

2 1

403 not relaying
by jaflong jaflong 09 Jun '15

09 Jun '15

Hi, I am getting a 403 not relaying response. This is the part of the config How would I print the value of var myself, from_uri and uri # 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; }

2 1

IP fragment
by Tuan Viet Nguyen 09 Jun '15

09 Jun '15

Hello, I would like to know if Kamailio handles the case where we receive a big SIP message fragmented in IP layer? Thanks,

2 2

Testing RPMs for v4.3.0
by Daniel-Constantin Mierla 09 Jun '15

09 Jun '15

Hello, in preparation for releasing v4.3.0 next week, I updated the specs for building RPMs from branch 4.3. However, I am not able to test the results, just made sure the build is ok. Packages are available at: - https://build.opensuse.org/package/show/home:kamailio:v4.3.x-rpms/kamailio43 I hope people here using RPMs will do some tests and provide feedback, interested mainly in: - it installs ok - after install, it starts ok with default configuration file - fifo and unix sockets are created now in /var/run/kamailio, so it has to be checked they are there - kamctl works ok for commands: ps, uptime and 'mi version' Some modules are not packaged in various versions, as I could not spot what are the packages needed for building them and if they are available with different names. For example, apparently jansson lib is on redhat, but not on centos, libev doesn't seem to be there. Also, I haven't had time to look at packaging erlang module. Help with the above is appreciated as well as helping building RPMs genrally. Maybe the init.d script needs to be reviewed and updated as well. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

2 4

Need information regarding kamilio setup to support IPSec
by Priyaranjan Nayak 09 Jun '15

09 Jun '15

Hi All, I have a ims client endpoint supporting rfc 3329(IPSec).Now I am sending register message to kamailio server with Security-Client header and not getting any Security-Server header in response. Could you please tell me below queries - Whether kamailio server supporting rfc 3329 (IPSec) or not ? - If yes,how can we setup kamailio server to support Security-Client,Security-Server and Security-varify(as per rfc 3329) header ? Thanks Priyaranjan

3 3

Private IP in SDP packet headers
by Alex 08 Jun '15

08 Jun '15

Hello! Please help to fix problem with sdp headers UAC Inet -> (X.X.X.X) Kamailio (192.168.30.250) -> Asterisk (192.168.30.2) When i call from UAC to 9002 i received INVITE/SDP from kamailio SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.52:27080 ;received=10.10.101.50;branch=z9hG4bK-d8754z-027c786dac17bf68-1---d8754z-;rport=27080 Record-Route: <sip :192.168.30.2;line=sr-mYtaP6eErk-dx6VfrLzfr6BaPGj0OHFfPYd0OHFfPYIQpHmFr9mQPKDEx9VlvZ8QO4ttma**> Record-Route: <sip:X.X.X.X;r2=on;lr=on;ftag=0748d948;nat=yes> From: <sip:user4@X.X.X.X>;tag=0748d948 To: <sip:9002@X.X.X.X>;tag=as3914e1d1 Call-ID: ZWU5YmFiNTNhNmNmYWQzYzhkZWUzZDNjOTU3MDFiNGU. CSeq: 2 INVITE Server: Virtel.net Node2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:192.168.30.2;line=sr-mYtaP62ar9nzrg20y6eYPA-LrA-0P6Bax6z*> Content-Type: application/sdp Content-Length: 278 v=0 o=root 732368067 732368067 IN IP4 X.X.X.X s=Asterisk PBX 11.17.1 c=IN IP4 X.X.X.X t=0 0 m=audio 15768 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=nortpproxy:yes Why Record-Route and Contact fields contain private IP of asterisk ?

2 1

Need to add extra header in 200 OK message by using kamailio.cfg file
by Priyaranjan Nayak 08 Jun '15

08 Jun '15

Hi All, I wanted to send a extra header(i.e. Security-Server ) in the 200 OK from kamailio server. Could you please suggest me how can I add extra header in 200 OK message by using kamailio.cfg file ? Thanks Priyaranjan

3 4

msilo storage of message id
by Shouvanik Chakrabarti 07 Jun '15

07 Jun '15

Hello. We have been using the msilo module to handle offline SIP messaging. I have noticed that the m_store function ends up storing all the records with the default message id which is zero. However m_dump uses this id to figure which messages have already been sent. Thus on calling m_dump on registration, no messages are dumped. We simply get a 'message already sent' debug log. Is there a way around this? Any help would be deeply appreciated. Thanks, Shouvanik.

2 2

Kamailio second contact header
by Ali Taher 05 Jun '15

05 Jun '15

Hello, I'm trying to build new contact header that will be sent from Kamailio as below : load_gws(1, $rU, $fu); append_to_reply("Contact:"); while (next_gw()) { append_to_reply("$ru"); } append_to_reply("\r\n"); sl_send_reply("300","Multiple Choices"); exit; yet I'm getting two Contact headers as shown in below SIP message. How can I remove second contact header (highlighted below in gray) Via: SIP/2.0/UDP 192.168.26.3:42528;branch=z9hG4bK-d8754z-4c2cc56c3145b446-1---d8754z-;rport= 42528;received=192.168.26.3 To: "9612"<sip:9612@192.168.110.181>;tag=b27e1a1d33761e85846fc98f5f3a7e58.1a8d From: "test"<sip:test@192.168.110.181>;tag=8571af74 Call-ID: Y2MzOTNiOGRhMTc2NDkwZjIxNzZhYTAzZmM0OGE3NDY CSeq: 2 INVITE Contact: sip:9612@C001,sip:9612@192.168.26.1,sip:9612@192.168.26.6,sip:9612@192.168.2 6.8 Contact: sip:9612@192.168.26.8 Server: kamailio (4.1.8 (x86_64/linux)) Content-Length: 0 Thanks, Ali

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sr-users June 2015