sr-users May 2018

sr-users@lists.kamailio.org

92 participants
138 discussions

Question about using Kamailio and Asterisk and flow of an "INVITE"

by Wilkins, Steve

Hello All, I am trying to resolve, in my mind, the flow of a WebRTC<=>WebRTC call using Kamailio and Asterisk. Each WebRTC client is registered in Kamailio and when I call WebTRC Client1 from WebRTC Client2 what I see is -> The Invite is sent from Kamailio to Asterisk and then Asterisk is sending the Invite back to Kamailio. Also depending on The version of Asterisk, the INVITE will then get forwarded to the AOR that is registered in Kamailio for the called number. Does this seem correct? It seems like there is an extra hop in there. The reason I am now very curious now is because everything works fine if using Kamailio 5.0 and Asterisk 14.6, but I switch to Asterisk 15.3 I get the extra hop and call is dropped after 30 seconds. I would appreciate any thoughts on this. Thank you in advance.

6 years

"Sleeping Giant - No Snore" Kam?

by KamDev Essa

We have noticed pesky SIP hackers already knocking on our POC servers. Kam is holding the fort by sending out the proverbial 401/407 (snore). I can SQLOPS to do a dip to check destination domain or Source IP and nix the snore for alien requests, but wanted to know if there was a native way to keep silent when we have a request from foes. KD

6 years

Understanding port changing

by Social Boh

Hello list, I'm looking for to understand a problem have with Kamailio. The configuration is: user (behind NAT) -> Kamilio (Public IP) -> user (behind NAT) When I make a call from user A to user B (both registered) , User B receive the INVITE correctly but in the ringing SIP message change the port present on the Contact Header and same think on 200OK so when Kamailio try to send it ACK SIP message to this port fail because the Softphone listen on the port where has received the INVITE from Kamailio. This happens with 3 different Softphone: BRIA, Xlite, Linphone on the same computer. Thank you Regards -- --- I'm SoCIaL, MayBe

6 years

WebRTC to SIP gateway

by Pan Christensen

Hello! It's been several years since I've used Kamailio. My current employer wants to implement WebRTC, which is currently not supported in our SIP backend, and asked if I could set up a Kamailio server as a gateway. I've been able to make calls in all directions between SIP and WebRTC clients registered locally on Kamailio. When I tried to connect the server to the SIP backend, I ran into an issue. I'm able to register SIP clients in the backend via the gateway and make calls everywhere. However, the WebRTC client fails to register. Here are the messages between the Kamailio gateway and the SIP backend: U 2018/05/09 10:12:58.316643 GATEWAY:15060 -> DOMAIN:5060 REGISTER sip:DOMAIN SIP/2.0. Via: SIP/2.0/UDP GATEWAY:15060;branch=z9hG4bK4fc6.04d1730be5b78d595c69a3aa137987c1.0. Via: SIP/2.0/AUTO lr2l9s72ehhc.invalid;rport=61353;received=CLIENT;branch=z9hG4bK5927151. Max-Forwards: 68. To: <sip:4777519304@DOMAIN>. From: <sip:4777519304@DOMAIN>;tag=9qhqhrnj3s. Call-ID: j5h7830ivr5dfc2mn5sov1. CSeq: 4 REGISTER. Contact: <sip:c3qkm4fv@lr2l9s72ehhc.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:0ef2deac-2d56-465a-840b-543b9fd01af8>";expires=600. Expires: 600. Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO. Supported: path,gruu,outbound. User-Agent: JsSIP 3.2.9. Content-Length: 0. Path: <sip:GATEWAY:15060;lr>. . U 2018/05/09 10:12:58.368409 DOMAIN:5060 -> GATEWAY:15060 SIP/2.0 400 Wrong transport. Provided transport either invalid or not supported.. Via: SIP/2.0/UDP GATEWAY:15060;branch=z9hG4bK4fc6.04d1730be5b78d595c69a3aa137987c1.0. Via: SIP/2.0/AUTO lr2l9s72ehhc.invalid;rport=61353;received=CLIENT;branch=z9hG4bK5927151. To: <sip:4777519304@DOMAIN>;tag=91334f57. From: <sip:4777519304@DOMAIN>;tag=9qhqhrnj3s. Call-ID: j5h7830ivr5dfc2mn5sov1. CSeq: 4 REGISTER. Content-Length: 0. I believe that this error message is caused by ';transport=ws' in the Contact header. I'm not allowed to modify this header. In the backend database, I found that some other clients have ';transport=UDP' in their path headers, so I tried to add that. (Why can I not add parameters in path module without adding username?) I still got the same error. How do I best proceed? For your information: We have outsourced the development of the WebRTC client, so we are able to change it. We also have the option of paying the supplier of the backend for development there. With kind regards Pan B. Christensen Developer Phonect AS Brugata 19, PB 9156 Grønland, N-0133 Oslo, Norway E-mail: pan.christensen(a)phonect.no<mailto:pan.christensen@phonect.no> Mobile: +47 41 88 88 00 [cid:image007.png@01D3A0E8.376921D0] <http://www.phonect.no/> [facebook_2]<https://www.facebook.com/phonectno> [LinkedIn_logo_initials (1)] <https://www.linkedin.com/company/44983?trk=tyah&trkInfo=clickedVertical%3Ac…>

6 years

"No Answer" time out after number of rings

by KamDev Essa

Does Kamailio have any way to control how long it tries to connect per call? KD

6 years

Is Media Release (signaling only) possible in Kamailio?

by KamDev Essa

What cfg files changes do I need to make to get Kamailio to be a signally only server, yet manipulate the SDP part of the INVITE message to allow remote parties to send media to each other? In Freeswitch terms "bypassmedia". KD

6 years

Build Kamailio with MariaDB failed.

by Ye, Charlie (NSB - CN/Qingdao)

Hi all, Does everyone use MariaDB in Kamailio? When I built Kamailio with MariaDB, the below error happened. Would anyone give a help? Thanks in advance! [root@clcmos /usr/local/src/kamailio-devel/kamailio]$ # make all make -C src/ all make[1]: Entering directory `/usr/local/src/kamailio-devel/kamailio/src' make[3]: `libsrdb2.so.1.0' is up to date. make[3]: `libsrdb1.so.1.0' is up to date. LD (gcc) [M db_mysql.so] db_mysql.so /usr/bin/ld: cannot find -lmariadb collect2: error: ld returned 1 exit status make[2]: *** [db_mysql.so] Error 1 make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/local/src/kamailio-devel/kamailio/src' make: *** [all] Error 2 Thanks and Best Regards, Charlie

6 years

Parse presece xml and put stats into asterisk database

by Aqs Younas

Greetings list, I need to put presence info of a device into asterisk database through sqlops module. Currently, i opt to use a lua script to parse xml body of PUBLISH packet issued by pua_dialoginfo and insert trying/ringing stats into asterisk database. Is there any better way to accomplish this? Best Regards, Aqs Younas

6 years

P-CSCF error: pv_parse_spec2(): error searching pvar "uac_req"

by John M

Hello, I upgraded Kamailio 4.4 to 5.2.0-dev5 and when I tried to start the service(p-cscf), I am getting below error. However I am able to start the service with 4.4 p-cscf example cfg. Also after upgrading I-CSCF and S-CSCF nodes are running without any issue. Any help would be greatly appreciated! [root@8bd3221bd31e kamailio]# kamailio -f /usr/local/src/kamailio-5.1/kamailio/misc/examples/ims/pcscf/kamailio.cfg loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(2464) ERROR: <core> [core/pvapi.c:903]: pv_parse_spec2(): error searching pvar "uac_req" 0(2464) ERROR: <core> [core/pvapi.c:1107]: pv_parse_spec2(): wrong char [m/109] in [$uac_req(method)] at [9 (5)] 0(2464) CRITICAL: <core> [core/cfg.y:3489]: yyerror_at(): parse error in config file /usr/local/src/kamailio-5.1/kamailio/misc/examples/ims/pcscf/kamailio.cfg, line 831, column 3-18: Can't get from cache: $uac_req(method) ERROR: bad config file (1 errors) 0(2464) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Regards, Uts

6 years

Use GStreamer instead of FreeSwitch or Asterisk?

by Chandramouli P

Hello, I am using Kamailio as registration server and FreeSwitch for signalling (RTP packet handling). Can I use GStreamer instead of Freeswitch or Asterisk? Thank you. Regards, CM

6 years

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sr-users May 2018