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17 participants
37528 discussions

Timeout for provisional response other that 100
by David Villasmil 09 Feb '21

09 Feb '21

Hello all, I’m trying to figure out if I can set. Timeout for specific provisional responses. I know I can set timeouts for INV, and restart the timer every time a provisional response is received. But this not what I actually need. What I need is to set a timeout from the 100 Trying to the next 180/183 (or a final response) and set back a 480 if not received. I’m not sure TM would allow me to do that, so maybe using a hash table to store transactions and calculate the time since the INV to the received 180/183. But there’s gotta be a simpler way. Many thanks! David -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Best SIP reply for service activation
by Duarte Rocha 09 Feb '21

09 Feb '21

Hello, I'm currently developing a service on Kamailio where it's possible to configure redirections for a subscriber using a code. In order to implement that client should send an INVITE with a specific code on the Request URI and the redirection is configured. What should be the best way to answer this INVITE? A 200 OK will make the call answered and doesn't seem the best way. A negative reply (4/5/6xx) would work, but it seems strange to have the operation concluded with success and answering with a negative reply. I found "199 - Early Dialog Terminated" to be suitable, but the documentation suggests it should be used upstream. Thanks in advance for any suggestion. Cheers

2 1

Kamailio inbound proxy -> Asterisk
by Mihai 09 Feb '21

09 Feb '21

Hi! Don't know if this is a Kamailio Question, maybe someone had deal with it in the past. I use Kamailio with dispatcher to send the request (registration, invites, all) to my asterisk (inbound) and then the asterisk has the outbound trunks, everything works, but this is little thing in asterisk when entering the "sip show peers" it shows me that all the peers are registered with the kamailio IP, if I set the nat option to no, it will show me the correct IP but breaks other clients. Any idea what to do ?

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How to drop 200 OK in onreply_route ?
by Denys Pozniak 09 Feb '21

09 Feb '21

Hello! In the configuration below Kamailio Proxy creates a transaction for the SIP PUBLISH to get info from the HTTP server in async mode. But before creating a transaction, a synthetic 200 OK is sent, so that I need somehow to drop the real 200 OK from the upstream Presence Server. If drop 200 OK in *reply_route*, tm module starts to retransmit. Those it is necessary that the 200 OK be consumed by the tm module, but does not go further. *if ( !is_method("PUBLISH") ) {* * sl_send_reply("200", "Ok"); t_newtran();* * $http_req(suspend) = 1; http_async_query("$var(url)", "CALLBACK");* *}* *route[CALLBACK] {* * <some header manipulation>* * t_on_reply("PUBLISH_REPLY"); route(RELAY); exit;* *}* *onreply_route[PUBLISH_REPLY] { if ( t_check_status("200") ) { drop; # Does not work!!! }}* Any advice is appreciated -- BR, Denys Pozniak

3 8

Kamailio config file for making Kamailio a Sip Proxy
by Derick Greenlee 08 Feb '21

08 Feb '21

Hello, I had some questions regarding the Kamailio config file. I am trying to make Kamailio just behave as a sip proxy between two sip trunks. Could you send me an example for this possibly?

2 1

authenticated INVITE badly managed when topos is activated
by frédéric Gaisnon 08 Feb '21

08 Feb '21

Hi, If topos is activated and kamailio responds with a 407 "proxy authentication required" to an initial INVITE, INVITE received after with credentials (MD5 response in Proxy-Authorization header) is badly managed. Credentials are never validated and Kamailio responds 407 again. If topoh is used instead topos all works fine and credentials are validated. call flow is: Phone sends INVITE to kamailio SBC kamailio respond 407 "proxy authentication required" because SBC wants to authenticate caller Phone resends INVITE with Proxy-Authorization header with all valid information kamailio responds 407 again instead forwarding INVITE. My code: if (!pv_auth_check("$fd", "$sht(auth_cache=>$var(key))", "0", "1")) { auth_challenge("$fd", “1”); exit; } # user authenticated - remove auth header consume_credentials(); So if topos is used, pv_auth_check always returns false even if the phone sets valid information. If topoh is activated, pv_auth_check always returns true (same phone and same kamailio SBC versioning) I made my tests with kamailio 5.4.3 on centos 7. Regards, Frédéric Gaisnon

1 0

Registered with sip user using user pass
by Gaurav Bmotra 08 Feb '21

08 Feb '21

Hi I m using kamailio 5.3 OS Ubuntu 18 I wan to Register on sip server and then send traffic to to it form kamailio. Like : Client ---(sip peer)-->kamailio------(sip peer)---->gateway Please help me Thank you Regards Gaurav Kumar

1 0

filter_body("application/sdp") on multipart/mixed; boundary with double Content-Length
by Karsten Horsmann 08 Feb '21

08 Feb '21

Hi List, I have some weird multipart/mixed;boundary issues. I use kamailio 5.3.6 with rtpengine 7.x.x (same with 8.x.x). If i try to remove a multipart/mixed and leave only the sdp with filter_body() that works except one thing. The sender used IMHO an non RFC compliant version of rfc5621 and wrote the Content-Length Header twice (one in the header, one in the sdp). After manipulating the SDP filter_body() lets Content-Length in the sdp (i guess). And rtpengine is not happy with the sdp after that. rtpengine [rtpengine.c:2588]: rtpp_function_call(): proxy replied with error: Failed to parse SDP Any ideas how to solve this? Kind regards Karsten Horsmann https://tools.ietf.org/html/rfc5621#section-3.1 route[REMOVE_MULTIPART] { if (has_body("multipart/mixed")) { if ($sel(cfg_get.remove.multipart) == 1) { if (filter_body("application/sdp")) { remove_hf("Content-Type"); append_hf("Content-Type: application/sdp\r\n"); msg_apply_changes(); xlog("L_WARN", "[$cfg(route)] Body part multipart/mixed with application/sdp found\n"); } else { xlog("L_ERROR", "[$cfg(route)] Body part multipart/mixed without application/sdp found\n"); } } else { xlog("L_INFO", "[$cfg(route)] Body part multipart/mixed found - manipulation disabled\n"); } } } ------------------- INVITE sip:+49XXXXXXXX@172.20.120.57:5060 SIP/2.0 Via: SIP/2.0/UDP 172.20.120.51:5061;branch=z9hG4bK-6589-1-0 From: sipp <sip:sipp@172.20.120.51:5061>;tag=6589SIPpTag001 To: sut <sip:+49XXXXXXXXX@172.20.120.57:5060> Call-ID: 1-6589(a)172.20.120.51 CSeq: 1 INVITE Contact: sip:sipp@172.20.120.51:5061 Max-Forwards: 70 Subject: Performance Test Content-Type: multipart/mixed;boundary=sonus-content-delim Content-Length: 1185 --sonus-content-delim Content-Type: application/sdp Content-Length: 305 v=0 o=Sonus_UAC 895311 862130 IN IP4 8.8.8.8 s=SIP Media Capabilities c=IN IP4 8.8.8.8 t=0 0 m=audio 26088 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=sendrecv a=rtcp:26089 a=ptime:20 --sonus-content-delim Content-Type: application/pidf+xml <?xml version="1.0" encoding="UTF-8"?><presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:gp="urn:ietf:params:xml:ns:pidf:geopriv10" xmlns:cl="urn:ietf:params:xml:ns:pidf:geopriv10:civicLoc" mlns:btd="http://btd.orange-business.com" entity=" pres:geotarget@btip.orange-business.com"><tuple id="XXXX"><status><gp:geopriv><gp:location-info><cl:civicAddress><cl:country>DE</cl:countr <cl:A1>Some State</cl:A1><cl:A3>Somewhere</cl:A3><cl:A4>Somewhere</cl:A4><cl:A6>Somestreet</cl:A6><cl:YYY>1</cl:YYY><cl:PC>9999</cl:PC><cl:EMERGENCYZONE>XXXXXXX</cl:EMERG CYZONE></cl:civicAddress></gp:location-info><gp:usage-rules></gp:usage-rules></gp:geopriv></status></tuple></presence> --sonus-content-delim-- --------------------- INVITE sip:+49XXXXXXXX@foo SIP/2.0 Via: SIP/2.0/TCP 212.XX.XX.XX;branch=z9hG4bKc228.6b8fdf20d79998d1d355feb2dcf5ba63.0 From: sipp <sip:sipp@foo>;tag=6589SIPpTag001 To: sut <sip:+49XXXXXXXX@foo> Call-ID: 1-6589(a)172.20.120.51 CSeq: 1 INVITE Max-Forwards: 69 Content-Length: 347 Content-Type: application/sdp P-Asserted-Identity: <sip:+49XXXXXXXX@212.XX.XX.XX> Contact: <sip:btpsh-60211bbe-674b-1@212.XX.XX.XX;transport=tcp> Content-Length: 305 v=0 o=Sonus_UAC 895311 862130 IN IP4 8.8.8.8 s=SIP Media Capabilities c=IN IP4 8.8.8.8 t=0 0 m=audio 26088 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=sendrecv a=rtcp:26089 a=ptime:20 -- Mit freundlichen Grüßen *Karsten Horsmann*

1 0

Retransmission of 407
by Marat Gareev 08 Feb '21

08 Feb '21

Hello, UAC sends INVITE to UAS, authorization procedure is required, therefore UAS responds with an 407 error code. At this point, the client does not respond with ACK, but the server does not send retransmits. Is this the expected behavior? * UAC -- SIPp, UAS -- Kamailio. Can the Kamailio be configured to send 407 code retransmissions? Thanks!

3 4

TO Header corrupted in RE-INVITE
by alirsm 07 Feb '21

07 Feb '21

Hi, Call to a number forwarded to a FAX by using two Kamailio proxies. (INVITE) --> KamailioA --> GW1 --> KamailioB --> GW2 --> FAX FAX --> (RE-INVITE) --> GW2 --> KamailioB --> (TO Header being corrupted) KamailioB is corrupting TO header when receiving RE-INVITE and causing the call to fail. 1) Only one "uac_replace_from()" is being used in each Kamailio. 2) GW1 is adding +1 (using E164 format) to URI and FROM headers. I was wondering if any of the above might cause the TO header corruption. Any help would be appreciated. AL

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