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37519 discussions

Trouble with 302 Redirect
by Jim Leahy 27 Feb '20

27 Feb '20

Hi All, I'm having some issues trying to properly send a '302 Moved Temporarily' reply to the calling UAC (UAC1) when the remote UAC (UAC2) returns a 486 status (Do Not Disturb). The goal is to redirect UAC1 to a voicemail server when UAC2 is in DND. Here are the pertinent parts of my config: route { # drop some things right away route(DROP); # perform sanity check route(SANITYCHECK); # handle registrations if(is_method("REGISTER")) { route(REGISTRATION); exit; } # handle invites if(is_method("INVITE")) { route("INVITE"); exit; } route(RELAY); } route[INVITE] { # add this proxy to the record-route so it stays in the loop record_route(); # if we can find this user in the location DB... if(lookup("location")) { # relay the invite to the new dURI t_on_failure("FOURDIGITFAIL"); t_on_reply("FOURDIGITREPLY"); route(RELAY); # go back to the main route return; } onreply_route[FOURDIGITREPLY] { if(t_check_status("4[0-9]{2}")) { t_reply("302", "Moved Temporarily"); } } I'm having issues with the t_reply in the onreply_route. It successfully sends the reply, but I can't seem to set the Contact header properly. I'm trying to set it to the location of the user's mailbox on the voicemail sever (ie sip:1000@vmpbx.domain.com), but nothing I try works. I've tried: remove_hf("Contact"); insert_hf("Contact: sip:1000@vmpbx.domain.com"); t_reply("302", "Moved Temporarily"); But as I understand it, that just changes the header in the reply from UAC2, not the reply I'm sending to UAC1 Then I tried: append_to_reply("Contact: sip:1000@vmpbx.domain.com" ); t_reply("302", "Moved Temporarily"); But I can't use append_to_reply in an onreply block. How can I change the Contact header in the reply that's sent? Am I just going about this the wrong way? Any help would be appreciated. Thanks

5 4

Rtpproxy not work in 2 kamailio + rtpproxy setup
by Voip support 27 Feb '20

27 Feb '20

Dear Community, I have a problem with rtpproxy not changing the SDP IP in connection information in following scenario: SIP1 -> kamailio1 + rtpproxy1 -> kamailio2 + rtpproxy2 -> SIP2 The first kamailio put SDP IP correctly (from rtpproxy) then the second kamailio does not change SDP connection information. I think its related to SDP body which has: a=nortpproxy:yes line added by kamailio1. So i tried to remove the a=nortpproxy:yes line from SDP before calling rtpproxy_manage. But it looks like no difference in working - i think the SDP sent to rtpproxy is still having the line even when i removed it. The issue is not happening when i remove kamailio1 + rtpproxy1. In that case my IP is put correctly in SDP in INVITE and in 200 OK. In case of two kamailio with rtpproxy (kamailio1 and kamailio2) in signalling path i am getting: rtpproxy [rtpproxy.c:2586]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy for replies (SIP2 IP address in 200OK SDP is present instead of kamailio2 IP address). For request INVITE i am not getting the error but the IP in SDP doesn't change. kamailio2 config is now like below (but the attempt to remove a=nortpproxy:yes, did not work for me) # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } # if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; -- commented this out because sometimes rtpproxy was not used and i wanted to use it always # added below code to remove a=nortpproxy:yes from SDP before rtpproxy_manage but its still not working if( has_body("application/sdp") && (1 == sdp_get_line_startswith("$avp(nortpproxy)", "a=nortpproxy" )) ){ sdp_remove_line_by_prefix("a=nortpproxy:yes"); xlog("removed nortpproxy:yes from SDP"); } if(nat_uac_test("8")) { xlog("rtpproxy CO"); rtpproxy_manage("co","MY_PUBLIC_IP_HERE"); } else { xlog("rtpproxy COR"); rtpproxy_manage("cor"," MY_PUBLIC_IP_HERE "); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } if(isbflagset(FLB_NATB)) { # no connect message in a dialog involving NAT traversal if (is_request()) { if(has_totag()) { set_forward_no_connect(); } } } #!endif return; } Best regards, Tom PS. thanks for any hints to resolve the issue.

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record_route_advertised_address causes null send socket error
by Daniel Greenwald 26 Feb '20

26 Feb '20

I am using record_route_advertised_address to insert a record route header with an SRV record. The header is added correctly and everything appears to be working fine however I get a CRITICAL log error when calling record_route_advertised_address('INTERNAL_SRV'); 2020-02-25T20:14:57.078235+00:00,ip-10-0-16-53: CRITICAL: cbbcfe06-d181-467c-bbf4-78772aa0562f 2 16777053 INVITE:<core> [core/msg_translator.c:514]: lump_check_opt(): null send socket 2020-02-25T20:14:57.078352+00:00,ip-10-0-16-53: CRITICAL: cbbcfe06-d181-467c-bbf4-78772aa0562f 2 16777053 INVITE:<core> [core/msg_translator.c:514]: lump_check_opt(): null send socket Any ideas? I'm considering using insert_hf() to manually add it and bypass whatever check is failing. Thoughts? Thanks

3 7

Access to CANCEL messages sent during parallel forking
by Marco Capetta 26 Feb '20

26 Feb '20

Hi All, I have a question regarding call forking and how to access failed branches: the ones for which kamailio sends out the CANCEL because on another one a 200OK was received. This is the scenario: - A calls B - 3 devices are registered on B, so a parallel forking is done to B1, B2 and B3 - B1, B2, and B3 all reply with 180 Ringing - B1 replies with 200OK - Kamailio CANCELs the branches with B2 and B3 What I need is access to each single CANCEL that Kamailio sends out to B2 and B3. In particular I need to know the branch-id and Via branch of all the cancelled branches. Additionally it would be perfect to have them inside the TM module, in order to have access also to internal AVPs. I tried with dedicated failure routes and also with per-branch failure routes, but I was never able to access to those values. I also didn't found a specific event-route that can help me solve this problem. Does anyone have an idea? Kamailio version 5.2.5 Thanks Regards Marco

3 5

Siremis v5.3.0 Released
by Daniel-Constantin Mierla 26 Feb '20

26 Feb '20

Siremis v5.3.0 is out – the open source web management interface for Kamailio SIP Server. This version comes with major changes and many new features, among them the removal of legacy flash charts (now uses pure JS), make the code compatible with PHP 7.x and web views to manage many new modules added up to Kamailio v5.3.x, including the replacement of MI control interface with JSONRPC. This is the first full release in the 5.x series of Siremis. Step by step installation tutorial, screenshots and changelog are available on the web at: * https://kb.asipto.com/siremis:install53x:main * https://kb.asipto.com/siremis:install32x:screenshots * https://github.com/asipto/siremis/blob/v5.3.0/ChangeLog The deployment of Siremis can be also done using Docker, see the guidelines at: * https://github.com/asipto/siremis/tree/master/misc/docker Siremis project is hosted on Github at: * https://github.com/asipto/siremis/ The web site for the project is at: * https://www.siremis.org Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 9-11, 2020, Berlin - www.asipto.com Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

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Understanding set_rtp_engine with two sets
by Anthony Alba 25 Feb '20

25 Feb '20

There is a use case mentioned "This is useful if you have a set of RTP proxies that the caller must use, and another distinct set of RTP proxies that the callee must use. This is supported by all rtpengine commands except rtpengine_manage(). " How do you actually implement this - is it something like this? Say caller supposed to target Set 1and callee supposed to target Set 2: # offer/request route set_rtpengine_set(1, 2) rtpengine_offer() # cannot use rtpengine_manage() # callee sees set 2 in SDP, right? # reply_route set_rtpengine_set(2, 1) # reverse the order? Is this correct?? # OR set_rtpengine_set(1, 2) # keep order, module does autoreversal magic # caller sees Set 1 in SDP My question: in the reply route do you keep the same order (module is clever enough to autoreverse so caller sees Set 1) OR do you explicitly reverse the order and THAT means caller sees Set 1. Cheer Anthony

3 2

kamcli postgres support
by Daniel-Constantin Mierla 25 Feb '20

25 Feb '20

Hello, anyone in the community with Python knowledge using Postgres that can spend a bit of time to add support in kamcli? * https://github.com/kamailio/kamcli It should be mainly about the content of the code corresponding to db subcommand: * https://github.com/kamailio/kamcli/blob/master/kamcli/commands/cmd_db.py The target for it is to replace kamdbctl and has some command line variants for couple of its functions, now being implemented for MySQL. The rest of the code relies on sqlalchemy, which should have transparent support for both mysql and postgress (even more). Testing the other commands connecting to database with postgres would be good, though. With more and more diversity in the default shell by main Linux/Unix distros, we hope to be able to replace in the near future the kamctl and kamdbctl tools, which are more like bash scripting (a few issue being reported and fixed lately). Besides that, using python allows better input/parameter checking/validation as well as more options for formatted output. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 9-11, 2020, Berlin - www.asipto.com Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

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Database errors with usrloc db_mode=1 and dmq_usrloc
by Joel Serrano 25 Feb '20

25 Feb '20

Hello, I have a setup with 2 kamailio in active/idle, locations are shared via dmq_usrloc, but database is also used as a cache where other systems can access and query "real-time" data. The config we have is: # ----- usrloc params ----- modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 1) modparam("usrloc", "db_load", 0) modparam("usrloc", "db_insert_update", 1) modparam("usrloc", "db_timer_clean", 1) modparam("usrloc", "server_id_filter", 1) modparam("usrloc", "timer_procs", 2) modparam("usrloc", "timer_interval", 60) # ----- dmq_usrloc params ----- modparam("dmq_usrloc", "enable", 1) modparam("dmq_usrloc", "sync", 1) modparam("dmq_usrloc", "batch_msg_contacts", 50) # 50 contacts / message modparam("dmq_usrloc", "batch_size", 10000) # 10000 contacts / batch modparam("dmq_usrloc", "batch_usleep", 500000) # one batch every 500ms With this setup, we assume: 1- almost real-time updated location info in the database. 2- as long as one of the two Kam is up, location information will not be lost (as they will sync with each other via DMQ) 3- if both servers go down, locations are lost (as they will not reload from database on startup). Our tests work well with the above config, with one exception, on the idle node, we see a lot of (only happens on the node receiving location via dmq): Feb 21 00:11:25 csbc02 csbc[13916]: ERROR: <core> [db.c:481]: db_use_table(): invalid parameter value Feb 21 00:11:25 csbc02 csbc[13916]: ERROR: usrloc [ucontact.c:1136]: db_update_ucontact_ruid(): sql use_table failed Feb 21 00:11:25 csbc02 csbc[13916]: ERROR: usrloc [ucontact.c:1657]: update_contact_db(): failed to update database I added debug logs to see if I could understand the reason: Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:478]: usrloc_dmq_handle_msg(): dmq message received from sip:usrloc@10.2.1.181:5050 Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:423]: usrloc_dmq_execute_action(): Received DMQ_UPDATE. Update contact info... Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:75]: add_contact(): aor: 1063362 Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:76]: add_contact(): ci->ruid: uloc-1-5e502cd3-5629-1 Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:77]: add_contact(): aorhash: 928261667 Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:115]: add_contact(): '1063362' Not found in usrloc Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:117]: add_contact(): Insert record Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: usrloc [ul_callback.h:83]: run_ul_callbacks(): contact=0x7fe41ff41f40, callback type 1/15, id 0 entered Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:800]: dmq_ul_cb_contact(): Callback from usrloc with type=1 Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:826]: dmq_ul_cb_contact(): Contact received from DMQ... skip Feb 21 14:38:34 csbc02 csbc[30796]: ERROR: <core> [db.c:481]: db_use_table(): invalid parameter value Feb 21 14:38:34 csbc02 csbc[30796]: ERROR: usrloc [ucontact.c:669]: db_insert_ucontact(): sql use_table failed Feb 21 14:38:34 csbc02 csbc[30796]: ERROR: usrloc [urecord.c:598]: insert_ucontact(): failed to insert in database Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:121]: add_contact(): Insert ucontact Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:124]: add_contact(): Release record Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:126]: add_contact(): Unlock udomain Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: sl [sl.c:305]: send_reply(): reply in stateless mode (sl) Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: <core> [core/msg_translator.c:161]: check_via_address(): (10.2.1.181, 10.2.1.181, 0) Can anyone help me understand what's the problem? I have the server_id setting, and the modparam server_id_filter, so shouldn't the receiving node just ignore any database tasks? Thanks, Joel.

3 12

Using the To header to route a call
by Dovid Bender 24 Feb '20

24 Feb '20

Hi, We are fighting with a vendor and trying to explain to them that call routing should be based on the RURI on the INVITE and *not* the To header. I went through RFC 3261 and while there are many "indicators" and it seems like its screaming that we should be using the RURI, is anyone aware of any specific text (that I may have missed) where it says specifically call routing should be done based on the RURI and *NOT* the to header? TIA. Dovid https://tools.ietf.org/html/rfc3261#page-78 1) 8.1.1.1 Request-URI *The initial Request-URI of the message SHOULD be set to the value of the URI in the To field.* - Note that it says SHOULD but is NOT mandatory. Also this is for the *INITIAL* request. *This may or may not be the ultimate recipient of the request.* - Again this is the To field. The RURI is what we are supposed to follow when we want to know where the call should be delivered to. 2) 8.1.2 Sending the Request -Usage of the URI from the *To and From fields in the original request within subsequent requests is done for backwards compatibility with RFC 2543*, which used the URI for dialog identification. *In this specification, only the tags are used for dialog identification*. It is expected that mandatory reflection of the original To and From URI in mid-dialog requests will be deprecated in a subsequent revision of this specification. - While this is for subsequent requests (and not an original one) you see that the only point of the To field is to identify the dialog. 3) 8.2.2.1 To and Request-URI The To header field identifies the *original recipient of the request* designated by the user identified in the From field. - Note that the TO field identifies the ORIGINAL recipient, this **CAN** change. This is basically telling you that you *can't* rely on the To header for routing. *However, the Request-URI identifies the UAS that is to process the request*. - As you can see the Request-RU and *NOT* the TO header tells you where the request should go. 5) 12.2.1.1 Generating the Request a. The UAC uses the *remote target* and route set to build the *Request-URI and Route header field of the request*. - This is self explanatory. it shows that the RURI is using for routing and *NOT* the to header. b. The procedures in Section 8.1.2 will normally result in the request being *sent to the address* indicated by the topmost Route header field value or the *Request-URI* if no Route header field is present. Subject to certain restrictions, they allow the request to be sent to an alternate address (such as a default outbound proxy not represented in the route set). Again self explanatory. 6) 16.5 Determining Request Targets Each target in the set is *represented as a URI*. - This clearly shows that how we route is based on the RURI and *NOT* the to header. 7) 26.5 Privacy Thus it MAY be desirable for privacy reasons to create a To header field that differs from the Request-URI. - Again this shows that the to may not match up the RURI. Think of the To header as the contact in someones phone and the RURI as where we are calling. The to header may be bob(a)1.1.1.1 but the URI is where we are actually calling which is 12346(a)1.1.1.1 or bob_smith(a)1.1.1.1.1

3 2

Kamailio presence when account not equal extension
by Zhan Bazarov 24 Feb '20

24 Feb '20

Hello, I'm using presence and pua modules, it works perfectly when username is equal to extension... But now I have case when username is not equal to the extension.. Like: extension: 1111 username: aythnbghhgt How I can implement this? Any ideas? -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

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