- sr-users - kamailio.org

Kamailio World 2020 - Call For Presentations
by Daniel-Constantin Mierla 02 Dec '19

02 Dec '19

Hello, I would like to announce that Call for Presentations at Kamailio World 2020 is now open. You can submit your proposal or see more details at: - https://www.kamailioworld.com/k08/call-for-speakers/ The 8th edition of the event takes place again in Berlin, Germany, during April 27-29, 2020. Expect over 150 participants, developers and community members as well as representatives of other popular open source VoIP projects such as Asterisk or FreeSwitch. Looking forward to meeting many of you there! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

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Possible conflict between t_newtran and setflag for ACC
by Patrick Wakano 01 Dec '19

01 Dec '19

Hello list, Hope you all doing well! I am using the ACC module and using the setflag() function as done in several examples. It works fine. However, I've added the t_newtran() function almost in the begging of the INVITE handler to help the retransmission detection and after that I noticed the ACC was not saving anything in DB. So after debugging I discovered that if I call the t_newtran() before setting the ACC flags, the module will not save the calls in DB, but if I call it after setting the ACC flags, it works.... So my question is, is this a bug or it is a expected side effect so when one is using t_newtran you must be careful and set all your transaction flags before? (ACC are the only transaction flags I am using so can't tell if other modules have the same problem) This is happening in Kamailio 5.2.2. Thank you! Kind regards, Patrick Wakano

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tcp limit, too many open files
by David Villasmil 01 Dec '19

01 Dec '19

Hello guys, I'm facing an issue where, if i start kamailio as root, all works properly, but if i start it as kamailio, i get hundreds of: handle_new_connect(): error while accepting connection(24): Too many open files My limits is like: kamailio soft core unlimited kamailio soft data unlimited kamailio soft fsize unlimited kamailio soft memlock unlimited kamailio soft nofile unlimited kamailio soft rss unlimited kamailio hard stack 240 kamailio soft cpu unlimited kamailio soft nproc unlimited kamailio soft as unlimited kamailio soft priority -11 kamailio soft locks unlimited kamailio soft sigpending unlimited kamailio soft msgqueue unlimited kamailio soft nice -11 kamailio soft nofile 65536 kamailio hard nofile 65536 root soft core unlimited root soft data unlimited root soft fsize unlimited root soft memlock unlimited root soft nofile unlimited root soft rss unlimited root hard stack 240 root soft cpu unlimited root soft nproc unlimited root soft as unlimited root soft priority -11 root soft locks unlimited root soft sigpending unlimited root soft msgqueue unlimited root soft nice -11 root soft nofile 65536 root hard nofile 65536 I can't figure out why this is happening... Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Kamailio as SBC for Microsoft Teams
by Henning Westerholt 29 Nov '19

29 Nov '19

Hello, I have created a "How-To" blog post on using Kamailio as "session border controller" for Microsoft Teams Direct Routing: https://skalatan.de/de/blog/kamailio-sbc-teams You can this way use all the existing possibilities that Kamailio provide to interact with MS Teams as well. Best regards, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

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Adding "q" value for the main branch
by Maxim M 28 Nov '19

28 Nov '19

Maxim Malygin <maxim.malygin(a)gmail.com> 17:08 (5 минут назад) кому: sr-users Hello, I need to form list of Contact headers with "q" parameters for "302 Moved Temporary". I do something like this: ... # $xavp(contacts) - contains list of contact URIs (uri) and Q values (q) $var(i) = 0; $var(num) = $cnt($xavp(contacts)); while($var(i) < $var(num)) { if($var(i) == 0) { # How to add q value to this branch? $ru = $xavp(contacts[$var(i)]=>uri); } else { append_branch("$xavp(contacts[$var(i)]=>uri)", "$xavp(contacts[$var(i)]=>q)"); } } send_reply("302", "Moved Temporarily"); ... Is it possible to set "q" value for the main branch? For additional branches I do it via append_branch("uri", q_value) but I didn't find any way to do it for the main branch pointed via $ru/$du. The pseudo-variable "$branch" also doesn't allow me to access the main branch. Thanks, Maxim

1 0

Get totally rid of H.264 SDP
by Kjeld Flarup 28 Nov '19

28 Nov '19

Hi I have a door telephone, with GSM fallback. Since telephone operators are going away from SS7, more and more traffic goes SIP. Thus we now have some numbers where we can see that doing an invite with H.264 causes them to fail. I have tried to remove coded 96 with sdp_remove_codecs_by_id, but it still leaves most of the video SDP in the invite. SDP received by kamailio Media Description, name and address (m): video 6100 RTP/AVP 96 Media Attribute (a): rtcp:6101 IN IP4 192.168.237.239 Media Attribute (a): rtpmap:96 H264/90000 Media Attribute (a): fmtp:96 profile-level-id=42000d;packetization-mode=1 Media Attribute (a): sendrecv After sdp_remove_codecs_by_id I send this. But without "96" Media Description, name and address (m): video 6534 RTP/AVP Media Attribute (a): sendrecv Media Attribute (a): rtcp:6535 Media Attribute (a): ice-ufrag:35ZngQdL Media Attribute (a): ice-pwd:DG6kUbcYwBFKNS6YzCG27wRlG4 Media Attribute (a): candidate:r9KO0wNPMTGktklS 1 UDP 2130706431 192.168.237.15 6534 typ host Media Attribute (a): candidate:r9KO0wNPMTGktklS 2 UDP 2130706430 192.168.237.15 6535 typ host My problem is that the door telephone does not do an ACK when it gets the 200 OK My suspicion is that it does not understand the video response Media Description, name and address (m): video 0 RTP/AVP 0 Media Attribute (a): ice-ufrag:35ZngQdL Media Attribute (a): ice-pwd:DG6kUbcYwBFKNS6YzCG27wRlG4 Media Attribute (a): sendrecv Is there a way to remove the whole video part and not just the coded id? It is not an option to disable video in the door telephone, as we need the video. It is just in the fallback is must be removed. -- --------------------- 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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Kamailio does not detect message
by Adesh Pandey 28 Nov '19

28 Nov '19

Hi, I have below scenario: User registers on Kamailio server, calls to a number which is forwarded to asterisk through dispatcher, asterisk server answers and sends a text message to the caller using sendText but first message is delivered to client with message "501 Not Implemented" but in the subsequent calls asterisk shows that the text was delivered but kamailio did not detect the message. Guys, any help would be appreciated, I am using the below mentioned config. https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg Kamailio version: 5.2

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Kamailio developers meeting - follow up remarks
by Daniel-Constantin Mierla 27 Nov '19

27 Nov '19

Hello, last week we had the 2nd annual Kamailio Developers Meeting hosted by Sipgate in Dusseldorf, Germany: * https://www.kamailio.org/w/developers-meeting/ 16 people were at the event in various roles. Giacomo Vacca published on his blog a good summary of what happened there: * https://www.giacomovacca.com/2019/11/my-notes-on-kamailio-developer-meeting… This year we had a lot of discussions, as well as work done on multiple planes, not only Kamailio code. So I am trying to list here some of the conclusions for future development, the technical aspects of the meeting, so everyone is aware and can provide feedback. 1) Effective work was done on: * kamailio code * kamailio rpm packaging * kamailio tools (kamctl) * kamailio release process * kamailio project keys (to be used to sign the packages) 2) Documentation 2.a) Wiki * it was somehow a rough consensus to move the wiki content to github, along with changing the format from dokuwiki markdown to the standard/github markdown. This should enable people to make pull requests so developers or community members can review and aprove new content. It also makes it easier to contribute using existing github account, now the kamailio.org/wiki is requiring to make a dedicated account, which many prefer not to do it. * the presentation can be done either by using mkdocs to generate html files hosted on kamailio.org or using the github provided wiki portal. 2.b) Docs for variables and transformations * there was a proposal to move them in the documentation the modules that export them, there are pros and cons, needs more discussions. Now they are in the wiki, so this probably has to resume after deciding on 2.a). 3) Kamailio Modules 3.a) replication (dmq) - several participants discussed about negotiations between nodes to take active role on some cases (e.g., active dialogs) 3.b) api integration - quite some interest in JSON-based API routing, concluding in extending rtjson to cover more use cases 3.c) security - have options to restrict the use of TLS1.3 or newer 4) Kamailio Releases * v5.3.2 was released during the event, allowing to document the process * work to automatize the process is planned, then eventually assing teams for takeing cares of releases from specific branches 5) Kamailio Testing * existing docker-based testing framework should be extended and integrated in CD/CI pipeline 6) Kamailio packages 6.a) rpms * rpm.kamailio.org has been prepared and is expected to take over the opensuse build service for building rpms and hosting them. Expected to provide support for hosting many kamailio versions in the same release series so one can do downgrade to older releases. Also, there is work in progress to provide nightly builds. 6.b) debs * work is planned to offer many kamailio versions in the same release series 7) Kamailio tools * kamctl/kamdbctl should be obsoleted in favor of kamcli, which offers a better framework for input validation and output formatting, as well as better portability, no longer depending on shell interpreter 8) Various discussions * kemi exports from C point of view and how to combine the documentation for modules and their kemi exports * how to make kamaiio friendlier in virtualized environments (ended up in the need of making the use of advertised address a bit more dynamic) * project organizatoric topics - to be approached separately * next events - Fosdem - someone should submit a proposal to present about Kamailio 9) Long term goals We speak here more or less about Kamailio 6.0 ... * change the behaviour of the native config interpreter to be consistent with the other programming languages in terms of handling the response code (change what is now: the evaluation of negative value to false and positive value to true and the hidden return 0 to exit) * make the pool of processes more generic, so they can handle traffic from more sockets (being sip traffic or something else) -- this should make better use of resources, as some sockets might be less busy that others I hope I covered the important topics, if I remember something else, I will reply on this thread. Or maybe other participants can contribute missing topics. Should anyone have comments or suggestions on the above topics, or new ones, let's discuss on sr-users because it impacts the long term use of Kamailio (sr-dev is cc-ed now for notification purposes). Overall, there were 2 very intensive days, however in a friendly and relaxed environment offered by Sipgate. Extremely useful discussions, not only about Kamailio, but about RTC ecosystem. At the evening social network event event, a couple of external people joined, some of them presenting very interesting use cases of Kamailio. Many thanks to all participants that allocated time and resources to come to Dusseldorf, as well as to the companies that covered expenses for participants. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

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Kamailio xavp_contact - Segmentation Fault
by Ilie Soltanici 27 Nov '19

27 Nov '19

Hello, Kamailio is crashing when i'm trying to set the parameter; modparam("tm|usrloc", "xavp_contact", "ulattrs") That's happening with Kamailio 5.3.1 and 5.2.5 too. Crash is happening when Im register 2 devices with the same extension. This is the core dump: [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib64/libthread_db.so.1". Core was generated by `/usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /usr/local/etc/ka'. Program terminated with signal 11, Segmentation fault. #0 0x000000000057cca2 in xavp_get_internal (name=0x8, list=0x0, idx=0, prv=0x0) at core/xavp.c:288 288 core/xavp.c: No such file or directory. (gdb) bt #0 0x000000000057cca2 in xavp_get_internal (name=0x8, list=0x0, idx=0, prv=0x0) at core/xavp.c:288 #1 0x0000000000581869 in xavp_insert (xavp=0x0, idx=0, list=0x0) at core/xavp.c:752 #2 0x00007f05be0ee3a1 in ki_t_next_contacts (msg=0x7f05cb76c218) at t_serial.c:552 #3 0x00007f05be0f09e8 in t_next_contacts (msg=0x7f05cb76c218, key=0x0, value=0x0) at t_serial.c:756 #4 0x0000000000434d41 in do_action (h=0x7ffe93c9cad0, a=0x7f05cb642250, msg=0x7f05cb76c218) at core/action.c:1067 #5 0x00000000004418e8 in run_actions (h=0x7ffe93c9cad0, a=0x7f05cb642250, msg=0x7f05cb76c218) at core/action.c:1572 #6 0x0000000000441fa9 in run_actions_safe (h=0x7ffe93c9de90, a=0x7f05cb642250, msg=0x7f05cb76c218) at core/action.c:1636 #7 0x000000000065734e in rval_get_int (h=0x7ffe93c9de90, msg=0x7f05cb76c218, i=0x7ffe93c9cf78, rv=0x7f05cb642570, cache=0x0) at core/rvalue.c:912 #8 0x000000000065b8fe in rval_expr_eval_int (h=0x7ffe93c9de90, msg=0x7f05cb76c218, res=0x7ffe93c9cf78, rve=0x7f05cb642568) at core/rvalue.c:1910 #9 0x000000000065bd51 in rval_expr_eval_int (h=0x7ffe93c9de90, msg=0x7f05cb76c218, res=0x7ffe93c9d42c, rve=0x7f05cb642c98) at core/rvalue.c:1918 #10 0x0000000000434807 in do_action (h=0x7ffe93c9de90, a=0x7f05cb645028, msg=0x7f05cb76c218) at core/action.c:1043 #11 0x00000000004418e8 in run_actions (h=0x7ffe93c9de90, a=0x7f05cb637e48, msg=0x7f05cb76c218) at core/action.c:1572 #12 0x0000000000431767 in do_action (h=0x7ffe93c9de90, a=0x7f05cb583858, msg=0x7f05cb76c218) at core/action.c:691 #13 0x00000000004418e8 in run_actions (h=0x7ffe93c9de90, a=0x7f05cb574750, msg=0x7f05cb76c218) at core/action.c:1572 #14 0x0000000000442071 in run_top_route (a=0x7f05cb574750, msg=0x7f05cb76c218, c=0x0) at core/action.c:1657 #15 0x00000000005874a9 in receive_msg (buf=0xa6ec00 <buf.6868> "OPTIONS sip:10@192.168.0.231:5060 SIP/2.0\r\nVia: SIP/2.0/UDP 192.168.0.231;branch=z9hG4bKd8a9.1b3abe656532065414a2f35a8916008d.1\r\nVia: SIP/2.0/UDP 192.168.0.131:5060;received=192.168.0.131;branch=z9hG"..., len=694, rcv_info=0x7ffe93c9e4d0) at core/receive.c:341 #16 0x000000000047bf81 in udp_rcv_loop () at core/udp_server.c:541 #17 0x0000000000424f9d in main_loop () at main.c:1669 #18 0x000000000042c688 in main (argc=13, argv=0x7ffe93c9ea18) at main.c:2710 Thank you.

2 1

Kamailio Parrallel Forking
by Ilie Soltanici 27 Nov '19

27 Nov '19

Hello, I'm trying to use parallel forking if one extension is registered from multiple devices. (using WebSockets) This is the related code from kamailio.cfg: ------------------------------------------------------------------------------------------------------------------------------------ if (!t_load_contacts()) { xlogl("L_WARN", "Error loading contacts for $rU\n"); sl_send_reply("500", "Server Internal Error (Code:$cfg(line))"); exit; } else { xlogl("L_INFO", "Contacts loaded for $rU\n"); } if (!t_next_contacts()) { xlogl("L_INFO", "t_next_contacts - Only one contact found for $rU, calling\n"); } else { xlogl("L_INFO", "t_next_contacts - Multiple contacts found, parallel forking\n"); } route(RELAY); ------------------------------------------------------------------------------------------------------------------------------------ I noticed that on Kamailio 5.2.5 - this code is working fine, and Kamailio is sending an INVITE to both Devices.But, on Kamailio 5.3.1 - that's not working anymore. For some reason, kamailio is sending just one INVITE to one device with wrong URI. The URI is set to the second device instead of the correct one. Was something changed in the t_next_contacts function related to this? Thank you

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Kamailio 5.2.3 segfaulting
by Daniel Greenwald 27 Nov '19

27 Nov '19

I'm seeing this in the log and kamailio is crashing every hour or so: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 33/var/log/kamailio/kamailio.log Any ideas? Should I emergency upgrade to 5.3.1?

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No Audio using RTP Proxy
by Sujit Roy 27 Nov '19

27 Nov '19

Hello I am facing a problem as below. Please suggest for the work around. My call flow is like this. SIP Gateway-1 (IP x.179) -> SIP Gateway-2 ( IP x.177) -> Kamalio+RTPProxy So when the call arrives at Kamalio+RTPProxy, i m getting below in log. Nov 26 23:25:31 rtpproxy[18508]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 1b7c870763616c6c15fff410@ 192.168.100.177, tag 1aa18fc201a68168;1 requested, type strong Nov 26 23:25:31 rtpproxy[18508]: INFO:1b7c870763616c6c15fff410@ 192.168.100.177:rtpp_command_ul_handle: new session on IPv4 port 15920 created, tag 1aa18fc201a68168;1 Nov 26 23:25:31 rtpproxy[18508]: INFO:1b7c870763616c6c15fff410@192.168.100.177:rtpp_stream_prefill_addr: pre-filling caller's RTP address with 192.168.100.177:27360 Nov 26 23:25:31 rtpproxy[18508]: INFO:1b7c870763616c6c15fff410@ 192.168.100.177:rtpp_stream_prefill_addr: pre-filling caller's RTCP address with 192.168.100.177:27361 But x.177 is working on signalling mode only ( Not routing Media ) . As a result, i m not getting any voice from IP x.179 What can be done to change the caller's RTP address to x.179 in RTPProxy ? Thanks -- Regards =================== Sujit Roy

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Encode host ip in Contact header for semi-stateless setup
by Daniel Greenwald 27 Nov '19

27 Nov '19

At kamailio world 2017, Jan Lorenz of Pascom explained how they use kamailio in a semi-stateless manner and encode the asterisk's IP into the contact header on the way out so that it can be decoded and used to route properly on the way back without Record-Routing. I would like to create such a setup. I'm looking for pointers if there is a particular module/method that would be ideal to to use for this. Any other gotcha's to consider in such a setup. My plan is to keep the transaction on the same kamailio by not altering the VIA but allow the subsequent in-dialog transactions to potentially hit another kamailio by changing the contact to a shared SRV record (and encoding the freeswitch box's IP into it for later retrieval.) Thanks in advance. Daniel Greenwald

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dialog module: Does dlg_manage() have to be called for all requests?
by George Diamantopoulos 27 Nov '19

27 Nov '19

Hello all, I've been reading through the dialog module docs again, and there seems to be a discrepancy in what's suggested in the docs. In the intro, it is stated that 'To create the dialog associated with an initial INVITE request, execute the function “dlg_manage()"'. Later on, in section "7.10" the following example code is provided: request_route { ... if(is_method("INVITE") && !has_totag()) { $dlg_ctx(timeout_route) = "DLGTIMEOUT"; $dlg_ctx(timeout_bye) = 1; } dlg_manage(); ... } In this example code, dlg_manage seems to be called for all requests, as it is not dependent on the conditional that only requests with no to-tag are to be handled. Previously, Daniel had suggested to me in person that one also run dlg_manage() for CANCELs, and in-dialog reINVITEs, BYEs and ACKs to mitigate some buggy corner cases. Is this still the case? What do you guys and gals do in production? Have you had to revise how you use dlg_manage in terms of calling it for initial INVITEs only vs all requests? Thanks! BR, George

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Modify SIP Message Header and Message Body parameter
by Sujit Roy 26 Nov '19

26 Nov '19

Hello Please see below SIP Message SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.1681.101:5060;branch=z9hG4bK5a61798d7c850d39;rport=5060 From: <sip:summona@ 192.1681.101>;tag=51a0dc1e620de16d To: <sip:101880171234567@ 192.1681.102> Call-ID: 6fd9b9d563616c6c15f2d7fa@ 192.1681.101 CSeq: 21701 INVITE *Server: kamailio (5.1.9 (x86_64/linux))* Content-Length: 0 *1. I want to Change the Server Parameter . How to do it ? * SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.1681.101:5060;rport=5060;branch=z9hG4bK5a61798d7c850d39 Record-Route: <sip: 192.1681.102;lr> From: <sip:summona@ 192.1681.101>;tag=51a0dc1e620de16d To: <sip:101880171234567 @ 192.1681.102>;tag=as728725b3 Call-ID: 6fd9b9d563616c6c15f2d7fa@ 192.1681.101 CSeq: 21701 INVITE Server: VOS3000 V2.1.4.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 600;refresher=uas *Contact: <sip:1018801751481417@104.216.39.66:5060 <http://sip:1018801751481417@104.216.39.66:5060>>* Content-Type: application/sdp Require: timer Content-Length: 289 v=0 o=root 912643161 912643161 IN IP4 192.1681.104 s=VOS3000 V2.1.4.0 c=IN IP4 192.1681.104 t=0 0 m=audio 10784 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:230 a=sendrecv *a=nortpproxy:yes* 2. I Want to hide the Contact Parameter and a= nortpproxy:yes ==> How to do it as well ? Thanks in advance. -- Regards =================== Sujit Roy

2 1

Beginner Help: Relay vs Forward
by Michael Iedema 26 Nov '19

26 Nov '19

Hi all, I am trying to place Kamailio between several SIP services which are differentiated by port and by the type of message they handle. To be concrete, here are the services: Originator: - reachable on port 5060 - originates all traffic to the services Service A: - reachable on port 5061 - handles REGISTER Service B: - reachable on port 5062 - handles INVITE Service C: - reachable on port 5063 - handles MESSAGE The services, originator and Kamailio instance may all be run on the same IP or may be split onto different addresses. Previously, the originator needed to know the address and port of each service. Now I want to make Kamailio responsible for that task so it can be done more intelligently. To start though I simply want to get traffic flowing between the services and this is where I’m struggling to climb the learning curve. In my test setup the originator and all services are on the same IP with only Kamailio running on a different IP. Address 1 Address 2 ======================== Originator <--> Kamailio:6060 | | Service A <-------> | Service B <-------> | Service C <-------> | My first attempt at the logic looks like this: # HOST DEFINITIONS #!substdef "/HOST_ORIGINATOR/192.168.86.110/" #!substdef "/HOST_REGISTER/192.168.86.110/“ #!substdef "/HOST_INVITE/192.168.86.110/" #!substdef "/HOST_MESSAGE/192.168.86.110/" # PORT DEFINITIONS #!substdef “/PORT_ORIGINATOR/5060/" #!substdef "/PORT_REGISTER/5061/“ #!substdef "/PORT_INVITE/5062/" #!substdef "/PORT_MESSAGE/5063/" # ROUTE LOGIC route { # ORIGINATOR sent us something if ( $(ct{nameaddr.uri}{uri.port}) == HOST_ORIGINATOR ) { xlog("L_INFO", "ORIGINATOR: sent $rm\n”); if (is_method("REGISTER")) { $du = "sip: HOST_REGISTER:PORT_REGISTER"; xlog("L_INFO", "ORIGINATOR -> REGISTER: Destination URI is now $du\n"); t_relay(); exit; } } … #### END SNIPPET #### My very newbie questions are: - Am I even on the right track here? I understand that t_relay() establishes a stateful dialog handler between the originator and register services. - Can I accomplish this bi-directional path statelessly with forward()? - Do I need to add logic to handle the messages from the REGISTER service back to ORIGINATOR? This config currently fails because of "no corresponding socket found” when the relay triggers. I believe this is because I’m running the Kamailio instance in a Docker container so my first step today is to set up a native install. I also looked at the dispatcher module which seems very close to what I want to do. Maybe that’s a more natural way to go? Anyway, apologies for the lengthy explanation. I hope it is at least clear what I intend to do. I will gladly research and do trial and error to solve this but need to know I’m going down roughly the right path. I’m not sure what I should even be googling even of the time :-) Thanks as always for your time and the great software! Regards, -Michael

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Re-use TCP connections on different transactions
by Joel Serrano 26 Nov '19

26 Nov '19

Hello, I'm setting up a Kamailio instance behind a TCP load balancer (with proxy protocol and NAT routing: meaning Kam stays in the flow all the time). I've managed to get working almost everything we need for our service, except for one thing, and that is for Kam to use existing connections for subsequent transactions: Following this example: [image: image.png] EXT & INT represent the external and internal interface of a LB between the UAC and Kamailio, using TLS on both legs and proxy protocol. Transaction 1: INVITE, 100, 180, 183, 200 OK UAC 1.1.1.1:1111 -> 2.2.2.2:443 (EXT) 3.3.3.3:3333 (INT) -> 7.7.7.7:5060 (Kamailio) Transaction 2: ACK UAC 1.1.1.1:1112 -> 2.2.2.2:443 (EXT) 4.4.4.4:4444 (INT) -> 7.7.7.7:5060 (Kamailio) Transaction 3: BYE Kam 7.7.7.7:5060 -> 3.3.3.3:3333 (INT) 2.2.2.2:443 (EXT) -> 1.1.1.1:1111 UAC My problem is with Transaction 3. In this case the BYE is originated by the callee, and Kam has to send it to the caller. As the TCP load balancer is between Kam and the UAC, Kam has to send it to the LB so then the LB can forward it back to the UAC. This works well for msgs that belong to the same transaction (INVITE, 100, 180, 183, 200 OK) but it fails when they don't belong to the same transaction. Thanks to the newly added $tcp(c_si) and $tcp(c_sp) pseudovars, I can save the internal IP:Port of the LB, so I can send stuff later to it, my problem is that Kam doesn't seem to allow this? On the original INVITE, I use the following to save where I have to reach the UAC: add_contact_alias("$tcp(c_si)", "$tcp(c_sp)", "tls"); Then, handle_ruri_alias() will take care of setting $du to the correct (internal LB) IP:Port so I can reach the UAC, this works. My problem is that Kamailio doesn't identify that there is a valid existing TLS connection still up (from the INVITE), and tries to create a new one (and this obviously doesn't gives all sorts of problems). So when I run handle_ruri_alias(), and $du is set to 3.3.3.3:3333 (from the example above), instead of using the existing connection, Kamailio tries to create a new one. I have a log statement right before with the result of tcp_conid_state(1) (the connid is 1 for this connection) and the $rc is 1 (Connection is OK), but when I tell Kamailio it has to use it I get this in the logs: DEBUG: {1 11726467 BYE gqR1qqNK8B} <core> [core\/tcp_main.c:2060]: tcp_send(): no open tcp connection found, opening new one"} And then the problems begin... I have tried playing around with: tcp_reuse_port tcp_connection_match But no luck..! I also thought it could be a problem of the connection being created on one worker, and a different worker handling BYE transaction, so tested with children=1 and tcp_children=1, but still same problem. A more detailed log: In blue my log statement checking for the status of conid "1", in red Kam not being able to find it, although it exists (as validated by tcp_conid_state(), and even in netstat I can see the connection established). In this log, 35.191.0.66:60271 would be the equivalent of 3.3.3.3:3333 and 104.175.176.242:28157 would be 1.1.1.1:1111 from the example above. ... "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/tcp_main.c:1657]: _tcpconn_find(): found connection by id: 1"} "message":" NOTICE: {1 11727734 BYE 5-LX4GdI9X} <script>: JOEL TEST New request - M=BYE TCP STATUS:1 ID=5-LX4GdI9X"} ... "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} nathelper [nathelper.c:1144]: handle_ruri_alias(): setting dst_uri to <sip:35.191.0.66:60271;transport=tls>"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} nathelper [nathelper.c:1166]: handle_ruri_alias(): rewriting r-uri to <sip:linphone@104.175.176.242:28157;transport=tls>"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_lookup.c:1328]: t_newtran(): msg (0x7f3c884259d0) id=534\/18664 global id=532\/18664 T start=0xffffffffffffffff"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_lookup.c:497]: t_lookup_request(): start searching: hash=63496, isACK=0"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_lookup.c:455]: matching_3261(): RFC3261 transaction matching failed - via branch [z9hG4bK808f.eee2444f92a02cb33e1b7a21f20bc6bb.0]"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_lookup.c:675]: t_lookup_request(): no transaction found"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_hooks.c:336]: run_reqin_callbacks_internal(): trans=0x7f3c83b8c598, callback type 1, id 0 entered"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/crypto\/md5utils.c:67]: MD5StringArray(): MD5 calculated: 3071029feb05962b26b53a9664a14210"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} siputils [checks.c:123]: has_totag(): totag found"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} rr [loose.c:1095]: check_route_param(): route params checking against [;r2=on;lr=on;ftag=Eb~TbdfTA;did=cab.01e2;nat=yes] (orig: [r2=on;lr=on;ftag=Eb~TbdfTA;did=cab.01e2;nat=yes])"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} rr [loose.c:1101]: check_route_param(): params are <;r2=on;lr=on;ftag=Eb~TbdfTA;did=cab.01e2;nat=yes>"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} siputils [checks.c:123]: has_totag(): totag found"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/msg_translator.c:161]: check_via_address(): (198.1.54.228, 198.1.54.228, 0)"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/tcp_main.c:2060]: tcp_send(): no open tcp connection found, opening new one"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/ip_addr.c:229]: print_ip(): tcpconn_new: new tcp connection: 35.191.0.66"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/tcp_main.c:1242]: tcpconn_new(): on port 60271, type 3"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/tcp_main.c:1561]: tcpconn_add(): hashes: 1446:2350:0, 5"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tls [tls_server.c:199]: tls_complete_init(): completing tls connection initialization"} ... So time to seek help from the community, any suggestions/ideas/comments? Sorry if all this sounds confusing, I've tried my best to put in text the whole scenario in and "understandable" way... Is this even doable? Thanks, Joel.

3 10

How to enforce REGISTER with AKAv1 (IMS)
by Moshe A 26 Nov '19

26 Nov '19

Hello, How can we enforce REGISTER with AKAv1? Here our setup 1. kamailio version 4.3.4 for IMS 2. Boghe IMS client v1.0.58.550 3. scscf.cfg is configure with: #!define REG_AUTH_DEFAULT_ALG "AKAv1-MD5" 4. Set Authentication Schemes in Private User Identity - IMPI for user bob to: Digest-AKAv1 (3GPP) results: First REGISTER fails with 401 Unauthorized Second REGISTER 200 OK, but the REGISTER is MD5 and not AKAv1 Attached is tcpdump. By the way: in which DB should we find Authentication Schemes? can't find it in hss_db, kamailio and not in scscf. Thank you, Moshe.

1 0

call not going to end point
by Gaurav Bmotra 26 Nov '19

26 Nov '19

hi i m using kamailio 5.4 dev Docker image when i try to call form calientA to clientB i got (Temporarily Unavailable (code: 480)) i m using freeswitch as media server so the flow and structure should be like this ClientA---->Kamailio------>FS---->Kamailio>---->ClientB but issue is (what i m getting ) ClientA---->Kamailio------>FS---->Kamailio>-------->FS-------->kamailio--------->FS IT CREATE THE CALL LOOP b/w KAMAILIO AND FREESWITCH *MY CONFIG FILE IS ATACHED * in logs i m getting -------------------------------------------------------- Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 38902: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:get_command: received command "18_4 Uc0,8,101 9857c00a-8947-1238-c6a7-12590b724429 3.82.96.70 22082 jKrKymKNg480p;1" Nov 24 10:25:36 ip-172-20-61-37 local0.info rtpproxy[6]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 9857c00a-8947-1238-c6a7-12590b724429, tag jKrKymKNg480p;1 requested, type strong Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 35888: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:get_command: received command "20_4 Uc0,8,101 98589959-8947-1238-c6a7-12590b724429 3.82.96.70 29846 m5a51aNvapN6D;1" Nov 24 10:25:36 ip-172-20-61-37 local0.info rtpproxy[6]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 98589959-8947-1238-c6a7-12590b724429, tag m5a51aNvapN6D;1 requested, type strong Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 37706: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:get_command: received command "13_4 Uc0,8,101 985962c1-8947-1238-c6a7-12590b724429 3.82.96.70 29432 pQXp50p3471BN;1" Nov 24 10:25:36 ip-172-20-61-37 local0.info rtpproxy[6]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 985962c1-8947-1238-c6a7-12590b724429, tag pQXp50p3471BN;1 requested, type strong Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 45078: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:get_command: received command "16_5 Uc0,8,101 985a3ce3-8947-1238-c6a7-12590b724429 3.82.96.70 25096 r9F88praZSeHc;1" Nov 24 10:25:36 ip-172-20-61-37 local0.info rtpproxy[6]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 985a3ce3-8947-1238-c6a7-12590b724429, tag r9F88praZSeHc;1 requested, type strong Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 58376: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" -------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- *here is my config file * #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ACCDB #!define WITH_NAT # # Kamailio (OpenSER) SIP Server v5.4 - default configuration script # - web: https://www.kamailio.org # - git: https://github.com/kamailio/kamailio # # Direct your questions about this file to: <sr-users(a)lists.kamailio.org> # # Refer to the Core CookBook at https://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Note: the comments can be: # - lines starting with #, but not the pre-processor directives, # which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef, # #!subst, #!substdef, ... # - lines starting with // # - blocks enclosed in between /* */ # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To use RTPEngine (instead of RTPProxy) for nat traversal execute: # - define WITH_RTPENGINE # - install RTPEngine: https://github.com/sipwise/rtpengine # - start RTPEngine: # rtpengine --listen-ng=127.0.0.1:2223 ... # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes for each UDP socket * - value inherited by tcp_children and sctp_children when not set explicitely */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* number of SIP routing processes for all TCP/TLS sockets */ # tcp_children=8 /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ # auto_aliases=no /* add local domain aliases */ # alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to * bind on a specific interface/port/proto (default bind on all available) */ # listen=udp:10.0.0.10:5060 listen=udp:LOCAL_IP:5091 advertise PUBLIC_IP:5091 /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 /* upper limit for TCP connections (it includes the TLS connections) */ tcp_max_connections=2048 #!ifdef WITH_TLS enable_tls=yes /* upper limit for TLS connections */ tls_max_connections=2048 #!endif ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" loadmodule "dispatcher.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" #!ifdef WITH_RTPENGINE loadmodule "rtpengine.so" #!else loadmodule "rtpproxy.so" #!endif #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- #-------------------dispatcher params------------- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/run/kamailio/kamailio_ctl") # ----- sanity params ----- modparam("sanity", "autodrop", 0) # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT #!ifdef WITH_RTPENGINE # ----- rtpengine params ----- modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223") #!else # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") #!endif # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains #route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN #route(PSTN); route(DISPATCH); # user location service } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); $rcv_addr="sip:"+$si+":"+$sp; if(starts_with("$rcv_addr","$du")) { route(LOCATION); } else { route(RELAY); } exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { xnotice("RTF_DISPATCH: $rm $rU [$ci]"); if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("(^5)") or (t_branch_timeout() and !t_branch_replied())) { xnotice("RTF_DISPATCH: WARNING $du is broken and marked as inactive"); # ds_mark_dst("ip"); if(ds_next_dst()) { #xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } # Per SIP request initial checks route[REQINIT] { # no connect for sending replies set_reply_no_connect(); #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if($ua =~ "friendly-scanner|sipcli|sipvicious|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP request from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) { sl_reply_error(); } exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # 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; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # 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; #!ifdef WITH_RTPENGINE if(nat_uac_test("8")) { rtpengine_manage("replace-origin replace-session-connection"); } else { rtpengine_manage("trust-address replace-origin replace-session-connection"); } #!else if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } #!endif 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; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies reply_route { if(!sanity_check("17604", "6")) { xlog("Malformed SIP response from $si:$sp\n"); drop; } } # Manage incoming replies in transaction context onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } -- *Regards:* Gaurav Kumar

2 2

How to get the audio codec used for each established call.
by Abdoul Osséni 25 Nov '19

25 Nov '19

Hello all, I try to save in the CDR the audio codec used for each established call. Can you help me? Regards Abdoul.I

3 4

SDP Modification
by Sujit Roy 25 Nov '19

25 Nov '19

Hello How can i change c= and o= in SDP using RTPProxy ? Please advise Thanks -- Regards =================== Sujit Roy

2 1

Passing media thru RTP Proxy
by Sujit Roy 25 Nov '19

25 Nov '19

Hello I have installed and configured both Kamailio and RTP Proxy and calls are going. But i m unable to force media to go thru RTP Proxy. How can i enforce media to go thru RTP Proxy ? Thanks -- Regards =================== Sujit Roy

3 2

Kamailio stopped relaying SIP text message from asterisk
by Adesh Pandey 25 Nov '19

25 Nov '19

Hi, I have been working on kamailio with following scenario: 1. Webrtc SIP Users register to Kamailio 2. Users initiate outgoing calls to some other numbers 3. Kamailio dispatches outgoing calls to Asterisk 4. Asterisk answers the call and then send a text message to the caller using SendText, which was working fine a few days back but all of sudden it has stopped working 5. At Asterisk end it is showing that message was delivered successfully but when I check SIP user logs, there is no message event. Please review my config file at following URL: https://gist.github.com/adeshpandey/2c338db7f5d992267f7415de0325e0b4 Developers, I also want to know if there is any way to log *text message* in kamailio server.

3 3

Unable to connect to sip server via proxy
by Ashwini Kumar 24 Nov '19

24 Nov '19

Hi I have two network interfaces on my system in which one interface is connected to my office WAN/LAN and other is connected to SIP private network connected to PSTN. Now, I want to access sip server using the internal IP of our office LAN as proxy. *192.168.6.2 (Office LAN IP as proxy) -> 10.50.185.2 (SIP)* So on machine 192.168.6.2, I setup kamailio as SIP proxy server so that our office LAN machine can connect to SIP server directly via SIP proxy. I used config defined in - *https://github.com/kamailio/kamailio/blob/master/misc/examples/mixed/kamailio-minimal-proxy.cfg* <https://github.com/kamailio/kamailio/blob/master/misc/examples/mixed/kamail…> and I changed the local domain alias with SIP IP -* 10.50.185.2*. When I try to connect to SIP via proxy using *linphone* (UDP) I got below errors in* /var/log/syslog* and shows *connection in progress* every time. Nov 23 15:15:49 ubantudesk /usr/sbin/kamailio[25784]: WARNING: {1 20 REGISTER ek82LguQ6z} sanity [sanity.c:585]: check_cl(): content length header missing in request Nov 23 15:15:53 ubantudesk /usr/sbin/kamailio[25785]: WARNING: {1 20 REGISTER ek82LguQ6z} sanity [sanity.c:585]: check_cl(): content length header missing in request Nov 23 15:15:57 ubantudesk /usr/sbin/kamailio[25784]: WARNING: {1 20 REGISTER ek82LguQ6z} sanity [sanity.c:585]: check_cl(): content length header missing in request Nov 23 15:16:01 ubantudesk /usr/sbin/kamailio[25785]: WARNING: {1 20 REGISTER ek82LguQ6z} sanity [sanity.c:585]: check_cl(): content length header missing in request Nov 23 15:16:05 ubantudesk /usr/sbin/kamailio[25784]: WARNING: {1 20 REGISTER ek82LguQ6z} sanity [sanity.c:585]: check_cl(): content length header missing in request Can you help me what I should do? -- *Thanks & Regards,* *Ashwini Kumar | Technology |* www.industrybuying.com | Ghitorni, Delhi - 110030 | India | È*+91-8802137782* | * ashwinikumar(a)industrybuying.com |

2 1

Avoid full table scan for dispatcher table on db_redis using ds_reload() from config script
by Joel Serrano 23 Nov '19

23 Nov '19

Hello, I'm trying out redis as db backend, and right now I only have dispatcher records there... Every so often, I do a ds_reload() from within Kam config script, and I see in logs: Nov 22 20:36:35 test COPS[25531]: WARNING: db_redis [redis_dbase.c:1098]: db_redis_perform_query(): performing full table scan on table 'dispatcher' while performing query After enabling debug logs: Nov 22 21:15:35 test COPS[26661]: DEBUG: db_redis [redis_dbase.c:1761]: db_redis_query(): querying prefix (table) 'dispatcher' Nov 22 21:15:35 test COPS[26661]: DEBUG: db_redis [redis_dbase.c:1811]: db_redis_query(): no columns given to build query keys, falling back to full table scan Nov 22 21:15:35 test COPS[26661]: DEBUG: <core> [db_res.c:119]: db_new_result(): allocate 56 bytes for result set at 0x7f0be4273898 Nov 22 21:15:35 test COPS[26661]: DEBUG: <core> [db_res.c:156]: db_allocate_columns(): allocate 40 bytes for result names at 0x7f0be4273938 Nov 22 21:15:35 test COPS[26661]: DEBUG: <core> [db_res.c:167]: db_allocate_columns(): allocate 20 bytes for result types at 0x7f0be42739c8 Nov 22 21:15:35 test COPS[26661]: WARNING: db_redis [redis_dbase.c:1098]: db_redis_perform_query(): performing full table scan on table 'dispatcher' while performing query Given the msg... "no columns given to build query keys, falling back to full table scan", I assume I'm missing a keys modparam.. but I don't know how to build it, and the docs (to me) seem confusing? https://kamailio.org/docs/modules/devel/modules/db_redis.html#db_redis.p.ke… Can anyone help me understand how to build the keys modparam for dispatcher table? Or if the reason for this is different, what do I need to do to avoid the full table scan and the warning in the log? Thanks, Joel.

1 1

Question about ims_ipsec_pcscf
by Woscek, Martin W. 22 Nov '19

22 Nov '19

Hi all, We are looking for information as to what the ims_ipsec_pcscf module was written for and tested against. What was the target UE/EPC/PCRF that Tsvetomir Dimitrov wrote the module towards? Has an emulated EPC been tested with this module? If so which one(s)? Thanks in advance, Martin

1 0

Cannot read an indexed xavp from Kemi-Lua
by Michel Pelletier 22 Nov '19

22 Nov '19

Hi, I am having trouble reading an indexed xavp from Kemi-Lua. The xavp contains 3 entries but I can only access the first one. The xavp is the one filled by the dispatcher module after a call to KSR.dispatcher.ds_select(group, 4) with the flags parameter set to 2. I can get the number of entries using KSR.pvx.xavp_child_get("ds_ctx", "cnt") which returns 3. I can also get the contents of the first element using, for example, KSR.pvx.xavp_child_get("ds_dst", "attrs"). But all my attempts at adding an index value using KSR.pvx.xavp_get() or KSR.pvx.xavp_child_get() return the value 'nil'; Below is the output of KSR.pvx.pv_xavp_print(): +++++ start XAVP list: (nil) ((nil)) (level=0) ----- end XAVP list: (nil) (level=0) +++++ start XAVP list: 0x7f94ce01f1f8 ((nil)) (level=0) *** (l:0 - 0x7f94ce01f1f8) XAVP name: ds_ctx XAVP id: 1762531321 XAVP value type: 6 XAVP value: <xavp:0x7f94ce01f150> +++++ start XAVP list: 0x7f94ce01f150 (0x7f94ce01f218) (level=1) *** (l:1 - 0x7f94ce01f150) XAVP name: cnt XAVP id: 7279890 XAVP value type: 1 XAVP value (int): 3 *** (l:1 - 0x7f94ce01f150) end ----- end XAVP list: 0x7f94ce01f150 (level=1) *** (l:0 - 0x7f94ce01f1f8) end *** (l:0 - 0x7f94ce01eaf8) XAVP name: ds_dst XAVP id: 1762531862 XAVP value type: 6 XAVP value: <xavp:0x7f94ce01ea30> +++++ start XAVP list: 0x7f94ce01ea30 (0x7f94ce01eb18) (level=1) *** (l:1 - 0x7f94ce01ea30) XAVP name: attrs XAVP id: 1835799177 XAVP value type: 2 XAVP value (str): duid=newjersey3;maxload=200 *** (l:1 - 0x7f94ce01ea30) end *** (l:1 - 0x7f94ce01e988) XAVP name: grp XAVP id: 7056653 XAVP value type: 1 XAVP value (int): 1 *** (l:1 - 0x7f94ce01e988) end *** (l:1 - 0x7f94ce01e8c8) XAVP name: uri XAVP id: 8122237 XAVP value type: 2 XAVP value (str): sip:123.123.123.123:5060 *** (l:1 - 0x7f94ce01e8c8) end ----- end XAVP list: 0x7f94ce01ea30 (level=1) *** (l:0 - 0x7f94ce01eaf8) end *** (l:0 - 0x7f94ce01edd0) XAVP name: ds_dst XAVP id: 1762531862 XAVP value type: 6 XAVP value: <xavp:0x7f94ce01ed08> +++++ start XAVP list: 0x7f94ce01ed08 (0x7f94ce01edf0) (level=1) *** (l:1 - 0x7f94ce01ed08) XAVP name: attrs XAVP id: 1835799177 XAVP value type: 2 XAVP value (str): duid=newjersey2;maxload=200 *** (l:1 - 0x7f94ce01ed08) end *** (l:1 - 0x7f94ce01ec60) XAVP name: grp XAVP id: 7056653 XAVP value type: 1 XAVP value (int): 1 *** (l:1 - 0x7f94ce01ec60) end *** (l:1 - 0x7f94ce01eba0) XAVP name: uri XAVP id: 8122237 XAVP value type: 2 XAVP value (str): sip:123.123.123.123:5060 *** (l:1 - 0x7f94ce01eba0) end ----- end XAVP list: 0x7f94ce01ed08 (level=1) *** (l:0 - 0x7f94ce01edd0) end *** (l:0 - 0x7f94ce01f0a8) XAVP name: ds_dst XAVP id: 1762531862 XAVP value type: 6 XAVP value: <xavp:0x7f94ce01efe0> +++++ start XAVP list: 0x7f94ce01efe0 (0x7f94ce01f0c8) (level=1) *** (l:1 - 0x7f94ce01efe0) XAVP name: attrs XAVP id: 1835799177 XAVP value type: 2 XAVP value (str): duid=newjersey1;maxload=200 *** (l:1 - 0x7f94ce01efe0) end *** (l:1 - 0x7f94ce01ef38) XAVP name: grp XAVP id: 7056653 XAVP value type: 1 XAVP value (int): 1 *** (l:1 - 0x7f94ce01ef38) end *** (l:1 - 0x7f94ce01ee78) XAVP name: uri XAVP id: 8122237 XAVP value type: 2 XAVP value (str): sip:123.123.123.123:5060 *** (l:1 - 0x7f94ce01ee78) end ----- end XAVP list: 0x7f94ce01efe0 (level=1) *** (l:0 - 0x7f94ce01f0a8) end ----- end XAVP list: 0x7f94ce01f1f8 (level=0) Regards, Michel Pelletier

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kamailio call loop issue B/W kamailio and freeswitch
by Gaurav Bmotra 22 Nov '19

22 Nov '19

hi im using kamailio 5.3 as register server and using dispatcher i m dendign the calls to freeswitch .. ibut the issue is then freeswitch send back call to kamailio after processing media ... it should go back to client but it again send call to freeswitch ,, which create the loop and hangup the call with (Temporarily Unavailable (code: 480)) *what i want *client1------>Kamailio------>FS-----Kamailio----->Client2 *but I am getting this * client1------>Kamailio------>FS-----Kamailio----->FS----->kamailio----->FS -- here is my config file data --- #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ACCDB #!define WITH_NAT # # Kamailio (OpenSER) SIP Server v5.2 - default configuration script # - web: https://www.kamailio.org # - git: https://github.com/kamailio/kamailio # # Direct your questions about this file to: <sr-users(a)lists.kamailio.org> # # Refer to the Core CookBook at https://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Note: the comments can be: # - lines starting with #, but not the pre-processor directives, # which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef, # #!subst, #!substdef, ... # - lines starting with // # - blocks enclosed in between /* */ # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes for each UDP socket * - value inherited by tcp_children and sctp_children when not set explicitely */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* number of SIP routing processes for all TCP/TLS sockets */ # tcp_children=8 /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ # auto_aliases=no /* add local domain aliases */ # alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to * bind on a specific interface/port/proto (default bind on all available) */ # listen=udp:10.0.0.10:5060 listen=udp:172.20.LOCAL_IP:5091 advertise 3.83.PUBLIC_IP:5091 /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 /* upper limit for TCP connections (it includes the TLS connections) */ tcp_max_connections=2048 #!ifdef WITH_TLS enable_tls=yes /* upper limit for TLS connections */ tls_max_connections=2048 #!endif ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" loadmodule "dispatcher.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- dispatcher params ----- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains #route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN #route(PSTN); route(DISPATCH); # user location service route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) { sl_reply_error(); } exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } *Regards:* Gaurav Kumar

1 0

Re: [SR-Users] disparcher making loop with call
by Gaurav Bmotra 22 Nov '19

22 Nov '19

hi thank you for reply i have check but same issue ,,, is the config file data plz check if you can help thank you ------------------------------------ #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ACCDB #!define WITH_NAT # # Kamailio (OpenSER) SIP Server v5.2 - default configuration script # - web: https://www.kamailio.org # - git: https://github.com/kamailio/kamailio # # Direct your questions about this file to: <sr-users(a)lists.kamailio.org> # # Refer to the Core CookBook at https://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Note: the comments can be: # - lines starting with #, but not the pre-processor directives, # which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef, # #!subst, #!substdef, ... # - lines starting with // # - blocks enclosed in between /* */ # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes for each UDP socket * - value inherited by tcp_children and sctp_children when not set explicitely */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* number of SIP routing processes for all TCP/TLS sockets */ # tcp_children=8 /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ # auto_aliases=no /* add local domain aliases */ # alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to * bind on a specific interface/port/proto (default bind on all available) */ # listen=udp:10.0.0.10:5060 listen=udp:172.20.LOCAL_IP:5091 advertise 3.83.PUBLIC_IP:5091 /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 /* upper limit for TCP connections (it includes the TLS connections) */ tcp_max_connections=2048 #!ifdef WITH_TLS enable_tls=yes /* upper limit for TLS connections */ tls_max_connections=2048 #!endif ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" loadmodule "dispatcher.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- dispatcher params ----- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains #route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN #route(PSTN); route(DISPATCH); # user location service route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) { sl_reply_error(); } exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } ------------------------------------------------------------------------- On Fri, Nov 22, 2019 at 4:57 PM Gaurav Bmotra <saigauravmehra91(a)gmail.com> wrote: > hi thank you for reply > i have check but same issue > i have attached my config file ... can you check if there is any issue > > thank you > > On Fri, Nov 22, 2019 at 4:41 PM David Villasmil < > david.villasmil.work(a)gmail.com> wrote: > >> Hello, >> >> You’re only pasting the dispatcher route. >> >> You need to follow the script logic. An invite comes in, how does it know >> the call is coming from the client and has to be sent to freeSWITCH? >> Conversely, how does it know it’s coming from freeswitch? >> >> Probably using the permissions module with >> >> https://kamailio.org/docs/modules/3.1.x/modules_k/permissions.html#id2528655 >> >> You can check if the $si is the freeswitch, then send to clientB >> >> On Fri, 22 Nov 2019 at 11:40, Daniel-Constantin Mierla <miconda(a)gmail.com> >> wrote: >> >>> Hello, >>> >>> there is something wrong in your config file, failing to detect the >>> traffic from freeswitch. Load debugger module and enable cfgtrace logging, >>> then see what actions are executed in the configuration file, maybe you can >>> spot what goes wrong there. >>> >>> Otherwise, to get more help from the mailing lists, the full config is >>> needed. >>> >>> Cheers, >>> Daniel >>> On 22.11.19 09:51, Gaurav Bmotra wrote: >>> >>> hi >>> i m using kamailio 5.3 as register server and using dispatcher i m >>> dendign the calls to freeswitch .. >>> >>> but the issue is then freeswitch send back call to kamailio after >>> processing media ... it should go back to client but it again send call to >>> freeswitch ,, which create the loop and hangup the call with (Temporarily >>> Unavailable (code: 480)) >>> >>> >>> *what i want *client1------>Kamailio------>FS-----Kamailio----->Client2 >>> >>> *but I am getting this * >>> >>> client1------>Kamailio------>FS-----Kamailio----->FS----->kamailio----->FS >>> >>> >>> *Routing Logic *------------------------------- >>> if(!ds_is_from_list()) { >>> >>> route(DISPATCH); >>> } >>> >>> -------- >>> # Dispatch requests >>> >>> >>> >>> route[DISPATCH] { >>> >>> >>> >>> # round robin dispatching on gateways group '1' >>> >>> >>> >>> if(!ds_select_dst("1", "4")) { >>> >>> >>> send_reply("404", "No destination"); >>> >>> >>> exit; >>> >>> >>> >>> } >>> >>> >>> >>> xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: >>> $xavp(_dsdst_=>attrs))\n"); >>> >>> >>> t_on_failure("RTF_DISPATCH"); >>> >>> >>> >>> route(RELAY); >>> >>> >>> >>> exit; >>> >>> >>> >>> } >>> >>> >>> >>> >>> >>> >>> >>> # Try next destionations in failure route >>> >>> >>> >>> failure_route[RTF_DISPATCH] { >>> >>> >>> >>> xnotice("RTF_DISPATCH: $rm $rU [$ci]"); >>> >>> >>> >>> if (t_is_canceled()) { >>> >>> >>> exit; >>> >>> >>> >>> } >>> >>> >>> >>> # next DST - only for 500 or local timeout >>> >>> >>> if (t_check_status("(^5)") >>> >>> >>> or (t_branch_timeout() and !t_branch_replied())) >>> { >>> >>> xnotice("RTF_DISPATCH: WARNING $du is broken and marked >>> as inactive"); >>> >>> >>> # ds_mark_dst("ip"); >>> >>> >>> >>> if(ds_next_dst()) { >>> >>> >>> >>> #xdbg("--- SCRIPT: retrying to <$ru> via <$du> >>> (attrs: $xavp(_dsdst_=>attrs))\n"); >>> >>> >>> t_on_failure("RTF_DISPATCH"); >>> >>> >>> >>> route(RELAY); >>> >>> >>> >>> exit; >>> >>> >>> >>> } >>> >>> >>> >>> } >>> >>> >>> >>> } >>> >>> >>> -- >>> thank you >>> >>> >>> >>> >>> >>> >>> >>> *Regards:* >>> Gaurav Kumar >>> >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >>> -- >>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>> Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> sr-users(a)lists.kamailio.org >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >> -- >> Regards, >> >> David Villasmil >> email: david.villasmil.work(a)gmail.com >> phone: +34669448337 >> > > > -- > > > > > > > > > *Regards:* > Gaurav Kumar > > -- *Regards:* Gaurav Kumar

1 0

disparcher making loop with call
by Gaurav Bmotra 22 Nov '19

22 Nov '19

hi i m using kamailio 5.3 as register server and using dispatcher i m dendign the calls to freeswitch .. but the issue is then freeswitch send back call to kamailio after processing media ... it should go back to client but it again send call to freeswitch ,, which create the loop and hangup the call with (Temporarily Unavailable (code: 480)) *what i want *client1------>Kamailio------>FS-----Kamailio----->Client2 *but I am getting this * client1------>Kamailio------>FS-----Kamailio----->FS----->kamailio----->FS *Routing Logic*------------------------------- if(!ds_is_from_list()) { route(DISPATCH); } -------- # Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { xnotice("RTF_DISPATCH: $rm $rU [$ci]"); if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("(^5)") or (t_branch_timeout() and !t_branch_replied())) { xnotice("RTF_DISPATCH: WARNING $du is broken and marked as inactive"); # ds_mark_dst("ip"); if(ds_next_dst()) { #xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } -- thank you *Regards:* Gaurav Kumar

3 2

How to retrieve local socket that will be used to send a message from the branch route
by Patrick Wakano 22 Nov '19

22 Nov '19

Hello list, Hope you are all doing well! I am trying to figure out a way to retrieve the local socket the t_relay() has decided to use in a mhomed=1 env. Is there a way to do that? I couldn't find any variable other than the $snd ones that maybe have this info, but I am not using onsend_route anyway..... I am after this value because this is what I can use to figure out if I need to engage RTPEngine with parameters "external, internal", "external, external", "internal, external" or "internal, internal". In my case, my server is in between two networks, and calls can traverse or can be routed back to the same interface depending on what was decide by the LCR module. Signalling is fine, but engaging the RTPEngine must be done with the correct parameters to get the correct IPs in the SDP, other else I get one way audio in the call..... Any help is much appreciated! Thank you! Kind regards, Patrick Wakano

4 7

Adding SIP Gateway Endpoint in Kamailio
by Sujit Roy 21 Nov '19

21 Nov '19

Hello Here is my call flow scenario. SIP Gateway (A) Kamailio (K) Asterisk (AST) SIP Gateway (B) Now i want to send calls from A -> B by using Asterisk as media server. Kamailio shall be used to authenticate A and allow A to send calls to B. What are the configurations i need to make in Kamailio and Asterisk ? Thanks in advance. -- Regards =================== Sujit Roy

3 6

How to access different uris in a header
by Duarte Rocha 21 Nov '19

21 Nov '19

Greetings, In order to identify Redirecting Number and Original Called Number on a given request i look for the top( Redirecting Number) and bottom (Original Called) Diversion headers using hdr(Diversion)[0] and hdr(Diversion)[-1]. However, i have a client that sends me all the Diversions URIS in the same header. Does kamailio has a similar tool for accessing the different URIS in the header? Best Regards,

2 1

CentOS Repo for Kamailio 5.3 - Strange Python23 Dependency?
by Zach Nielsen 21 Nov '19

21 Nov '19

Hi all, Working with some of our Centos-based Kamailio boxes lately has produced a strange result when using the provided yum repos (download.opensuse.org). When trying to yum update the system or install some additional modules, we're getting feedback from yum indicating that it's expecting /usr/bin/python23 --> Finished Dependency Resolution Error: Package: kamailio-5.3.1-3.1.x86_64 (home_kamailio_v5.3.x-rpms) Requires: /usr/bin/python23 You could try using --skip-broken to work around the problem You could try running: rpm -Va --nofiles --nodigest Tried my best to do my due diligence here to make sure there was nothing up with our local env that was causing this, but this issue persists across multiple systems, including a new (no frills) one that we just build out. Can I get some confirmation if this is a "me" issue, or if maybe there's an issue with what made it into the repo? Also, just a general question... what's the typical way of you folks deploying Kamailio from a sysadmin perspective? Do you generally compile from source, or do any of you use the provided repositories to manage it? Thanks! Zach

5 6

kamailio Dispatcher error
by Gaurav Bmotra 20 Nov '19

20 Nov '19

hi i want to use dispatcher Module and following is the config.. --------------------------------------- loadmodule "dispatcher.so" -------------- modparam("dispatcher", "db_url", "mysql://kamailio:kamailiorw@localhost /kamailio") modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("dispatcher", "ds_hash_size", 2) -------------------------------------------------------------- but when i hit $ kamailio out put: ERROR: bad config file (4 errors) ------------------------------------------------------ logs : kamailio.service - Kamailio (OpenSER) - the Open Source SIP Server Loaded: loaded (/lib/systemd/system/kamailio.service; enabled; vendor preset: enabled) Active: failed (Result: exit-code) since Wed 2019-11-20 15:16:34 IST; 16s ago Process: 25747 ExecStart=/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f $CFGFILE -m $SHM_MEMORY -M $PKG_MEMORY -u $USER -g $GROUP (code=exited, status=255/EXCEPTION) Nov 20 15:16:34 gaurav-Latitude-E7240 systemd[1]: kamailio.service: Service RestartSec=100ms expired, scheduling restart. Nov 20 15:16:34 gaurav-Latitude-E7240 systemd[1]: kamailio.service: Scheduled restart job, restart counter is at 5. Nov 20 15:16:34 gaurav-Latitude-E7240 systemd[1]: Stopped Kamailio (OpenSER) - the Open Source SIP Server. Nov 20 15:16:34 gaurav-Latitude-E7240 systemd[1]: kamailio.service: Start request repeated too quickly. Nov 20 15:16:34 gaurav-Latitude-E7240 systemd[1]: kamailio.service: Failed with result 'exit-code'. Nov 20 15:16:34 gaurav-Latitude-E7240 systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. ------------------------------------------------------------- plz help me -- Thank you *Regards:* Gaurav Kumar

4 6

Dynamic routing and REGISTER
by Igor Olhovskiy 20 Nov '19

20 Nov '19

Hi! I’m trying to get `drouting` module working for REGISTER. But seems it’s not passing one of internal checks, cause in REGISTER R-URI does not contains username and I’m getting errors drouting [drouting.c:567]: build_ruri(): stripping 0 makes username <> null drouting [drouting.c:970]: do_routing(): failed to build ruri So, as I got, this module is mainly for calls (means INVITE) routing, rather than to use it like one more dispatcher, or it’s also a way to handle this part here? Thanks, Igor

3 5

Query regarding Kamailio Sip Server for IMS supporting VOLTE
by Shubhendu Das 20 Nov '19

20 Nov '19

Hi , I have a query regarding Kamailio open source SIP server. Does it have any release, where its supports the feature of IMS supporting VOLTE with real commercial mobile phones(IMSI based) ? If so, what is the release number ? Please also let me know about the supported mobile phones list. BR, Shubhendu -- _Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._

1 0

How to intercept an INVITE after (ts_append)
by Oleg Podguyko 20 Nov '19

20 Nov '19

Hello everybody! Kamailio works as a Registrar. Additionally, rtpengine is deployed next to it. Using rtpgngine module I proxy media stream via rtpengine(rtpengine_manage()). I change SDP. SiP2SIP calls work fine. Now I would like to have possibility to "wake up" sip-clients(at the android and ios phones) which are usually not on-line via push notification. Trying to use TSILO module. # User location service route[LOCATION] { lookup("location"); $var(rc) = $rc; if (is_method("INVITE")) { if (t_newtran()) { ts_store(); # if it a new transuction ,it will be store for some time. } } switch ($var(rc)) { case 1: route(RELAY); break; case -1: break; case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if (!save("location")) { sl_reply_error(); } ts_append("location", "$tu"); # if transuction for this sip-client is exist. The invite will be send. exit; } If the client is not registered at the kamailio during the call. Then such calls are stored through the "ts_store" function for some time. And if during this time the client send register, kamailio will be sent INVITE to it. But I have the problem. Such calls after "wake up"go through a special branch route that I don't know how to intercept. As a result, I can't change SDP(using rtpengite_manage commands) and rtp stream do not proxy via rtpengine… and call failed. /usr/sbin/kamailio[9957]: exec: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} *** cfgtrace:request_route=[REGISTRAR] c=[/etc/kamailio/kamailio.cfg] l=428 a=26 n=ts_append /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} tsilo [ts_append.c:72]: ts_append(): transaction 36343:92182831 found for 777777777, going to append branches /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} tm [t_lookup.c:1629]: t_lookup_ident_filter(): transaction found /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} registrar [lookup.c:255]: lookup_helper(): contact for [777777777] found by address /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} registrar [lookup.c:456]: lookup_helper(): instance is /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} tm [t_append_branches.c:77]: t_append_branches(): transaction 36343:92182831 in status 0 /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} tm [t_append_branches.c:99]: t_append_branches(): Call uvAzTGsEV9N2ReDjmWoo8A..: 0 (1) outgoing branches /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} tm [t_append_branches.c:126]: t_append_branches(): Current uri sip:77777777@95.165.146.181:63618;transport=TCP;rinstance=8b1865b95767b7ba /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} <core> [core/msg_translator.c:2933]: create_via_hf(): id added: <;i=1>, rcv proto=2 /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} tm [t_append_branches.c:149]: t_append_branches(): added branch [sip:777777777@95.165.146.181:63618;transport=TCP;rinstance=8b1865b95767b7ba] with ruid [uloc-5dd2faa0-26e5-1] /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} tm [t_append_branches.c:163]: t_append_branches(): Call uvAzTGsEV9N2ReDjmWoo8A..: 0 (0) outgoing branches after clear_branches() /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} <core> [core/tcp_main.c:2238]: tcpconn_send_put(): send from reader (9957 (22)), reusing fd /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} <core> [core/tcp_main.c:2473]: tcpconn_do_send(): sending... -- Oleg Podguyko

2 2

TLS v 1.3 failed call
by Andrew Chen 19 Nov '19

19 Nov '19

Hi guys, I was wondering if someone can help decipher what these few lines mean? Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21501]: DEBUG: tls [tls_server.c:431]: tls_accept(): tls_accept: new connection from 10.94.98.18:51698 using TLSv1.3 TLS_AES_256_GCM_SHA384 256 Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21501]: DEBUG: tls [tls_server.c:434]: tls_accept(): tls_accept: local socket: 206.81.191.45:443 Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21501]: DEBUG: tls [tls_server.c:445]: tls_accept(): tls_accept: client did not present a certificate Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21501]: DEBUG: tls [tls_server.c:1189]: tls_read_f(): Reading on a renegotiation of connection (n:1652) (0) Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21501]: DEBUG: <core> [core/tcp_read.c:1527]: tcp_read_req(): EOF Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21501]: DEBUG: <core> [core/io_wait.h:602]: io_watch_del(): DBG: io_watch_del (0x56367f8b8f80, 12, -1, 0x10) fd_no=2 called Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21501]: DEBUG: <core> [core/tcp_read.c:1680]: release_tcpconn(): releasing con 0x7fcfc20c57b0, state -1, fd=12, id=23 ([10.94.98.18]:51698 -> [10.94.98.18]:443) Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21501]: DEBUG: <core> [core/tcp_read.c:1684]: release_tcpconn(): extra_data 0x7fcfc2089f88 Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21517]: DEBUG: <core> [core/tcp_main.c:3320]: handle_tcp_child(): reader response= 7fcfc20c57b0, -1 from 16 Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21517]: DEBUG: tls [tls_server.c:683]: tls_h_close(): Closing SSL connection 0x7fcfc2089f88 Bottom line is kamailio closed the connection and I have method setting to this: root@sjomainkama51:/etc/kamailio # grep method tls.cfg method = TLSv1.1+ Supposedly this should work? -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 2

Fwd: Websocket immediately disconnecting
by Tom D 19 Nov '19

19 Nov '19

Gentlemen, Thanks for your feedback. Yes, I've doubled the RAM last night (between my posts) and I am monitoring the behavior. It *feels* that it is definitely working better. I haven't enabled memjoin yet - are there any repercussions that should be expected for enabling memjoin=yes? Thanks for your guidance. Tom

3 2

Websocket immediately disconnecting
by Tom D 19 Nov '19

19 Nov '19

Hi David, Thanks for your reply. After increasing the verbosity of the debugging, I found the following entries. So am I understanding the log correctly: the websocket module is trying to get some free memory (from Kamilio's general pool?). It doesn't receive any yet proceeds with parsing the response (I get this same parsing error even when it's working) and ultimately, it doesn't register the websocket? I appreciate your (or anyone's) thoughts. Regards, Tom Nov 19 02:44:44 MY-HOSTNAME /usr/sbin/kamailio[14928]: CRITICAL: tls [tls_domain.c:737]: sr_ssl_ctx_info_callback(): SSL handshake started Nov 19 02:44:44 MY-HOSTNAME /usr/sbin/kamailio[14928]: CRITICAL: tls [tls_domain.c:749]: sr_ssl_ctx_info_callback(): SSL handshake done Nov 19 02:44:44 MY-HOSTNAME /usr/sbin/kamailio[14928]: CRITICAL: tls [tls_domain.c:753]: sr_ssl_ctx_info_callback(): SSL disable renegotiation Nov 19 02:44:44 MY-HOSTNAME /usr/sbin/kamailio[14928]: CRITICAL: tls [tls_server.c:431]: tls_accept(): tls_accept: new connection from HI.DD.E.N:63356 using TLSv1/SSLv3 AES256-GCM-SHA384 256 Nov 19 02:44:44 MY-HOSTNAME /usr/sbin/kamailio[14928]: CRITICAL: tls [tls_server.c:434]: tls_accept(): tls_accept: local socket: HI.DD.E.N:8089 Nov 19 02:44:44 MY-HOSTNAME /usr/sbin/kamailio[14928]: CRITICAL: tls [tls_server.c:445]: tls_accept(): tls_accept: client did not present a certificate Nov 19 02:44:45 MY-HOSTNAME /usr/sbin/kamailio[14928]: CRITICAL: <script>: 857:HTTP Request Received Nov 19 02:44:45 MY-HOSTNAME /usr/sbin/kamailio[14928]: ERROR: <core> [core/mem/q_malloc.c:291]: qm_find_free(): qm_find_free(0x7f8b8cdcd000, 65728); Free fragment not found! Nov 19 02:44:45 MY-HOSTNAME /usr/sbin/kamailio[14928]: ERROR: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7f8b8cdcd000, 65728) called from websocket: ws_conn.c: wsconn_add(197), module: websocket; Free fragment not found! Nov 19 02:44:45 MY-HOSTNAME /usr/sbin/kamailio[14928]: ERROR: websocket [ws_conn.c:199]: wsconn_add(): allocating shared memory Nov 19 02:44:45 MY-HOSTNAME /usr/sbin/kamailio[14928]: ERROR: <core> [core/parser/parse_fline.c:262]: parse_first_line(): parse_first_line: bad message (offset: 22) Nov 19 02:44:45 MY-HOSTNAME /usr/sbin/kamailio[14928]: ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 101 Switching Protocols#015#012Sia: SIP/2.0/TLS HI.DD.E.N:63356#015#012Access-Control-Allow-Origin: https://www.doubango.org#015#012Sec-WebSocket-Protocol: sip#015#012Upgrade: websocket#015#012Connection: upgrade#015#012Sec-WebSocket-Accept: taS0s3Z7t4tOhklSbJDLGmSSYsY=#015#012Server: kamailio (5.2.5 (x86_64/linux))#015#012Content-Length: 0#015#012#015#012> Nov 19 02:44:45 MY-HOSTNAME /usr/sbin/kamailio[14928]: ERROR: websocket [ws_frame.c:583]: ws_frame_receive(): WebSocket connection not found

3 3

Asterisk text messages not delivered to SIP client
by Adesh Pandey 19 Nov '19

19 Nov '19

Hi Guys, I have been using Kamailio as dispatcher but I have been facing a weird situation that when asterisk SendText command says that message was delivered successfully but webrtc sip client did not receive the message. My deployment has Kamailio as registrar where sip users are registered and when they make any outgoing call, the call is proxied by RTPEngine and then dispatched to Asterisk boxes. Asterisk boxes send some text to the client using SendText but the message is not received by the SIP client. My config as follows: https://gist.github.com/adeshpandey/2c338db7f5d992267f7415de0325e0b4

1 0

Kamailio meeting Istanbul and Cairo
by Henning Westerholt 19 Nov '19

19 Nov '19

Hello, I will be in about 2 weeks in Istanbul and in Cairo a few days later. I am thinking of organizing a Kamailio/SIP/VoIP social networking event (perhaps a dinner or a pub meeting). Possible dates are the 19th- 20th November in Istanbul and the 23th - 24th November in Cairo. Are people from the community interested in joining such event? If yes, write me your preferred date -- based on feedback in the next days, it will be decided whether it is going to happen and when. Best regards, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

2 2

Re: [SR-Users] call disconnected in 45 min
by Liu, Charles 18 Nov '19

18 Nov '19

Hi All, I have install kamailio IMS using the example config files for P/S/I-cscf with version 5.2... call randomly disconnected between 15 to 45 minutes. Wireshark log show scscf sent BYE to the caller and callee. Any idea what maybe the problem? Scscf debug output ......... ............ 20(25) INFO: ims_registrar_scscf [cxdx_avp.c:138]: cxdx_get_avp(): cxdx_get_experimental_result_code: Failed finding avp 20(25) INFO: ims_registrar_scscf [cxdx_sar.c:84]: create_return_code(): created AVP successfully : [saa_return_code] - [1] 13(18) DEBUG: ims_dialog [dlg_handlers.c:1923]: print_all_dlgs(): ********************13(18) DEBUG: ims_dialog [dlg_handlers.c:1924]: print_all_dlgs(): printing 4096 dialogs 13(18) DEBUG: ims_dialog [dlg_handlers.c:1862]: internal_print_all_dlg(): Trying to get lock for printing 13(18) DEBUG: ims_dialog [dlg_handlers.c:1868]: internal_print_all_dlg(): ----------------------------13(18) DEBUG: ims_dialog [dlg_handlers.c:1869]: internal_print_all_dlg(): Dialog h_entry:h_id = [3396 : 2798] 13(18) DEBUG: ims_dialog [dlg_handlers.c:1870]: internal_print_all_dlg(): Dialog age: 2749 13(18) DEBUG: ims_dialog [dlg_handlers.c:1871]: internal_print_all_dlg(): Dialog call-id: ATokIPEjouNpuwSo6MpHUw.. 13(18) DEBUG: ims_dialog [dlg_handlers.c:1872]: internal_print_all_dlg(): Dialog state: 4 13(18) DEBUG: ims_dialog [dlg_handlers.c:1873]: internal_print_all_dlg(): Dialog ref counter: 2 13(18) DEBUG: ims_dialog [dlg_handlers.c:1874]: internal_print_all_dlg(): Dialog did: 13(18) DEBUG: ims_dialog [dlg_handlers.c:1875]: internal_print_all_dlg(): Dialog from_tag: 8975d611 13(18) DEBUG: ims_dialog [dlg_handlers.c:1876]: internal_print_all_dlg(): Dialog from_uri: sip:bob@ims.mnc001.mcc001.3gppnetwork.org;transport=UDP 13(18) DEBUG: ims_dialog [dlg_handlers.c:1877]: internal_print_all_dlg(): Dialog caller contact: sip:bob@192.168.164.23:39180;transport=UDP 13(18) DEBUG: ims_dialog [dlg_handlers.c:1878]: internal_print_all_dlg(): Dialog first request cseq: 1 13(18) DEBUG: ims_dialog [dlg_handlers.c:1879]: internal_print_all_dlg(): Dialog caller route set: <sip:mo@10.1.1.1;lr=on;ftag=8975d611;rm=8> 13(18) DEBUG: ims_dialog [dlg_handlers.c:1880]: internal_print_all_dlg(): Dialog lifetime: 43200 13(18) DEBUG: ims_dialog [dlg_handlers.c:1881]: internal_print_all_dlg(): Dialog bind_address: udp:172.26.1.60:6060 13(18) DEBUG: ims_dialog [dlg_handlers.c:1882]: internal_print_all_dlg(): Dialog-creating transaction pointer [0x7f8bb2a51858] 13(18) DEBUG: ims_dialog [dlg_handlers.c:1888]: internal_print_all_dlg(): ----------13(18) DEBUG: ims_dialog [dlg_handlers.c:1889]: internal_print_all_dlg(): Dialog out h_entry:h_id = [3296 : 0] 13(18) DEBUG: ims_dialog [dlg_handlers.c:1890]: internal_print_all_dlg(): Dialog out did: 13(18) DEBUG: ims_dialog [dlg_handlers.c:1891]: internal_print_all_dlg(): Dialog out to_tag: c9eb5e55 13(18) DEBUG: ims_dialog [dlg_handlers.c:1892]: internal_print_all_dlg(): Dialog out caller cseq: 1 13(18) DEBUG: ims_dialog [dlg_handlers.c:1893]: internal_print_all_dlg(): Dialog out callee cseq: 1 13(18) DEBUG: ims_dialog [dlg_handlers.c:1894]: internal_print_all_dlg(): Dialog out callee contact: sip:alice@192.168.164.84:49153;alias=192.168.164.84~49153~1 13(18) DEBUG: ims_dialog [dlg_handlers.c:1895]: internal_print_all_dlg(): Dialog out callee route set: <sip:mt@172.26.1.60:6060;lr;ftag=8975d611;did=44d.eea>,<sip:mt@10.1.1.1;lr;ftag=8975d611;rm=7> 13(18) DEBUG: ims_dialog [dlg_handlers.c:1896]: internal_print_all_dlg(): Dialog out branch: z9hG4bKb7d5.6f11b9b34a8a493a3a1b7949831e2f47.0 13(18) DEBUG: ims_dialog [dlg_handlers.c:1897]: internal_print_all_dlg(): Dialog out state (deleted): 0 13(18) DEBUG: ims_dialog [dlg_handlers.c:1899]: internal_print_all_dlg(): ----------13(18) DEBUG: ims_dialog [dlg_handlers.c:1903]: internal_print_all_dlg(): Releasing lock for dlgout 13(18) DEBUG: ims_dialog [dlg_handlers.c:1906]: internal_print_all_dlg(): ----------------------------13(18) DEBUG: ims_dialog [dlg_handlers.c:1934]: print_all_dlgs(): ********************13(18) DEBUG: ims_auth [authorize.c:187]: reg_await_timer(): Looking for expired/useless at 44469727 13(18) DEBUG: ims_auth [authorize.c:192]: reg_await_timer(): Slot 121 <alice(a)ims.mnc001.mcc001.3gppnetwork.org<mailto:alice@ims.mnc001.mcc001.3gppnetwork.org>> 13(18) DEBUG: ims_auth [authorize.c:197]: reg_await_timer(): .. AV 1 - 3 Exp 44469993 0x7f8bb2a5c988 13(18) DEBUG: ims_auth [authorize.c:192]: reg_await_timer(): Slot 914 <bob(a)ims.mnc001.mcc001.3gppnetwork.org<mailto:bob@ims.mnc001.mcc001.3gppnetwork.org>> 13(18) DEBUG: ims_auth [authorize.c:197]: reg_await_timer(): .. AV 1 - 3 Exp 44470022 0x7f8bb2a54e88 13(18) DEBUG: ims_auth [authorize.c:197]: reg_await_timer(): .. AV 1 - 2 Exp 44469974 0x7f8bb2a506a8 13(18) DEBUG: ims_auth [authorize.c:203]: reg_await_timer(): ... dropping av 1 - 2 13(18) DEBUG: ims_auth [authorize.c:232]: reg_await_timer(): [DONE] Looking for expired/useless at 44469727 13(18) DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7f8bb2a551f8 with 1 -> 3 13(18) DEBUG: ims_dialog [dlg_hash.c:889]: lookup_dlg(): dialog id=2798 found on entry 3396 13(18) DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7f8bb2a551f8 with 1 -> 2 13(18) DEBUG: ims_dialog [dlg_req_within.c:379]: dlg_terminate(): terminating confirmed dialog 13(18) DEBUG: ims_dialog [dlg_req_within.c:272]: send_bye(): sending BYE to caller 13(18) DEBUG: ims_dialog [dlg_hash.c:1047]: ref_dlg(): ref dlg 0x7f8bb2a551f8 with 1 -> 3 13(18) DEBUG: ims_dialog [dlg_req_within.c:289]: send_bye(): BYE sent to caller 13(18) DEBUG: ims_dialog [dlg_req_within.c:272]: send_bye(): sending BYE to callee 13(18) DEBUG: ims_dialog [dlg_hash.c:1047]: ref_dlg(): ref dlg 0x7f8bb2a551f8 with 1 -> 4 13(18) DEBUG: ims_dialog [dlg_req_within.c:289]: send_bye(): BYE sent to callee 1(6) ERROR: <script>: BYE (sip:bob@ims.mnc001.mcc001.3gppnetwork.org;transport=UDP (172.26.1.60:6060) to sip:alice@ims.mnc001.mcc001.3gppnetwork.org, ATokIPEjouNpuwSo6MpHUw..) 1(6) DEBUG: ims_dialog [dlg_handlers.c:844]: dlg_onroute(): route param is '44d.eea' (len=7) 1(6) DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7f8bb2a551f8 with 1 -> 5 1(6) DEBUG: ims_dialog [dlg_hash.c:889]: lookup_dlg(): dialog id=2798 found on entry 3396 1(6) DEBUG: ims_dialog [dlg_hash.h:459]: match_dialog(): searching dlg_outs 1(6) DEBUG: ims_dialog [dlg_hash.h:496]: match_dialog(): p: 0x7f8bb2a68c48 ft[8975d611] tt [c9eb5e55] 1(6) DEBUG: ims_dialog [dlg_hash.h:520]: match_dialog(): dout: tt[c9eb5e55] 1(6) DEBUG: ims_dialog [dlg_profile.c:419]: set_current_dialog(): setting current dialog [3396:2798] 1(6) DEBUG: ims_dialog [dlg_hash.c:1293]: next_state_dlg(): dialog 0x7f8bb2a551f8 changed from state 4 to state 6, due event 7 CONFIDENTIALITY NOTICE: This email and any attachments are for the sole use of the intended recipient and may contain material that is proprietary, confidential, privileged or otherwise legally protected or restricted under applicable government laws. Any review, disclosure, distributing or other use without expressed permission of the sender is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies without reading, printing, or saving.

1 0

Re: [SR-Users] Saving REGISTER status in kamailio database while REGSITER packets are forwarded.
by Voip support 18 Nov '19

18 Nov '19

I figured this finally out and all works for me very well. In usrloc dbmode i set value of 3 (its saving immediately changes into DB). In request route for REGISTER method i set t_on_reply("MANAGE_REPLY_REGISTER" and for REGISTER with expires set to 0 i set t_on_reply("MANAGE_REPLY_UNREGISTER") (used pseudovariable $expires(max)). I stored Contact header in $avp(contact) for unREGISTER request for further use when calling unregister(). Then in onreply_route for MANAGE_REPLY_REGISTER i call save lcoation and in MANAGE_REPLY_UNREGISTER for reply code 200 i call unregister function with the stored contact (i had to remove < > and display name using re.subst). So far it looks to work perfect. pon., 18 lis 2019 o 13:38 Voip support <voipexpert0(a)gmail.com> napisał(a): > > Thanks again for all replies... I need to try it out - did not have time > for it yet. > > I think i will go for the onreply_route answer and try to save location in > this place. > > pon., 18 lis 2019 o 06:23 Joel Serrano <joel(a)textplus.com> napisał(a): > >> Yes sir! Works perfectly! >> >> To give you a quick example: >> >> 1) Add in the appropriate place for the REGISTER request: >> t_on_reply("MANAGE_REG_REPLY"); >> 2) Create a new onreply route: >> >> # Manage registration incoming replies >> onreply_route[MANAGE_REG_REPLY] { >> if(status=~"2[0-9][0-9]") { >> if (save("location", "0x04")) { >> xlog("L_NOTICE", "[usrloc] saved user location\n"); >> } >> } >> } >> >> >> Note1: In my setup, I needed to strictly simulate asterisk (chan_sip) >> registrations, so only one AoR was allowed, thus the flags 0x04, if you are >> using PJSIP you might not need it. >> Note2: With this example code, a "unregister" (REGISTER with expires=0) >> will not remove the AoR from Kamailio, to achieve that I check >> independently if the header Expires exists and is 0, or if the contact >> param Expires exists and is 0, and in that case I use the unregister() >> function instead of the save(). >> >> Hope it helps! >> Joel. >> >> >> >> On Sun, Nov 17, 2019 at 5:42 PM Ovidiu Sas <osas(a)voipembedded.com> wrote: >> >>> Did it work calling save() on the reply route? >>> >>> -ovidiu >>> >>> On Thu, Nov 14, 2019 at 6:20 PM Voip support <voipexpert0(a)gmail.com> >>> wrote: >>> > >>> > I am using kamailio 5.3. >>> > >>> > Ovidiu Sas, i will try this tomorrow today is bit late. Maybe it could >>> work. >>> > >>> > David Villasmil , i know i can use SQLOPS but i was thinking of easier >>> method of doing this. >>> > By successful REGISTER i mean save only REGISTER attempt that >>> authenticated correctly at SIP server and in 200 OK a binding was returned. >>> > Same if the binding is later removed i would like remove it from >>> database. I need this because when i run REGISTER over kamailio to >>> destination SIP server i see all register from same IP. >>> > I need to be able to check what was the real source IP of client >>> REGISTER. >>> > >>> > Regards, >>> > Tomasz >>> > >>> > >>> > >>> > pt., 15 lis 2019 o 00:16 Mike from ApoloCore <mike(a)apolocore.com> >>> napisał(a): >>> >> >>> >> I am using kamailio 5.3. >>> >> >>> >> Ovidiu Sas, i will try this tomorrow today is bit late. Maybe it >>> could work. >>> >> >>> >> David Villasmil , i know i can use SQLOPS but i was thinking of >>> easier method of doing this. >>> >> By successful REGISTER i mean save only REGISTER attempt that >>> authenticated correctly at SIP server and in 200 OK a binding was returned. >>> >> Same if the binding is later removed i would like remove it from >>> database. I need this because when i run REGISTER over kamailio to >>> destination SIP server i see all register from same IP. >>> >> I need to be able to check what was the real source IP of client >>> REGISTER. >>> >> >>> >> Regards, >>> >> Tomasz >>> >> >>> >> >>> >> >>> >> >>> >> czw., 14 lis 2019 o 23:33 Ovidiu Sas <osas(a)voipembedded.com> >>> napisał(a): >>> >>> >>> >>> Try to use save() when you process the 200 ok reply for REGISTER in >>> >>> the reply route. >>> >>> >>> >>> Regards, >>> >>> Ovidiu Sas >>> >>> >>> >>> On Thu, Nov 14, 2019 at 3:40 PM Voip support <voipexpert0(a)gmail.com> >>> wrote: >>> >>> > >>> >>> > I setup kamailio to forward REGISTER packets using dispatcher to >>> two servers main and backup. >>> >>> > >>> >>> > The routing logic is quite simple i use the dispatcher probing >>> mechanism to detect if my main dispatcher set id (one server) is active or >>> inactive once it detect a failure all packets are sent to secondary SIP >>> server to achieve High Availability. >>> >>> > >>> >>> > I am looking for module or a way to save into kamailio database >>> client information such as IP address, register username (from FROM >>> header), digest username (authorization username), call-id for successful >>> registration. >>> >>> > >>> >>> > I tried to use registrar module but once i used function >>> save("location") the client was registered on kamailio and packets was not >>> forwarded to main/secondary SIP server (like dispatcher stopped working, >>> the destination did not override). >>> >>> > >>> >>> > REGISTER is sent to SIP URI of kamailio. >>> >>> > >>> >>> > Please let me know if you have any idea how to make this data >>> update in database (insert an entry on fresh successfull REGISTER, delete >>> on expired or removed binding). >>> >>> > >>> >>> > Best regards, >>> >>> > Tomasz >>> >>> > >>> >>> > _______________________________________________ >>> >>> > Kamailio (SER) - Users Mailing List >>> >>> > sr-users(a)lists.kamailio.org >>> >>> > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >>> >>> >>> >>> >>> >>> -- >>> >>> VoIP Embedded, Inc. >>> >>> http://www.voipembedded.com >>> >>> >>> >>> _______________________________________________ >>> >>> Kamailio (SER) - Users Mailing List >>> >>> sr-users(a)lists.kamailio.org >>> >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> > >>> > _______________________________________________ >>> > Kamailio (SER) - Users Mailing List >>> > sr-users(a)lists.kamailio.org >>> > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >>> >>> -- >>> VoIP Embedded, Inc. >>> http://www.voipembedded.com >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> sr-users(a)lists.kamailio.org >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >>

1 0

Websocket immediately disconnecting
by Tom D 18 Nov '19

18 Nov '19

Hi Everyone, We're running Kamailio 5.2.5 with XHTTP, Websocket and TLS extensions - for the purpose of being able to provide WebRTC capability. The settings for the HTTP request are quite default as set out below. The SIP client we're using is SIP5ML in different web browsers. After a successful start of Kamailio, it is possible to use SIP5ML and its features - i.e. it REGISTER's properly, calls can be made, etc. After about a few hours of running, Kamailio refuses to interact and abruptly closes the TCP connection. The only solution (discovered so far) is to restart Kamailio, which is not feasible on a server that is serving production needs. (On our development server, where the load is only a fraction - this problem does not occur). When the problems does start to occur, in Wireshark I see: 1) Normal TLS negotiation and certificate exchange; followed by 2) SIP5ML initiating the initial HTTP GET 3) Kamailio responds to the upgrade with a 101/Switch Protocols. (4- SIP5ML outputs a SIP REGISTER on a defunct socket as its was closed) Immediately after the 101/Switching Protocol message is received, Kamailio closes down the TCP connection with a FIN message. Nothing in the message or Kamailio log files help to identify what module is responsible for this. Would anyone have any advise where to look further? BTW, I don't see the TCP connection being Reset in the exchanges - so any advise will be appreciated. Tom event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != 8089) { xlogl("L_CRIT", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden - 0x24", "", ""); exit; } xlogl("L_CRIT", "HTTP Request Received\n"); if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm=~"GET") { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xlogl("L_CRIT", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden - 0x25", "", ""); exit; } # ws_handle_handshake() exits (no further configuration file # processing of the request) when complete. if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("404", "Resource Not Found - 0x27", "", ""); } event_route[websocket:closed] { xlogl("L_CRIT", "WebSocket connection from $si:$sp has closed\n"); }

2 1

Kamailio stop responding to AWS R53 health check
by Andrew Chen 18 Nov '19

18 Nov '19

Hi all, R53 reported healthcheck failure on tcp port 5060. Looking at the system, all the ports that kamailio listens on can't be monitored remotely. To test this theory, I can telnet successfully to consul port. I ran tcpdump and see no responses returned back to R53 monitoring hosts: tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on ens5, link-type EN10MB (Ethernet), capture size 262144 bytes 19:11:49.368378 IP ec2-54-241-32-75.us-west-1.compute.amazonaws.com.10072 > sjoprodkama51.fuzemeeting.com.sip: Flags [S], seq 1252984048, win 29200, options [mss 1460,sackOK,TS val 3770272726 ecr 0,nop,wscale 7], length 0 19:11:49.558877 IP ec2-107-23-255-43.compute-1.amazonaws.com.60794 > sjoprodkama51.fuzemeeting.com.sip: Flags [S], seq 3484910496, win 29200, options [mss 1460,sackOK,TS val 1780607162 ecr 0,nop,wscale 7], length 0 19:11:50.572870 IP ec2-107-23-255-43.compute-1.amazonaws.com.60794 > sjoprodkama51.fuzemeeting.com.sip: Flags [S], seq 3484910496, win 29200, options [mss 1460,sackOK,TS val 1780608175 ecr 0,nop,wscale 7], length 0 Fixed the issue after restarting kamailio process. Given this system is support production traffic, we have minimized the logging on it. The events that we thought may lead to this is as follows: Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd960380 r: 0x7f33fd960400 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda01dc0 r: 0x7f33fda01e40 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20310]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20310]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdd829e8 r: 0x7f33fdd82a68 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdb70738 r: 0x7f33fdb707b8 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20310]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20310]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20310]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdd829e8 r: 0x7f33fdd82a68 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda01dc0 r: 0x7f33fda01e40 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdaec8f0 r: 0x7f33fdaec970 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd93e6c0 r: 0x7f33fd93e740 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdaec8f0 r: 0x7f33fdaec970 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20307]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20307]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd960380 r: 0x7f33fd960400 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd960380 r: 0x7f33fd960400 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20307]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20307]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdaec8f0 r: 0x7f33fdaec970 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20307]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20307]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdd829e8 r: 0x7f33fdd82a68 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20309]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20309]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20309]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20309]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda69958 r: 0x7f33fda699d8 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20309]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20309]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdd829e8 r: 0x7f33fdd82a68 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda25fd8 r: 0x7f33fda26058 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20311]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20311]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda69958 r: 0x7f33fda699d8 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20309]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20309]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdd829e8 r: 0x7f33fdd82a68 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20311]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20311]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda25fd8 r: 0x7f33fda26058 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20311]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20311]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda25fd8 r: 0x7f33fda26058 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20311]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20311]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda25fd8 r: 0x7f33fda26058 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20312]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20312]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20312]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20312]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda25fd8 r: 0x7f33fda26058 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20312]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20312]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdbd5d78 r: 0x7f33fdbd5df8 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdb822d0 r: 0x7f33fdb82350 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdb822d0 r: 0x7f33fdb82350 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda01dc0 r: 0x7f33fda01e40 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdaec8f0 r: 0x7f33fdaec970 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda25fd8 r: 0x7f33fda26058 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdacac30 r: 0x7f33fdacacb0 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdaec8f0 r: 0x7f33fdaec970 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20317]: BUG: tls [tls_server.c:1303]: tls_read_f(): IO error (0) Success Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20317]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdacac30 r: 0x7f33fdacacb0 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdbb40b8 r: 0x7f33fdbb4138 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda01dc0 r: 0x7f33fda01e40 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdaa8f70 r: 0x7f33fdaa8ff0 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdb70738 r: 0x7f33fdb707b8 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdc7f490 r: 0x7f33fdc7f510 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20317]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20317]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdb70738 r: 0x7f33fdb707b8 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdbb40b8 r: 0x7f33fdbb4138 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdacac30 r: 0x7f33fdacacb0 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd960380 r: 0x7f33fd960400 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda01dc0 r: 0x7f33fda01e40 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda01dc0 r: 0x7f33fda01e40 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd960380 r: 0x7f33fd960400 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdacac30 r: 0x7f33fdacacb0 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20307]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdaa8f70 r: 0x7f33fdaa8ff0 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20307]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd93e6c0 r: 0x7f33fd93e740 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdb70738 r: 0x7f33fdb707b8 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20315]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20315]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdb822d0 r: 0x7f33fdb82350 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20316]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20316]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda25fd8 r: 0x7f33fda26058 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdc196f8 r: 0x7f33fdc19778 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Any thoughts on what would cause this issue? Thanks -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

4 14

Re: [SR-Users] Parsing Contact Port Field
by Michael Iedema 18 Nov '19

18 Nov '19

> Oops. No, it’s not a bug, it’s just that the URI isn’t the only value present in $ct. > > Try $(ct{nameaddr.uri}{uri.host}) Yup, that fixed it. Thanks again! I have another question about how I’m applying these to some routing logic but I’ll start a new thread for that in a bit.

1 0

http_async_client and HTTP2
by Igor Olhovskiy 18 Nov '19

18 Nov '19

HI! Seems to be, http_async client is not compatible with http2 servers. I'm having like kamailio_1 | 15(20) INFO: http_async_client [http_multi.c:238]: debug_cb(): [cURL] Connection #0 to host relay-api.pbxservice.com left intact kamailio_1 | 15(20) ERROR: <core> [core/parser/parse_fline.c:264]: parse_first_line(): parse_first_line: bad message (offset: 11) kamailio_1 | 15(20) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<HTTP/2 404 kamailio_1 | date: Thu, 14 Nov 2019 18:31:34 GMT .... kamailio_1 | 15(20) ERROR: <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [(null)] kamailio_1 | 15(20) ERROR: <core> [core/parser/msg_parser.c:1274]: get_body(): failed to parse to end of headers But plain http_client worked good (not async for sure) kamailio 5.3.0 -- Best regards, Igor

3 3

Problem with URI transformations
by Duarte Rocha 18 Nov '19

18 Nov '19

Greetings, I'm trying to parse some URI info on Kamailio but i'm having some troubles the URI transformations. Here i have two examples of URIS from which i'm trying to extract username(with user parameters included) and URI Parameters : - sip:123456789;test=param@1.1.1.1 - sip:123456789;test=param@1.1.1.1;user=phone In order to test this i made the following code : $var(URI) = "sip:123456789;test=param@1.1.1.1"; xerr("DEBUG 1 : URI $var(URI)"); $var(User) = $(var(URI){uri.user}); $var(Parameters) = $(var(URI){uri.params}); xerr("DEBUG 1 : User = $var(User)"); xerr("DEBUG 1 : Parameters = $var(Parameters)"); $var(URI) = "sip:123456789;test=param@1.1.1.1;user=phone"; xerr("DEBUG 2 : URI $var(URI)"); ; $var(User) = $(var(URI){uri.user}); $var(Parameters) = $(var(URI){uri.params}); xerr("DEBUG 2 : User = $var(User)"); xerr("DEBUG 2 : Parameters = $var(Parameters)"); The result is the following : DEBUG 1 : URI sip:123456789;test=param@1.1.1.1 DEBUG 1 : User = 123456789;test=param DEBUG 1 : Parameters = DEBUG 2 : URI sip:123456789;test=param@1.1.1.1;user=phone DEBUG 2 : User = 123456789 DEBUG 2 : Parameters = user=phone If the URI has parameters, the User parameters won't be included in the User part. Is this a bug or is it working as intended? I'm running Kamailio 5.2.0 . Thank you for your time. Best Regards, Duarte Rocha

3 6

Parsing Contact Port Field
by Michael Iedema 18 Nov '19

18 Nov '19

Hello everyone, I have a potentially silly beginners question: how can I parse the contact port field and act on it in my routing logic? I know that the $ct variable contains the entire contact header and I can print it in an xlog() call. However, I’d like to do something like the following in my routing logic: . . . SIP CONTENT Contact: <sip:someaccount@192.168.86.104:5062>;expires=1800 . . . CONFIG LOGIC #!define PORT_SERVICE1 5061 #!define PORT_SERVICE2 5062 route { if ( src_port == PORT_SERVICE1 ) { xlog(“L_INFO”, “Received $rm from SERVICE1\n”); } else if ( src_port == PORT_SERVICE2 ) { xlog(“L_INFO”, “Received $rm from SERVICE2\n”); } } I realize that src_port is not the correct value to compare against. I want to compare against the originating contact’s port value. How can I extract the port field from $ct? Many thanks in advance and apologies for the beginners question. I’ve googled for what I think I’m trying to do without any real results on functions or tokenizers, etc. I’m using Kamailio 5.3.0. Regards, -Michael

3 3

Re: [SR-Users] Saving REGISTER status in kamailio database while REGSITER packets are forwarded.
by Voip support 18 Nov '19

18 Nov '19

I am using kamailio 5.3. Ovidiu Sas, i will try this tomorrow today is bit late. Maybe it could work. David Villasmil , i know i can use SQLOPS but i was thinking of easier method of doing this. By successful REGISTER i mean save only REGISTER attempt that authenticated correctly at SIP server and in 200 OK a binding was returned. Same if the binding is later removed i would like remove it from database. I need this because when i run REGISTER over kamailio to destination SIP server i see all register from same IP. I need to be able to check what was the real source IP of client REGISTER. Regards, Tomasz pt., 15 lis 2019 o 00:16 Mike from ApoloCore <mike(a)apolocore.com> napisał(a): > I am using kamailio 5.3. > > Ovidiu Sas, i will try this tomorrow today is bit late. Maybe it could > work. > > David Villasmil , i know i can use SQLOPS but i was thinking of easier > method of doing this. > By successful REGISTER i mean save only REGISTER attempt that > authenticated correctly at SIP server and in 200 OK a binding was returned. > Same if the binding is later removed i would like remove it from database. > I need this because when i run REGISTER over kamailio to destination SIP > server i see all register from same IP. > I need to be able to check what was the real source IP of client REGISTER. > > Regards, > Tomasz > > > czw., 14 lis 2019 o 23:33 Ovidiu Sas <osas(a)voipembedded.com> napisał(a): > >> Try to use save() when you process the 200 ok reply for REGISTER in >> the reply route. >> >> Regards, >> Ovidiu Sas >> >> On Thu, Nov 14, 2019 at 3:40 PM Voip support <voipexpert0(a)gmail.com> >> wrote: >> > >> > I setup kamailio to forward REGISTER packets using dispatcher to two >> servers main and backup. >> > >> > The routing logic is quite simple i use the dispatcher probing >> mechanism to detect if my main dispatcher set id (one server) is active or >> inactive once it detect a failure all packets are sent to secondary SIP >> server to achieve High Availability. >> > >> > I am looking for module or a way to save into kamailio database client >> information such as IP address, register username (from FROM header), >> digest username (authorization username), call-id for successful >> registration. >> > >> > I tried to use registrar module but once i used function >> save("location") the client was registered on kamailio and packets was not >> forwarded to main/secondary SIP server (like dispatcher stopped working, >> the destination did not override). >> > >> > REGISTER is sent to SIP URI of kamailio. >> > >> > Please let me know if you have any idea how to make this data update in >> database (insert an entry on fresh successfull REGISTER, delete on expired >> or removed binding). >> > >> > Best regards, >> > Tomasz >> > >> > _______________________________________________ >> > Kamailio (SER) - Users Mailing List >> > sr-users(a)lists.kamailio.org >> > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> >> >> -- >> VoIP Embedded, Inc. >> http://www.voipembedded.com >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users(a)lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >

3 2

Saving REGISTER status in kamailio database while REGSITER packets are forwarded.
by Voip support 15 Nov '19

15 Nov '19

I setup kamailio to forward REGISTER packets using dispatcher to two servers main and backup. The routing logic is quite simple i use the dispatcher probing mechanism to detect if my main dispatcher set id (one server) is active or inactive once it detect a failure all packets are sent to secondary SIP server to achieve High Availability. I am looking for module or a way to save into kamailio database client information such as IP address, register username (from FROM header), digest username (authorization username), call-id for successful registration. I tried to use registrar module but once i used function save("location") the client was registered on kamailio and packets was not forwarded to main/secondary SIP server (like dispatcher stopped working, the destination did not override). REGISTER is sent to SIP URI of kamailio. Please let me know if you have any idea how to make this data update in database (insert an entry on fresh successfull REGISTER, delete on expired or removed binding). Best regards, Tomasz

5 5

dispatcher, pvar_hash and multiple kamailio instances
by Barry Flanagan 15 Nov '19

15 Nov '19

Hi, I have multiple Kamailio instances using dispatcher using alg “7” - hash over the content of PVs string, both using the same dispatcher list. Is it reasonable to expect both Kamailio, given the same hash_pvar, to select the same host from the same set? Looking at the source I do not see any reason why not, but I am getting different results from the two Kamailio servers. The only difference between them is different minor version numbers - 4.3.4 and 4.3.7 I want to ensure that calls with the same hash_pvar value go to the same backend no matter which Kamailio processes the request. Thanks! -Barry Flanagan

3 2

Re: [SR-Users] Question on Database Version Table
by JR Richardson 15 Nov '19

15 Nov '19

Yep version_table Set the name of the table holding the table version. Useful if the proxy is sharing a database within a project and during upgrades. Default value is “version”. Example of usage: version_table="version44" That's what I was looking for, perfect! Yes, I added accountcode and notes to my table for internal use, was just making notes on columns I need on the new version upgrade. Thanks. JR On Tue, Jun 18, 2019 at 2:17 PM Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > > Hello, > > like Alex said, you can have two version tables, one for each versions > of kamailio you play with and use in config the core parameter to set > the name of that table. > > I am actually writing to say that address table never had accountcode > and notes columns. Likely they were added in your deployment for other > purposes or either the default column names were changed. The default > column names are listed at: > > - > https://kamailio.org/docs/db-tables/kamailio-db-5.2.x.html#gen-db-address > > These were the same for rather long time, iirc the recent version number > changed mainly due to with modifications done to the size, not due to > new columns. > > Cheers, > Daniel > > On 18.06.19 20:50, JR Richardson wrote: > > Hey All, > > > > I'm doing some upgrades and noticed database table version numbers > > have increased. > > > > database table adjustments: > > | address | production ver 3 | upgrade 5.2 ver 6 | need to add > > accountcode and notes column > > | dispatcher | production ver 3 | upgrade 5.2 ver 4 | need to add attr column > > | trusted | production ver 5 | upgrade 5.2 ver 6 | need to add > > ruri_pattern and priority column > > > > I run mysql master/slave so I'm concerned about the master database > > not having the correct version numbers or can I just update the > > version number in the table when I do the upgrade of the kamailio > > versions? > > > > Thanks. > > > > JR

4 6

Processes blocking - troubleshooting slow actions
by Julien Chavanton 15 Nov '19

15 Nov '19

I am troubleshooting intermittent latency on a server, several processes will report slow action at the same time every now and then, something odd it that some of the actions are most likely not using any mutexes, I can not imagine how they can end up blocking unless the process is actually suspended by the OS, however the CPU usage and load average is quite low on this server. Is there any reason why assigning to a PKG var could be delayed 3 seconds. 2019-11-13T21:58:04.018631614Z 23(42) WARNING: <core> [core/action.c:1586]: run_actions(): alert - action [corefunc (16)] cfg [/etc/kamailio/kamailio.cfg:115] took too long [2579486 us] $var(reply) = 0; // I wonder how this can endup blocking ? 2019-11-13T21:58:04.018748710Z 42(61) WARNING: <core> [core/action.c:1586]: run_actions(): alert - action [corefunc (16)] cfg [/etc/kamailio/kamailio.cfg:221] took too long [2222010 us] } // Samething here, simply exiting a scope 2019-11-13T21:58:04.015675095Z 11(30) WARNING: <core> [core/action.c:1586]: run_actions(): alert - action [t_relay_cancel (24)] cfg [/etc/kamailio/kamailio.cfg:86] took too long [2588818 us] } if (!t_relay_cancel()) { // Indeed a t_relay_cancel 2019-11-13T21:58:04.018647988Z 26(45) WARNING: <core> [core/action.c:1586]: run_actions(): alert - action [t_relay (24)] cfg [/etc/kamailio/kamailio.cfg:289] took too long [2739366 us] t_relay(); // Indeed a t_relay 2019-11-13T21:58:04.018651243Z 15(34) WARNING: <core> [core/action.c:1586]: run_actions(): alert - action [sl_send_reply (26)] cfg [/etc/kamailio/kamailio.cfg:112] took too long [2587812 us] sl_send_reply("100", "Trying"); // Indeed a sl_send_reply Not sure which mailing list is best for this question Regards Julien

2 4

kamailio 5.2.4 unable to load db_mysql.so
by Rupesh Kumar 15 Nov '19

15 Nov '19

I am trying to compile and load kamailio 5.2.4 on centos 7.5. I am getting below errors whenever I start kamailio . I have the latest openssl and openssl-devel packages installed on my centos . Not sure what other dependencies . Not starting Kamailio SIP Server: invalid configuration fil[FAILED] [FAILED]0(23235) ERROR: <core> [core/sr_module.c:498]: load_module(): could not open module </usr/local/lib64/kamailio/modules/db_mysql.so>: /usr/local/lib64/kamailio/modules/db_mysql.so: undefined symbol: SSL_get_error 0(23235) CRITICAL: <core> [core/cfg.y:3508]: yyerror_at(): parse error in config file //usr/local/etc/kamailio/kamailio.cfg, line 229, column 12-24: failed to load module ERROR: bad config file (1 errors) 0(23235) INFO: <core[FAILED]sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized [FAILED]

1 1

Next online devel meeting - irc/video conference - Thursday, Nov 14, 15:00UTC
by Daniel-Constantin Mierla 14 Nov '19

14 Nov '19

Hello, on Thursday and Friday this week over a dozen of Kamailio developers will be in Dusseldorf for the annual coding meeting, so we thought of doing an online meeting via some video conferencing bridge, with IRC as a backup chatting channel, to allow also other community members to join remotely and discuss the development plans for short/medium term. It is more or less what we used to do during the past IRC devel meetings, but to make the discussions go faster and shorter the meeting, so people in Dusseldorf spend more time coding, this time we will try video conferencing. Very likely we will use Jitsi Meet, so anyone that wants to join should be prepared with a WebRTC capable web browser (all major ones support it) -- link and details how to join will be provided short before the meeting starts, right now the plan is to do it on Thursday, Nov 14, at 15:00UTC/London (16:00Berlin-Paris, 10:00New York). If someone wants to address questions but doesn't have the possibility to join the video conf bridge, then join the IRC #kamailio channel on freenode.net and post your question there during the meeting. The IRC channel will be monitored and used for text chatting whenever needed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

2 3

Kamailio v5.3.1 Released
by Daniel-Constantin Mierla 14 Nov '19

14 Nov '19

Hello, Kamailio SIP Server v5.3.1 stable release is out. This is a maintenance release of the latest stable branch, 5.3, that includes fixes since the release of v5.3.1. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.3.x. Deployments running previous v5.3.x versions are strongly recommended to be upgraded to v5.3.1. For more details about version 5.3.1 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/11/kamailio-v5-3-1-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

1 0

Releasing Kamailio v5.3.1
by Daniel-Constantin Mierla 14 Nov '19

14 Nov '19

Hello, the plan right now is to release v5.3.1 during the next Kamailio Developers Meeting, if we succeed to coordinate on site, therefore expect it to happen next week during Nov 14-15. Besides just doing this specific release, to goal is to try to build a more automatic process for releases, document how is done, etc... updates about the results will follow after the event. Anyhow, for now the usual the pre-release notice: if anyone is aware of some issue that is not yet reported, do it asap in order to give it a chance to be analyzed and fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

1 1

event_route[core:msg-received] native vs kemi/lua
by GG GG 14 Nov '19

14 Nov '19

Hi! I just made a simple test performance between native script and kemi/lua with the event_route[core:msg-received] callback. # lua script children=1 cpu.process=100% function ksr_core_received() exit; end # native script children=1 cpu.process=70% event_route[core:msg-received] { exit; } # tested with udp 5060, 1000 Mbits/sec iperf -c 127.0.0.1 -p 5060 -u -b1000m I don't know if it's normal and what we should expect from kemi, but I would like to share this with you. Maybe someone could confirm the performance difference. Have a good day !

2 8

kemi framework design
by GG GG 12 Nov '19

12 Nov '19

Hi, looking at the documentation and the code, I couldn't answer myself. - does kamailio use one lua interpreter per fork ? - about the safety, I just made a bad lua module in c, to force a segmentation fault in the module. I would expect kamailio to protect itself from the crash, but kamailio restarted.

2 1

Change SDP Session Name
by Alvaro Araos 11 Nov '19

11 Nov '19

Hi everyone, Exists some way to change the SDP Session Name value (s=)??? Saludos, Alvaro Araos

3 2

Change SDP Session Name
by Alvaro Araos 11 Nov '19

11 Nov '19

Hi everyone, Exists some way to change the SDP Session Name value (s=)??? Saludos, Alvaro Araos

1 0

Change SDP Session Name
by Alvaro Araos 11 Nov '19

11 Nov '19

Hi everyone, Exists some way to change the SDP Session Name value (s=)??? Saludos, Alvaro Araos

1 0

Kamailio in cloud environments
by Alejandro Recarey 11 Nov '19

11 Nov '19

Hi, I am trying to get kamailio working cloud provider, autoscaling them behind a TCP load balancer. Is there a way for kamailio to discover its Public IP at startup? Or do people generally program their own startup scripts that modify the kamailio config file before starting it? Is there any gotchas you have hit while configuring this setup? Thank you! Alex

8 13

uac_replace_from(uri) alters display name
by Sergiu Pojoga 10 Nov '19

10 Nov '19

Hi ppl, Need to replace origination carrier's domain with my own, so I do: branch_route[MANAGE_PSTN_IN_BRANCH] { uac_replace_from("sip:$fU@mydomain.net"); } The dialog-forming INVITE is relayed properly with ONLY the domain part being replaced. However, subsequent messages such as ACK and BYE have the From DISPLAY NAME (_uac_fdp) stripped off. Not that it affects real functionality in any known way (CLID is still good for callee), but curious if this is intentional or not, as *uac_replace_from(uri)* clearly states that: "*Replace in FROM header the URI part without altering the display name*" # kamcmd dlg.list variables: { { _uac_fdpnew: } { _uac_fdp: "JOHN DOE" } { _uac_funew: sip:18661112222@mydomain.net } { _uac_fu: sip:18661112222@carrierdomain.net } Thanks!

2 4

OPTIONS request not authenticating
by Gustavo A. M. 10 Nov '19

10 Nov '19

Greetings, I'm trying to configure kamailio to get OPTIONS response, but it does not working properly. After a successful REGISTER, I send OPTIONS first time, get Proxy- Authenticate header to calculate Proxy-Authorization, and resend OPTIONS with Proxy-Authenticate header, but server follows answering with a 407 response, apparently ignoring Proxy-Authenticate. Note that same password is used in REGISTER request, and first time REGISTER response with a 401, not a 407 response. I think messages sent from client are right, and problem is on the server configuration. Follow messages and kamailio.cfg. Thanks in advance, Gustavo Adolfo Moennich OPTIONS sip:6000@kamailio.memorea.com SIP/2.0 Call-ID: A6DB9DCAA18049E784034FB6F5F95680 Contact: <sip:6000@kamailio.memorea.com> Content-Length: 0 CSeq: 0 OPTIONS Expires: 20 From: "Bob" <sip:6000@kamailio.memorea.com>;tag=0B3D2156FF9E46DAAD7FC43 B62C9E2FC Max-Forwards: 70 To: "Bob" <sip:6000@kamailio.memorea.com> User-Agent: Memorea 1.0 (SIP 2.0) Build 1.1.1.0, Windows x64 Build 6.1.7601 Via: SIP/2.0/UDP 192.168.122.21:5060;rport=58905 SIP/2.0 407 Proxy Authentication Required Call-ID: A6DB9DCAA18049E784034FB6F5F95680 CSeq: 0 OPTIONS From: "Bob" <sip:6000@kamailio.memorea.com>;tag=0B3D2156FF9E46DAAD7FC43 B62C9E2FC To: "Bob" <sip:6000@kamailio.memorea.com>;tag=9dd61ff61e802d8e2bef5f146 21ef3c2.467d Via: SIP/2.0/UDP 192.168.122.21:5060;rport=58905;received=192.168.122.12 Proxy-Authenticate: Digest realm="kamailio.memorea.com", nonce="Xcc1wV3HNJXdfUDchNlEUkT6kHCtqBAn" Server: kamailio (5.2.1 (x86_64/linux)) Content-Length: 0 OPTIONS sip:6000@kamailio.memorea.com SIP/2.0 Call-ID: A6DB9DCAA18049E784034FB6F5F95680 Contact: <sip:6000@kamailio.memorea.com> Content-Length: 0 CSeq: 1 OPTIONS Expires: 20 From: "Bob" <sip:6000@kamailio.memorea.com>;tag=0B3D2156FF9E46DAAD7FC43 B62C9E2FC Max-Forwards: 70 Proxy-Authorization: Digest uri=sip:6000@kamailio.memorea.com,nonce="Xc c1wV3HNJXdfUDchNlEUkT6kHCtqBAn",realm="kamailio.memorea.com",response=" 5bcd184eea53284f3ec548c2412ce36a",username=6000 To: "Bob" <sip:6000@kamailio.memorea.com> User-Agent: Memorea 1.0 (SIP 2.0) Build 1.1.1.0, Windows x64 Build 6.1.7601 Via: SIP/2.0/UDP 192.168.122.21:5060;rport=58905 SIP/2.0 407 Proxy Authentication Required Call-ID: A6DB9DCAA18049E784034FB6F5F95680 CSeq: 1 OPTIONS From: "Bob" <sip:6000@kamailio.memorea.com>;tag=0B3D2156FF9E46DAAD7FC43 B62C9E2FC To: "Bob" <sip:6000@kamailio.memorea.com>;tag=9dd61ff61e802d8e2bef5f146 21ef3c2.32de Via: SIP/2.0/UDP 192.168.122.21:5060;received=192.168.122.12;rport=58905 Proxy-Authenticate: Digest realm="kamailio.memorea.com", nonce="Xcc1xl3HNJqRobJFuZggPSGvNdtp+GnP" Server: kamailio (5.2.1 (x86_64/linux)) Content-Length: 0 kamailio.cfg content's: #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB # # Kamailio (OpenSER) SIP Server v5.2 - default configuration script # - web: https://www.kamailio.org # - git: https://github.com/kamailio/kamailio # # Direct your questions about this file to: <sr-users(a)lists.kamailio.or g> # # Refer to the Core CookBook at https://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Note: the comments can be: # - lines starting with #, but not the pre-processor directives, # which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef, # #!subst, #!substdef, ... # - lines starting with // # - blocks enclosed in between /* */ # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:temp@database.memorea.com/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ # auto_aliases=no /* add local domain aliases */ # alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to * bind on a specific interface/port/proto (default bind on all available) */ # listen=udp:10.0.0.10:5060 #!ifdef WITH_TLS enable_tls=yes #!endif /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=$si) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if (uri==$ru) { if ((method==OPTIONS) && (! uri=~"sip:.*[@]+.*")) { options_reply(); } } if(!sanity_check("17895", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == $ru) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) { sl_reply_error(); } exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==$fu) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } if (is_method("OPTIONS") || from_uri==$fu) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("OPTIONS|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=$fu && uri!=$ru) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=$ru) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==$ru) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=$fu) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif }

2 1

dispatch REGISTER over TLS
by sthustfo 10 Nov '19

10 Nov '19

I have a basic setup where kamailio receives SIP over websocket (no WSS) and forwards to SIP server over TLS. I have enabled TLS in kamailio.cfg and added dispatcher node as sips:SIP_SERVER:5061 and transport=tls. +----+-------+------------------------+-------+----------+---------------+----------------+ | id | setid | destination | flags | priority | attrs | description | +----+-------+------------------------+-------+----------+---------------+----------------+ | 4 | 1 | sips:10.0.0.100:5061 | 0 | 0 | transport=tls | SIP SERVER | +----+-------+------------------------+-------+----------+---------------+----------------+ Now when REGISTER is received over websocket, kamailio is responding with error code 500 and phrase "500 I'm terribly sorry, server error occurred (7/SL)". And on the console I see the following error messages. 12(33858) DEBUG: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} <core> [core/md5utils.c:67]: MD5StringArray(): MD5 calculated: f1ecf7bcb659b07fe81e332e100044e5 12(33858) ERROR: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} tm [ut.h:315]: uri2dst2(): no corresponding socket found for "10.0.0.100" af 2 (tls: 10.0.0.100:5061) 12(33858) ERROR: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} tm [t_fwd.c:467]: prepare_new_uac(): can't fwd to af 2, proto 3 (no corresponding listening socket) 12(33858) ERROR: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} tm [t_fwd.c:1735]: t_forward_nonack(): failure to add branches 12(33858) DEBUG: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} tm [t_funcs.c:334]: t_relay_to(): t_forward_nonack returned error -7 (-7) 12(33858) DEBUG: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} tm [t_funcs.c:352]: t_relay_to(): -7 error reply generation delayed 12(33858) exec: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} *** cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio.cfg] l=587 a=24 n=sl_reply_error 12(33858) DEBUG: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} <core> [core/msg_translator.c:162]: check_via_address(): (10.0.0.14, hsvmphm3ps12.invalid, 0) 12(33858) DEBUG: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} websocket [ws_conn.c:452]: wsconn_get(): wsconn_get for id [3] *tls.cfg contents* [client:default] method = TLSv1 verify_certificate = yes require_certificate = yes private_key = /home/test/kamailio/internal.key certificate = /home/test/kamailio/internal.crt ca_list = /home/test/kamailio/ca_list.pem Any reason why this error is seen? Any inputs appreciated. Thanks.

4 20

Re: [SR-Users] _w_t_relay_to(): t_forward_noack failed when forwarding call
by Konstantinos Merentitis 08 Nov '19

08 Nov '19

I added $du=$null; before route(RELAY) as you suggested but I get the same results: Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: <script>: transaction canceled Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: <script>: #012 RDNIS = sip:297@192.168.1.134 Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: <script>: =============forwarding to voicemail1, RDNIS: sip:297@192.168.1.134 =========== Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: tm [tm.c:1679]: _w_t_relay_to(): t_forward_noack failed Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: sl [sl_funcs.c:392]: sl_reply_error(): stateless error reply used: transaction canceled (487/SL) Regards, Konstantinos > On Fri Nov 8 10:24:06 CET 2019 Daniel Tryba wrote: > >> On Thu, Nov 07, 2019 at 04:53:43PM +0200, Konstantinos Merentitis wrote: >> >> I tried to modify the failure route in order to also forward not answered (failed?) calls to asterisk mailbox but i get the following error and the call is never forwarded: >> > You appear to do the same thing I do, except setting $du to null. If the user is REGISTERed $du will be set to their endpoint, changing $ru will have no effect on routing. So add $du=$null; in the TOVOICEMAIL route before route(RELAY) and see what happens. >

3 4

_w_t_relay_to(): t_forward_noack failed when forwarding call
by Konstantinos Merentitis 08 Nov '19

08 Nov '19

I would like to forward all incoming calls that are not answered from kamailio, to my asterisk voicemail server. - I have not setup any Realtime Integration. I am using the "Redirected Dialed Number Information Service" header and calling "554(a)192.168.1.134" which is my voicemail server. Forwarding succeeds when the subscriber is not registered to kamailio: Nov 7 16:19:45 spock /usr/sbin/kamailio[17606]: ERROR: <script>: =============User sip:6297@192.168.1.131 not found here, going to voicemail============= Nov 7 16:19:45 spock /usr/sbin/kamailio[17606]: ERROR: <script>: #012 RDNIS = sip:297@192.168.1.134 Nov 7 16:19:45 spock /usr/sbin/kamailio[17606]: ERROR: <script>: =============forwarding to voicemail1, RDNIS: sip:297@192.168.1.134 =========== I tried to modify the failure route in order to also forward not answered (failed?) calls to asterisk mailbox but i get the following error and the call is never forwarded: Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: <script>: transaction canceled Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: <script>: #012 RDNIS = sip:297@192.168.1.134 Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: <script>: =============forwarding to voicemail1, RDNIS: sip:297@192.168.1.134 =========== Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: tm [tm.c:1679]: _w_t_relay_to(): t_forward_noack failed Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: sl [sl_funcs.c:392]: sl_reply_error(): stateless error reply used: transaction canceled (487/SL) failure and voicemail routes are slightly modified from the default config: route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { xlog("=============User $ru not found here, going to voicemail=============\n"); $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { $var(RDNIS)=$di; xlog("\n RDNIS = $var(RDNIS) \n"); if($avp(oexten)==$null) return; # $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip + ";transport=udp") # + ":" + $sel(cfg_get.voicemail.srv_port); xlog("=============forwarding to voicemail1, RDNIS: $di =========== "); $ru = "sip:554@192.168.1.134:5060"; } else { if($rU==$null) return; # $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip + ";transport=udp") # + ":" + $sel(cfg_get.voicemail.srv_port); xlog("=============forwarding to voicemail2, RDNIS: $di =========== "); $ru = "sip:554@192.168.1.134:5060"; } route(RELAY); exit; #!endif return; } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { #modified here xlog("transaction canceled\n"); route(TOVOICEMAIL); #exit; #/modified } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; xlog("============= Route failure, forwarding to voicemail ============="); route(TOVOICEMAIL); exit; } #!endif }

2 1

How to read some data inside the INVITE ..
by Laura 07 Nov '19

07 Nov '19

Dears.. how is possibile to read that data 9052562462354464 inside the first INVITE message === Example === U 2019/11/07 14:46:59.320936 192.168.1.1:5060 -> 192.168.1.242:5060 INVITE sip:9052562462354464@192.168.1.242 SIP/2.0. Record-Route: <sip:192.168.1.1;lr;ftag=105854496;did=24e.e751;vst=AAAAAAAAAwQAAAAAAAAAAAAAAAAADQIAAwEDAAAHDwAGBAg1MDYw;nat=yes>. Max-Forwards: 66. Via: SIP/2.0/UDP 51.254.158.40:5060;branch=z9hG4bK501c.87a8e402c8f1cf93192ddd8e6744da57.1. From: "38617774738" <sip:38617774738@10.1.1.1>;tag=105854496. To: <sip:562462354464@192.168.1.242>. Call-ID: 1573138018-1602381967(a)10.1.1.182. CSeq: 20215490 INVITE ..... ETC ETC... Thanks to all

4 4

Re-invite after RTPengine pacemaker switching?
by Жан Базаров 07 Nov '19

07 Nov '19

I need to send re-invite after pacemaker fails over on new rtpengine server. Because new rtpengine dont participate in DTLS handshake and i hear nothing, but silence. I think, may me its would be work. Do you have any idea on this issue?

3 5

multihomed kamailio and sip-reinvites SRTP to RTP
by Karsten Horsmann 07 Nov '19

07 Nov '19

Hello Mailinglist, i try to figure out, how to solve RE-INVITES and SRTP: i have an kamailio route names route[MEDIAPROXY] that i used to controll rtpengine in an private and public ip setup. This works fine for me now. It use the permission module and the adress table to figure out $var(mediagroup) = allow_address_group($Ri, 0); received from internal or external ip. The idea is from SBC-OS chamless stolen, so thanks for that. So fare so good. I adapted SRTP also to this. But on RE-INVITES it stops handling SRTP correctly to the encryption upstream. And i am sure, its my fault. But i have no clue how to detect that correctly. Any hints where i can read more about an correct handling on this? Here my main-config file / kamailio 5.3.x and rtpengine. https://gist.github.com/khorsmann/02bd98a628a190caee68fc3d876a3448 -- Mit freundlichen Grüßen *Karsten Horsmann*

2 2

Re: [SR-Users] Group module
by Daniel Tryba 07 Nov '19

07 Nov '19

On Wed, Nov 06, 2019 at 07:15:42PM +0100, Igor Olhovskiy wrote: > > Hm... Maybe there is other module to achieve such functions? > Best if it would be with cache :) > But if no - regex also fine. But if it works it works! I don't think there is a specific module to do this but it is easy to implement yourself. What I don't like personally about your solution is the use of the permission tables, you need atleast 1 per user (unless users have the same ip). Personally I do this by direct database queries in usr_preferences on INVITEs and REGISTERs. You could use any database (depending on how you store the ACL) and cache it in a htable. My solution is mysql specific (inet_aton) and forces users to have atleast 1 CIDR notated subnet: if(!avp_db_query("select value from usr_preferences where username='$au' and attribute='acl' and inet_aton(substring_index(value,'/',1))&(1 << 32) - 1 & ~((1 << (32 - substring_index(value,'/',-1))) - 1)=inet_aton('$si')&(1 << 32) - 1 & ~((1 << (32 - substring_index(value,'/',-1))) - 1)")) { t_reply("404","Go away"); exit; }

3 2

MongoDB findone(). Is projection supported?
by Lars Olsson 06 Nov '19

06 Nov '19

Hi, I am using NDB_MONGODB to ask some questions to a database. My database structure is pretty heavy, so I like to return as little data as required to Kamailio. I have tried the following MongoDB syntax for projection to include only given (my_field) column in the response. mongodb_find_one("srv_name", "database", "collection", "{ \"user_id\" : \"$var(user_id)\" }, { \"my_field\" : 1}", "mgr1") I always get full DB entry back. Is this supported? The same string works OK in the vanilla mongo client. How can I limit the result? Thanks Lars

2 3

Group module
by Igor Olhovskiy 06 Nov '19

06 Nov '19

Hi! I'm trying to solve quite simple task and want to solve it as simple as possible. Idea is to have restriction to allow register certain username/domain only from certain subnets. So, I decided to use group and permissions modules. But stuck on group one. Code is fairly simple modparam("group", "db_url", DBURL) modparam("group", "use_domain", 1) ... if (get_user_group("From", "$avp(user_groups)")) { xlog("$fu from $si:$sp is matched group $avp(user_groups)\n"); if (allow_source_address("$avp(user_groups)")) { ...allow furter } } else { xlog("$fu from $si:$sp is not matched any groups\n"); } grp table is looks like kamctl db exec 'select * from grp' +----+-------------+-------------------+-----+---------------------+ | id | username | domain | grp | last_modified | +----+-------------+-------------------+-----+---------------------+ | 1 | 61388140395 | sip.localhost.net | 5 | 2000-01-01 00:00:01 | +----+-------------+-------------------+-----+---------------------+ But actual line of debug in logs saying this kamailio_1 | 10(18) exec: {1 1 REGISTER 4oQTNaaVlEWQC2VMK6YAKg..} *** cfgtrace:request_route=[PERMISSIONS] c=[/etc/kamailio/kamailio.cfg] l=720 a=26 n=get_user_group kamailio_1 | 10(18) DEBUG: {1 1 REGISTER 4oQTNaaVlEWQC2VMK6YAKg..} group [re_group.c:188]: get_user_group(): getting groups for <sip:61388140395@sip.localhost.net> kamailio_1 | 10(18) exec: {1 1 REGISTER 4oQTNaaVlEWQC2VMK6YAKg..} *** cfgtrace:request_route=[PERMISSIONS] c=[/etc/kamailio/kamailio.cfg] l=727 a=25 n=xlog kamailio_1 | 10(18) ERROR: {1 1 REGISTER 4oQTNaaVlEWQC2VMK6YAKg..} <script>: sip:61388140395@sip.localhost.net;transport=UDP from 172.21.0.1:56787 is not matched any groups What am I missing here? Or module is too old? Thanks!

1 0

RFC: default config with jsonrpc over http instead of xmlrpc
by Daniel-Constantin Mierla 06 Nov '19

06 Nov '19

Hello, I think it was discussed during past IRC meetings or at some events across the world, with jsonrpc being these days more popular than xmlrpc, specially due to easier integration with web based apps, I am considering to replace the example of using xmlrpc over http(s) with jsonrpc in the default configuration file. It is not enabled by default for xmlrpc (requiring to define WITH_XMLRPC), so it will stay also as a define option for jsonrpc. The jsonrpcs module is already loaded for kamctl, it will require the xhttp module to be added. As for those looking for xmlrpc in the future, there is another example kamailio cfg which has the option with it: misc/examples/pkg/kamailio-oob.cfg Any other opinions? Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

4 3

$jsonrpl(body) empty or malfunctioned during higher traffic
by Michal Škuta 05 Nov '19

05 Nov '19

Hello, Kamailio community, we are implementing evapi communication to our Kamailio server. We have implemented and tested it on our testing server. Everything was working OK. Now we are testing it on our production servers with a lot more calls and we have some problems here. Every n seconds we are requesting from Kamailio list of all dialogs. And this list of dialogs (from jsonrpc *dlg.list*) is sent to evapi connection. Here is a snippet of our *Kamailio code*: route[CGR_DLG_LIST] { xlog("L_INFO", "XLOG: $ci [CGR_DLG_LIST] Requested dialog list"); # CGRateS connection is still alive, set check to true $sht(cgrconn=>check) = 1; if $sht(cgrconn=>cgr) == $null { sl_send_reply("503","Charging controller unreachable"); exit; } jsonrpc_exec('{"jsonrpc":"2.0","id":1, "method":"dlg.list","params":[]}'); xlog("L_INFO", "XLOG: $ci [CGR_DLG_LIST] Test1: $jsonrpl(body)"); evapi_relay("{\"event\":\"CGR_DLG_LIST_REPLY\", \"jsonrpl_body\":$jsonrpl(body)}"); } During a higher number of calls Kamailio is generating ERROR messages from evapi module. Here is the syslog: Oct 30 12:02:15 kam-prod /opt/ipgate/kamailio/sbin/kamailio[19512]: INFO: <script>: XLOG: 123 [CGR_DLG_LIST] Requested dialog list Oct 30 12:02:15 kam-prod /opt/ipgate/kamailio/sbin/kamailio[19512]: INFO: <script>: Oct 30 12:02:15 kam-prod /opt/ipgate/kamailio/sbin/kamailio[19512]: DEBUG: evapi [evapi_dispatch.c:740]: _evapi_relay(): relaying event data [ Oct 30 12:02:15 kam-prod /opt/ipgate/kamailio/sbin/kamailio[19512]: ERROR: evapi [evapi_dispatch.c:764]: _evapi_relay(): cannot serialize event Oct 30 12:02:15 kam-prod /opt/ipgate/kamailio/sbin/kamailio[19512]: ERROR: evapi [evapi_mod.c:265]: w_evapi_relay(): failed to relay event: *We are expecting something like this when nobody is calling* Oct 30 14:12:22 kam-test /opt/ipgate/kamailio/sbin/kamailio[32090]: INFO: <script>: XLOG: 123 [CGR_DLG_LIST] Requested dialog list Oct 30 14:12:22 kam-test /opt/ipgate/kamailio/sbin/kamailio[32090]: INFO: <script>: XLOG: 123 [CGR_DLG_LIST] Test1: { #011"jsonrpc":#011"2.0", #011"result":#011[], #011"id":#0111 } Oct 30 14:12:22 kam-test /opt/ipgate/kamailio/sbin/kamailio[32090]: DEBUG: evapi [evapi_dispatch.c:740]: _evapi_relay(): relaying event data [{"event":"CGR_DLG_LIST_REPLY", #011#011#011"jsonrpl_body":{ #011"jsonrpc":#011"2.0", #011"result":#011[], #011"id":#0111 }}] (96) Oct 30 14:12:22 kam-test /opt/ipgate/kamailio/sbin/kamailio[32090]: DEBUG: evapi [evapi_dispatch.c:778]: _evapi_relay(): sending [0x169ff674] [96:{"event":"CGR_DLG_LIST_REPLY", #011#011#011"jsonrpl_body":{ #011"jsonrpc":#011"2.0", #011"result":#011[], #011"id":#0111 }},] (100) Oct 30 14:12:22 kam-test /opt/ipgate/kamailio/sbin/kamailio[32090]: DEBUG: evapi [evapi_dispatch.c:789]: _evapi_relay(): dispatching [0x169ff674] [96:{"event":"CGR_DLG_LIST_REPLY", #011#011#011"jsonrpl_body":{ #011"jsonrpc":#011"2.0", #011"result":#011[], #011"id":#0111 }},] (100) We enabled debug messages for module evapi and are trying to understand where is the problem. To make this work to this state we needed to increase the *pv_buffer* to really big size (2097152). It looks like $jsonrpl(body) is not ready and everything concated with it is not working. Any help or suggestion is welcomed. Sincerely Michal

3 6

Module acc - How to get sip_reason/code on log_write_cdr() messages
by David VILLAUME 05 Nov '19

05 Nov '19

Hello, I'm trying to get the reason in my cdrs in log_write_cdr() messages while I see It in acc_log_request() messages. Is there a way to retrieve it in log_write_cdr() messages ? Bellow is my config : modparam("acc", "log_level", 1) modparam("acc", "log_facility", "LOG_LOCAL3") modparam("acc", "cdr_log_enable", 1) modparam("acc", "log_flag", 1) modparam("acc", "failed_transaction_flag", 2) modparam("acc", "log_missed_flag", 3) modparam("acc", "report_cancels", 1) modparam("acc", "cdr_on_failed", 1) modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_facility", "LOG_LOCAL3") modparam("acc", "cdr_expired_dlg_enable", 1) modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "cdr_extra", "call_id=$ci;src_ip=$si;fU=$fU;")" David VILLAUME

2 2

Re: [SR-Users] driver error: Can't connect to MySQL server with kamailio 5.3.x and mysql-config-groups
by Karsten Horsmann 04 Nov '19

04 Nov '19

Hello Henning, is there an planed release date for this 5.3.x version? Unfortunately without this change i cant deploy it without changing all my mysql user credentials to be sure there is no "kamailio DBURL separator" within the password field. Cheers Karsten [...] -- Mit freundlichen Grüßen *Karsten Horsmann*

3 2

kamailio - failover setup
by Youssef Boujraf 04 Nov '19

04 Nov '19

Dear All, I have seen in the new kamailio release 5.3.0, kamailio support for haproxy protocol – both the v1 (human-readable) and v2 (binary) versions of the protocol are supported ( https://www.kamailio.org/w/kamailio-v5-3-0-release-notes/ ) but it seems to be for Database, webserver failover not for kamailio itself. Haproxy is ONLY supporting TCP but not UDP. My question is : I am looking for failover service of kamailio servers looks like haproxy but udp & tcp. - one domain name : sip.secure.com pointing my public ip address. - two kamailio server pointing same database content (user, whitelist, ...) no asterisk or pbx. Kamailio is used Only for signaling (users, whitelist & rtp) Monitoring service : identify who is the kamailio server online, offline and the ability to force all the SIP/RTP/MANAGEMENT connection from Kamailio Srv1 to Kamailio Srv2 and vice-versa ? Thanks for your suggestions. -- Youssef

4 4

How to use sdp_remove_media() ?
by Denys Pozniak 04 Nov '19

04 Nov '19

Hello! I'm trying to remove m=video stream from incoming SIP/SDP request in th way below: modparam("rtpengine", "read_sdp_pv", "$avp(read_sdp_pv)") modparam("rtpengine", "write_sdp_pv", "$avp(write_sdp_pv)") ... sdp_remove_media("video"); msg_apply_changes(); sdp_get("$avp(read_sdp_pv)"); xlog("L_WARN","[$cfg(route)] *m=video removed* ci| rb=/$rb/ \n"); xlog("L_WARN","[$cfg(route)] *m=video removed* ci| avp(read_sdp_pv)=/$avp(read_sdp_pv)/ \n"); rtpengine_manage("$avp(rtpengine_offer)"); set_body("$avp(write_sdp_pv)", "application/sdp"); But no luck, xlog shows m=video in $rb and $avp(rtp_sdp_read) as well. -- BR, Denys Pozniak

3 8

Phonenum module
by Igor Olhovskiy 01 Nov '19

01 Nov '19

Hi! I'm playing around libphonenum to get validation accross e.164 numbers. So, idea is simple if (phonenum_match($fU, "src_num")) { if ($phn(src_num=>valid) != 1) { send_reply("400", "Invalid e.164 format"); xlog("L_ALERT","ALERT: FU: $fU is not e.164 number $phn(src_num=>error)\n"); exit; } } But on every (actually valid e.164 number like 380442459966 or 61388141397) I got Parsing number failed in error result. Tried both phonenum_match($fU, "src_num") and phonenum_match("$fU", "src_num") Actually phonenum_match("380442459966", "src_num") also not working Debian 10 Buster version: kamailio 5.2.5 (x86_64/linux) libphonenumber7 is already the newest version (7.1.0-5+b4).

2 6

kamailio on kubernetes
by Gaurav Bmotra 01 Nov '19

01 Nov '19

hello everyone i m working on a project kamailio on kubernetes for high availability and auto scalability but as i m able to run it in side the cluster with using kubenetes feature but having issue to access the service form outside the cluster i m using the ingress in kubernetse to point the service its flow like that domain.com:80 ------ingress-controler--------kamailio-service:80:5060------- POD(kamailio)5060 ISSUE IS I M NOT GETING CONNECTION BITWEEN DOMAIN.COM TO POD(kamailio) is there any one ho can help me in this thank you -- *Regards:* Gaurav Kumar

2 1

Routing Blocks - Missing in latest core wiki
by Joel Serrano 31 Oct '19

31 Oct '19

Hi, I was looking at the documentation and noticed that "Routing Blocks" section is missing in latest releases... http://www.kamailio.org/wiki/cookbooks/5.2.x/core#routing_blocks is OK But: http://www.kamailio.org/wiki/cookbooks/5.3.x/core#routing_blocks http://www.kamailio.org/wiki/cookbooks/devel/core#routing_blocks Don't exist, have they been moved elsewhere? Thanks! Joel.

3 8

Command Line SIP client
by Gaurav Bmotra 31 Oct '19

31 Oct '19

hi i m looking for Command Line SIP client , where i can register user form CLI for testing thank you -- *Regards:* Gaurav Kumar

9 11

How to intercept an INVITE after waking up (TSILO module) to change SDP
by Oleg Podguyko 31 Oct '19

31 Oct '19

Hello everybody! Kamailio works as a Registrar. Additionally, rtpengine is deployed next to it. Using the module rtpgngine I change SDP. Webrtc <-> SIP calls work fine. RTPengine proxies media. It is necessary to understand that when the call comes, I determine what Protocol the call came wss/tcp/udp, and then through the command "rtpengine_manage" change SDP accordingly. Now I have a new task. Wake up mobile customers through push notification. This client may be as WebRTC and also regular SIP. Trying to use TSILO module. # User location service route[LOCATION] { lookup("location"); $var(rc) = $rc; if (is_method("INVITE")) { if (t_newtran()) { route(SEND_PUSH); xlog("L_INFO", "tsilo store\n"); ts_store(); } } switch ($var(rc)) { case 1: xlog("L_INFO", "LOCATION, Subscriber found, RC: $var(rc)\n"); route(RELAY); break; case -1: xlog("L_INFO", "LOCATION, Subscriber doesn't found, RC: $var(rc)\n"); exit; break; case -3: xlog("L_INFO", "LOCATION, Subscriber doesn't found, RC: $var(rc)\n"); send_reply("404", "Not Found"); exit; case -2: xlog("L_INFO", "LOCATION, Method Not Allowed, RC: $var(rc)\n"); send_reply("405", "Method Not Allowed"); exit; } } If a call occurs when the called client is registered, then we go to the route(RELAY), which creates a branch in which I change the SDP. Everything works. route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (!t_relay()) { sl_reply_error(); } exit; } And already in MANAGE BRANCH the necessary SDP is put. If the called party is not registered. In the beginning everything goes well. I find this caller through a separate server. Send push notification. The client successfully registers on kamailio. # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); setbflag(FLB_NATSIPPING); } if(isflagset(FLT_SRC_WSS)) { setbflag(FLB_DST_WS); } if (!save("location")) { sl_reply_error(); } ts_append("location", "$tu") exit; } Next I better show a piece of the log: окт 31 14:55:08 /usr/sbin/kamailio[17488]: exec: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} *** cfgtrace:request_route=[REGISTRAR] c=[/etc/kamailio/kamailio.cfg] l=429 a=26 n=ts_append окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} tsilo [ts_append.c:72]: ts_append(): transaction 8800:338205132 found for 1234567890, going to append branches окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} tm [t_lookup.c:1629]: t_lookup_ident_filter(): transaction found окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} registrar [lookup.c:255]: lookup_helper(): contact for [1234567890] found by address окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} registrar [lookup.c:456]: lookup_helper(): instance is окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} tm [t_append_branches.c:77]: t_append_branches(): transaction 8800:338205132 in status 0 окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} tm [t_append_branches.c:99]: t_append_branches(): Call uefn7rvufi112jj5lt91: 0 (1) outgoing branches окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} tm [t_append_branches.c:126]: t_append_branches(): Current uri sip:1234567890@10.10.10.1:58027;transport=TCP;rinstance=727a64a3a59c508f окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} <core> [core/msg_translator.c:162]: check_via_address(): (50.7.93.27, uas7i396ceb5.invalid, 0) окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} tm [t_append_branches.c:149]: t_append_branches(): added branch [sip:1234567890@10.10.10.1:58027;transport=TCP;rinstance=727a64a3a59c508f] with ruid [uloc-5dbacb09-4450-7] окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} tm [t_append_branches.c:163]: t_append_branches(): Call uefn7rvufi112jj5lt91: 0 (0) outgoing branches after clear_branches() окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} <core> [core/tcp_main.c:2238]: tcpconn_send_put(): send from reader (17488 (21)), reusing fd окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} <core> [core/tcp_main.c:2473]: tcpconn_do_send(): sending... окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} <core> [core/tcp_main.c:2507]: tcpconn_do_send(): after real write: c= 0x7fb39dde4c68 n=3131 fd=12 окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} <core> [core/tcp_main.c:2508]: tcpconn_do_send(): buf= Thus, the transaction wakes up and the initial INVITE is sent to the new registered client without any SDP change! Which eventually leads to 415 error. Now the question itself. Tell me! As me at this moment intercept this INVITE, to it has passed through in advance prepared path for changing SDP. Thank you. -- Oleg Podguyko

1 0

kamailio - failover setup
by Ilie Soltanici 31 Oct '19

31 Oct '19

We are using keepalived with virtual IP Address for Kamailio, dispatcher module for Asterisk, 2 MySQL Databases master-master replicated and 2 Servers with ProxySQL. So far so good.

2 3

[abalashov@evaristesys.com: binrpc reply sizes]
by Alex Balashov 31 Oct '19

31 Oct '19

Hi, What is the relationship between the `ctl` modparams `binrpc_buffer_size`, `binrpc_max_body_size`, and `binrpc_struct_max_body_size`? More especially, how interdependent are they? Is there a set of circumstances in which one would wish for the structure buffer to be smaller than the reply body buffer? What about the internal buffer? Is it used only for some kind of intermediate formatting, or does it also hold the entire reply in some aspect? The reasons for asking are predictable: RPC command 'dlg.list' throws out "reply too big" on some occasions, and rather than simply increasing all of these drastically, I would like to learn more about which ones it makes sense to increase, and in what proportion with respect to each other. I also think this idea should be more clearly developed in the documentation. Thanks, -- 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/

2 4

setbflag and isbflagset usage.
by Володимир Іванець 30 Oct '19

30 Oct '19

Hello! I'm using branch flags but not sure if the behavior is correct. There is a logic in branch route that decides if branch flag must be set and does it with this command: *setbflag(FLB_FLAG, $T_branch_idx)*. Then I'm checking if the flag is set with *if* statement later in the configuration file for requests and in onreply route for replies: *if (isbflagset(FLB_FLAG, $T_branch_idx))*. The thing is that *if* statement always returns true. The only way I can fix this behavior is to use *resetbflag* at the beginning of branch route: *resetbflag(FLB_SIPTOPUB, $T_branch_idx)*. Thank you! Regards, Volodya Ivanets.

1 0

strange conditional "if ($au!=$tU)" with previous exit
by PICCORO McKAY Lenz 30 Oct '19

30 Oct '19

i have this in asterisk integration how to, and i noted the "exit" before the "if($au!=$tU)" .. i dont understan the conditional and the exit there! please can someon xplain me that!? # authenticate the REGISTER requests (uncomment to enable auth) #!ifdef WITH_ASTERISK if (!www_authorize("$td", "sipusers")) #!else if (!www_authorize("$td", "subscriber")) #!endif { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } i investigate at the kamailio cgf documentation and there's no clear topic related! http://www.kamailio.org/wiki/cookbooks/5.2.x/pseudovariables#tu_-_to_uri Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

3 3

Re: [SR-Users] [sr-dev] Kamailio as SBC for Microsoft Teams
by Henning Westerholt 30 Oct '19

30 Oct '19

Hello Rhys, I already replied to you yesterday, maybe it got into Spam on your side. :-) Quote below: > thanks. About the presence integration question - according to my research currently the only possibility would be to use the > existing Skype APIs and then use this (per interop-mode) also in MS Teams. > > Microsoft announced the availability of a API to set the presence status also for Teams, this is now scheduled for Q1 2020. There > are other limitations which also needs to be solved from MS and they are also scheduled for Q1 2020. > Feel free to contact me directly if you are interested in this topics. Cheers, Henning Am 29.10.19 um 08:51 schrieb Rhys Hanrahan: Hi Henning, This is a great article, thanks for sharing. I'm wondering if you've experimented with the possibility of integration of presence/dialog information with teams as well? Pretty much every integration guide I've seen on MS's UC solutions, including the previous ones (Lync, Communicator, SFB ) always seem limited to integrating calling via trunks. To me, the integration is not complete unless you can integrate presence things like: * If a user is on the phone then their status should be "busy" on Teams. When off call, set back to Available. * If a user is on a call in teams, our PBX should consider their PBX extension as "in use". Not sure it's possible, because I haven't seen anything of the sort being done yet, but I'm curious if this is something you looked into at all? Thanks. Rhys Hanrahan Chief Information Officer Nexus One Pty Ltd E: support(a)nexusone.com.au<mailto:support@nexusone.com.au> P: +61 2 9191 0606 W: http://www.nexusone.com.au/ M: PO Box 127, Royal Exchange NSW 1225 A: Level 10 307 Pitt St, Sydney NSW 2000 [http://quintus.nexusone.com.au/~rhys/nexus1-email-sig.jpg] ________________________________ From: sr-users <sr-users-bounces(a)lists.kamailio.org><mailto:sr-users-bounces@lists.kamailio.org> on behalf of Henning Westerholt <hw(a)skalatan.de><mailto:hw@skalatan.de> Sent: Tuesday, 29 October 2019 3:21 AM To: Kamailio (SER) - Users Mailing List Cc: Kamailio (SER) - Development Mailing List Subject: [SR-Users] Kamailio as SBC for Microsoft Teams Hello, I have created a “How-To” blog post on using Kamailio as “session border controller” for Microsoft Teams Direct Routing: https://skalatan.de/en/blog/kamailio-sbc-teams<https://skalatan.de/de/blog/kamailio-sbc-teams> You can this way use all the existing possibilities that Kamailio provide to interact with MS Teams as well. Best regards, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services _______________________________________________ Kamailio (SER) - Development Mailing List sr-dev(a)lists.kamailio.org<mailto:sr-dev@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev -- Kamailio Merchandising - https://skalatan.de/merchandising/ Kamailio services - https://skalatan.de/services Henning Westerholt - https://skalatan.de/blog/

2 2

Fwd: [CFP] FOSDEM 2020, RTC devroom, speakers, volunteers neeeded
by fosdem-rtc-admin＠freertc.org 28 Oct '19

28 Oct '19

FOSDEM - Real Time Communications devroom CfP ============================================= Overview -------- [FOSDEM](https://fosdem.org) is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2020 takes place 1-2 February 2020 in Brussels, Belgium. This document contains information about: - Real-Time Communications developer room (devroom) and lounge - speaking opportunities - volunteering in the devroom and lounge - social events (the legendary FOSDEM Beer Night and Saturday night dinners provide endless networking opportunities) - the Planet aggregation sites for RTC blogs **NEW:** Save yourself entering Free-RTC events and CFP deadlines into your calendar and task list, follow our iCalendar feed: https://freertc.org/events.ics Call for participation - Real Time Communications (RTC) ------------------------------------------------------- The Real-Time devroom and Real-Time lounge are about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, peer-to-peer, privacy and encryption. **We are looking for speakers for the devroom and volunteers and participants for the tables in the Real-Time lounge.** The devroom is only on Sunday, 2nd of February 2020. The lounge will be present for both days. To discuss the devroom and lounge, please join the [Free-RTC mailing list](http://lists.freertc.org/mailman/listinfo/discuss). ### Speaking opportunities Note: if you used FOSDEM Pentabarf before, please use the same account/username Real-Time Communications devroom: deadline 23:59 UTC on 15th of December. Please use the [Pentabarf](https://penta.fosdem.org/submission/FOSDEM20/) system to submit a talk proposal for the devroom. On the "General" tab, please look for the "Track" option and choose "Real Time Communications devroom". Other devrooms and lightning talks: some speakers may find their topic is in the scope of more than one devroom. It is encouraged to apply to more than one devroom and also consider proposing a lightning talk, but please be kind enough to tell us if you do this by filling out the notes in the form. Here you can find the [full list of devrooms](https://www.fosdem.org/2020/schedule/tracks/) and here you can apply for a [lightning talk](https://fosdem.org/submit). ### First-time speaking? FOSDEM devrooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the devroom administrators personally if you would like to ask any questions about it. ### Submission guidelines The Pentabarf system will ask for many of the essential details. Please remember to re-use your account from previous years if you have one. In the "Submission notes", please tell us about: - The purpose of your talk - Any other talk applications (devrooms, lightning talks, main track) - Availability constraints and special needs You can use HTML and links in your bio, abstract and description. If you maintain a blog, please consider providing us with the URL of a feed with posts tagged for your RTC-related work. We will be looking for relevance to the conference and devroom themes, presentations aimed at developers of free and open source software about RTC-related topics. Please feel free to suggest a duration between 20 minutes and 55 minutes but note that the final decision on talk durations will be made by the devroom administrators based on the number of received proposals. As the two previous devrooms have been combined into one, we may decide to give shorter slots than in previous years so that more speakers can participate. Please note FOSDEM aims to record and live-stream all talks. The CC-BY license is used. Volunteers needed ----------------- To make the devroom and lounge run successfully, we are looking for volunteers: - FOSDEM provides video recording equipment and live streaming, volunteers are needed to assist in this - Organizing one or more restaurant bookings (dependending upon number of participants) for the evening of Saturday, 1 February - Participation in the Real-Time lounge - Circulating this Call for Participation to other mailing lists Social events and dinners ------------------------- The traditional FOSDEM beer night occurs on Friday, 31st of January. On Saturday night, there are usually dinners associated with each of the devrooms. Most restaurants in Brussels are not so large so these dinners have space constraints and reservations are essential. Please subscribe to the [Free-RTC mailing list](http://lists.freertc.org/mailman/listinfo/discuss) for further details about the Saturday night dinner options and how you can register for a seat. Related events around FOSDEM ---------------------------- As per usual, the [XMPP Summit](https://wiki.xmpp.org/web/Conferences/Summit_24) is happening ahead of FOSDEM. This time it will take place on the 30th and 31st of January also in Brussels. All details are available on the [summit website](https://wiki.xmpp.org/web/Conferences/Summit_24). Spread the word and discuss --------------------------- If you know of any mailing lists where this CfP would be relevant, please forward this document. If this devroom excites you, please blog or microblog about it, especially if you are submitting a talk. If you regularly blog about RTC topics, please send details about your blog to the planet site administrators: - All projects https://planet.freertc.org planet(a)freertc.org - XMPP https://planet.jabber.org ralphm(a)ik.nu - SIP https://planet.sip5060.net planet(a)sip5060.net - (Español) https://planet.sip5060.net/es/ planet(a)sip5060.net Please also link to the Planet sites from your own blog or web site as this helps everybody in the free real-time communications community. Contact ------- For any private queries, contact us directly using the address **fosdem-rtc-admin(a)freertc.org** and for any other queries please ask on the [Free-RTC mailing list](http://lists.freertc.org/mailman/listinfo/discuss). The devroom administration team: - Saúl Ibarra Corretgé <s(a)saghul.net> - Ralph Meijer <ralphm(a)ik.nu> - Daniel-Constantin Mierla <miconda(a)gmail.com> - Daniel Pocock <daniel(a)pocock.pro> - Guus der Kinderen <guus.der.kinderen(a)gmail.com>

1 0

Rewrite contact address on 200 OK?
by Oded Arbel 28 Oct '19

28 Oct '19

Hi, we are having a problem with a network change use case. It goes on something like this: 1. Kamailio receives INVITE from caller. 2. Kamailio sends INVITE to callee MUA. 3. MUA response with 100, then 183. 4. MUA then loses network connectivity and re-establishes access with another IP address. 5. MUA sends 200 OK from the new IP address, but with `Contact` still set as the old address (which I think it should, because you aren't allowed to change contact in the middle of dialog?) 6. Kamailio forwards 200 OK to caller and receives ACK. 7. Kamailio sends ACK to old IP address of MUA. The last step is obviously a problem as the MUA will not be able to see the ACK. I'm trying to see how to resolve this issue. I can capture the "$si" and "$sp" of the new address in the reply route (I also tried to get the MUA to send OPTIONS immediately after reestablishing connectivity, and capture "$si"/"$sp" there, which also works), but I can't figure out how to update the destination of the ACK. I've tried running `fix_nated_contact()` as well as trying to write into various pseudo variables - in either the OPTIONS request handler or the 200 reply handler - which either errors or does nothing. Any suggestions? -- Oded Arbel

2 4

Kamailio to capture mirrored traffic
by Igor Olhovskiy 28 Oct '19

28 Oct '19

Hi! I'm trying to get Kamailio working as a traffic capture on a same machine with other PBX software installed. Actually, traffic is mirrored with iptables -A PREROUTING -t mangle -i eth0 -p udp --dport 5060 -j TEE --gateway 127.0.0.2 (https://link.getmailspring.com/link/AB5F9D36-533D-4A52-ADE3-FB76B813163C@ge…) iptables -t nat -A PREROUTING -d 127.0.0.2 -p udp --dport 5060 -j DNAT --to 127.0.0.1:5062 (https://link.getmailspring.com/link/AB5F9D36-533D-4A52-ADE3-FB76B813163C@ge…) Kamailio request route is super simple request_route { xlog("L_ALERT", "[SIP-PACKET] Got packet [F=$fu R=$ru D=$du M=$rm IP=($si:$sp $Ri:$Rp) ID=$ci]\n"); drop; } I was trying to get Kamailio just listen on interface 127.0.0.1:5062, but no luck listen=udp:127.0.0.1:5062 Next was to use sipcapture module with following parameters loadmodule "sipcapture.so" modparam("sipcapture", "db_url", "text:///tmp/") modparam("sipcapture", "raw_socket_listen", "127.0.0.1:5060-5062") modparam("sipcapture", "raw_interface", "lo") modparam("sipcapture", "promiscious_on", 1) Also no luck. Means Kamailio can't see packets, but I see em with wireshark on lo interface. What is best way to get it working? Or I'm missing something? Thanks!

5 7

Cancel / OK handling
by Oz Mortimer 26 Oct '19

26 Oct '19

Hi, I have an issue where a client is sending cancel after the call has been answered but before the client received the 200OK, I know they should then respond with a BYE based on the 200 final result, but they don’t (Meaning the call / billing stays up).. Is there a way I can timeout a call/force BYE if ACK is not received to the 200 OK? I know technically this is the clients issue but they can’t / won’t fix Thanks is advance Oz.

1 0

No audio after hold and unhold event
by arish haque 25 Oct '19

25 Oct '19

Hi all, No Audio flows between the two parties when one side resumes the call after putting on hold. Configuration - WebRTC <--> Kamailio <--> Asterisk Below is the warning I'm getting in asterisk console WARNING[32388]: chan_sip.c:10425 process_sdp: Declining non-primary audio stream: audio 10596 UDP/TLS/RTP/SAVPF 107 103 104 9 0 8 106 105 13 110 112 113 101 Is this a codec issue or something else. Thanks, Arish

1 0

Kamailio 3.3.7
by Brandon Armstead 24 Oct '19

24 Oct '19

Has anyone seen this error while compiling 3.3.7 raw_sock.c:431: undefined reference to `udpv4_vhdr_sum' system: Linux 4.9.0-11-amd64 #1 SMP Debian 4.9.189-3 (2019-09-02) x86_64 GNU/Linux lsmod: Module Size Used by bonding 147456 0 intel_rapl 20480 0 x86_pkg_temp_thermal 16384 0 intel_powerclamp 16384 0 coretemp 16384 0 kvm_intel 200704 0 kvm 598016 1 kvm_intel irqbypass 16384 1 kvm ttm 98304 0 drm_kms_helper 155648 0 crct10dif_pclmul 16384 0 crc32_pclmul 16384 0 joydev 20480 0 evdev 24576 3 ghash_clmulni_intel 16384 0 drm 360448 2 ttm,drm_kms_helper i2c_algo_bit 16384 0 iTCO_wdt 16384 0 shpchp 36864 0 ipmi_si 57344 0 iTCO_vendor_support 16384 1 iTCO_wdt sg 32768 0 pcspkr 16384 0 mei_me 36864 0 ipmi_msghandler 49152 1 ipmi_si mei 102400 1 mei_me video 40960 0 button 16384 0 acpi_power_meter 20480 0 acpi_pad 24576 0 ib_iser 49152 0 rdma_cm 53248 1 ib_iser iw_cm 49152 1 rdma_cm ib_cm 45056 1 rdma_cm ib_core 208896 4 ib_iser,ib_cm,rdma_cm,iw_cm configfs 40960 2 rdma_cm iscsi_tcp 20480 0 libiscsi_tcp 24576 1 iscsi_tcp libiscsi 53248 3 ib_iser,libiscsi_tcp,iscsi_tcp scsi_transport_iscsi 102400 4 ib_iser,libiscsi,iscsi_tcp ip_tables 24576 0 x_tables 36864 1 ip_tables autofs4 40960 2 ext4 589824 2 crc16 16384 1 ext4 jbd2 106496 1 ext4 crc32c_generic 16384 0 fscrypto 28672 1 ext4 ecb 16384 0 mbcache 16384 3 ext4 sd_mod 49152 4 hid_generic 16384 0 usbhid 53248 0 hid 122880 2 hid_generic,usbhid crc32c_intel 24576 2 aesni_intel 167936 0 aes_x86_64 20480 1 aesni_intel glue_helper 16384 1 aesni_intel lrw 16384 1 aesni_intel gf128mul 16384 1 lrw ablk_helper 16384 1 aesni_intel cryptd 24576 3 ablk_helper,ghash_clmulni_intel,aesni_intel ahci 40960 3 xhci_pci 16384 0 libahci 32768 1 ahci ixgbe 294912 0 xhci_hcd 188416 1 xhci_pci i2c_i801 24576 0 i2c_smbus 16384 1 i2c_i801 dca 16384 1 ixgbe ptp 20480 1 ixgbe libata 249856 2 ahci,libahci pps_core 16384 1 ptp mdio 16384 1 ixgbe usbcore 253952 3 usbhid,xhci_pci,xhci_hcd scsi_mod 225280 7 ib_iser,sd_mod,libiscsi,libata,scsi_transport_iscsi,iscsi_tcp,sg usb_common 16384 1 usbcore fan 16384 0 thermal 20480 0 I believe I've crossed this issue once before and it was a simple apt-get package to include compile options: make include_modules="mi_fifo db_mysql tm dialog sl tmx kex rr maxfwd textops textopsx nathelper acc avpops xlog uac alias_db siputils utils usrloc registrar pv regex htable sqlops sdpops topoh ctl db_mysql mysql" prefix="/usr/local" Thanks - Brandon

3 3

environmental variables in kamailio
by Gaurav Bmotra 24 Oct '19

24 Oct '19

hi i using kamailio 5.2 is it possible to use environmental variables value in kamailio.cfg like i want to use local ip and public ip in config file listen=udp:$(local_ip):5060 advertise $(public_ip):5060 if iti is possible then how i can use this thank you -- *Regards:* Gaurav Kumar

4 3

driver error: Can't connect to MySQL server with kamailio 5.3.x and mysql-config-groups
by Karsten Horsmann 23 Oct '19

23 Oct '19

Hi List, i updated one of my testing kamailios from 5.2.x to latest 5.3.x. After restarting the process, i got an mysql error that the server "kamailio" is not found. I use the config-feature of my.cnf config-group names within the kamailio.cfg I can start my kamailio 5.3.x installation with "normal username:password@host" definitions for DBURL. Did anyone have the same issue? 0(19625) ERROR: db_mysql [km_my_con.c:139]: db_mysql_new_connection(): driver error: Can't connect to MySQL server on 'kamailio' (4) 0(19625) ERROR: <core> [db.c:319]: db_do_init2(): could not add connection to the pool 0(19625) ERROR: dialplan [dp_db.c:179]: dp_connect_db(): unable to connect to the database 0(19625) ERROR: dialplan [dialplan.c:191]: mod_init(): could not initialize data 0(19625) ERROR: <core> [core/sr_module.c:850]: init_mod(): Error while initializing module dialplan (/usr/lib64/kamailio/modules/dialplan.so) ERROR: error while initializing modules from my kamailio.cfg i defined the DBURL like this: # Use /etc/my.cnf.d/kamailio.cnf to store credentials # within a mysql-db-group #!define DBURL "mysql://[kamailio]/kamailio" and special modules can use it like this (or via DBURL) modparam("sqlops", "sqlcon", "cb=>mysql://[kamailio]/kamailio") cat /etc/my.cnf.d/kamailio.cnf # MySQL configuration group for kamailio [kamailio] user=kamailio password="kamailiorw" host=172.20.100.22 rpm -qa | grep kam kamailio-dialplan-5.3.0-13.1.x86_64 kamailio-websocket-5.3.0-13.1.x86_64 kamailio-jansson-5.3.0-13.1.x86_64 kamailio-mysql-5.3.0-13.1.x86_64 kamailio-tls-5.3.0-13.1.x86_64 kamailio-ims-5.3.0-13.1.x86_64 kamailio-statsd-5.3.0-13.1.x86_64 kamailio-5.3.0-13.1.x86_64 kamailio-redis-5.3.0-13.1.x86_64 cat /etc/redhat-release CentOS Linux release 7.6.1810 (Core) -- Cheers *Karsten Horsmann*

3 7

MS Teams Direct routing with kamailio
by Karsten Horsmann 23 Oct '19

23 Oct '19

Hello List, i search for the right place in kamailio to manipulate the dispatcher OPTIONS packets to add an Contact header. For uac generated traffic i can use the event_route[tm:local-request] from the tm module. But how can i manipulate kamailio *self* generated OPTION pings from the dispatcher? OPTIONS request for MS Teams must contain the Contact header of such a format: Contact: <sip:SBC_FQDN:5061;transport=tls> I follow this guide (for opensips) and try to adapt that step by step to kamailio: https://blog.opensips.org/2019/09/16/opensips-as-ms-teams-sbc/ CentOS 7 / kamailio 5.2.x (and maybe soon 5.3.x) / rtpengine -- Cheers *Karsten Horsmann*

2 2

Unable to use BLF
by Cehes (Free) 23 Oct '19

23 Oct '19

Hi all, This is many days we are trying many things and reading many threads :) We also re installed a simple brand new kamailio to be sure everything is ok. In fact, with sngrep we can see "subscribe" arriving on kamailio but on syslog we have a critical error : Oct 23 10:08:42 GTO-Temp01 kamailio[2004]: 20(2027) exec: {1 29 SUBSCRIBE 3135373137353930373733323337-joash9r9lnre} *** cfgtrace:request_route=[PRESENCE] c=[/usr/local/etc/kamailio/kamailio.cfg] l=770 a=24 n=handle_subscribe Oct 23 10:08:42 GTO-Temp01 kamailio[2004]: 20(2027) CRITICAL: {1 29 SUBSCRIBE 3135373137353930373733323337-joash9r9lnre} <core> [core/mem/q_malloc.c:494]: qm_free(): *BUG: bad pointer 0xa (out of memory block!) *called from presence: *subscribe.c*: handle_subscribe(1040) - ignoring We tried to start kamailio with many more memory (x4) and this is the same. We tried with only "presenceXXX" plugins, and also with "puaXXX" plugins below is our config file : https://gist.github.com/Dyriolon/e894b9f2a67f9dfb7117421717ad6178 Any idea of what is wrong ? Thanks!

1 0

Call Pickup and Call Parking
by Zhan Bazarov 22 Oct '19

22 Oct '19

Hello, What is the Best way to implement call parking and call pickup in setup Kamailio(active-active) and dispatcher with a few asterisk round-robin? Could someone share some best practices, please? Thank you!

1 0

mysql create user and grant privileges
by Daniel-Constantin Mierla 22 Oct '19

22 Oct '19

Hello, the mysql server version 8 is no longer supporting creating the user and granting privileges in the same statement, they have to be separate. I checked the mariadb specs and they seem to support also the separate creation of user and granting privileges in two statements. I pushed the changes in kamdbctl to master branch via commit: - https://github.com/kamailio/kamailio/commit/c0879add55bef77703f2259f224b4a1… The reason I am writing here is to see if anyone is aware of some (old) versions of mariadb or mysql servers that do not support the creation and grant in two steps. If yes, then we can add a config option in kamctlrc to do one or the other variants. But for now I did it only with the new style, as it seems to be supported everywhere by latest versions, to avoid unnecessary option to keep also the old style. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Oct 21-23, 2019, Berlin, Germany -- https://asipto.com/u/kat

1 0

[Dialog] Removing entries from dialog memory
by Julien Klingenmeyer 21 Oct '19

21 Oct '19

Hi, I am wondering if there is any way to remove an entry from the active dialog list? I mean, assuming that Kamailio did not get any BYE related to a call for some reason and that I want that call to be “killed” from dialog memory before reaching the lifetime timeout. What I already tried with no success: sending a request to Kamailio catched by a dedicated route of the script then: * using the dlg_get function (with the target call as parameter) then send_bye function (then dropping the BYE request generated by the dialog module in the onsend_route with the drop function): no success. Dialog module correctly generates BYE requests but dialog related to the call is kept in memory until its lifetime timeout (and BYE requests are not dropped in the onsend_route either). * using the dlg_get function then dlg_set_timeout to set a very low timeout value for that particular call. No successs: Trying to update a bogus dlg as critical error. Could some RPC commands or whatever help me to accomplish what I try to do? Thanks Julien

4 9

pv_buffer_size limit
by Sergiu Pojoga 20 Oct '19

20 Oct '19

Hi ppl, I'm testing this external app (cgrates, you all know it) that talks to Kamailio via evapi. Every *n* minutes the app sends an evapi request to Kamailio which returns back the list of all dialogs like so: jsonrpc_exec('{"jsonrpc":"2.0","id":1, "method":"dlg.list","params":[]}'); evapi_relay("{\"event\":\"CGR_DLG_LIST_REPLY\", \"jsonrpl_body\":$jsonrpl(body)}"); The problem is that the size of *$jsonrpl(body)* PV is limited by Kamailio's *pv_buffer_size* setting. My math shows that a single dialog is about 1KiB of json data. *Problem*: Kamailio fails to start with a *pv_buffer_size *higher than 117 KiB (119808 bytes), which in turn rends the system's limit to about 117 simultaneous calls... pretty sad. Below find the error when trying to start Kamailio with higher values: 0(17080) ERROR: <core> [core/mem/q_malloc.c:291]: qm_find_free(): qm_find_free(0x7f02275c5010, 120832); Free fragment not found! 0(17080) ERROR: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7f02275c5010, 120832) called from core: core/pvapi.c: pv_init_buffer(2069), module: core; Free fragment not found! 0(17080) ERROR: <core> [core/pvapi.c:2072]: pv_init_buffer(): cannot init PV print buffer slot[39] Is there a workaround for this? Thanks, --Sergiu

2 1

Get value from header and check value type
by Karsten Horsmann 18 Oct '19

18 Oct '19

Hi List, I use Kamailio 5.2.x and I want to fetch an value from an X Header and do some checks like "is not empty" and "should be an integer". AFAIK per default all values are strings with the hdr function. What are the best way for some validations within Kamailio script language? if(is_present_hf("X-someheader")) { $avp(routing_target) = $(hdr(X-someheader){s.int}); if(!ds_list_exists("$avp(routing_target)")) Cheers Karsten Horsmann

2 2

How to get the dispatcher value dynamically
by Narayan P 18 Oct '19

18 Oct '19

Hello, We are using kamailio for load balancing as stateless proxy as mentioned below. {$du = "sip:192.168.130.XXX;transport=udp"; forward();} We have set $du to the correct media server value in the dispatcher module.Can you please tell me how to get the correct dispatcher value(Media server IP) and print this into log file. I am using $ru and $du to print,but it is not giving the correct one. for example: Suppose for the first call ; $du = "sip:192.168.130.XXX:5060;transport=udp"; $du = "sip:192.168.130.XXX:5062;transport=udp"; and so on. How to fetch the correct dispatcher value dynamically. Any response will be highly appreciated. Thanks, Rudra

3 2

need different Contact with the UAC modul.
by "Hans-Jürgen Brand" 18 Oct '19

18 Oct '19

3 4

Cancel / OK handling
by Oz Mortimer 18 Oct '19

18 Oct '19

Hi, I have an issue where a client is sending cancel after the call has been answered but before the client received the 200OK, I know they should then respond with a BYE based on the 200 final result, but they don’t (Meaning the call / billing stays up).. Is there a way I can timeout a call if ACK is not received to the 200 OK? I know technically this is the clients issue but they can’t / won’t fix Thanks is advance Oz.

1 0

Changing $fU or $fn in branch_route
by Jurijs Ivolga 18 Oct '19

18 Oct '19

Hi! I have small problem. When I assign $fU or $fn in request-route and then one more time I make new assignment in branch_route. Instead of rewriting it is just concatenate needed value at the end. lets assume $avp(cli)=1234567 and $avp(cliplus)=+1234567. So if in request_route I do $fU=$avp(cli); and then in branch_route $fU=$avp(cliplus);. $fU becomes one string: "1234567+1234567". Is it a bug or I missing something? Here is config snippet: request_route { ... route(MY); ... } route[MY] { ... $avp(cli) = "1234567" $avp(cliplus) = "+"+$avp(cli); $fU=$avp(cli); # here $fU=1234567 $fn=$avp(cli); # here $fn=1234567 route(LCR_ROUTE); exit; } route[LCR_ROUTE] { if(!is_method("INVITE")) return; if (!load_gws("1",$rU,$avp(cli))) { send_reply("503", "Error loading gateways"); exit; } $var(i)=0; while($(avp(lcr_gw_uri)[$var(i)])!= $null){ xlog("L_INFO", "loaded gw_uri_avp[$var(i)]=$(avp(lcr_gw_uri)[$var(i)]) \n"); $var(i) = $var(i)+1; } if (!next_gw()) { send_reply("503", "No available gateways"); exit; } xlog("L_INFO", "request-uri $ru \n"); xlog("L_INFO", "$avp(lcr_gw_uri), $avp(lcr_id)\n"); xlog("L_INFO", "flag $avp(lcr_flag) \n"); t_set_fr(0, 4000); t_on_failure("RTF_LCR_ROUTE"); t_on_branch("BRANCH_CUST"); route(RELAY); exit; } branch_route[BRANCH_CUST] { if ( $avp(lcr_flag) == 4) { $fU=$avp(cliplus); #here $fU=1234567+1234567 $fn=$avp(cliplus); #here $fn=1234567+1234567 } } Jurijs

2 5

do MOD_<module name> keywords exist?
by Juha Heinanen 18 Oct '19

18 Oct '19

I have used in config file these kind of constructs: #!ifdef MOD_<module name> ... #!endif Is the MOD_xxx keyword automatically set when module xxx is loaded? Perhaps I have invented the whole concept in a dream? -- Juha

2 4

Failed AKAv1-MD5 authentication with Boghe
by Bao, Yin 18 Oct '19

18 Oct '19

I have a Boghe (v2.0.153.836) working fine with Kamailio with MD5 algorithm however fails to authenticate if I configure S-CSCF to use AKAv1-MD5. We have the same configuration on private user identity in FHoSS on both the succ MD5 run and failed AKAv1-MD5 run. I wonder if anyone has faced this issue before. Here are some details on message exchanges: * S-CSCF included the challenge in 401 Unauthorized – Challenging the UE: WWW-Authenticate: Digest realm="example.example.org", nonce="rzGnTBGPw3hE+mDPHrZ1PAAAAAAApAAAmfftHqYUuUA=", algorithm=AKAv1-MD5, ck="629d6d9a6a6befa509b1a9bb17a9c2a3", ik="32fcd14a102279de2e38200a2257efcb", qop="auth,auth-int"\r\n * The same header was seen when 401 Unauth travelled through I-CSCF and P-CSCF (shouldn’t P-CSCF strips out the ck and ik fields before sending to UE?) * UE responded with another register with response: Authorization: Digest username="bob(a)example.example.org",realm="example.example.org",nonce="rzGnTBGPw3hE+mDPHrZ1PAAAAAAApAAAmfftHqYUuUA=",uri="sip:example.example.org",response="6445f9df2b785eab3fa461849880b48d",algorithm=AKAv1-MD5,cnonce="34baf441e99a23e0fcb2bc001355c4bf",qop=auth-int,nc=00000001\r\n * S-CSCF failed the calculation and sent back 403 Authentication Failed I tried AKAv2-MD5 setting as default algorithm on S-CSCF and the result is the same as above. It seems that only MD5 works on my setup. What are the different configs in HSS (or other places) AKAv1-MD5 need? I checked the AMF/OP on both HSS and Boghe and they match. Boghe does not have SQN setting but I set HSS to start with 0. Any pointers are appreciated! Yin

2 1

IMC through HTTP
by Péter Barabás 18 Oct '19

18 Oct '19

Hi, I want to implement conference messaging functionality into a client. IMC module seems perfect. In our scenario, the rooms and members are managed in a different server bacause of other user management functionalities and the kamailio only uses the rooms to broadcast messages to members. Is it possible anyway to create rooms and join members from a separate server like through http protocol with XHTTP module? The idea is: 1. Create a room on server A with adding members. Server A store the room data. 2. Server A sends http message to kamailio with proper format, like POST .../createRoom with jsonBody { name: „room1” } 3. Kamailio receives the http request and process it with creating a room ’room1’. 4. Server A sends members of room like POST .../addMembers with json body { room: „room1”, members: [ username: „user1”, username: „user2” ] } 5. Kamailio receives the https request and process it with joining memers into room All commands should have the belonging http request. Is it possible? Regards, Peter

2 2

MySQL vs MariaDB support
by Sergey Safarov 18 Oct '19

18 Oct '19

At present time we create RPM packages using MySQL libs. May be need switch to MariaDB libs. What you think about it?

2 1

Need to suppress "200 cancelling" in kamailio
by amit Kumar 18 Oct '19

18 Oct '19

Hi All, As per our requirement, when A party send a cancel to B party we do not want "200 cancelling" message from Kamailio.We try to find any high-level flag to disable it. but could not able to find. Can you please help us on how to achieve this. A Party Kamailio B Party ---------------->INVITE------------------------> <---------180 Ringing<------------------------ ------------>CANCEL--------------------------> <---------200 canceling [we want to suppress this message and do not want kamialio to inform A party about progress status of cancel] <------------OK----------------------------------- Thanks, Amit

2 1

Need to suppress "200 cancelling" in kamailio
by amit Kumar 18 Oct '19

18 Oct '19

Hi All, As per our requirement, when A party send a cancel to B party we do not want "200 cancelling" message from Kamailio. We try to find any high-level flag to disable it. but could not able to find it. Can you please help us on how to achieve this. A Party Kamailio B Party ---------------->INVITE------------------------> <---------180 Ringing<------------------------ ------------>CANCEL--------------------------> <---------200 canceling [we want to suppress this message and do not want kamialio to inform A party about progress status of cancel] <------------OK----------------------------------- Thanks, Amit

1 0

Kamailio v5.3.0 Released - new major version is out
by Daniel-Constantin Mierla 17 Oct '19

17 Oct '19

Kamailio v5.3.0 is out – it comes with 6 new modules and a considerable set of improvements touching more than 100 existing modules. You can read detailed release notes at: * https://www.kamailio.org/w/kamailio-v5-3-0-release-notes/ Many thanks to all developers and community members that made possible this release. v5.3.0 brings more flexibility and optimizations for KEMI interpreters, enhancements to load balancer, dialog tracking, uac remote registration and tls with libssl 1.1.x implementations, new variables, transformations and plenty of other new features. Enjoy Kamailio v5.3.0! Thank you for flying Kamailio! Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 1

Releasing Kamailio v5.3.0
by Daniel-Constantin Mierla 17 Oct '19

17 Oct '19

Hello, if there are no known major issues that need longer time to take care of them, in my opinion we could consider releasing the next major version 5.3.0 next week, for me Thursday, October 10 seems to work the best, still allowing a bit more than one week for testing. Any other suggestions? Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Oct 21-23, 2019, Berlin, Germany -- https://asipto.com/u/kat

3 4

catching tls errors
by Juha Heinanen 17 Oct '19

17 Oct '19

Sometimes I see in syslog errors like this: Oct 15 16:44:57 salmon /usr/bin/sip-proxy[2064]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417C086:SSL routines:tls_process_client_certificate:certificate verify failed Is it possible somehow to catch the error in config file, for example, to figure out from which IP address the connection attempt came from? -- Juha

3 6

Kamailio weighted dispatch problems
by Jasen Hall 17 Oct '19

17 Oct '19

I am trying to change our request dispatching from round-robin to a weight-based method (ds_select_dst() algorithm 9 vs. algorithm 4) and am having a hard time getting an asymmetrical dispatch pattern enabled. In testing, I'm dispatching to two Ringswitch servers and trying to send three times the traffic to one instance. My destinations list file previously set both endpoints in the same group and simply provided the SIP URI for each. My new list file adds the attribute `weight` to each endpoint and looks like this: >1 sip:10.0.0.1:5060 weight=3 >1 sip:10.0.0.2:5060 weight=1 However, when I throw a couple thousand calls at Kamailio, I always end up with an almost perfect split of calls to both ringswitches. Regardless of the weights I choose, I never see variation in the dispatched load. Does anyone have experience with this issue and have some guidance? Thanks in advance. -- Jasen Hall | Invoca *Site Reliability Engineer* www.invoca.com [image: See the list] <https://invoca.sigstr.net/uc/5cba111d3527a200469b9afd>

3 4

IMS Kamailio Diameter Charging Server
by Jack R 14 Oct '19

14 Oct '19

Hi All, We have installed Kamailio as IMS setup pcscf icscf scscf - 3 instances running on Ubuntu.. Referring to the link below, it mentioned about Diameter based Kamailio Charging Server. https://www.kamailio.org/events/2016-KamailioWorld/Day2/15-Carsten.Bock-Dia… Where to find it? How to install it? to make it work in Kamailio IMS set up as per below articles: https://www.kamailio.org/docs/modules/stable/modules/ims_charging.html https://kamailio.org/docs/modules/devel/modules/ims_ocs.html Regards, Jack.

2 2

Kamailio v5.2.5 Released
by Daniel-Constantin Mierla 10 Oct '19

10 Oct '19

Hello, Kamailio SIP Server v5.2.5 stable release is out. This is a maintenance release of the latest stable branch, 5.2, that includes fixes since the release of v5.2.4. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.2.x. Deployments running previous v5.2.x versions are strongly recommended to be upgraded to v5.2.5. For more details about version 5.2.5 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/10/kamailio-v5-2-5-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Oct 21-23, 2019, Berlin, Germany -- https://asipto.com/u/kat

3 3

SSL Offloading with AWS ELB or Google GSLB and Kamailio
by Joel Serrano 10 Oct '19

10 Oct '19

Hi everyone, I was giving a try to setup Kamailio with a Cloud TCP load balancer in front, taking advantage of the newly added proxy protocol compatibility and my initial tests went very well. Flow: client -> (tcp) -> load balancer -> (tcp) -> Kamailio TCP socket I then did another quick test and enabled TLS, also with good results: Flow: client -> (tls) -> load balancer -> (tls) -> Kamailio TLS socket So far so good, proxy protocol works as expected. I wanted to go one step further and see if I could somehow offload SSL operations at the load balancer level, and leave kamailio handling plain tcp. Flow: client -> (tls) -> load balancer -> (tcp) -> Kamailio TCP socket This partially worked, and before I start digging into what I have to do to get it completely working, I'd like to know if anyone already has a similar setup, or even if Kamailio is able to handle such a scenario, the reason I'm asking is because of the headers, etc. In this last scenario, I receive in a TCP socket, a request with TLS headers all over the place.. INVITE sip:14a84f2016944eb0854ef0e9b71bfa10@app.mydomain.com:60655 SIP/2.0 Via: SIP/2.0/TLS 192.168.1.16:60717;branch=z9hG4bK.KmUpamn5P;rport From: ... To: ... CSeq: 21 INVITE Call-ID: -j1QSnam9o Max-Forwards: 70 Route: <sip:sbc-test2.mydomain.com:443;lr> <http://sbc-test2.sbx.gii.me:443;lr>/> Supported: replaces, outbound Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE Content-Type: application/sdp Content-Length: 436 Contact: <sip:linphone@A.B.C.D:60717;transport=tls>;+sip.instance="<urn:uuid:fabcb441-a348-49a7-948d-72448d6840eb>" I then forward this request via UDP to subsequent proxies for further processing, on the replies, my payload information back to the client should be TLS, although sent via a TCP socket.. Is this something that will not work by design? Is there any hack I can take advantage of? The goal would be for Kamailio to handle TLS headers via TCP socket, as the client expects TLS information, but the actual traffic should go in plan TCP, and the load balancer will take care of re-encrypting before replying to the client. Any ideas/suggestions/comments? I hope this email is understandable, I find it complicated to detail the exact problem, feel free to ask any questions if you don't understand anything. Thanks, Joel.

3 5

PUBLISH ERROR
by Social Boh 10 Oct '19

10 Oct '19

Hello, Kamailio 5.2.5. I'm testing presence on my Kamailio using different Clients (Jitsi, Bria5, Linphone). when they send the PUBLISH to Kamailio, all receive the same answer: 400 Invalid Request On the LOG file: ERROR: {1 21 PUBLISH O8-BtqLnzw} presence [publish.c:421]: ki_handle_publish_uri(): No E-Tag and no body found I'm pretty sure same configuration work with 5.2.3 version. Any hint? Regards -- --- I'm SoCIaL, MayBe

2 4

centos7 rpm for evapi
by Aidar Kamalov 10 Oct '19

10 Oct '19

Hello, how can I install evapi for kamailio using rpm repo? # ls /usr/lib64/kamailio/modules | grep evapi | wc -l 0 # yum provides *evapi.so Loaded plugins: fastestmirror Loading mirror speeds from cached hostfile * base: mirror.docker.ru * extras: mirror.linux-ia64.org * updates: mirror.corbina.net No matches found -- Aydar A. Kamalov

2 1

Releasing Kamailio v5.2.5
by Daniel-Constantin Mierla 10 Oct '19

10 Oct '19

Hello, I am considering to release Kamailio v5.2.5 out of the latest branch 5.2 on Thursday, October 10. As usual, if you noticed any patch that should be backported or you are aware of some issue not yet reported, announce it to sr-dev or via bug tracker. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Oct 21-23, 2019, Berlin, Germany -- https://asipto.com/u/kat

1 1

Testing kxlibssl prng for tls module
by Daniel-Constantin Mierla 09 Oct '19

09 Oct '19

Hello Marco, I am writing to see if you can test the kxlibssl prng that I just added for tls module. I want to see if exposes the same issue you reported in: * https://github.com/kamailio/kamailio/issues/2077 If you can't test with master branch, you need to backport two commits: * https://github.com/kamailio/kamailio/commit/99eafac2d92533ba93cd8244173aef0… * https://github.com/kamailio/kamailio/commit/a52f05087a211bfecd36300907d0bff… Then set: modparam("tls", "rand_engine", "kxlibssl") The latest branch 5.2 has the code for setting custom prng backported. The idea behind kxlibssl prng is to reuse the function of the default libssl v1.1.x prng, but guarded by a kamailio specific mutex. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Oct 21-23, 2019, Berlin, Germany -- https://asipto.com/u/kat

2 2

Kamailio Developers Meeting, Nov 14-15, 2019, in Dusseldorf
by Daniel-Constantin Mierla 09 Oct '19

09 Oct '19

Hello, Kamailio SIP Server project is organizing another meeting of its developers during November 14-15, 2019, hosted again by sipgate.de in Dusseldorf, Germany. The event is intended to facilitate the interaction between Kamailio developers and to offer a convenient environment for working together on several topics of high interest for the project, including writing code for Kamailio and its tools, improving documentation, or discuss about future development. Everyone from the community is welcome to join. Please note we have a limited capacity of 20 seats in the meeting room. Also, very important to be aware that this is not an event to learn how to use Kamailio. More details about the event, the venue, how to register, are available at: * https://www.kamailio.org/w/developers-meeting/ Looking forward to those two intensive hacking Kamailio days in Dusseldorf! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Oct 21-23, 2019, Berlin, Germany -- https://asipto.com/u/kat

1 1

Fred on FLOSS Weekly 16 Oct
by jungle boogie 09 Oct '19

09 Oct '19

Hi All, I'm happy to see friendly Fred Posner on FLOSS Weekly next Wednesday (16 October, 2019) to discuss Kamailio. If you can, tune in live and see how the interview goes. Recording is at 9:30am Pacific (-7 UTC) / 16:30 UTC. https://twit.tv/shows/floss-weekly Upcoming guests: https://docs.google.com/spreadsheets/d/1_dOHpYlzRlMb0JEzwUcAUflPz8PLq7dKkos… Thanks for being on the show, Fred! Best, j.b.

1 0

exit after www_challenge
by PICCORO McKAY Lenz 08 Oct '19

08 Oct '19

i have the code with an exit, i dont know if are correct that "exit" in that line? or not? help me please? # authenticate the REGISTER requests (uncomment to enable auth) #!ifdef WITH_ASTERISK if (!www_authorize("$td", "sipusers")) #!else if (!www_authorize("$td", "subscriber")) #!endif { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } consume_credentials(); Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

4 4

Error parsing parameter value
by Duarte Rocha 08 Oct '19

08 Oct '19

Greetings, I'm testing some parameters actions and i found a weird behaviour This is my code for the test : $var(Parameter_List) = "blue=twelve;green=twe\"lve\";yellow=\"twelve\""; xerr("Parameter List = $var(Parameter_List)"); xerr("Value of Parameter blue = $(var(Parameter_List){param.value,blue})"); xerr("Value of Parameter green = $(var(Parameter_List){param.value,green})"); xerr("Value of Parameter yellow = $(var(Parameter_List){param.value,yellow})"); The printed logs are the following : ERROR: <script>: Parameter List = blue=twelve;green=twe"lve";yellow="twelve" ERROR: <script>: Value of Parameter blue = twelve ERROR: <script>: Value of Parameter green = twe"lve" ERROR: <script>: Value of Parameter yellow = twelve I have also tried setting the string as : $var(Parameter_List) = 'blue=twelve;green=twe"lve";yellow="twelve"'; But the results were similar The "" are not recognized in some scenarios. Is this intended behaviour or should i submit it as a bug? Best Regards, Duarte Rocha

3 8

Bad BYE request
by Duarte Rocha 08 Oct '19

08 Oct '19

Greetings, This is a repeated problem, but i have more info now. Please help understand if there is a problem on my code or if there is any kind of Kamailio bug. On the first INVITE of every call i apply uac_replace_from() and uac_replace_to() in order to normalize numeration. I have the restore_mode set as "auto". The original numeration is : To : 351250000000 From : 351211455000 The INVITE is sent to the client with To and From like this and with the Record-Route headers (Kamailio has 2 Ips and inserts 2 Record-Routes) containing vsf and vst parameters to help the future restores : To: <sip:+351250000000@111.111.111.111> From: <sip:+351211455000@111.111.111.111 >;tag=BD583030373631320511EAE6 Record-Route: <sip:111.111.111.111;r2=on;lr;ftag=BD583030373631320511EAE6;tbk_i=2_2_Y;tbk_o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQAJCnNyAwMcHwIdHQsWHwUMOA--;vst=AAAAABgGBAMHDQgJDQECA3RyAwMcHwIdGgQeHAIFOQ--;did=18e.5d62> Record-Route: <sip:111.111.111.111;r2=on;lr;ftag=BD583030373631320511EAE6;tbk_i=2_2_Y;tbk_o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQAJCnNyAwMcHwIdHQsWHwUMOA--;vst=AAAAABgGBAMHDQgJDQECA3RyAwMcHwIdGgQeHAIFOQ--;did=18e.5d62> My client, upon receiving the INVITE, also does some changes in the numeration and sends a 183 reply with the following headers : Record-Route: <sip:111.111.111.111;r2=on;lr;ftag=BD583030373631320511EAE6;tbk_i=2_2_Y;tbk_o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQAJCnNyAwMcHwIdHQsWHwUMOA--;vst=AAAAABgGBAMHDQgJDQECA3RyAwMcHwIdGgQeHAIFOQ--;did=18e.5d62> Record-Route: <sip:111.111.111.111;r2=on;lr;ftag=BD583030373631320511EAE6;tbk_i=2_2_Y;tbk_o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQAJCnNyAwMcHwIdHQsWHwUMOA--;vst=AAAAABgGBAMHDQgJDQECA3RyAwMcHwIdGgQeHAIFOQ--;did=18e.5d62> From: "+351211455000" <sip:I2116446I_500@111.111.111.111 >;tag=BD583030373631320511EAE6 To: <sip:+351250000000@111.111.111.111>;tag=as3e46f684 After Kamailio processes the reply, To and >From are restored to it's original values and the reply is relayed : From: <sip:351211455000@111.111.111.111 >;tag=BD583030373631320511EAE6 To: <sip:351250000000@111.111.111.111>;tag=as3e46f684 Here, the restore works perfectly. By the end of the call the client sends me a BYE request with Route, From and To like this : Route: <sip:111.111.111.111;r2=on;lr;ftag=BD583030373631320511EAE6;tbk_i=2_2_Y;tbk_o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQAJCnNyAwMcHwIdHQsWHwUMOA--;vst=AAAAABgGBAMHDQgJDQECA3RyAwMcHwIdGgQeHAIFOQ--;did=18e.5d62>,<sip:212.13.40.205;r2=on;lr;ftag=BD583030373631320511EAE6;tbk_i=2_2_Y;tbk_o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQAJCnNyAwMcHwIdHQsWHwUMOA--;vst=AAAAABgGBAMHDQgJDQECA3RyAwMcHwIdGgQeHAIFOQ--;did=18e.5d62> From: <sip:+351250000000@111.111.111.111 >;tag=as3e46f684 To: "+351211455000" <sip:I2116446I_500@111.111.111.111 >;tag=BD583030373631320511EAE6 Here, the difference between 183 and BYE seems to be only in the Route\Record-Route are built (2 Headers or one header with 2 uri). Now, there are a couple of problems relaying the BYE that seem to me to be related to auto replace. This BYE message isn't logged on the onsend-route and it logs a couple of errors : ERROR: <core> [core/parser/msg_parser.c:164]: get_hdr_field(): bad to header ERROR: <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [To: "+351211455000" <sip:Q4525417L_<:C212.13.38.148>;tag=3DC130303736313205201F18#015#012Call-ID: 020FAAC7] ERROR: dialog [dlg_handlers.c:666]: pre_match_parse(): bad request or missing CALLID/TO hdr :-/ ERROR: <core> [core/parser/parse_addr_spec.c:719]: parse_addr_spec(): unexpected char [<] in status 6: ["+351211455000" <sip:Q4525417L_] . Despite not being logged on onsend_route and creating some errors, the BYE is still relayed like this : From: < sip:351250000000@111.111.111.111>;tag=as3e46f684 To: "+351211455000" <sip:Q4525417L_<:C111.111.111.111>;tag=BD583030373631320511EAE6 This, naturally, generates a "400 Bad Request" reply from the next equipment. I've also logged the $tU values along the code for BYE requests and it always logs the "I2116446I_500", so i don't know if my code can be the cause of this. A couple of questions and comments - What can i do to better debug this issue? I want to be sure if the problem comes from my code of if it is a Kamailio issue. Right now i'm not sure where to look. - Can a request be sent without passing on onsend_route ? I have used this code to "catch" the BYE, which works for other BYEs - if( $rm == "BYE") {xerr("On send : To User = $tU");}

3 2

SIP-links adjacency
by Oleg Belousov 07 Oct '19

07 Oct '19

Hello. In our setup kamailio perfoms some logic depending on call direction, inbound or outbound. Call direction is identified by 2 SBC IP (IP_inbound/IP_outbound), we currently use for that 2 dispatcher groups and that is working fine. Now number of peer nodes increased from one SBC till few tens, but we still need to keep SIP-links adjacency, meaning that call which arrived from SBC_N IP_inbound need to be submitted via same SBC_N IP_outbound and vice versa. There could be few approaches to implement that requirement, I'm wondering if can use existing dispatcher function/select destination logic to support SIP-links adjacency. -- obelousov.tel

3 4

q_malloc crash on Kamailio 5.2.3 with http_async_query function
by Michal Škuta 07 Oct '19

07 Oct '19

Hello everyone, I am adding some new features to our Kamailio server, one of them is sending HTTP POST request after calls end. We created simple route like this and we are calling it from event_route[dialog_end]: route[HTTP_TEST_CALL_END] { t_newtran(); $http_req(body) = "TEST"; http_async_query("http://192.168.6.23:2080/cdr_http", "HTTP_TEST_REPLY"); } Problem is that Kamailio is crashing on line *http_async_query*, the log from crash: Oct 3 16:13:37 sip_server /opt/ipgate/kamailio/sbin/kamailio[5331]: CRITICAL: <core> [core/mem/q_malloc.c:149]: qm_debug_check_frag(): BUG: qm: prev. fragm. tail overwritten(c0003032, abcdefed)[0x97a7ff3c:0x97a7ff58]! Memory allocator was called from http_async_client: hm_hash.c:82. Fragment marked by http_async_client: hm_hash.c:71. Exec from core/mem/q_malloc.c:384. In older Kamailio 4.4 I am using http_async_query with no problems. I opened issue on GitHub: https://github.com/kamailio/kamailio/issues/2091 I hope that is ok, more logs and core dump is in the issue. Any help or suggestion is welcomed. Sincerely Michal

2 2

using http_async_query
by David Villasmil 05 Oct '19

05 Oct '19

Hello guys, So, i'm using http_async_query to send out requests. It may happen that the request times out, in this case, i would like to try a new destination, i.e. with ds_select (NOT ds_delect_dest or ds_next, because i don't want the $du to be set, since in my case i don't actually have a SIP packet). Is is possible to get the original http request url in the http_async_query timeout route? Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

1 1

storing vars in htable
by David Villasmil 05 Oct '19

05 Oct '19

Hello guys, I'm storing some data in an htable, like so: $var(http_query) = "/default/en_US/send.html?u=user&p=pass&l=$var(next_channel)&n=$var(to)&m=$var(message)"; then i need to store it: $sht(http_requests=>$http_req_id) = $var(http_query); But what's stored is the actual vars, not the the values, it's stored like: sh store: /default/en_US/send.html?u=user&p=pass&l=$var(next_channel)&n=$var(to)&m=$var(message) Why is that? shouldn't the vars be parsed? Thanks Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 2

Re: [SR-Users] Call limit through shared database
by Olli Attila 04 Oct '19

04 Oct '19

Hello Henning, I tried the dialog profile sharing with dmq&dialog modules and now the syncing works between the two kam nodes. Thank you! Is there a way to keep the dialog profiles persistent (load from db if kam restarts)? I was thinking a scenario where kam1 has already the call limit maxed out and kam2 suddenly reboots and after reboot, kam2 should be aware of dialogs on kam1 to be able to limit calls again correctly. Cheers, Olli

3 3

Re: [SR-Users] Call limit through shared database
by Olli Attila 04 Oct '19

04 Oct '19

Hello, I had the following enabled in db_mode param and now it is actually working: *1 - REALTIME* - any dialog information changes will be reflected into the database immediately. Restarting either of the kam1 or kam2 proxies "on the fly" will result in retrieving dialog profiles correctly from database. Cheers, Olli

1 0

Record-Route management
by Duarte Rocha 03 Oct '19

03 Oct '19

Greetings, I have multiple IPs in my Kamailio. In some scenarios, the caller and the called connect to different IPs and in others, they use the same IP. Currently i'm running with the RR's module "enable_double_rr" parameter set so i can have one Record-Route for each IP used on the call. However, in the scenario that both caller and called have the same IP, i would like to only have on Record-Route in order to save on the pactet's size. Is this possible to have those two scenarios on the same Kamailio ? Best Regards

2 2

string manipulation
by David Villasmil 03 Oct '19

03 Oct '19

Hello guys, I'm receiving an http request via xhttp, something like: http://ip.com/request?value1=123&value2=2313&value3... Is it possible to get those values somehow? I also have a suri coming from a ds_select in _dsdst_, I need to get the host from that _dsdst_ Any way of doing this? Thanks all Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 2

Get destination out of _dsdst_
by David Villasmil 03 Oct '19

03 Oct '19

Hello guys, I've never used "ds_select" as opposed to "ds_select_dst", and i would like to get the actual destination out of _dsdst_ I've got modparam("dispatcher", "xavp_dst", "_dsdst_") and then i'm trying to do: if(!ds_select("100", "4")) { send_reply("404", "No destination"); exit; }else{ xlog("L_ERR", "[XHTTP]: The next destination for SMS is: [$xavp(_dsdst_)]\n"); ds_mark_dst("d"); } But that's coming up with The next destination for SMS is: [<<xavp:0x7fbe87acdbf0>>] I need to get the actual ip address out of the struct. Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 2

Development is open for future release of v5.4.x series
by Daniel-Constantin Mierla 03 Oct '19

03 Oct '19

Hello, the branch 5.3 was created late last evening, therefore the master branch is open for adding new features, to be part of future release series v5.4.x. Any bug fix committed to master that applies to 5.3.x or older stable branches should be backported as usual with "git cherry-pick -x ..." to appropriate branches like 5.3 or 5.2. Expect that v5.3.0 will be released in a few weeks from now. Based on the workflow used during the past years, the next future release v5.4.0 should be out after another 8-10 months of development, plus 1-2 months of testing, so sometime in close to the end of spring 2020 or afterwards. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Oct 21-23, 2019, Berlin, Germany -- https://asipto.com/u/kat

1 0

Branch 5.3 has been created
by Daniel-Constantin Mierla 03 Oct '19

03 Oct '19

Hello, the branch 5.3 has been created, to be used for releasing v5.3.x series. To check out this branch, the following commands can be used: git clone https://github.com/kamailio/kamailio kamailio-5.3 cd kamailio-5.3 git checkout -b 5.3 origin/5.3 Pushing commits in this branch: git push origin 5.3:5.3 Note that 5.3 is an official stable branch, so only bug fixes, missing kemi exports (discuss on sr-dev if not sure) or improvements to documentation or helper tools can be pushed to this branch. You may notice a bit of more activity on tls (some issue is under work) and siptrace (a bit of cleanup for coherence might be done) modules, but over all no new features will be pushed there. As usual, if there is a bug fixed, commit and push first to master branch and then cherry pick to 5.2 branch: git cherry-pick -x COMMITID In few weeks, the first release from branch 5.3 will be out, respectively Kamailio v5.3.0. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Oct 21-23, 2019, Berlin, Germany -- https://asipto.com/u/kat

1 0

Branching 5.3
by Daniel-Constantin Mierla 02 Oct '19

02 Oct '19

Hello, if there are no strong opinions against, then it is time to branch 5.3 on git repository. I am considering to do it by the end of the day on Wednesday, October 2, 2019. After that, the master will be again open for new features, while 5.3 branch will continue to get the only fixes and improvements to documentation and tools. 2-3 weeks later we should release v5.3.0, the first stable version in 5.3.x series. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Oct 21-23, 2019, Berlin, Germany -- https://asipto.com/u/kat

1 1

Unresolvable destination for new added users...
by Zhan Bazarov 02 Oct '19

02 Oct '19

I can't make call through kamailio between new added users. I get 478 ... But it works for older users (who was added couple weeks ago e.g.) SIP flow, location and subscriber table the same in both cases. Could you please shed any light on this issue? *2019/10/02 11:11:20.719834 10.2.1.198:5068 <http://10.2.1.198:5068> -> 10.4.1.14:50600 <http://10.4.1.14:50600>* SIP/2.0 *478* Unresolvable destination (478/SL) Via: SIP/2.0/UDP asterisk_ip:asterisk_port;branch=z9hG4bK18626d14;rport=50600;received=10.4.1.14 From: "Admin Admin" <sip:some_user@asterisk_ip:asterisk_port>;tag=as09b1f1f3 To: <sip:some_user@kamailio_ip :kamailio_port>;tag=a1384aaeda4d71b5edb13882669b943e.8468 Call-ID: *20e023254f1b3c942a61e47d63badfd6@10.4.1.14:50600 <http://20e023254f1b3c942a61e47d63badfd6@10.4.1.14:50600>* CSeq: 102 INVITE Server: MS Lync Content-Length: 0

3 5

BLF 'NOTIFY' not working
by Petarr Jones 02 Oct '19

02 Oct '19

Hi I am using Kamailio 5.2 with the following PRESENCE modules and parameters: loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "presence_dialoginfo.so" loadmodule "presence_reginfo.so" modparam("presence", "db_url", DBURL) modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) modparam("presence_xml", "disable_bla", 0) modparam("presence_xml", "integrated_xcap_server", 1) I have a Yealink T28P phone which has two lines/extensions successfully registered with Kamailio 5.2. When I setup the BLF light to monitor one of the extensions, I get the correct SUBSCRIBE record but the NOTIFY record is sent over and over again until it fails. ----------------------------------------------------------------------------------------------------------- NOTIFY sip:101@192.168.1.123:5062 SIP/2.0 Via: SIP/2.0/UDP 45.45.45.45;branch=z9hG4bKd16c.03b5bc97000000000000000000000000.0 To: <sip:101@45.45.45.45>;tag=1366853594 From: <sip:104@45.45.45.45>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-e6c3 CSeq: 2 NOTIFY Call-ID: 594386350(a)192.168.1.123 Content-Length: 0 User-Agent: kamailio (5.2.4 (x86_64/linux)) Max-Forwards: 70 Event: dialog Contact: <sip:45.45.45.45:5060;transport=udp> ----------------------------------------------------------------------------------------------------------- When I register the phone with Freeswitch instead and run the same test, the NOTIFY works and I get a much longer NOTIFY message ----------------------------------------------------------------------------------------------------------- NOTIFY sip:100@192.168.1.123:5064 SIP/2.0 Via: SIP/2.0/UDP 66.66.66.66;rport;branch=z9hG4bK1mXej9r2605UB Route: <sip:55.55.55.55:5064>;transport=udp Max-Forwards: 70 From: <sip:102@66.66.66.66>;tag=xUoVghM5syHc To: "100" <sip:100@66.66.66.66>;tag=1720892352 Call-ID: 2583553127(a)192.168.1.123 CSeq: 1178415701 NOTIFY Contact: <sip:102@66.66.66.66:5060> User-Agent: FreeSWITCH-mod_sofia/1.10.1-release-12-f9990221e6~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Event: dialog Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=1800 Content-Type: application/dialog-info+xml Content-Length: 149 xml version="1.0"?> dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:102@66.66.66.66"> dialog-info> ----------------------------------------------------------------------------------------------------------- Should the message coming from Kamailio contain more data or is that not the issue. Should I be including other modules also? All guidance would be appreciated.

2 4

Call limit through shared database
by Olli Attila 01 Oct '19

01 Oct '19

Hello, I have 2 kam instances using one shared database and both kam servers running the same routing logic code. Kamailio servers are configured to use the same database as dialog backend. I am limiting the concurrent dialogs in the following way but it seems that kamailio is using its own memory to read profile size rather than the db. Second kam is not aware of the dialog profiles that the first kam wrote to the shared db. Is there a way to get multiple kam instances to read profile size from the same database when calling the get_profile_size function? modparam("dialog", "profiles_with_value", "concurrent_calls") modparam("dialog", "dlg_flag", 4) modparam("dialog", "db_url", DBURL) modparam("dialog", "db_mode", 1) route[LIMIT_CALLS] { if (!dlg_isflagset("1")) { # $td – gateways, $fu - user if (get_profile_size("concurrent_calls", "$fu", "$avp(calls)")) { if ($avp(calls) >= 3) { xlog("L_INFO", "Concurrent calls $fu at limit"); send_reply("503", "Calls limit reached"); exit; } else { dlg_manage(); dlg_setflag("1"); set_dlg_profile("concurrent_calls", "$fu"); } } } } Cheers, Olli

5 8

Kamailio Charging
by Jack R 01 Oct '19

01 Oct '19

Hi, Does Kamailio provide some charging functionality? If yes, then how do we go about setting it up? like what module/configuration? and Database tables to see/touch? I see these links: https://www.kamailio.org/docs/modules/4.2.x/modules/ims_charging.html https://www.kamailio.org/w/2013/07/documentation-on-charging-in-ims/ Or Do we need separate software like example cgrates for charging? (also a side question do we have any other open-source like cgrates) Any information would be helpful. Thanks and Regards, Jack.

3 3

Relay Request
by Maryam Baghdadi 29 Sep '19

29 Sep '19

Hello everyone, Is it possible to relay INVITE request to another server after receiving "180 Ringing" response ??? I would like to activate CRBT(ring back tone) service and according to my research after receiving 180 ringing media playing request should be sent to a media server .

3 2

Re: [SR-Users] Relay Request
by Mojtaba 29 Sep '19

29 Sep '19

Would you please keep "Users Mailing List" <sr-users(a)lists.kamailio.org> in tracking. :) That is straight froward. There is no change in sending the request from Kamailio to Asterisk. In Asterisk you could use Progress command to support Media-Relay. In additional use Dial command (with r or m option for supporting CRBT ). With Best Regards On Sun, Sep 29, 2019 at 8:52 AM Maryam Baghdadi <maryam.baghdadi(a)gmail.com> wrote: > Hello Mojtaba, > I chose asterisk as media server and some services like voice mail > provided ... about CRBT I know asterisk have this service but I don't know > how to send this request from kamailio to asterisk.. > do you know kamailio can provide this service itself or not??? > could you pls explain more > > thanks > > > On Sat, Sep 28, 2019, 1:48 PM Mojtaba <mespio(a)gmail.com wrote: > >> Hello, Some consideration would be needed: >> 1) SS7 for PSTN subscriber to support media-relay >> 2)Progress capability for SIP subscriber. >> 3) Media server to support ring-back VAS service >> You should check enabling this service or not during MO procedure. >> With Best Regards. >> >> On Sat, Sep 28, 2019 at 1:19 PM Maryam Baghdadi < >> maryam.baghdadi(a)gmail.com> wrote: >> >>> Hello everyone, >>> >>> Is it possible to relay INVITE request to another server after receiving >>> "180 Ringing" response ??? >>> I would like to activate CRBT(ring back tone) service and according to >>> my research after receiving 180 ringing media playing request should be >>> sent to a media server . >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> sr-users(a)lists.kamailio.org >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >> >> >> -- >> --Mojtaba Esfandiari.S >> > -- --Mojtaba Esfandiari.S

1 0

tcp_accept_haproxy questions
by Grant Bagdasarian 27 Sep '19

27 Sep '19

Hello, The docs mention regular TCP connections not being accepted anymore once this is enabled. I assume UDP is unaffected and both proxyprotocol and udp may live side by side? Also, does this play nicely with WebSockets? Taking into account the Websocket connection being setup through a proxy protocol supporting proxy. Regards, Grant

1 0

wiki on $sht(htable=>key)
by Juha Heinanen 27 Sep '19

27 Sep '19

Kamailio wiki does not tell what result $sht(htable=>key) gives if the key is not found in htable. It would be nice if it did. -- Juha

1 0

KEMI dialplan dp_replace returns not implemented yet
by Grant Bagdasarian 26 Sep '19

26 Sep '19

Hello, The dp_replace KEMI function is returning the message "not implemented yet". Function call: _.dialplan.dp_replace(1, _.pv.get("$rU"), "$var(dpr)") app_python3 ¡apy_kemi.c:872¿: sr_apy_kemi_f_pv_get_mode(): pv get: $rU <core> ¡core/pvapi.c:398¿: pv_cache_lookup(): pvar ¡$rU¿ found in cache app_python3 ¡apy_kemi.c:193¿: sr_apy_kemi_exec_func_ex(): execution of method: dp_replace app_python3 ¡apy_kemi.c:338¿: sr_apy_kemi_exec_func_ex(): not implemented yet app_python3 ¡apy_kemi.c:872¿: sr_apy_kemi_f_pv_get_mode(): pv get: $var(dpr) <core> ¡core/pvapi.c:371¿: pv_cache_add(): pvar ¡$var(dpr)¿ added in cache Using Kamailio 5.2.4 Docker image. version: kamailio 5.2.4 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 6.3.0 Regards, Grant

2 4

Kamailio Rasberry Pi benchmark results
by Henning Westerholt 26 Sep '19

26 Sep '19

Hello, as promised here the report on my Rasberry Pi benchmarks. The numbers were quite impressive! - 3000 registration requests per second (in-memory usrloc) - 300 calls per second for a p2p call (default cfg, only SIP traffic) - about 18 MBit/s incoming and 16 MBit/s outgoing SIP peak traffic could be processed All the details about the test setup and more numbers in the blog post: https://skalatan.de/blog/kamailio-rpi-benchmark Cheers, Henning -- Kamailio Merchandising - https://skalatan.de/merchandising/ Kamailio services - https://skalatan.de/services Henning Westerholt - https://skalatan.de/blog/

1 0

Kamailio v5.1.9 Released
by Daniel-Constantin Mierla 26 Sep '19

26 Sep '19

Hello, Kamailio SIP Server v5.1.9 stable release is out. This is a maintenance release of the previous stable branch, 5.1, that includes fixes since the release of v5.1.8. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.1.x. Deployments running previous v5.1.x versions are strongly recommended to be upgraded to v5.1.9 or to v5.2.x series. For more details about version 5.1.9 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/09/kamailio-v5-1-9-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the branch 5.1 is the previous stable branch. The latest stable branch is 5.2, at this time with v5.2.4 being released out of it. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020 -- www.kamailioworld.com

2 1

ims_ipsec_pcscf: maybe core dump due to maximum count of connections
by Valentin Christoph 26 Sep '19

26 Sep '19

Hi, Doing some tests with the module ims_ipsec_pcscf, we encountered a core dump during startup of Kamailio and would like to understand the correct behavior. We assume the ipsec_max_connections parameter was the root cause --> how many max connections have been actually tested? Can it be that ipsec_client-port and ipsec_server_port were too close together (some overlapping)? Thx Christoph

2 1

install with kazoo
by Benoit Chesneau 26 Sep '19

26 Sep '19

Hi all, Can anyone share a simple configuration to use with kazoo or a link to it? I'm struggle in having it setup without using the defaults packages. Ideally it should dispatch messages to freeswitch. Any help is welcome :) Benoît

3 2

Planning the release of v5.1.9
by Daniel-Constantin Mierla 26 Sep '19

26 Sep '19

Hello, I am considering to release v5.1.9 out of the branch 5.1 during the next week, on Tuesday (Sep 24) or Thursday (Sep 26). As usual, if you are aware of issues not reported on tracker or commits that should be backported to branch 5.1, do report them asap in order to get them a chance to be sorted out. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Oct 21-23, 2019, Berlin, Germany -- https://asipto.com/u/kat

1 2

Kamailio ARM support improved, tests
by Henning Westerholt 25 Sep '19

25 Sep '19

Hello, as part of the testing for 5.3.0 I looked into the ARM support in Kamailio. Previously the core used some deprecated assembler functions for atomic operations (locking etc..). There was also no native support for the ARM v7 architecture available. After several fixes and extensions both topics should be now much improved. I did today a lot of tests on Rasberry Pi and also some benchmarking (will send another e-mail when the report is ready). If you use Kamailio on another embedded ARM (>= v6) system it would be great if you also give it a try. Use git master revision ccc0eb6d12a1d (or newer) for your tests. Please report eventual issues to the sr-dev list. Cheers, Henning -- Kamailio Merchandising - https://skalatan.de/merchandising/ Kamailio services - https://skalatan.de/services Henning Westerholt - https://skalatan.de/blog/

1 0

Development for v5.3.x is frozen
by Daniel-Constantin Mierla 25 Sep '19

25 Sep '19

Hello, the formal notification that the development for the next major version 5.3.0 is now frozen. The focus has to be on testing the master branch. Also, the master branch should not get commits with new features till the branch 5.3 is created, expected to happen in 2-4 weeks, a matter of how testing goes on. Meanwhile, the commits with new features in the C code can be pushed to personal branches, pull requests can still be done, but they will be merged after branching 5.3. Can still be done commits with documentation improvements, enhancements to related tool (e.g., kamctl, kamcmd) and exporting missing KEMI functions to the modules already exporting most of the other functions (done for previous releases, because likely that was a fail to notice, all functions should be exported for a module at the same time). Once the branch 5.3 is created, new features can be pushed again to master branch as usual. From that moment, the v5.3.0 should be out very soon, time used for further testing but also preparing the release of packages. If someone is not sure if a commit brings a new feature, just make a pull request and it can be discussed there on github portal or via sr-dev mailing list. A summary of what is new in upcoming 5.3 is collected at: * https://www.kamailio.org/wiki/features/new-in-5.3.x Upgrade guidelines will be collected at: * https://www.kamailio.org/wiki/install/upgrade/5.1.x-to-5.3.0 Everyone is more than welcome to contribute to the above wiki pages, especially to the upgrade guidelines, to help everyone else during the migration process from v5.2.x to 5.3.x. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

3 8

Extract user from AOR
by Grant Bagdasarian 25 Sep '19

25 Sep '19

Hello, I've been trying to extract the user part from an AoR using transformations in Python (KEMI), but without success. The reg_fetch_contacts function has been called successfully before trying to access the AoR. None of the below statements work for me. They all return None. user = _.pv.get("$(ulc(profile=>aor){uri.user})") user = _.pv.get("$(ulc(profile=>aor){tobody.user})") or _.pv.sets("$var(aor)", _.pv.get("$(ulc(profile=>aor))")) user = _.pv.get("$(var(aor){tobody.user})") user = _.pv.get("$(var(aor){uri.user})") I'm not sure what's causing the None result. Could it be the curly braces being used for string interpolation in Python? Or are these transformations not able to extract the value since the AoR has the following format: user@domain? If the same can be accomplished in another way, please let me know. Regards, Grant

2 1

Question
by Shahdee Nelson 25 Sep '19

25 Sep '19

With your software would I be able to create, and modify VOIP lines? As in saying would I be able to provide VOIP numbers that can make outbound calls to USA from USA?

2 1

Problem restoring To and From (uac_replace)
by Duarte Rocha 24 Sep '19

24 Sep '19

Greetings, I'm getting some errors with the auto restore of From and To from UAC Module. After the restore, the URI has lots garbage and it's not like the original. I think the problem lies on the way the client sends the Route Header to me, but i'm not sure. When Kamailio sends the INVITE Record-Route are like this : Record-Route: <sip:xxx.xxx.xxx.xxx;................> Record-Route: <sip:aaa.bbb.aaa.bbb;................> However, when my client sends a BYE to finish the call, the Route are on this mode : Route: <sip:aaa.bbb.aaa.bbb;................>,<sip:xxx.xxx.xxx.xxx;................> The module's documentation says this : "One uses the rr module and appends to the Record-Route header a parameter containing some strings from which the original and new URI can be computed. The problem with this mode is that it relies on the fact the parties will send the Route exactly as it was received. In case there is a change, the resulting URIs will not be correct." Can this change by the client be the cause of this issue? Is there anything i can do to avoid this? Best Regards, Duarte Rocha

1 0

Can I load balance Kamailio with Kamailio?
by Ne Mo 24 Sep '19

24 Sep '19

3 2

Parsing json file on kamailio.cfg file
by Melek Oktay 24 Sep '19

24 Sep '19

Hi, I am trying to monitor *uac *channels status periodically in kamailio config file. I call uac.reg_dump rpc command and result will be inside in $var(uac_channel) variable jsonrpc_exec('{"jsonrpc": "2.0", "method": "uac.reg_dump"}'); $var(uac_channel) = $jsonrpl(body); it returns me json formatted result like below, it has a several channels info and their status *flags* {"jsonrpc":"2.0", "result":[ {"l_uuid":"YY","l_username":"YY","l_domain":"YY","r_username":"YY","., " *flags*":16 }, {"l_uuid":"XX","l_username":"XX","l_domain":"XX","r_username":"XX",,... " *flags*":1 }}]} I could not parse this result on kamailio.cfg file ! I want to learn channel that has a 16 value in flag attribute for example !! Would you help me to figure out how I am gonna parse this json file please ?

2 2

qm_find_free() Free fragment not found, called from xcap_server, no more pkg
by Cody Herzog 23 Sep '19

23 Sep '19

Hello. Recently, for the first time, we experienced an apparent memory issue in our production environment. Here's an example of the relevant log messages: Sep 13 18:55:22 SIPCOMM /usr/local/sbin/kamailio[2302]: ERROR: <core> [core/mem/q_malloc.c:286]: qm_find_free(): qm_find_free(0x7fd8bbcfc010, 134648); Free fragment not found! Sep 13 18:55:22 SIPCOMM /usr/local/sbin/kamailio[2302]: ERROR: <core> [core/mem/q_malloc.c:419]: qm_malloc(): qm_malloc(0x7fd8bbcfc010, 134648) called from xcap_server: xcap_server.c: ki_xcaps_put(549), module: xcap_server; Free fragment not found! Sep 13 18:55:22 SIPCOMM /usr/local/sbin/kamailio[2302]: ERROR: xcap_server [xcap_server.c:552]: ki_xcaps_put(): no more pkg Sep 13 18:55:22 SIPCOMM /usr/local/sbin/kamailio[2252]: ERROR: app_perl [kamailioxs.xs:487]: XS_Kamailio_log(): 500 Server error The failed operation was the XCAP server module trying to generate a very large RLS contact list for a user. The issue only impacted users with very large lists, as though a large contiguous block of memory could not be found, whereas other smaller allocations continued to work fine. We believe the requested allocation was around 112 KB in size. The server had been up for 14 days. We were able to work around the issue temporarily by just restarting the Kamailio service. It's unusual, because our production server is often up for months, and we've never seen this issue before. The load on production is increasing slowly due to an increased concurrent user count, so that might be related. Before restarting the service on production, we captured the output of the following commands: kamctl stats shmem kamcmd mod.stats all shm kamcmd pkg.stats kamcmd mod.stats all pkg Here's the shared mem output: { "jsonrpc": "2.0", "result": [ "shmem:fragments = 27240", "shmem:free_size = 447203296", "shmem:max_used_size = 116175576", "shmem:real_used_size = 89667616", "shmem:total_size = 536870912", "shmem:used_size = 68824240" ], "id": 6934 } Here's the pkg output for the particular PID which was throwing the errors: { entry: 34 pid: 2302 rank: 14 used: 2415864 free: 4949688 real_used: 3438920 total_size: 8388608 total_frags: 1951 } We didn't see anything obvious in the stats output which explains the issue. We've been trying to reproduce the issue in a dev environment using simulated higher than production load running for many days, but so far we've had no luck. We've been monitoring memory stats over time, but we don't see any obvious leaks or issues. We've searched various past threads, but didn't find any obvious answers. Here are some of the documents and the threads we've been reading: https://www.kamailio.org/wiki/tutorials/troubleshooting/memory https://www.kamailio.org/wiki/cookbooks/3.3.x/core#mem_join https://sr-users.sip-router.narkive.com/3TEDs3ga/tcp-free-fragment-not-found https://lists.kamailio.org/pipermail/sr-users/2012-June/073552.html https://lists.kamailio.org/pipermail/sr-users/2017-February/096132.html https://lists.kamailio.org/pipermail/sr-users/2017-September/098607.html https://github.com/kamailio/kamailio/issues/1001 https://lists.kamailio.org/pipermail/sr-users/2016-April/092592.html https://lists.kamailio.org/pipermail/sr-users/2010-July/064832.html Regarding our Kamailio version and build options, here's the output of 'kamailio -v': ---- version: kamailio 5.1.0 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 20:07:31 Jan 4 2018 with gcc 5.4.0 ---- We are running on a 64-bit Ubuntu Server virtual machine. Any help would be greatly appreciated. Thanks very much.

3 6

Using static entries with usrloc
by George Diamantopoulos 23 Sep '19

23 Sep '19

Hello, Normally I wouldn't be asking this but I was surprised to find no reference to anyone doing this online. I need to have a few AoRs with static entries in kami's location table, which will always be reachable at their contacts without ever having to register with Kamailio. Am I right in thinking that merely INSERTing respective rows in kamailio's location DB backend will do the trick? Is there anything I need to worry about, like contacts being periodically purged? If yes, how do I prevent that from happening? And will kamailio always query the DB when performing location lookup or do I need to restart upon inserting new records (to purge caches, for example)? Lastly, I'm not sure how to treat some fields present in usrloc tables like ruid, expires (can I set this to NULL?), callid and cseq (which is irrelevant since there's no REGISTER), flags, cflags and methods. Thanks, George

5 7

xtrange error but those rules that already work, some bug?
by PICCORO McKAY Lenz 23 Sep '19

23 Sep '19

i have many xtrange errors in the log.. but the code seems as i guess completed each rule without error. In the kamailio i have that error by example: Sep 23 13:38:29 ser-server-02 kamailio[2847]: 4(2855) ERROR: *** cfgtrace:request_route=[REGISTRAR] c=[/etc/kamailio/kamailio-devel.cfg] l=692 a=25 n=save that line represents a supposed error in the following code, BUT that code seem work.. kamailio it’s working perfectly.. so i’m worry about it, what it happened? # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) sl_reply_error(); #!ifdef WITH_ASTERISK route(REGFWD); #!endif exit; } why that happened? Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

2 1

kamailio run in docker not working
by Gaurav Bmotra 23 Sep '19

23 Sep '19

hi i m trying to run kamailio in docker this kamailio/kamailio-ci:master-alpine but inside the container kamailio not running if i rung this / # /etc/init.d/kamailio status sh: /etc/init.d/kamailio: not found plz help me thanks -- *Regards:* Gaurav Kumar

2 3

rate-limiting enabled on kamailio.org
by Henning Westerholt 22 Sep '19

22 Sep '19

Hello, just to inform you that rate-limiting on kamailio.org web server has been enabled. The motivation is to throttle robots and other automated services to not overload the server. If you notice any new errors in accessing the service (like page not found, or a 429 Too Many Requests) let us know. Cheers, Henning -- Kamailio Merchandising - https://skalatan.de/merchandising/ Kamailio services - https://skalatan.de/services Henning Westerholt - https://skalatan.de/blog/

1 0

TLS in-dialog set_forward_no_connect()and upstream TLS LCR gateway
by Anthony Joseph Messina 21 Sep '19

21 Sep '19

In preparation for the 5.3 release, I've been testing the following configuration change for TCP/TLS connections: https://github.com/kamailio/kamailio/commit/ 8bba208fe6ae7ccb4c92362b8c33f1530b9f56da route[REQINIT] { # no connect for sending replies set_reply_no_connect(); if(has_totag()) { # no connect for requests within dialog set_forward_no_connect(); } This change creates issues when a UAC TLS INVITE routes to an upstream gateway using TLS to port 5061 (via the LCR module). Kamailio sends the initial outbound TLS connection from a local ephemeral port. The TCPOPS tcp_keepalive_enable function issues keepalives from the local ephemeral port to the gateway port 5061: https://kamailio.org/docs/modules/stable/modules/ tcpops#tcpops.f.tcp_keepalive_enable Even so, the TLS connection eventually times out, after which in-dialog requests from the UAC are no longer able to reach the upstream gateway. ERROR: tm [../../core/forward.h:293]: msg_send_buffer(): tcp_send failed WARNING: tm [t_fwd.c:1570]: t_send_branch(): sending request on branch 0 failed ERROR: sl [sl_funcs.c:372]: sl_reply_error(): stateless error reply used: Unfortunately error on sending to next hop occurred (477/SL) I figure I must be doing something wrong with my TCPOPS here. Is a TLS connection to an upstream gateway supposed to be maintained throughout the duration of a call? -- Anthony - https://messinet.com F9B6 560E 68EA 037D 8C3D D1C9 FF31 3BDB D9D8 99B6

2 6

Info: initialization of libssl 1.1.+ with PTHREAD_PROCESS_SHARED attribute for pthread mutex and rwlock
by Daniel-Constantin Mierla 21 Sep '19

21 Sep '19

Hello, after couple of months waiting for feedback about the workaround with the preloaded shared library to set the PTHREAD_PROCESS_SHARED attribute for pthread mutex/rwlock in order to avoid deadlocks and as it seems to be the solution, I just pushed a few commits to move that code in the core of kamailio. It this way the initialization with the PTHREAD_PROCESS_SHARED attribute is done by default when libssl v1.1 is detected at compile time, without the need of using the approach with a preloaded library. The shared library is still kept in the tls module: * https://github.com/kamailio/kamailio/tree/master/src/modules/tls/utils/open… Probably is going to be removed in the near future. I am asking the community members that use Kamailio on systems with libssl v1.1+ to test the master branch and give feedback. The master branch is anyhow in testing phase for preparing the next major release v5.3.0, so it will help for that as well. If no issue is going to be reported soon, the commits will be backported to stable branches, to re-enable smooth use of Kamailio with libssl v1.1+. When libssl v1.1+ is detected, the core links with pthread, but that was already present in the Makefiles.defs for different compile options. The pthread more or less a default library on linuxes/unixes, so this should not introduce any major dependency to the Kamailio core. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

6 21

Possible locking issue in usrloc module in file udomain.c
by Rupesh Kumar 18 Sep '19

18 Sep '19

Hi , Currently I am running kamailio 4.2 and now upgrading to kamailio 5.2 and was just going thru the changes. In the file modules/usrloc/udomain.c inside the function free_udomain( ) , it appears that we are not protecting via lock_ulslot and unlock_ulslot doing free. Could it cause some race condition ? May be its intentional but going thru the github history its there since day 1. I checked all branches in kamailio 4.x and we have the protection. In 4.x void free_udomain(udomain_t* _d) { int i; if (_d->table) { for(i = 0; i < _d->size; i++) { lock_ulslot(_d, i); deinit_slot(_d->table + i); unlock_ulslot(_d, i); } shm_free(_d->table); } shm_free(_d); } In 5.x void free_udomain(udomain_t* _d) { int i; if (_d->table) { for(i = 0; i < _d->size; i++) { deinit_slot(_d->table + i); } shm_free(_d->table); } shm_free(_d); }

2 2

How to distinguish is from Asterisk or from the client
by przeqpiciel 18 Sep '19

18 Sep '19

Hi, I looking information how to distinguish whether the packet is directed to the Astersik (my platform) or to the customer (softphone / sip client ).

5 5

NAT problem
by Arnout Van Den Kieboom 18 Sep '19

18 Sep '19

Hi, First of all, I'm really new to Kamailio. So sorry if I ask a stupid question, or perhaps a really weird one. I started using kamailio in a test environment about a week ago. for setup i have the situation for inbound calls: Carrier --> FW (NAT) --> Kamailio for outbound to an asterisk box i have: Kamailio --> FW (NAT) --> Asterisk for outbound to users i have: Kamailio --> FW (NAT) --> sip-phone (yealink) or grandstream During that time i was able to : Set up calls to users, Use the dispatcher module use the avpops module to do data lookups. Get calls from asterisk to kamailio user to work. Also calls between users (even though behind nat) are having good audio. I have installed rtpproxy and it works nicely. There is just one situation where it fails: Carrier --> FW (NAT) Kamailio --> asterisk There is only one way audio (from asterisk to carrier) but not from carrier to asterisk. I believe I need to do something with the rtp proxy module ... and tried different things (forcing the r flag, forcing the w flag) Any idea where I might need to start looking? It's been driving me crazy... Thanks Pemp

4 7

app_lua: invalid Lua environment attributes or parameters
by Yuriy Gorlichenko 18 Sep '19

18 Sep '19

Hi! Im using kamailio 5.1.8 with app_lua and getting this message during startup of kamailio. Looks like I forgot to add something at the module setup, but cant figure out what is wrong. My setup of app_lua is: modparam("app_lua", "load", "/etc/kamailio/extended/main.lua") modparam("app_lua","reload",1) modparam("app_lua","register","textops") app_lua [app_lua_sr.c:1599]: sr_kemi_lua_exec_func_ex(): invalid Lua environment attributes or parameters starting kamailio service with kamailio -f /etc/kamailio/extended/main.lua Also here is little bit confusing me that in case of use "KSR" everywhere I'm getting message about "sr"... Thx in advice

2 5

dedicated bearer - Kamailio settings
by Jack R 18 Sep '19

18 Sep '19

Hi, We have installed Kamailio as IMS setup pcscf icscf scscf - 3 instances running on Ubuntu. I have done the attached (image) settings to create dedicated bearer in nextepc as advised by this link: https://github.com/open5gs/nextepc/issues/180 Do I need to do some setting in Kamailio server side (maybe in pcscf cfg files etc.to take some information from PCRF) to make dedicated bearer work end to end or that is all (just configure the new apn in the phone) rest will be taken care itself? Thanks and Regards, Jack.

2 1

core dump if silo table > ~5000 records
by David Villasmil 18 Sep '19

18 Sep '19

Hello guys, i've got a kamailo: version: kamailio 5.0.0 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 19:50:58 Mar 28 2017 with gcc 4.9.2 which was core dumping every 20 seconds and we had no idea why. We emptied out the silo table, which had: mysql> select count(*) from silobackup; +----------+ | count(*) | +----------+ | 5240 | +----------+ records. Once we truncated it, kamailio started working again. Is there a limit to how many records msilo can handle? I will look into the data in the table, but in case there's a limit, i'd like to know. core available on request. Thanks Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 3

Kamailio+B2BUA on IPv4-only and NAT64 Apple app store tests
by Anthony Alba 17 Sep '19

17 Sep '19

Hi folks, I'm wondering what you do to pass Apple app store tests (IPv6-only NAT64/DNS64 scenario) with the iOS app likely to put IPv6 addresses in the SDP, when Kamailio and your B2BUA(Asterisk, FreeSWITCH,...) are IPv4-only endpoints. By "pass" I mean media flowing + SIP working - not sure the app store gatekeepers would actually test media, but it would be nice to have a robust solution. Strictly speaking, this is not a Kamailio's problem but a media SDP/<your-rtp-proxy-of-choice> problem. But since everything is so closely intertwined I'm wondering where you choose to the location the "solution". I have noticed that if Kamailio is IPv4-only and it "sees" IPv6-addresses(Contact, Via headers??) it will try to resolve the IPv6-address as a DNS name! The Wikipedia article on NAT64 mentions 464XLAT(RFC 6877) and Port Control Protocol (RFC 7225), but those seem to be deep rabbit-holes... Cheers Anthony Alba

4 5

Kamailio merchandising shop
by Henning Westerholt 16 Sep '19

16 Sep '19

Hello, there is now a Kamailio merchandising shop available: https://skalatan.de/merchandising/ You find there many options e.g. different Kamailio T-Shirts, Hoodies, Bags, Caps, Mugs, etc.. It provides shipping in many countries world-wide (based from Europe and also USA). Feedback via (direct) e-mail welcome. Best regards, Henning

1 1

3GPP compliance
by Tihomir Culjaga 15 Sep '19

15 Sep '19

Hi, is there any 3GPP compliance list for Kamailio IMS ? Regards, Tihomir.

2 1

RUN kamailio in kubernetes
by Gaurav Bmotra 14 Sep '19

14 Sep '19

hi i want to run kamailio in kubernetes deployment and i want to run this kamailio/kamailio-ci:5.2 image in to the kubernetse .. can anyone help thank you -- *Regards:* Gaurav Kumar

5 6

Problem with incoming audio from Asterisk
by przeqpiciel 14 Sep '19

14 Sep '19

Hi, After a month of fight with SIP signalling I made it but now I have problem with incoming audio from Asterisk. Kamailio and Asterisk are their own public IPs I dont use rtpproxy. When Asterisk sends rtp I'm able to see RTP on my router but it not going to my computer. Maybe somebody have idea why ?

2 1

Using $shv within current call only
by Zhan Bazarov 13 Sep '19

13 Sep '19

Hello, we ran into a trouble with sharing variable into reply route... We are going to match asterisk's uniqueid with rtpengine records like this: route[from_asterisk]{ $var(UniqueId)=$hdr(X-UniqueId); $var(rtp_flags) = $var(rtp_flags) + " label={uid:$var(UniqueId),stream:outgoing} record-call=on"; rtpengine_manage($var(rtp_flags)); } it nicely work for outgoing request, but doesn't for reply.... reply_route[from_pstn]{ rtpengine_manage("replace-origin replace-session-connection ICE=remove RTP/AVP rtcp-mux-demux label={uid:$var(UniqueId),stream:outgoing"); } and it doesn't work... The one possible way what we found its using $shv instead of $var/$avp... But its unsecure way, cuz it may confus our bussines logic... Do you have any idea how to implement this, how we can assign $shv within current call only? Many thanks!

3 2

Issue with <null> and 0 when comparing vars
by Joel Serrano 13 Sep '19

13 Sep '19

Hello, I'm running into what I believe a weird situation when comparing <null> to 0. I have the following lines: xlog("L_NOTICE", "DEBUG: sel(contact.expires)=$sel(contact.expires) - hdr(Expires)=$hdr(Expires)\n"); if ($hdr(Expires)==0 || $sel(contact.expires)==0) { xlog("L_NOTICE", "[end] removed user location\n"); } else { xlog("L_NOTICE", "[end] saved user location\n"); } I registered with 3 devices, one sends expires as header, another as contact param, and the last uses both header AND contact param. My surprise was this: 1- Both header and contact param: Sep 12 19:37:48 csbc01 csbc[8036]: NOTICE: {1 4 REGISTER 02c3ea6ee623916d37532188031017c9@0:0:0:0:0:0:0:0} <script>: DEBUG: sel(contact.expires)=600 - hdr(Expires)=600 Sep 12 19:37:48 csbc01 csbc[8036]: NOTICE: {1 4 REGISTER 02c3ea6ee623916d37532188031017c9@0:0:0:0:0:0:0:0} <script>: [end] saved user location 2- Only contact param: Sep 12 19:36:15 csbc01 csbc[8035]: NOTICE: {1 40629 REGISTER f2cbaa53-eabfd633(a)84.121.150.224} <script>: DEBUG: sel(contact.expires)=600 - hdr(Expires)=<null> Sep 12 19:36:15 csbc01 csbc[8035]: NOTICE: {1 40629 REGISTER f2cbaa53-eabfd633(a)84.121.150.224} <script>: [end] removed user location 2- Only header: Sep 12 19:29:33 csbc01 csbc[8033]: NOTICE: {1 402 REGISTER 2_1804628355(a)192.168.30.31} <script>: DEBUG: sel(contact.expires)=<null> - hdr(Expires)=3600 Sep 12 19:29:33 csbc01 csbc[8033]: NOTICE: {1 402 REGISTER 2_1804628355(a)192.168.30.31} <script>: [end] removed user location I can workaround the header by using is_present_hf() before comparing, I can also probably parse the $ct to see if it has some Expires in it, but I really want to know if my understanding is incorrect and I'm doing something wrong or if this is a bug? Thanks! Joel.

2 2

Testing server up with SIP OPTIONS/netcat
by Andrew White 13 Sep '19

13 Sep '19

Hi all, As part of a custom monitoring solution, I’m writing a simple script that sends an OPTIONS via netcat to a Kamailio server. However the server doesn’t appear to like it. Here’s the text - options.sip: OPTIONS sip:10.0.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.20:5060 From: <sip:monitoring@10.0.0.20> To: sip:monitoring@10.0.0.10 Call-ID: monitoring(a)10.0.0.20 CSeq: 1 OPTIONS I’m running the following command from the 10.0.0.20 machine: nc -u 10.0.0.10 5060 < options.sip And I get the following error in the Kamailio logs: ERROR: <core> [core/parser/parse_cseq.c:56]: parse_cseq(): method terminated unexpectedly ERROR: <core> [core/parser/parse_cseq.c:92]: parse_cseq(): bad cseq ERROR: <core> [core/parser/msg_parser.c:144]: get_hdr_field(): bad cseq ERROR: <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [CSeq: 1 OPTIONS] ERROR: <core> [core/parser/parse_cseq.c:56]: parse_cseq(): method terminated unexpectedly ERROR: <core> [core/parser/parse_cseq.c:92]: parse_cseq(): bad cseq ERROR: <core> [core/parser/msg_parser.c:144]: get_hdr_field(): bad cseq ERROR: <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [CSeq: 1 OPTIONS] WARNING: <core> [core/receive.c:319]: receive_msg(): parsing relevant headers failed ERROR: <core> [core/parser/parse_cseq.c:56]: parse_cseq(): method terminated unexpectedly ERROR: <core> [core/parser/parse_cseq.c:92]: parse_cseq(): bad cseq ERROR: <core> [core/parser/msg_parser.c:144]: get_hdr_field(): bad cseq ERROR: <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [CSeq: 1 OPTIONS] ERROR: pv [pv_core.c:1893]: pv_get_hdr(): error parsing headers ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/parse_cseq.c:56]: parse_cseq(): method terminated unexpectedly ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/parse_cseq.c:92]: parse_cseq(): bad cseq ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/msg_parser.c:144]: get_hdr_field(): bad cseq ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [CSeq: 1 OPTIONS] ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/parse_cseq.c:56]: parse_cseq(): method terminated unexpectedly ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/parse_cseq.c:92]: parse_cseq(): bad cseq ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/msg_parser.c:144]: get_hdr_field(): bad cseq ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [CSeq: 1 OPTIONS] ERROR: {1 <null> monitoring(a)13.236.91.139} maxfwd [mf_funcs.c:51]: is_maxfwd_present(): parsing MAX_FORWARD header failed! ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/parse_cseq.c:56]: parse_cseq(): method terminated unexpectedly ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/parse_cseq.c:92]: parse_cseq(): bad cseq ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/msg_parser.c:144]: get_hdr_field(): bad cseq ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [CSeq: 1 OPTIONS] ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/msg_translator.c:2361]: build_res_buf_from_sip_req(): alas, parse_headers failed As far as I can see, the message itself is valid. It appears parse_cseq() is expecting a higher cseq, or more headers? I’ve looked at another OPTIONS message we get regularly from one of our upstream, and it’s almost identical to this one. Can anyone point me the right way here? Thanks! Andrew

3 2

Presence and BLF
by Bill Neely 12 Sep '19

12 Sep '19

Hi all: I am trying to set up presence and BLF notifies on Kamailo 5.2.4 I am following the instructions laid out in http://kb.asipto.com/kamailio:presence:k43-blf Don't know if these instructions are valid, but I am definitely not getting any light action (using Grandstream 2130 and Grandstream Wav for testing.) Entries in active_watchers appear to be correct. Here is my cfg file: #!KAMAILIO # # Kamailio (OpenSER) SIP Server v5.1 - default configuration script # - web: https://www.kamailio.org # - git: https://github.com/kamailio/kamailio # # Direct your questions about this file to: <sr-users(a)lists.kamailio.org> # # Refer to the Core CookBook at https://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Note: the comments can be: # - lines starting with #, but not the pre-processor directives, # which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef, # #!subst, #!substdef, ... # - lines starting with // # - blocks enclosed in between /* */ # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" #!define WITH_DEBUG #!define WITH_NAT #!define WITH_AUTH #!define WITH_MYSQL #!define WITH_USRLOCDB #!define WITH_PRESENCE ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLT_DLG 9 #!define FLT_DLGINFO 10 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=3 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ # auto_aliases=no /* add local domain aliases */ # alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to * bind on a specific interface/port/proto (default bind on all available) */ listen=udp:62.151.xx.xx:50060 # allow http requests listen=tcp:127.0.0.1:8080 tcp_accept_no_cl=yes #!ifdef WITH_TLS enable_tls=yes #!endif /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "104.192.xx.xx" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib64/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "presence_dialoginfo.so" loadmodule "pua.so" loadmodule "pua_dialoginfo.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # load the module loadmodule "xhttp.so" loadmodule "json.so" loadmodule "pua_json.so" loadmodule "dialog.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) modparam("presence", "server_address", "sip:62.151.xx.xx:50060" ) modparam("presence", "send_fast_notify", 0) modparam("presence", "db_update_period", 20) modparam("presence", "clean_period", 40) modparam("presence", "subs_db_mode", 2) modparam("presence", "fetch_rows", 1000) modparam("presence", "local_log_level", 3) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) # ----- presence_dialoginfo params ----- modparam("presence_dialoginfo", "force_single_dialog", 0) # ----- pua params ----- modparam("pua", "db_url", DBURL) modparam("pua", "db_mode", 2) modparam("pua", "update_period", 60) modparam("pua", "dlginfo_increase_version", 0) modparam("pua", "reginfo_increase_version", 0) modparam("pua", "check_remote_contact", 1) modparam("pua", "fetch_rows", 1000) # ----- pua_dialoginfo params ----- modparam("pua_dialoginfo", "include_callid", 1) modparam("pua_dialoginfo", "send_publish_flag", FLT_DLGINFO) modparam("pua_dialoginfo", "caller_confirmed", 0) modparam("pua_dialoginfo", "include_tags", 1) modparam("pua_dialoginfo", "override_lifetime", 124) #!endif # ----- dialog params ----- modparam("dialog", "db_url", DBURL) modparam("dialog", "enable_stats", 1) modparam("dialog", "db_mode", 1) modparam("dialog", "dlg_flag", FLT_DLG) #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { xlog("L_ALERT","ALERT: REQUEST ROUTE ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^6 $rm from $fu (IP:$si:$sp) $du\nMESSAGE BUFFER:\n$mb\n\n\nRU:$ru\n"); $var(i)=0; while($var(i)<$branch(count)) { xlog("$(branch(uri)[$var(i)])\n"); $var(i) = $var(i) + 1; } # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle registrations route(REGISTRAR); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); xlog("L_ALERT","ALERT: line 492******************************** $rm from $fu (IP:$si:$sp)\n"); if (is_method("INVITE|SUBSCRIBE")) { xlog("L_ALERT","ALERT: line 494******************************** $rm from $fu (IP:$si:$sp)\n"); record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } event_route[xhttp:request] { if ($hu != "/presence/") { xlog("L_INFO", "invalid request [$hu]"); xhttp_reply("404", "Not Found", "text/html", ""); return; } $var(call-id) = $(rb{json.parse,Call-ID}); xlog("L_INFO", "$var(call-id)|log|[$hu] payload $rb"); if ($(rb{json.parse,Event-Package}) == "dialog") { xlog("L_INFO", "$var(call-id)|log|received $(rb{json.parse,Event-Package}) update for $(rb{json.parse,From}) $rb"); pua_json_publish($rb); } xhttp_reply("200", "OK", "text/html", "$rb"); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if(is_method("INVITE|BYE|UPDATE|CANCEL|ACK")) { setflag(FLT_DLGINFO); dlg_manage(); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) { sl_reply_error(); } exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # 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; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { xlog("ALERT:NATMANAGE********************* (IP:$si:$sp)\n(DEST:$du @ $di:$dp)\n"); ; #!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; if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("corwie"); } 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(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains # Routing to foreign domains route[SIPOUT] { xlog("ALERT: SIPOUT (IP:$si:$sp)\n(DEST:$du @ $di:$dp)\n"); ; if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } route[TOAST] { rewritehostport("45.33.xx.xx:5060"); $fu = "sip:7028743833@62.151.182.83:5012"; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif }

2 2

Kamailio IMS setup - user online - unable to call
by Jack R 12 Sep '19

12 Sep '19

Hi All, We have installed Kamailio as IMS setup pcscf icscf scscf - 3 instances running on Ubuntu.. Users alice and bob are online however unable to call. When we check the logs the FHoSS is hunting for values in sh_subscription table for alice / bob user. Can you please share: 1. A sample database table values to be inserted across all tables in FHoSS helpful for the call and 2, Maybe, if possible, some description to understand table structures/ importance of fields of FHoSS. Thanks and Regards, Jack.

2 1

Help with "routines:ssl3_read_bytes:sslv3 alert bad certificate" kamailio TLS error
by david＠aslo.us 12 Sep '19

12 Sep '19

Hello everyone, I am trying to configure TLS in kamailio (5.2.4) following this guide: [ http://www.kamailio.org/dokuwiki/doku.php/tls:create-certificates ]( http://www.kamailio.org/dokuwiki/doku.php/tls:create-certificates ) Modules: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_PRESENCE #!define WITH_ALIASDB #!define WITH_IMC #!define WITH_TLS When i try to connect via command line, this is the result (just including relevant parts): $ openssl s_client -connect 192.X.X.X:5061 -tls1 CONNECTED(00000003) depth=1 C = XX, ST = XXXX, L = XXXXXX, O = XXX CA, CN = XXX CA verify error:num=19:self signed certificate in certificate chain verify return:0 --- No client certificate CA names sent --- SSL handshake has read 2550 bytes and written 336 bytes --- --- Start Time: 1567787935 Timeout : 7200 (sec) Verify return code: 19 (self signed certificate in certificate chain) --- read:errno=0 Now, when I setup my clients, they connect to the server, but they can't send messages or make calls. This is the TLS startup LOG: Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_mod.c:372]: mod_init(): With ECDH-Support! Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_mod.c:375]: mod_init(): With Diffie Hellman Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: CRITICAL: tls [tls_init.c:671]: init_tls_h(): installed openssl library version is too different from the library the kamailio tls module was compiled with: installed "OpenSSL 1.1.1 11 Sep 2018" (0x1010100f), compiled "OpenSSL 1.1.0k 28 May 2019" (0x101000bf).#012 Please make sure a compatible version is used (tls_force_run in kamailio.cfg will override this check) Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: WARNING: tls [tls_init.c:680]: init_tls_h(): tls_force_run turned on, ignoring openssl version mismatch Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: WARNING: tls [tls_init.c:778]: init_tls_h(): openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 12582912 and 6291456 bytes Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: <core> [core/cfg/cfg_ctx.c:595]: cfg_set_now(): tls.low_mem_threshold1 has been changed to 12582912 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: <core> [core/cfg/cfg_ctx.c:595]: cfg_set_now(): tls.low_mem_threshold2 has been changed to 6291456 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: <core> [main.c:2669]: main(): processes (at least): 24 - shm size: 67108864 - pkg size: 8388608 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: <core> [core/udp_server.c:153]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:303]: ksr_tls_fill_missing(): TLSs<default>: tls_method=12 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:315]: ksr_tls_fill_missing(): TLSs<default>: certificate='/etc/certs/192.X.X.X/cert.pem' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:322]: ksr_tls_fill_missing(): TLSs<default>: ca_list='/etc/certs/demoCA/cert.pem' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:329]: ksr_tls_fill_missing(): TLSs<default>: crl='(null)' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:333]: ksr_tls_fill_missing(): TLSs<default>: require_certificate=0 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:340]: ksr_tls_fill_missing(): TLSs<default>: cipher_list='(null)' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:347]: ksr_tls_fill_missing(): TLSs<default>: private_key='/etc/certs/192.X.X.X/key.pem' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:351]: ksr_tls_fill_missing(): TLSs<default>: verify_certificate=0 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:354]: ksr_tls_fill_missing(): TLSs<default>: verify_depth=9 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: NOTICE: tls [tls_domain.c:1087]: ksr_tls_fix_domain(): registered server_name callback handler for socket [:0], server_name='<default>' ... Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:707]: set_verification(): TLSs<default>: No client certificate required and no checks performed Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:303]: ksr_tls_fill_missing(): TLSc<default>: tls_method=12 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:315]: ksr_tls_fill_missing(): TLSc<default>: certificate='(null)' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:322]: ksr_tls_fill_missing(): TLSc<default>: ca_list='(null)' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:329]: ksr_tls_fill_missing(): TLSc<default>: crl='(null)' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:333]: ksr_tls_fill_missing(): TLSc<default>: require_certificate=0 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:340]: ksr_tls_fill_missing(): TLSc<default>: cipher_list='(null)' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:347]: ksr_tls_fill_missing(): TLSc<default>: private_key='(null)' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:351]: ksr_tls_fill_missing(): TLSc<default>: verify_certificate=0 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:354]: ksr_tls_fill_missing(): TLSc<default>: verify_depth=9 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:710]: set_verification(): TLSc<default>: Server MAY present invalid certificate Sep 6 16:41:58 aslo-kamailio /usr/sbin/kamailio[5862]: INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/5862 Sep 6 16:41:58 aslo-kamailio /usr/sbin/kamailio[5866]: INFO: ctl [io_listener.c:214]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) This is my tls.cfg file: [server:default] method = TLSv1 verify_certificate = no require_certificate = no private_key = /etc/certs/192.X.X.X/key.pem certificate = /etc/certs/192.X.X.X/cert.pem ca_list = /etc/certs/demoCA/cert.pem #crl = /etc/kamailio/tls/crl.pem # --- # This is the default client domain profile. # Settings in this domain will be used for all outgoing # TLS connections that do not match any other # client domain in this configuration file. # We require that servers present valid certificate. # [client:default] method = TLSv1 verify_certificate = no require_certificate = no These are the relevant parts of my kamailio.cfg: # alias="sip.mydomain.com" alias=192.X.X.X:5060 alias=192.X.X.X:5061 /* uncomment and configure the following line if you want Kamailio to * bind on a specific interface/port/proto (default bind on all available) */ listen=udp:192.X.X.X:5060 listen=tcp:192.X.X.X:5060 listen=tls:192.X.X.X:5061 #!ifdef WITH_TLS enable_tls=yes /* upper limit for TLS connections */ tls_max_connections=2048 #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") modparam("tls", "tls_force_run", 1) #!endif These are the errors that show up everytime i try to connect with a client: Sep 6 16:53:42 aslo-kamailio /usr/sbin/kamailio[5870]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14094412:SSL routines:ssl3_read_bytes:sslv3 alert bad certificate Sep 6 16:53:42 aslo-kamailio /usr/sbin/kamailio[5870]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f7c4e3ddd00 r: 0x7f7c4e3ddd80 (-1) Sep 6 16:53:43 aslo-kamailio /usr/sbin/kamailio[5874]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14094412:SSL routines:ssl3_read_bytes:sslv3 alert bad certificate Sep 6 16:53:43 aslo-kamailio /usr/sbin/kamailio[5874]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f7c4e3ddd00 r: 0x7f7c4e3ddd80 (-1) Sep 6 16:53:44 aslo-kamailio /usr/sbin/kamailio[5875]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14094412:SSL routines:ssl3_read_bytes:sslv3 alert bad certificate Sep 6 16:53:44 aslo-kamailio /usr/sbin/kamailio[5875]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f7c4e3ddd00 r: 0x7f7c4e3ddd80 (-1) Any help would be greatly appreciated. Regards.

2 1

Kamailio status get Active (Exited)
by Gaurav Bmotra 11 Sep '19

11 Sep '19

hi i m using kamailio 5.1, with -m 512 -M 8 it stop registering user SIP anf in status it shows ------------------------------ ● kamailio.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio; generated) Active: active (exited) since Sat 2019-09-07 11:05:16 UTC; 3 days ago Docs: man:systemd-sysv-generator(8) Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17708]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17677]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17714]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17685]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17681]: {1 1094155754 NOTIFY TwGwmzx4wINnraJs1Gh3Kg..} INFO: <core> [main.c:819]: sig_usr(): s… received Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17704]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17689]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17693]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17691]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17674]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Hint: Some lines were ellipsized, use -l to show in full. --------------------------------------------------------------------- plz help thank you -- *Regards:* Gaurav Kumar

2 1

Can't connect to db
by Andrew Chen 10 Sep '19

10 Sep '19

Hi all, I just installed kamailio 5.2.4 on two separate aws instances in two separate regions. Both regions uses the same configuration template and share the same db in AWS RDS. The issue I see is this: - SJO location was able to launch fine without any issues. - ASH location could not start up kamailio process due to db connection failure: Sep 10 21:24:13 ashmainkama51 /usr/sbin/kamailio[8043]: ERROR: db_mysql [km_my_con.c:139]: db_mysql_new_connection(): driver error: Too many connections Sep 10 21:24:13 ashmainkama51 /usr/sbin/kamailio[8043]: ERROR: <core> [db.c:319]: db_do_init2(): could not add connection to the pool Sep 10 21:24:13 ashmainkama51 /usr/sbin/kamailio[8045]: ERROR: db_mysql [km_my_con.c:139]: db_mysql_new_connection(): driver error: Too many connections Sep 10 21:24:13 ashmainkama51 /usr/sbin/kamailio[8045]: ERROR: <core> [db.c:319]: db_do_init2(): could not add connection to the pool Sep 10 21:24:13 ashmainkama51 /usr/sbin/kamailio[8045]: ERROR: carrierroute [db_carrierroute.c:141]: carrierroute_db_open(): can't connect to database. Sep 10 21:24:13 ashmainkama51 /usr/sbin/kamailio[8045]: ERROR: <core> [core/sr_module.c:847]: init_mod_child(): error while initializing module carrierroute (/usr/lib/x86_64-linux-gnu/kamailio/modules/carrierroute.so) (idx: 30 rank: 18 desc: [tcp receiver (generic) child=9]) - In ASH instance, I can manually mysql to the RDS host using the credentials in the config file. - I checked db user permissions and it's opened for the subnet in our ASH location (which is also proven using manual connection). So I don't get why one works and the other doesn't Thoughts? -- Andy Chen Sr. Telephony Lead Engineer -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 2

LCR lcr_id short or int?
by Grant Bagdasarian 10 Sep '19

10 Sep '19

Hello, The database tables definition for 5.2.x mentions the lcr_id parameter being of unsigned short type, while in the code I only see definitions of type int. Is it safe to assume the parameter is of type int, so the database column can also be set to an integer type? Regards, Grant

2 7

Share variable into reply route.
by Zhan Bazarov 09 Sep '19

09 Sep '19

Hello, we ran into a trouble with sharing variable into reply route... We are going to match asterisk's uniqueid with rtpengine records like this: route[from_asterisk]{ $var(UniqueId)=$hdr(X-UniqueId); $var(rtp_flags) = $var(rtp_flags) + " label={uid:$var(UniqueId),stream:outgoing} record-call=on"; rtpengine_manage($var(rtp_flags)); } reply_route[from_pstn]{ rtpengine_manage("replace-origin replace-session-connection ICE=remove RTP/AVP rtcp-mux-demux label={uid:$var(UniqueId),stream:outgoing"); } and it doesn't work... The one possible way what we found its using $shv instead of $var... But its unsecure way, cuz it may confus our bussines logic... Do you have any idea how to implement this? Many thanks!

3 2

How to install kamailio symbols for debugging
by YASIN CANER 09 Sep '19

09 Sep '19

Hello, I want to debug kamailio with gdb or try to get core.dump file but i failed. GDB wants to kamailio symbols. How to install them? on Ubuntu - raspberry PI Best regards. Yasin CANER

4 3

Error Users - 604 Does not exist anywhere - HSS User Unknown
by Jack R 08 Sep '19

08 Sep '19

Hi, We have installed Kamailio as IMS setup pcscf icscf scscf - 3 instances running on Ubuntu.. 1. 2 Sip client registers successfully (shows Online) as 'bob(a)net1.test' and 'alice(a)net1.test' 2. When we check server logs: Invite request shows 'bob@IP address' i.e. ' bob(a)10.0.0.10' 3. However, in HSS it is registered as 'bob(a)net1.test'. 4. When the request from SCSCF goes to ISCSF it gives Error: Users - 604 Does not exist anywhere - HSS User Unknown. As it compares 'bob(a)10.0.0.10' with 'bob(a)net1.test'. Please help and advise how everywhere (SCSCF, PCSCF) during query/comparisons it can take 'bob(a)net1.test' and not 'bob(a)10.0.0.10' Thanks and Regards, Jack.

1 1

Kamailio - PfSense Firewall - inbound/outbound rules
by Youssef Boujraf 06 Sep '19

06 Sep '19

Dear All, I am evaluating the kamailio proxy 5.2 server with rtpproxy on an internal network. Everything are ok. Now, I would like to evalute Kamailio proxy from outside (internet) but I have some difficulties to setup my PfSense firewall to enable inbound & outbound rules, network settings and kamailio.cfg. The issues are : Setup correctly Network files : * /etc/hostname 127.0.0.1 localhost ip_private codecsip.mydomain codecsip ip_public codecsip.mydomain codecsip * /etc/hosts file codesip * dns * /etc/resolv.conf nameserver 8.8.8.8Setup /etc/kamailio/kamailio.cfg :#!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_NAT #!define WITH_TLS /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ alias="codecsip.mydomain" #!ifdef WITH_NAT # ----- rtpproxy params ----- # modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") modparam("rtpproxy", "rtpproxy_sock", "unix:/var/run/rtpproxy/rtpproxy.sock") Setup etc/default/rtpproxy : CONTROL_SOCK=udp:127.0.0.1:7722 EXTRA_OPTS="-l <IP-address>" Where <IP-address> is the external IP address of your host. Regarding the firewall rules, it will be great to identify which rules I need to create : Inbound and outbound rules and NAT 1:1 ? If anyone have any information it could be great Best regards, Youssef

2 1

Kamailio HA
by Christian Attanasio 06 Sep '19

06 Sep '19

Hi everyone I use kamailio 5.1 with rtpengine and I would like to set up a simple warm standby or hot standby system for my sip server. on the web I found various discussions and posts that led me to deepen them the dispatcher, dmq, dmq_usrloc, db_cluster and p_usrloc modules. now, though, I'm a little confused: what approach do you recommend for this system? I thought of using two identical servers: Kam01 10.10.10.2 Kam02 10.10.10.3 Hostname cluster.mydomain.com each with its own mariadb and implementing data replication (with the dmq module? Or db_cluster and sqlops?). in this scenario is it necessary to also use the dispatcher module or implement a high available system with keepalives and floating ip? thanks, Christian

1 0

Kamalio & Asterisk integration erroneous behavior, setup issue
by Aristeidis Tsitras 05 Sep '19

05 Sep '19

new to the area and trying to setup Kamailio with Asterisk in a single machine. All users will register to Kamailio's port and in case of need for media, it will be forwarded to Asterisk, that is my intention. All of my work is based on the following link https://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb. Here is what i have done: - Debian 8, 64 bit machine with mysql and odbc - *root@kamast: ~ $ lsb_release -a No LSB modules are available. Distributor ID: Debian Description: Debian GNU/Linux 8.11 (jessie) Release: 8.11 Codename: jessie root@kamast: ~ $ uname -a Linux kamast 3.16.0-10-amd64 #1 SMP Debian 3.16.72-1 (2019-08-13) x86_64 GNU/Linux root@kamast: ~ $ * - Kamailio 5.2 installed from Kamailio's deb repository - Asterisk 13LTS installed from source - Used the same passwords such as kamailiorw and asterisk_password, since this is a test system, for proof of concept. I did import to the mysql>asterisk database 3 users 2200, 2201 and 2202. Then created in sip.conf the same 3 users with the same credentials. Then on 3 PCs i used softphones (Jitsi, Zoiper) and registered each account to a softphone. Problems: - Cannot see the users in the Asterisk's cli, sip show peers - I can see users only in Kamailio with kamctl ul show - A call between the extensions goes to voicemail. It never reaches the other destination eg 2200 calls 2201 and in Asterisk's console i am getting a message that 2201 is absent and it goes to voicemail. The same with any other extension. Attached you can find: 1. Kamailio.cfg 2. Asterisk's sip.conf 3. Asterisk's extension.conf 4. The import that i have done to mysql for the user creation. I would appreciate if someone could point me to the error and help me fix it please?

3 7

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