- sr-users - kamailio.org

NatHelper's Keep-alive timeout behind proxy
by Asgaroth 17 May '18

17 May '18

Hi All, We are looking to enable nathelpers keepalive timeout parameter where it will remove an aor's contact if it doesn't receive a reply from the ua. The problem we have is that our registrar's are seperated from our proxies, so this means nathelper is sending options messages via the proxy to the ua. In this case, if the us has lost connectivity, the kamailio proxy sends a 408 Request Timeout back to the registrar and the nathelper module sees that as a response to the initial options message it sends, so it never actually removes the contact from the location table. Is there a way I can stop the proxy from sending any responses for these specific nat ping/options messages, or, is there a way I can tell the nathelper modules that a 408 response should be treated as the ua being down and then after the timeout interval remove the contact from the location table? I've had a read of the nathelper, tm and sl modules but I didnt see anything that jumps out at me that I could achieve one or the other possible solutions I mentioned above. Does anyone have any suggestions or pointers on how we can achieve this with the registrars being seperate from the edge proxies. Any pointrs/tips etc would be greatly appreciated. Thanks

2 2

Updated NAT traversal article
by Alex Balashov 17 May '18

17 May '18

Hi, I have updated this article with some other topics: http://blog.csrpswitch.com/server-side-nat-traversal-with-kamailio-the-defi… "The definitive guide" is of course an ambitious and moving target, so there may be more evolution in the future. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

5 7

t_load_contacts() and t_next_contacts() guts
by KamDev Essa 16 May '18

16 May '18

I am tackling Hunting and see that there are two functions recommended to accomplish this. Before I jump into using them I wanted to know a bit about each regard to the call flow. >From where does t_load_contacts() load contacts and how does it order then in case of a serial fork flow, Is there a generic way of doing it say with a loop in the script. Docs on these two functions are a bit cryptic for me H

2 7

Contact Header
by KamDev Essa 16 May '18

16 May '18

My scenario is UA <====> Kamailio <====> Freeswitch UA calls a number. Kam delivers to FS and FS plays a wav file. My problem is that Kam relays the 200 OK from FS back to UA without change and UA ACKs back to FS because the FS IP is in the Contact Header. - Why does Kam not change the contact header since the ACK should come back to Kam - If this is normal what do I need to do to change the contact header for the 200K to the UA to make this work. KD

3 4

Whats is wrong in this call
by Sergey Safarov 15 May '18

15 May '18

Hello I have a call with not passed ACK by Kamailio. http://prntscr.com/jhtobo Same behaviour i looked before when SIP endpoint contact string is wrong In this call look as contact string correct is correct (packets 7 and 8) and i not understand why ACK (packet 9) not passed to callee. Could you help me understand this. PCAP file is create using homer tracer and TCP transport packets is stored as UDP. Sergey

3 4

Last negative 5XX reply not getting relayed
by Sergiu Pojoga 15 May '18

15 May '18

Not sure if it's a default Kamailio behavior or something I'm doing wrong... When receiving a negative reply for an INVITE (more like in the 5xx range), proxy doesn't relay the message to the originator but instead issues it's own "500 Service Unavailable". That doesn't happen, from observations so far, to "603 Declined" or "486 Busy Here" or "404 Not Found" or "200 OK" replies. *For example, here's the negative reply proxy receives from the callee:* 2018/05/14 10:52:37.716627 65.XX.XX.166:5060 -> 65.XX.XX.167:5060 SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 65.xx.xx.167;branch=z9hG4bK9223.00b4f3143d245a895c4bb6a107add258.0;received=65.xx.xx.167;rport=5060 Via: SIP/2.0/UDP 205.xx.xx.221;received=205.xx.xx.221;rport=5060;branch=z9hG4bKQ4XpXga0vgZ8B From: "514XXXXXXX" <sip: 514XXXXXXX @205.xx.xx.221>;tag=tXj0NHrje9jNS To: <sip:450XXXXXXX@65.xx.xx.167>;tag=as18e452b7 Call-ID: 48eac296-d229-1236-a685-005056a149a4 CSeq: 122818714 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas X-Asterisk-HangupCause: Subscriber absent X-Asterisk-HangupCauseCode: 20 Content-Length: 0 *Here's what the originator gets in the final message:* 2018/05/14 10:52:37.718143 65.xx.xx.167:5060 -> 205.xx.xx.221:5060 SIP/2.0 500 Service Unavailable Via: SIP/2.0/UDP 205.xx.xx.221;rport=5060;branch=z9hG4bKQ4XpXga0vgZ8B;received=205.xx.xx.221 From: "514XXXXXXX" <sip:514XXXXXXX@205.xx.xx.221>;tag=tXj0NHrje9jNS To: <sip:450XXXXXXX@65.xx.xx.167>;tag=31a2c69b5d434ef4a60796e26f568bf0-9343 Call-ID: 48eac296-d229-1236-a685-005056a149a4 CSeq: 122818714 INVITE Server: Kamailio 5.1 Content-Length: 0 I'd like any reply to be relayed back as is, unless I do some t_check_status() => t_reply() for specific reasons (which I have none at this point). Thanks Sergiu

3 5

ACC_CDRS table with src and dst data are inverted
by Emanuele Gambaro 15 May '18

15 May '18

Hi to all, I have some troubles with acc_db module and acc_cdrs table. First scenario: Alice calls Bob and after a while, when a call is established, Alice hangs up, sending a BYE. The acc_cdrs table is filled up with the right data: * src_user, src_domain, src_ip are from Alice account (the caller) * dst_user etc etc are from Bob account (the callee) In this scenario all works as expected: the src data are from who starts the call, and the dst data are from who receives the call. In the following scenario src and dst data are inverted. Alice calls Bob and after a while, when a call is established, Bob hangs up, sending a BYE. The acc_cdrs table now is: * src_user, src_domain, src_ip are from Bob account (the callee that sends the BYE message) * dst_user etc etc are from Alice account (the caller that receives the BYE) This is the intended behavior or I’m missing something? How I can record the CDR data with the right CALLER and CALLEE data ? Emanuele

2 1

UAC Module Registration Table
by Greg Adams 15 May '18

15 May '18

Hey there, I am currently using the UAC module to maintain remote registrations. I am just curious if it is possible to register new users immediately rather than waiting for reg_timer_interval? I am not seeing any functionality like that but just wondering if I am missing something or if it would be possible to add in. For example an RPC command that takes a key and instantly registers that user. Thanks, Greg

2 1

Parallel Forking - 1 Ring on extension to notify that Call Forward is in place
by KamDev Essa 14 May '18

14 May '18

I am trying to simulate a Broadsoft feature that blips the extension with a single ring that basically heralds the user that their extension has CF turned on. I know parallel forking can do that. But how do I set the fr timer on the call to the extension @ just 1 ring (5 seconds) and let the other one the default timeout set in fr_inv_timer. KD

2 1

Tagging a call with custom pseudo variables - cfg 101
by KamDev Essa 14 May '18

14 May '18

Being spoiled by Freeswitch profiles and dialplans, the stock 1 piece Kamailio cfg file is cool @ first to get certain call flows knocked out but quite a maze when you get in deeper. Right now, using the 1 piece stock cfg file, I am faced with the fact that my carrier outbound calls started failing after I enabled VM. Basically WITH_VOICEMAIL in route LOCATION started sending the carrier calls to TOVOICEMAIL and it basically dies there. I can see the route(TOVOICEMAIL) line and can condition it out with an if but I really don't now how to differentiate between a ext to ext call or a outbound call in the cfg file. However given SQL I could. Is there a way for me to place custom pseudo variables in the call right @ the start and use them in the script as needed. For example I could mark the call as ext to ext or carrier bound right @ the start and dodge VM in the LOCATION route. KD

2 5

Question about using Kamailio and Asterisk and flow of an "INVITE"
by Wilkins, Steve 13 May '18

13 May '18

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.

3 6

"Sleeping Giant - No Snore" Kam?
by KamDev Essa 11 May '18

11 May '18

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

3 6

Understanding port changing
by Social Boh 11 May '18

11 May '18

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

2 1

WebRTC to SIP gateway
by Pan Christensen 11 May '18

11 May '18

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…>

2 2

"No Answer" time out after number of rings
by KamDev Essa 11 May '18

11 May '18

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

2 5

Is Media Release (signaling only) possible in Kamailio?
by KamDev Essa 10 May '18

10 May '18

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

2 14

Build Kamailio with MariaDB failed.
by Ye, Charlie (NSB - CN/Qingdao) 10 May '18

10 May '18

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

4 6

Parse presece xml and put stats into asterisk database
by Aqs Younas 09 May '18

09 May '18

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

1 0

P-CSCF error: pv_parse_spec2(): error searching pvar "uac_req"
by John M 09 May '18

09 May '18

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

2 1

Use GStreamer instead of FreeSwitch or Asterisk?
by Chandramouli P 09 May '18

09 May '18

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

2 1

Connecting UAs behind Firewall/CgNat with Kamailio without using a Stun Server
by Pinter, Gerd. 09 May '18

09 May '18

Hi there, unfortunately I am personally not a complete IT geek, I am more in to Pro Audio. Anyway, I have to maintain a SIP Server for high quality Audio transmission with special UA's. Also we use Software UA's running as Apps (Luci Live) on mobile devices such as I Phones and I pads which are by nature situated in mobile networks. Due to "lack of IP address issues" most providers use private ranges separated by CGNAT from the real Internet. This causes a lot of trouble. In some cases stun server entries on the clients helps, in other cases don't! But these days I read in announcement of a german company (Mayah Communications), which is in the ProAudio biz since decades, that they offer a SIP Service where no stun is recommended whether the clients are behind NAT or not. After I acknowledged that IT business is not witchcraft (almost :-) ) , I thought that there might exist a solution for my Kamailio SIP server to make it work like the "Mayah" Sip Server. What settings have to be done that Kamailio uses IP Header Information for SIP signalling instead of the content of SIP packages themselves. Without stun you find the Local Address in the SIP packages, and with Stun the external Address but mostly with the wrong portnumber, but in the IP header you'll find the correct IP:port. I already read a lot of the Kamailio documentation but due to the fact that I am not too deep in IT I am not sure if I got it all. If someone could give me a useful hint on that, that would make my day. Hope this description is unterstandable. Nice Regards Gerhard Pinter

4 8

Kamailio + RTPEngine + RTPEngine Recording daemon forward-to option
by Balázs Radák 08 May '18

08 May '18

Dear all, I would need some help setting up Kamailio with RTP engine. I'm trying to make use of RTPEngine recording daemon's forward-to option, to have the calls streams sent to 3rd party application. Have someone managed to have this forward-to option work? If I understood well, when RTPEngine's forward-to option is turned on, the call metainfo + streams can be sent to a socket server? Thanks in advance! Regards, Balazs

2 1

Copy $xavp
by Duarte Rocha 08 May '18

08 May '18

Greetings, I'm getting a query result in a $xavp. I'm trying to copy the entire structure to a new $xavp like this : $xavp(Valid_Acd_Results) = $xavp(ACD_Query_Result[$var(iterator)]); However, i can't retrieve the value with : xnotice(" Mode = $xavp(Valid_Acd_Results=>TBK_AcdMode)"); since it returns NULL. Am i doing something wrong? Thanks

1 0

Record-route and proxy_challenge
by Евгений Голей 08 May '18

08 May '18

Hi! Tell me why when using the save ('') function in the response generated by the server, there are the Record-route headers that were in the request. And when using the following functions: - proxy_challenge (), - www_challenge (), - send_reply (), - sl_send_reply there is no Record-route header. How can you add them? -- Evgeniy

2 2

Garbage in to field
by Kjeld Flarup 08 May '18

08 May '18

I'm having problems with my To field. I'm doing an async_route, and afterwards sends the call with auth to a SIP server. My problem is that in the first invite I see the expected To field sip:004520202020@isp.com But when the Auto is send the To field is changed to the, To field for the PBX sip:1234@pbx.local I tried to fix this with uac_replace_to("$ru"), but that ends up in this To field: sip:004520202020@isp.comsip:004520202020@isp.com It seems like uac_replace_to sometimes appends rather that replaces. If I run it twice, then it for sure appends. Is this expected behaviour. -- -------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

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