- sr-users - kamailio.org

How to get the id or ip in the multiple rtproxies for LB with weights。
by mm e 15 Jul '24

15 Jul '24

modparam("rtpengine", "rtpengine_sock","udp:192.168.1.3:12221=2 udp:192.168.1.1:12222=1 udp:192.168.1.5:12221=7") When rtpengine_sock is set to multiple rtproxies mode, how to get the id or ip of the rtpengine used in the current call? There is no information returned which rtpengine kamailio selected for the call.

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Re: (no subject)
by Sergio Charrua 12 Jul '24

12 Jul '24

Hi Brett! From what I could understand, my client (a Carrier) assigns services and products to its customers identified by trunkgroups (depending on the customer and the service requested, as well as the ingress trunkgroup the call will be forwarded to a specific egress trunkgroup). . The INVITE is sent by customer to the SBC (ingress), where some kind of process (which i have no details) will decide what IP:Port to forward the messages to Kamailio with additional infos (Kamailio is listening on different NICs/IPs/Ports). Then, Kamailio will check on the (external) routing engine the route to take and forwards to the outbound endpoint. Depending on the IP:Port that Kamailio receives the call (SBC may choose one NIC or the other) the request is then received by Kamailio and verified/processed by the routing engine replying back to kamailio with the route to take, and then Kamailio forwards to the outbound endpoint (egress SBC...). It is a bit confusing, and I want to simplify things as much as possible, but apparently the old/current solution is set this way and they pretend to keep the new solution as similar as possible to the old/current solution. This is mostly due to the billing process, which is very (overly?) complex and they do not pretend to change it. IMHO, the routing engine should be much simpler but it is a much more complex process the Carrier currently has...and wishes to keep. (sorry for posting without a subject. I did, however, re-posted with corrected subject) Atenciosamente / Kind Regards / Cordialement, *Sérgio Charrua* On Fri, Jul 12, 2024 at 8:11 PM Brett Nemeroff <brett(a)voicefoxtelephony.com> wrote: > Is there a reason you have to identify the trunk group based on received > IP/Port? I've seen this as a requirement from some more antiquated systems > and it's always a pain. It's always better to use source IP or even a trunk > prefix (dialed number prefix) which isn't really secure, but works when > security isn't an issue. > > If you have to do it that way, I'd probably turn the $Ri/$Rp into some > sort of cache key like $Ri_$Rp and map it to something like a dispatcher > setid. > > There may be a more modern way to do this, but I don't think there is. > > Good luck! > -Brett > > > On Fri, Jul 12, 2024 at 1:40 PM Sergio Charrua via sr-users < > sr-users(a)lists.kamailio.org> wrote: > >> Hi all! >> >> for a project under development, one requirement is to be able to >> forward/relay the call to a specific destination depending on which IP >> Address and Port the call was received by Kamailio. >> This also means that Kamailio will be listening on multiple IP Address >> and Ports >> listen=udp:IP_1:port_1 # trunkgroup 1 >> listen=udp:IP_2:port_2 # trunkgroup 2 >> listen=udp:IP_3:port_3 # trunkgroup 3 >> >> I know the pv $Ri and $Rp can be used to identify from which IP address >> and port the message reached Kamailio and having those details, depending >> on the value, I can define some conditions that allows Kamailio to relay >> the call to different destination endpoints. >> For example: >> - calls from +32 to + 39 received on trunkgroup1 needs to go to Italy SBC >> address A.B.C.D >> - but calls from +32 to +39 received on trunkgroup2 need to go to Germany >> on SBC address 1.2.3.4 (as it is cheaper) >> >> Is there a better way of doing this without using IF/THEN/ELSE or SWITCH >> blocks and "hard code" destination SBC addresses? >> I have read the DRouting module's documentation but not sure if it could >> help and how... >> >> Any suggestions? >> >> >> >> Atenciosamente / Kind Regards / Cordialement, >> >> >> *Sérgio * >> __________________________________________________________ >> Kamailio - Users Mailing List - Non Commercial Discussions >> To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org >> Important: keep the mailing list in the recipients, do not reply only to >> the sender! >> Edit mailing list options or unsubscribe: >> > > > -- > > ======================================================== > Brett Nemeroff > Voice Fox Telephony LLC > Office: 512-670-8369 > Email: brett(a)voicefoxtelephony.com >

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Change the To Header in the failure_route
by sadik.oualla.mohamed＠gmail.com 12 Jul '24

12 Jul '24

Hello, I need help with modifying the "To" header in the "failure_route[ID]{...}". Specifically, I want to add the "to tag" parameter from the previous SIP messages of the current dialog(e.g., 180 Ringing, PRACK or final responses). Here's the scenario: 1. Kamailio receives and relays the SIP messages: INVITE, 180 Ringing, PRACK, 200 OK, 486 Busy Here. 2. In my case, the 180 Ringing includes the "to tag" parameter. 3. When Kamailio receives the 486 response, it triggers "failure_route[ID]" using "t_on_failure(ID)" method. 4. I extract the SIP response code and reason from the final response of the INVITE transcation in this failure_route. 5. After processing and applying some instructions, I send a reply using "t_send_reply" with the code and reason extracted. The problem: - The failure_route[ID] holds the initial INVITE, which has no "to tag" parameter. - When I send a reply using "t_send_reply()", Kamailio generates a random "to tag" different from the previous SIP messages, which confuses my UAC. I think adding the "to tag" parameter manually might be a solution, but I'm not sure if I'm doing it correctly or if there is a better approach. I also tried using uac_replace_to() method, but it doesn't work in the failure_route[ID]. How can I make sure the reply in failure_route[ID] uses the correct "to tag" from the previous SIP messages? Are there any other possibilities to achieve this? Any assistance would be appreciated. Thank you!

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Kemi Python3 throws a TypeError
by Michel Pelletier 12 Jul '24

12 Jul '24

Hello, Please help. I am using Kemi with the python3 module. I get random TypeError exceptions thrown within the Kamailio code. For example I have the following code: event = str(KSR.pv.get("$hdr(Event)")) log.logInfo( "KamailioStateful.ksr_request_route_loopback(): " + "Passing the request to the presence " + "publish handler for event: " + event ) rc = KSR.presence.handle_publish() if rc != 1: log.logError( "KamailioStateful.ksr_request_route_loopback(): " + "The presence publish handler returned " + "an error code: " + str(rc) ) This is what I see in the logs: 21(120) ERROR: PY3 {PUBLISH}: app_python3 [python_support.c:167]: python_handle_exception(): apy_exec: ksr_request_route((null)): Unhandled exception in the Python code: TypeError: expected bytes, str found The above exception was the direct cause of the following exception: Traceback (most recent call last): File "/etc/kamailio/kamailio.py", line 91, in ksr_request_route return ksf.ksr_request_route(msg, "Loopback") ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ File "/etc/kamailio/kamailio_stateful.py", line 100, in ksr_request_route return self.ksr_request_route_loopback(msg) ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ File "/etc/kamailio/kamailio_stateful.py", line 240, in ksr_request_route_loopback rc = KSR.presence.handle_publish() ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ SystemError: <built-in function handle_publish> returned a result with an exception set If I put the offending line within a Try/Except it hides it but I would rather not see it not happen at all. Any Kemi Python3 users experiencing something similar? What could be causing this? Regards, Michel Pelletier

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routing call depending on which IP address SIP message reached Kamailio
by Sergio Charrua 12 Jul '24

12 Jul '24

Hi all! For a project under development, one requirement is to be able to forward/relay the call to a specific destination depending on which IP Address and Port the call was received by Kamailio. This also means that Kamailio will be listening on multiple IP Address and Ports listen=udp:IP_1:port_1 # trunkgroup 1 listen=udp:IP_2:port_2 # trunkgroup 2 listen=udp:IP_3:port_3 # trunkgroup 3 I know the pv $Ri and $Rp can be used to identify from which IP address and port the message reached Kamailio and having those details, depending on the value, I can define some conditions that allows Kamailio to relay the call to different destination endpoints. For example: - calls from +32 to + 39 received on trunkgroup1 needs to go to Italy SBC address A.B.C.D - but calls from +32 to +39 received on trunkgroup2 need to go to Germany on SBC address 1.2.3.4 (as it is cheaper) Is there a better way of doing this without using IF/THEN/ELSE or SWITCH blocks and "hard code" destination SBC addresses? I have read the DRouting module's documentation but not sure if it could help and how... Any suggestions? Atenciosamente / Kind Regards / Cordialement, *Sérgio *

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(no subject)
by Sergio Charrua 12 Jul '24

12 Jul '24

Hi all! for a project under development, one requirement is to be able to forward/relay the call to a specific destination depending on which IP Address and Port the call was received by Kamailio. This also means that Kamailio will be listening on multiple IP Address and Ports listen=udp:IP_1:port_1 # trunkgroup 1 listen=udp:IP_2:port_2 # trunkgroup 2 listen=udp:IP_3:port_3 # trunkgroup 3 I know the pv $Ri and $Rp can be used to identify from which IP address and port the message reached Kamailio and having those details, depending on the value, I can define some conditions that allows Kamailio to relay the call to different destination endpoints. For example: - calls from +32 to + 39 received on trunkgroup1 needs to go to Italy SBC address A.B.C.D - but calls from +32 to +39 received on trunkgroup2 need to go to Germany on SBC address 1.2.3.4 (as it is cheaper) Is there a better way of doing this without using IF/THEN/ELSE or SWITCH blocks and "hard code" destination SBC addresses? I have read the DRouting module's documentation but not sure if it could help and how... Any suggestions? Atenciosamente / Kind Regards / Cordialement, *Sérgio *

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deb package build error (v5.7.6) [error: debian/rules binary subprocess returned exit status 2]
by Denys Pozniak 12 Jul '24

12 Jul '24

Hello! I'm trying to build standard deb packages for Kamailio v5.7.6 for Debian 12.6, but I get an error. By the way, there are no problems with building v5.7.2 ... #git clone --single-branch --branch 5.7.6 https://github.com/kamailio/kamailio.git /usr/src/kamailio #cd /usr/src/kamailio && ln -s pkg/kamailio/deb/bookworm debian #make cfg && RADCLI=1 make deb .... Makefile.defs defs skipped make[6]: 'libkamailio_ims.so.0.1' is up to date. gcc -shared -g -m64 -Wl,-O2 -Wl,-E -Wl,-z,relro -pthread -rdynamic "-ldl" -Wl,-Bsymbolic-functions cmd.o ims_ipsec_pcscf_mod.o ipsec.o sec_agree.o spi_gen.o spi_list.o spi_list_tests.o -lm -lmnl -L../../lib/ims/ -lkamailio_ims -Wl,-rpath,/usr/src/kamailio/src/lib/ims -o ims_ipsec_pcscf.so make[4]: Leaving directory '/usr/src/kamailio/src' make[3]: Leaving directory '/usr/src/kamailio' make[2]: Leaving directory '/usr/src/kamailio' *dpkg-buildpackage: error: debian/rules binary subprocess returned exit status 2* make[1]: Leaving directory '/usr/src/kamailio/src' -- BR, Denys Pozniak

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New Member Introduction
by Maria Jonas 11 Jul '24

11 Jul '24

Hi everyone, I am new to this forum and excited to be here. I'm eager to learn more about Kamailio and its features and I'm looking forward to participating in discussions here. Could someone please guide me on how to ask questions here ? Where should I post if I have a query ? Looking forward to your advice and connecting with you all. Thanks,

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usrloc ka_interval timer is not correct in kamailio 5.8
by Michal Popovič 11 Jul '24

11 Jul '24

Hello everybody, I am using kamailio in AWS behind load balancer and after upgraded from version 5.7 to 5.8 kamailio is not able to keep connection open. After quick investigation I found out that usrloc module got new parameter ka_interval that replaced timer_interval. Change is here. https://github.com/kamailio/kamailio/commit/53ace443020075f7ecd12c72f154193… Problem is that this interval is not reliable, as it is most probably still called only when timer_interval is executed. If I set values to timer_interval = 30 ka_interval = 30 ka_randomize = 20 Maximum time for keep-alive is not 30+20 seconds but 30+30+20 seconds. That is too late for AWS load balancer. I have fixed it by enabling tcp keepalives on kernel level. Is there any way how to specify exact time in version 5.8? Thanks. Michal

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HTTP Client configuration path
by Sergio Charrua 11 Jul '24

11 Jul '24

Hi all! I am using HTTP_CLIENT module, and have created a http configuration file. The module's config_file parameter is set as: modparam("http_client", "config_file", "/etc/kamailio/http_client.cfg") and it works OK on my DEV environment, because there is a symlink /etc/kamailio pointing to physical folder /usr/local/etc/kamailio/ (using Kamailio 5.7.4) However in our integration environment (also using Kamailio 5.7.4), which is identical to production, and due to limitations on access rights, we cannot create a symlink /etc/kamailio pointing /usr/local/etc/kamailio/, so all cfg files are in the installation folder. So I thought that the config_file parameter should be a relative path, starting with " ./ " or by just setting the config file name, without path (as per documentation at http_client (kamailio.org) <https://kamailio.org/docs/modules/5.1.x/modules/http_client.html#idp38265140> . The issue is that even though the file http_client.cfg is present in /usr/local/etc/kamailio/, as well as other cfg files, the cfg_parser fails to load it: Jul 11 09:32:09 lab-kamailio kamailio[44114]: ERROR: <core> > [core/cfg_parser.c:632]: cfg_parser_init(): cfg_parser: Unable to open file > './http_client.cfg' > Jul 11 09:32:09 lab-kamailio kamailio[44114]: ERROR: http_client > [curlcon.c:782]: http_client_load_config(): Failed to init http_client > config file parser > Jul 11 09:32:09 lab-kamailio kamailio[44114]: ERROR: http_client > [http_client.c:307]: mod_init(): Failed to load http_client connections > from [./http_client.cfg] > Jul 11 09:32:09 lab-kamailio kamailio[44114]: ERROR: <core> > [core/sr_module.c:1030]: init_mod(): Error while initializing module > http_client (/usr/local/lib64/kam > ailio/modules/http_client.so) I have set the parameter to > modparam("http_client", "config_file", "./http_client.cfg") and also to > modparam("http_client", "config_file", "http_client.cfg") but it only works if I add it as : > modparam("http_client", "config_file", > "/usr/local/etc/kamailio/http_client.cfg") or by adding > chroot="/usr/local/etc/kamailio" in the main cfg script, which was my final solution. Also checked for permissions, and all is fine. Note that this only seems to happen with http_client config_file parameter. I have other include commands in the main cfg file with relative paths and all works fine. I have also analysed the cfg_parser.so file, which loads the configuration file for http_client module, but could not find anything wrong with it (at 1st glance....) What am I doing wrong? Any clue? Atenciosamente / Kind Regards / Atentamente / Cordialement, *Sérgio *

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SIP Question: B2BUA not ACKing 200 OK response
by Ghulam Mustafa 11 Jul '24

11 Jul '24

Hi Community, We are setting up a new provider in our VoIP setup for outgoing calls (FreeSWITCH --> Kamailio --> Provider). When the provider sends a 200 OK in response to our INVITE, FreeSWITCH's SIP stack discards the message, and we are unable to determine the cause of this behavior. I am sharing the SIP message exchange below. Could any SIP experts assist us in understanding why FreeSWITCH is not sending an ACK in response to the 200 OK message? We understand that the issue is not related to Kamailio but seems to be within the SIP stack or FreeSWITCH itself. Below are the actual INVITE and its response, which was discarded by FreeSWITCH: send 1203 bytes to tcp/[10.xxx.x.30]:5060 at 15:48:23.001054: ------------------------------------------------------------------------ INVITE sip:+86xxxxxxxxxxx@35.xxx.xxx.xxx;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 10.xxx.x.10:5080;rport;branch=z9hG4bKD8p0KmvB8ZKma Max-Forwards: 69 From: "+1xxxxxxxxxx" <sip:+1xxxxxxxxxx@10.xxx.x.10>;tag=ZSvBBtypKypDS To: <sip:+86xxxxxxxxxxx@35.xxx.xxx.xxx> Call-ID: 21b7b9b5-b8a5-123d-63a6-ecf4bbd64f6c CSeq: 85714459 INVITE Contact: <sip:gw+commsgroup@10.xxx.x.10:5080;transport=tcp;gw=commsgroup> User-Agent: FreeSWITCH-mod_sofia/1.10.11-release~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 347 X-FS-Support: update_display,send_info Remote-Party-ID: "+1xxxxxxxxxx" <sip:+1xxxxxxxxxx@10.xxx.x.10 >;party=calling;screen=yes;privacy=off v=0 o=Z 5395246197255529491 693195311 IN IP4 10.xxx.x.225 s=Z c=IN IP4 10.xxx.x.225 t=0 0 m=audio 55319 RTP/AVP 106 9 98 101 0 8 3 a=rtpmap:106 opus/48000/2 a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1 a=rtpmap:98 telephone-event/48000 a=fmtp:98 0-16 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-mux tport.c:2316 tport_set_secondary_timer() tport(0x150e6c0291c0): reset timer nta.c:8400 outgoing_send() nta: sent INVITE (85714459) to tcp/10.xxx.x.30:5060 tport.c:4191 tport_pend() tport_pend(0x150e6c0291c0): pending 0x150e6c0c52f0 for tcp/10.xxx.x.30:5060 (already 2) nua_session.c:4131 signal_call_state_change() nua(0x150d9c107460): call state changed: init -> calling, sent offer soa.c:1268 soa_get_local_sdp() soa_get_local_sdp(static::0x150e6c0c0210, [0x150e91772798], [0x150e91772790], [(nil)]) called nua_stack.c:299 nua_stack_event() nua(0x150d9c107460): event i_state INVITE sent nua_stack.c:389 nua_application_event() nua: nua_application_event: entering nua.c:1170 nua_handle_unref_user() nua nua_handle_unref_user(0x150d9c107460): entering nua.c:365 nua_handle_magic() nua: nua_handle_magic: entering nua_stack.c:558 nua_signal() nua(0x150d9c107460): sent signal r_handle_unref nua.c:1157 nua_unref_user() nua: nua_unref_user: entering 2024-07-09 15:48:22.985772 100.00% [DEBUG] sofia.c:7493 Channel sofia/external/+86xxxxxxxxxxx entering state [calling][0] nua_stack.c:599 nua_stack_signal() nua(0x150d9c107460): recv signal r_handle_unref nua.c:365 nua_handle_magic() nua: nua_handle_magic: entering nua_stack.c:558 nua_signal() nua((nil)): sent signal r_unref nua.c:1170 nua_handle_unref_user() nua nua_handle_unref_user(0x150d9c107460): entering nua_stack.c:599 nua_stack_signal() nua((nil)): recv signal r_unref nua_stack.c:558 nua_signal() nua(0x150d9c107460): sent signal r_handle_unref nua_stack.c:599 nua_stack_signal() nua(0x150d9c107460): recv signal r_handle_unref nua.c:1157 nua_unref_user() nua: nua_unref_user: entering nua_stack.c:558 nua_signal() nua((nil)): sent signal r_unref nua_stack.c:599 nua_stack_signal() nua((nil)): recv signal r_unref 2024-07-09 15:48:22.985772 100.00% [DEBUG] switch_core_state_machine.c:659 (sofia/external/+86xxxxxxxxxxx) State CONSUME_MEDIA 2024-07-09 15:48:22.985772 100.00% [DEBUG] switch_core_state_machine.c:659 (sofia/external/+86xxxxxxxxxxx) State CONSUME_MEDIA going to sleep tport.c:2796 tport_wakeup() tport_wakeup(0x150e6c0291c0): events IN tport.c:2900 tport_recv_event() tport_recv_event(0x150e6c0291c0) tport.c:3237 tport_recv_iovec() tport_recv_iovec(0x150e6c0291c0) msg 0x150e6c0cb390 from (tcp/10.xxx.x.30:5060) has 352 bytes, veclen = 1 recv 352 bytes from tcp/[10.xxx.x.30]:5060 at 15:48:23.004029: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/TCP 10.xxx.x.10:5080;rport=37555;branch=z9hG4bKD8p0KmvB8ZKma;received=10.xxx.x.10 From: "+1xxxxxxxxxx" <sip:+1xxxxxxxxxx@10.xxx.x.10>;tag=ZSvBBtypKypDS To: <sip:+86xxxxxxxxxxx@35.xxx.xxx.xxx> Call-ID: 21b7b9b5-b8a5-123d-63a6-ecf4bbd64f6c CSeq: 85714459 INVITE Server: Kamailio 5.8 Content-Length: 0 tport.c:3055 tport_deliver() tport_deliver(0x150e6c0291c0): msg 0x150e6c0cb390 (352 bytes) from tcp/10.xxx.x.30:5060/sip next=(nil) nta.c:3378 agent_recv_response() nta: received 100 Trying for INVITE (85714459) nta.c:3445 agent_recv_response() nta: 100 Trying is going to a transaction nta.c:9697 outgoing_estimate_delay() nta_outgoing: RTT is 3.038 ms tport.c:4253 tport_release() tport_release(0x150e6c0291c0): 0x150e6c0c52f0 by 0x150e6c0be9a0 with 0x150e6c0cb390 (preliminary) tport.c:2316 tport_set_secondary_timer() tport(0x150e6c0291c0): reset timer nta.c:5904 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), 0x150e91973a50) nta.c:7270 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, 0/0 term, 2/3 free nta.c:1310 agent_timer() nta: timer not set tport.c:2796 tport_wakeup() tport_wakeup(0x150e6c0291c0): events IN tport.c:2900 tport_recv_event() tport_recv_event(0x150e6c0291c0) tport.c:3237 tport_recv_iovec() tport_recv_iovec(0x150e6c0291c0) msg 0x150e6c0c9d90 from (tcp/10.xxx.x.30:5060) has 1195 bytes, veclen = 1 recv 1195 bytes from tcp/[10.xxx.x.30]:5060 at 15:48:27.070278: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TCP 10.xxx.x.10:5080;received=10.xxx.x.10;rport=37555;branch=z9hG4bKD8p0Km From: "+1xxxxxxxxxx" <sip:+1xxxxxxxxxx@10.xxx.x.10>;tag=ZSvBBtypKypDS To: <sip:+86xxxxxxxxxxx@35.xxx.xxx.xxx >;tag=7bQ0OSl9Dl-39455-57712329-1057554-14 Call-ID: 21b7b9b5-b8a5-123d-63a6-ecf4bbd64f6c CSeq: 85714459 INVITE Contact: <sip:127.0.0.8;line=sr-H46V9.fbsXNDlkfEJjGbujK3J4aDo1GzP5GOJ8hcsXwV9LAWw1abHkqWlXczwL7*> Record-Route: <sip:13x.xxx.xxx.xx;transport=tcp;lr;ftag=ZSvBBtypKypDS;did=cbd.e581;nat=yes> Supported: sdp-anat Allow: INVITE,PRACK,ACK,UPDATE,CANCEL,BYE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY Require: timer Session-Expires: 2000;refresher=uas Server: Mediant VE SBC/v.7.40A.250.265 P-Asserted-Identity: <sip:4006100666@hk.chinatelecom.com;user=phone> Accept: application/sdp, application/isup, multipart/mixed, application/dtmf, application/dtmf-relay Content-Type: application/sdp Content-Length: 219 User-Agent: Kamailio 5.8 v=0 o=- 846721169 1919912032 IN IP4 13x.xxx.xxx.xx s=- c=IN IP4 13x.xxx.xxx.xx t=0 0 m=audio 49708 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:49709 tport.c:3055 tport_deliver() tport_deliver(0x150e6c0291c0): msg 0x150e6c0c9d90 (1195 bytes) from tcp/10.xxx.x.30:5060/sip next=(nil) nta.c:3378 agent_recv_response() nta: received 200 OK for INVITE (85714459) nta.c:3494 agent_recv_response() nta: 200 OK was discarded Any insights or suggestions would be greatly appreciated, Specifically, why is the 200 OK response being discarded?. Thank you! Regards, Mustafa

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Re: Possible Kamailio memmory leak - version 5.8.2
by Vanderlei Torres Batistela 11 Jul '24

11 Jul '24

Hi, Daniel. Thank you for the quick response! I will update you after testing the fix. Regards, Vanderlei

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Re: Possible Kamailio memmory leak - version 5.8.2
by Vanderlei Torres Batistela 10 Jul '24

10 Jul '24

Hello Henning, Thank you for your quick response. The increase happens in the core module (create-avp). See bellow: [root@labcom070 kamailio]# kamcmd mod.stats all shm Module: core { create_avp(178): 52257504 <---------------------------- counters_prefork_init(211): 53760 cfg_clone_str(132): 112 cfg_shmize(221): 832 main_loop(1381): 16 init_pt(104): 16 init_pt(103): 16 init_pt(102): 6224 cfg_register_ctx(47): 96 init_tcp(5197): 8192 init_tcp(5191): 32768 init_tcp(5182): 16 init_tcp(5175): 16 init_tcp(5167): 16 init_tcp(5161): 16 init_tcp(5149): 16 init_avps(92): 16 init_avps(91): 16 init_dst_blocklist(435): 16384 init_dst_blocklist(427): 16 timer_alloc(494): 96 init_dns_cache(368): 16 init_dns_cache(359): 16384 init_dns_cache(351): 16 init_dns_cache(343): 16 init_timer(264): 16 init_timer(263): 16384 init_timer(262): 16 init_timer(261): 16 init_timer(250): 16 init_timer(218): 16 init_timer(207): 278544 init_timer(206): 16 init_timer(194): 16 cfg_child_cb_new(832): 64 sr_cfg_init(371): 16 sr_cfg_init(364): 16 sr_cfg_init(356): 16 sr_cfg_init(344): 16 sr_cfg_init(332): 16 ksr_shutdown_phase_init(123): 16 rpc_hash_add(118): 16 qm_shm_lock_init(1463): 16 Total: 52687808 <------------------ } [root@labcom070 kamailio]# kamctl stats shmem { "jsonrpc": "2.0", "result": [ "shmem:fragments = 900", "shmem:free_size = 6180368", "shmem:max_used_size = 66980272", "shmem:real_used_size = 60928496", "shmem:total_size = 67108864", "shmem:used_size = 54884944" ], "id": 393115 } These reports above are shown after the end of calls (4000) where the outgoing INVITE is not answered. In normal calls the issue does not occur and the reports are shown below: [root@labcom070 kamailio]# kamcmd mod.stats all shm Module: core { counters_prefork_init(211): 53760 cfg_clone_str(132): 112 cfg_shmize(221): 832 main_loop(1381): 16 init_pt(104): 16 init_pt(103): 16 init_pt(102): 6224 cfg_register_ctx(47): 96 init_tcp(5197): 8192 init_tcp(5191): 32768 init_tcp(5182): 16 init_tcp(5175): 16 init_tcp(5167): 16 init_tcp(5161): 16 init_tcp(5149): 16 init_avps(92): 16 init_avps(91): 16 init_dst_blocklist(435): 16384 init_dst_blocklist(427): 16 timer_alloc(494): 96 init_dns_cache(368): 16 init_dns_cache(359): 16384 init_dns_cache(351): 16 init_dns_cache(343): 16 init_timer(264): 16 init_timer(263): 16384 init_timer(262): 16 init_timer(261): 16 init_timer(250): 16 init_timer(218): 16 init_timer(207): 278544 init_timer(206): 16 init_timer(194): 16 cfg_child_cb_new(832): 64 sr_cfg_init(371): 16 sr_cfg_init(364): 16 sr_cfg_init(356): 16 sr_cfg_init(344): 16 sr_cfg_init(332): 16 ksr_shutdown_phase_init(123): 16 rpc_hash_add(118): 16 qm_shm_lock_init(1463): 16 Total: 430304 } [root@labcom070 kamailio]# kamctl stats shmem { "jsonrpc": "2.0", "result": [ "shmem:fragments = 6", "shmem:free_size = 64143184", "shmem:max_used_size = 4070816", "shmem:real_used_size = 2965680", "shmem:total_size = 67108864", "shmem:used_size = 2627856" ], "id": 683948 } Regars, Vanderlei

2 1

Intercepting OPTIONS from nathelper
by Denys Pozniak 10 Jul '24

10 Jul '24

Hello! I need advice on how to intercept local SIP OPTIONS requests from the nathelper module for some adjustments? As far as I tested, it does not fall into the *event_route[tm:local-request]* Thanks in advance! # kamailio -v version: kamailio 5.7.1 (x86_64/linux) 1cf389-dirty -- BR, Denys Pozniak

2 5

handle_subscribe(), handle_publish() and DB usage
by Володимир Іванець 04 Jul '24

04 Jul '24

Hello everyone! According to the documentation functions handle_subscribe() and handle_publish() update information not only in memory but also in the database. Isn't there a way to disable database usage for this functionality? Setting subs_db_mode to 0 does not make a change. Thank you!

1 0

Contact does not exist in PCSCF usrloc error in log
by Shantanu K Roy 03 Jul '24

03 Jul '24

Hello, We are trying to integrate Kamailio as the IMS server with our 5G core emulator by following the steps listed out at https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/. In our case, we have our own 5G core emulator instead of the Open5GS core. We are using sipp scripts to generate the SIP calls. The PCRF component of the 5G core is emulated by us and is capable of handling Diamater calls from Kamailio over the Rx interface. We have got the setup working to the point where the SIP Register calls are landing on the PCSCF which in turn is sending out AA Requests to PCRF and the PCRF is responding. However there is an internal error in the Kamailio log "4(3682133) ERROR: ims_qos [ims_qos_mod.c:1305]: w_rx_aar_register(): This contact does not exist in PCSCF usrloc - error in cfg file" Please help us in identifying if we are missing anything in the Kamailio set up. The packet capture and syslog excerpts from the VM where Kamailio is running are attached; here we are trying to perform SIP Registration for three SIP endpoints 001010123456791, 001010123456792 and 001010123456793. If there is any additional information required, please let us know. Regards, Shantanu Kumar Roy

3 6

Kamailio v5.6.6 Released
by Daniel-Constantin Mierla 03 Jul '24

03 Jul '24

Hello, Kamailio SIP Server v5.6.6 stable release is out. This is a maintenance release of the latest stable branch, 5.6, that includes fixes since the release of v5.6.5. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.6.x. Deployments running previous v5.6.x versions are strongly recommended to be upgraded to v5.6.6. Note that 5.6 is now the third last stable branch, v5.6.6 being the last planned released in 5.5.x series. The latest two stable branch are 5.7 and 5.8, with v5.8.2 being released a while ago. For more details about version 5.6.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/07/kamailio-v5-6-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy and Development Services Kamailio Advanced Training -- asipto.com

1 0

Releasing v5.6.6
by Daniel-Constantin Mierla 03 Jul '24

03 Jul '24

Hello, I am considering to release Kamailio v5.6.6 (out of branch 5.6) later this week (likely on Wednesday, July 3, 2024). If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. This is going to be the last release out of branch 5.6 to mark the end of official/packaging maintenance. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 1

Extracting and Handling Via Headers in failure_route
by sadik.oualla.mohamed＠gmail.com 02 Jul '24

02 Jul '24

Hello, I am currently working on a scenario where I need to use Kamailio in the failure_route to process SIP responses, specifically handling negative SIP replies (e.g., 486 Busy Here) that Kamailio may receive. Here is the topology: ##B2BUA (acts as a UAC with the Kamailio, because it sends the INVITE) -> Kamailio -> Callee## In my setup, Kamailio needs to extract all Via headers from the received SIP response in the failure_route, focusing particularly on the bottom Via header (from the B2BUA' UAC part). However, I am encountering difficulties in achieving this. Here is what I have tried so far: 1. To extract the Via headers, I used the following line of code in the failure_route: ```$var(viaHeaderValues) = $T_rpl($hdr(Via));``` However, this only returns the topmost Via header. 2. I attempted to get the first and last Via headers values with: ```$var(viaHeaderValueFirst) = $T_rpl($hdr(Via)[0]); $var(viaHeaderValueLast) = $T_rpl($hdr(Via)[1]);``` Unfortunately, this also only returns the topmost Via header, which contains the address of Kamailio. My goal is to make sure that when Kamailio sends the SIP reply to the UAC with send_reply method on the failure_route, it uses the extracted *Via header* of the incoming SIP response (only the bottom one). Here is an example scenario: - Received SIP Response in kamailio: ``` SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 200.200.200.4;branch=z9hG4bK9157.090080a91105c0ec6279bb56882d1dc8.0 Via: SIP/2.0/UDP 200.200.200.3;received=200.200.200.3;rport=5060;branch=z9hG4bKF1aaZ7ea7yc2m From: <sip:caller@200.200.200.3>;tag=222222 To: <sip:callee@200.200.200.4:5060>;tag=4545454 Call-ID: 65695DSQ(a)200.200.200.3 CSeq: 1 INVITE Content-Length: 0 ``` - Desired SIP Response from kamailio: ``` SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 200.200.200.3;received=200.200.200.3;rport=5060;branch=z9hG4bKF1aaZ7ea7yc2m From: <sip:caller@200.200.200.3>;tag=222222 To: <sip:callee@200.200.200.4:5060>;tag=4545454 Call-ID: 65695DSQ(a)200.200.200.3 CSeq: 1 INVITE Content-Length: 0 ``` The current issue is that Kamailio modifies the Via header when sending the reply with send_reply method, resulting in: ``` SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 200.200.200.3;rport=5060;branch=z9hG4bKF1aaZ7ea7yc2m;received=200.200.200.3 From: <sip:caller@200.200.200.3>;tag=222222 To: <sip:callee@200.200.200.4:5060>;tag=4545454 Call-ID: 65695DSQ(a)200.200.200.3 CSeq: 1 INVITE Content-Length: 0 ``` Due to this change, the UAC does not correctly react to or understand the SIP response "SIP Reply follows the RFC standards, but my current UAC waits for a Via that looks like Via of the previous SIP messages in the structure of parameters.". It is crucial for the Via header to maintain the same important order such as the branch, received, and rport values as the previous exchanged SIP messages. Could you please provide guidance on how to force Kamailio to send the SIP reply using send_reply method with the extracted bottom Via header from the incoming SIP response? Any advice or examples on handling this situation would be greatly appreciated. Thank you for your time and assistance.

1 0

Kamailio v5.7.6 Released
by Daniel-Constantin Mierla 02 Jul '24

02 Jul '24

Hello, Kamailio SIP Server v5.7.6 stable release is out. This is a maintenance release of the latest stable branch, 5.7, that includes fixes since the release of v5.7.5. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.7.x. Deployments running previous v5.7.x versions are strongly recommended to be upgraded to v5.7.6. For more details about version 5.7.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/07/kamailio-v5-7-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note that 5.7.x is currently the previous stable release series, the latest is 5.8.x and v5.8.2 was released on June 12, 2024. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 0

Releasing v5.7.6
by Daniel-Constantin Mierla 02 Jul '24

02 Jul '24

Hello, I am considering to release Kamailio v5.7.6 (out of branch 5.7) on Tuesday, July 2, 2024. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 1

SRV in advertise?
by David Villasmil 01 Jul '24

01 Jul '24

Hello guys, I'm trying to use an SRV as an advertised address but kamailio fails to start. Is this not possible? Thanks, David Villasmil email: david.villasmil.work(a)gmail.com

4 7

SecSIPID Assistance
by Blake Ivey 01 Jul '24

01 Jul '24

Hi everyone. Wanting to see if someone could point me in the right direction. Still very knew to Kamailio but I am beginning to understand it better. I'm making an outbound proxy and have everything working well besides stir/shaken. I'm looking at the module page and have went back and forth with chatGPT and can't seem to figure this part out. I keep getting errors on the modparam lines. Obviously this is a self signed cert because I'm just testing. I am able to reach and download the cert from the Web server. Thank you for any assistance. # SECSIPID for Stir/Shaken modparam("secsipid", "private_key", "/etc/kamailio/secsipid/private.key") modparam("secsipid", "certificate", "/etc/kamailio/secsipid/cert.crt") modparam("secsipid", "authority_cert", "/etc/kamailio/secsipid/ca.crt") modparam("secsipid", "expire", 600) modparam("secsipid", "timeout", 2) route[STIRSHAKEN] { if (is_method("INVITE")) { if (!secsipid_add_identity("$fU", "$rU", "A", "", " http://myIPaddress.com/stir_shaken_cert.crt <http://myipaddress.com/stir_shaken_cert.crt>", "/etc/kamailio/secsipid/private.key")) { xlog("L_ERR", "Failed to sign call with ID: $ci - From: $fU\n"); send_reply("500", "Internal Server Error"); exit; } else { xlog("L_INFO", "Successfully signed call with ID: $ci - From: $fU\n"); } } # Relay the call after signing route(RELAY); }

3 8

removing Via Header with received=
by mosbah abdelkader 30 Jun '24

30 Jun '24

Hello All, I am trying to build a simple proxy that forwards SIP requests to a remote portaone server. But, kamailio is adding a Via header with received=@IP to replies got from portaone. How to disable this behavior as Asterisk (client in my setup) is not accepting this header. Thank you.

1 0

misctest module not found in kamailio debian packages
by Patrick Karton 29 Jun '24

29 Jun '24

Hello, how do i get misctest module in debian packages ? it seems not included in a package. is compilation the only way to get it ? im using kamailio 5.7.5 on debian (bullseye) 11.8 version: kamailio 5.7.5 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, MEM_JOIN_FREE, 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_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED 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 10.2.1 Thanks.

1 0

Ubuntu 24.04 Packages for Kamailio 5.8 on deb.kamailio.org
by Xenofon Karamanos 28 Jun '24

28 Jun '24

Any news on regarding the availability and status of Kamailio packages for Ubuntu 24.04 on the official Kamailio repository at deb.kamailio.org. Are there any plans to support this version of Ubuntu with updated packages? If so, is there an estimated timeline for when these packages might become available? Thanks, Xenofon ________________________________ From: Xenofon Karamanos via sr-dev <sr-dev(a)lists.kamailio.org> Sent: Wednesday, June 19, 2024 11:51 To: sr-dev(a)lists.kamailio.org Cc: Xenofon Karamanos <xk(a)gilawa.com> Subject: [sr-dev] Ubuntu 24.04 Packages for Kamailio 5.8 on deb.kamailio.org Hey everybody, While attempting to use the official Ubuntu package for Kamailio version 5.7.4 on Ubunut 24.04, we encountered an issue where TLS functionality does not seem to work as expected but 5.8 branch does. Given this, I am inquiring about the availability and status of Kamailio packages for Ubuntu 24.04 on the official Kamailio repository at deb.kamailio.org. Are there any plans to support this version of Ubuntu with updated packages? If so, is there an estimated timeline for when these packages might become available? Thank you in advance for your assistance. Best regards, Xenofon

1 0

replies using the wrong protocol
by sarah.martin＠nextiva.com 27 Jun '24

27 Jun '24

I am extremely new at this, but trying to set up TLS with a carrier. TLS connection is good, Invite goes out, we get the 100 and the 200, but subsequent messages (ACK and BYE) are being sent with UDP and I cannot figure out how to get them to maintain the TLS transport. Any suggestions? I think this is the section I'm looking for. # Manage incoming replies in transaction context onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } if (has_body("application/sdp")) { xdbg("rtpengine_manage loop-protect MANAGE_REPLY"); #rtpengine_manage("loop-protect"); } }

6 21

Kamailio cannot forward request - invalid sendtoparameters
by serkan celik 27 Jun '24

27 Jun '24

Hi all, I face a problem when trying to forward an UPDATE request to the IMS which communicates with UDP transport protocol on one side UDP and SCTP on other side. When I try to send UPDATE towards the UDP side of IMs I see following error on kamailio side and request cannot be forwarded Jun 25 16:19:18 prashant-0lk-sbc-1 /usr/sbin/kamailio[1223341]: ERROR: <core> [core/udp_server.c:597]: udp_send(): sendto(sock, buf: 0x7fc180deb5f0, len: 1509, 0, dst: (172.18.0.20:5060), tolen: 16) - err: Invalid argument (22) Jun 25 16:19:18 prashant-0lk-sbc-1 /usr/sbin/kamailio[1223341]: CRITICAL: <core> [core/udp_server.c:603]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net The UPDATE message which is tried to be forwarded is as follow: Here there are double rr one is UDP and one is sctp (Route: < sip:mt@172.18.0.20;lr;did=6aa.aaa;ftag=49IZjY2irfb;r2=on>, < sip:mt@172.18.0.20;lr;transport=sctp;did=6aa.aaa;ftag=49IZjY2irfb;r2=on>) ) That would be great if you can help why this UPDATE cannot be forwarded. By the way double rr is on on kamailio config Jun 25 16:19:18 prashant-0lk-sbc-1 /usr/sbin/kamailio[1223341]: DEBUG: <core> [core/forward.c:577]: forward_request(): Sending:#012UPDATE sip:alice@10.10.141.188;ob;sbc-id=xpllrk47odqh4pb2ofpib6vsym SIP/2.0#015#012Via: SIP/2.0/UDP 127.0.0.1:50601;branch=z9hG4bKddab.3554948ce3cfe321729448a2b082c2f3.0#015#012Via: SIP/2.0/UDP 127.0.0.1:7060;branch=z9hG4bKnHN8Q3ya#015#012To: < sip:alice@ims.wingcon.com>;tag=49IZjY2irfb#015#012From: < sip:bob@ims.wingcon.com>;tag=hfdqjj2pmui#015#012Call-Id: 3mzzecWt3Cbgvy#015#012CSeq: 1 UPDATE#015#012Max-Forwards: 70#015#012Contact: <sip:bob@10.10.141.188;ob;sbc-id=o7dpxe7qu2r34zwzio5pis2vqi>#015#012Content-Type: application/sdp#015#012Route: <sip:mt@172.18.0.20;lr;did=6aa.aaa;ftag=49IZjY2irfb;r2=on>, <sip:mt@172.18.0.20;lr;transport=sctp;did=6aa.aaa;ftag=49IZjY2irfb;r2=on>, < sip:mo@172.18.0.20;lr;did=6aa.9aa;ftag=49IZjY2irfb;r2=on>, < sip:mo@172.18.0.20;lr;transport=sctp;did=6aa.9aa;ftag=49IZjY2irfb;r2=on>, < sip:mo@172.18.0.120;lr>#015#012Session-Expires: 180;refresher=uas#015#012Supported: replaces, 100rel, timer, norefersub#015#012Min-SE: 90#015#012P-Charging-Vector: icid-value=e277783432fd11efba03fa163eed425f;icid-generated-at=172.18.0.120;orig-ioi=172.18.0.120;term-ioi=172.18.0.120#015#012User-Agent: PJSUA v2.14 Linux-5.4.0.172/x86_64/glibc-2.35#015#012Content-Length: 409#015#012#015#012v=0#015#012o=- 3928313948 3928313950 IN IP4 172.18.0.120#015#012s=pjmedia#015#012b=AS:84#015#012t=0 0#015#012a=X-nat:0#015#012m=audio 30106 RTP/AVP 8 96 97 121 120#015#012c=IN IP4 172.18.0.120#015#012b=TIAS:64000#015#012a=ssrc:1986855169 cname:1fca52e812ced466#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:96 AMR/8000#015#012a=rtpmap:97 AMR-WB/16000#015#012a=rtpmap:121 telephone-event/16000#015#012a=fmtp:121 0-16#015#012a=rtpmap:120 telephone-event/8000#015#012a=fmtp:120 0-16#015#012a=sendrecv#015#012a=rtcp:30107#015#012. Regards Serkan

1 0

Handling final response retransmissions after a reInvite
by hollis.rhonda＠gmail.com 26 Jun '24

26 Jun '24

I have a scenario where Kamailio is receiving a retransmit of a 200 OK to a late offer INVITE after it has sent the ACK, and after it has begun to handle a reInvite from the calling party. This results in the following commands to the rtpengine: Offer (reInvite), Offer (200 retransmit), Answer (ACK to retransmitted 200). At this point, I see STUN binding errors both in Chrome, where the webRTC client (called party) is running, and in the rtpengine logs. ERROR:port.cc(498)]: Received STUN BINDING error response: class=4 number=1 reason='Unauthorized' [core] STUN authentication mismatch from x.x.x.x:63396 [ice] Received ICE/STUN response code 487 for candidate pair dl2efuRG06eK4nFs:2713745946:1 from x.x.x.x::64509 to x.x.x.x [ice] ICE role change, now controlled [ice] Recalculating all ICE pair priorities [ice] Triggering check for dl2efuRG06eK4nFs:2713745946:1 [ice] ICE/STUN response with unknown transaction received (from x.x.x.x:64509 on interface x.x.x.x:16184) So, briefly, the call flow up to the point of error looks like this: INVITE w/o SDP 200 OK w/SDP, Offer to rtpengine ACK w/SDP, Answer to rtpengine reInvite w/SDP, Offer to rtpengine retransmission of 200 OK, Offer to rtpengine ACK to retransmitted 200, Answer to rtpengine I have dialog tracking enabled in my config, and it seems like Kamailio is behaving as expected when the retransmitted 200 and ACK are forwarded on, but I'm curious if other people have run into a scenario like this and how they have dealt with it.

1 0

Need help on how to implement presence_xml
by Michel Pelletier 25 Jun '24

25 Jun '24

Hello, I need to implement Event: presence, application/pidf+xml as described in RFC-3856. I understand I need to use the presence_xml module and generate PUBLISH requests to update the information and ensure that the NOTIFY get sent. I know nothing abour XCAP or wether I need it or not. Any information source out there that could help me? Regards, Michel Pelletier

1 0

TLS: client doesn't check certificate CN
by Marat Gareev 20 Jun '24

20 Jun '24

Hello, I noticed that Kamailio can route either to IP and FQDN. This means the server certificate CN is not checked by the client. How to enable something like the 'verify_peer' option? Thanks

1 0

Possibility to print values of Core Keywords in Kamailio configuration file
by sadik.oualla.mohamed＠gmail.com 20 Jun '24

20 Jun '24

Hello everyone, I have a question regarding the usage of core keywords, specifically the possibility of printing their values in log messages. While I understand that core keywords such as 'dst_port' and 'af' etc... can be accessed directly in if conditions, I am interested to know if there is a way to print the values of these core keywords for logging purposes. Thank you.

4 6

Re: exec concurrency
by Jon Bonilla (Manwe) 19 Jun '24

19 Jun '24

El Wed, 19 Jun 2024 11:54:05 -0500 Brett Nemeroff <brett(a)voicefoxtelephony.com> escribió: > What you should expect using this method is blocking while the exec runs > which could run you out of child processes while they complete. I'd also > expect to see an "unusual amount of CPU activity" which will be the result > of the forked shells being created and destroyed. > > All of that being said, yes it works. It's architecturally overly simple > and very inefficient. If your load is tiny, it might be ok. Anything > moderate like Henning and others have said, will use an inordinate amount > of cpu. I would not personally risk it. > > Thank you Brett, Henning and Alex Sadly the load won't be tiny and cannot risk it. For now I'll deploy a local nginx and fpm to run the scripts, which are in php, and will use the http_async method. Sadly we have to "http all the things" nowadays. Let's see if I'm able to write down all the logic tomorrow and do some testing. cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

1 0

Adding custom headers to HTTP server response xhttp
by Sasivarunan 19 Jun '24

19 Jun '24

HI Team, I use tls and xhttp module to enable websocket interface for webrtc client to connect. I wanted to add "Strict-Transport-Security" header to xhttp response to add security. But currently I don't see a way to add it anywhere. I can see only server_header as an option, But this will get added to SIP messages as well. It 'd be great if anybody has a suggestion around this. Thanks Varunan

2 2

Re: exec concurrency
by Jon Bonilla (Manwe) 19 Jun '24

19 Jun '24

El Tue, 18 Jun 2024 13:54:41 -0500 Brett Nemeroff <brett(a)voicefoxtelephony.com> escribió: > Just want to add that exec is heavy and slow. I would not recommend it. > > Is there a reason you want to do this over http-ifying your script and > using async? > > Not really. The scripts are provided. Maybe I'm against "http all the things" which seems to be what everyone does nowadays. Why would http client be lighter than executing a script? That would be a design problem, right? -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 1

exec concurrency
by Jon Bonilla (Manwe) 18 Jun '24

18 Jun '24

Hi all I've been working with async http client and async db queries in the past but now I have to execute scripts and store the return values in avps. Since I have no experience with that, I wonder how to achieve concurrency with that scenario. Is there a way to exec async or suspend until the exec has finished and resume later? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 2

Psresence dialoginfo question
by Stefan Mititelu 17 Jun '24

17 Jun '24

Hello, I am doing BLF but I want different users to be able to setup different dialog info initial states. My question is if there's an already implemented way for doing that? I've looked over docs and C code so far and this is what I've found: 1. modparam("presence_dialoginfo", "force_dummy_dialog", 1) => will force sending initial NOTIFY with dialog state "terminated" 2. Function calls: dlginfo_agg_nbody()->dlginfo_agg_nbody_empty() which will build the pre-defined xml body with that "terminated" state. If not implemented so far, maybe this can be used to set a different initial state for a user (e.g. export a cfg function or take that info from DB?) Any opinions on this? Thank you, Stefan

2 2

db_mysql startup warnings
by Juha Heinanen 17 Jun '24

17 Jun '24

When I start Kamailio master version that has been build on Debian 12 with default-libmysqlclient-dev depending on libmariadb-dev-compat, I get several warnings like this: 2024-06-17T14:48:27.702827+03:00 siika /usr/bin/sip-proxy[297246]: WARNING: db_mysql [km_my_con.c:187]: db_mysql_new_connection(): opt_ssl_ca option not supported by mysql version (value (null)) - ignoring How can I prevent them? -- Juha

1 1

How to Check Source Port in Application Layer with Kamailio
by sadik.oualla.mohamed＠gmail.com 16 Jun '24

16 Jun '24

Hello kamailio lovers, I am currently working on a Kamailio configuration where I need to check the source port of incoming SIP messages. Initially, I used the $sp pseudo-variable to get the source port, which worked perfectly while testing with SIPp. Here is the snippets code that I use " if ($sp != 5080 && $sp != 8080 && $sp != 5060){ sl_send_reply(403, "forbidden"); xlog("this $rm received, is forbidden\n"); exit; } " However, I encountered an issue when I switched to using sipsak for testing. It seems that sipsak uses a randomly assigned port for the actual network connection, even though it sets the source port in the SIP headers (e.g., Via, To and Contact headers) to a specified port using the -l option. Here is the sipsak command that I use "sipsak -vv -s sip:jack@192.168.1.2:5060-l 5080" My goal is to have Kamailio check the source port as specified in the application layer, rather than the transport layer port (i.e., the ephemeral port assigned by the OS randomly like 54691, 33017...). Could anyone guide me on how to properly configure Kamailio to extract and check the port from the application layer or SIP headers, or am I doing this in the wrong way? Any advice or examples would be greatly appreciated. "I am not a native english speak, so I use chatGPT to make my questions clear" Thank you!

1 0

topos module issue on hangup
by Arun K R 13 Jun '24

13 Jun '24

Hello, I am using topos module to hide Record-Route and Route header as per the requirement from carrier. But the topos module hides, contact, via and routing headers also. Due to this reason when I am getting bye from the carrier I am not able to route the bye back to the agent. Bye from Agent to carrier is working fine. Bye from carrier to my internal network interface is not routing to client through public interface. I assume this may be because Via / Contact header has been hidden and changed by the topos module with the invite please help me for this Thanks Arun

3 5

Kamailio v5.8.2 Released
by Daniel-Constantin Mierla 12 Jun '24

12 Jun '24

Hello, Kamailio SIP Server v5.8.2 stable release is out. This is a maintenance release of the latest stable branch, 5.8, that includes fixes since the release of v5.8.1. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.8.x. Deployments running previous v5.8.x versions are strongly recommended to be upgraded to v5.8.2. For more details about version 5.8.2 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/06/kamailio-v5-8-2-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 0

Releasing v5.8.2
by Daniel-Constantin Mierla 12 Jun '24

12 Jun '24

Hello, I am considering to release Kamailio v5.8.2 (out of branch 5.8) on Wednesday, June 12, 2024. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

2 2

New features in v.5.8
by Leonid Fainshtein 10 Jun '24

10 Jun '24

Hi, In the "Kamailio v5.8.0 Release Notes" I see: "SIP overload control support for server (RFC7339)". Where can I find some more details about the feature? Also, the similar question about the new "influxdbc" module. What are the possible use cases for this module? Thank you, Leonid Fainshtein Xorcom Ltd

2 2

handling 300 redirect requests
by Rob Green 09 Jun '24

09 Jun '24

I am trying to proxy a request from server A to server C with me being server B. i send the invite from A to B, then i need to send to server R which returns back to me a 300 redirect, which i then need to take the answer from the Contsct and then proxy the original invite from A to C. server A can not handle 3xx responses and must be proxied. I havent been able to find any examples where the 300 is intercepted and a new invite to the target is sent. i have the following code but it results in no new INVITE and the original session is hung until it is canceled. Any idea what i may be doing wrong? In the example below i am just hardcoding the uri for the new invite instead of getting from the Contact as a debug exercise. I am using Kamailio 5.7.3 route[HANDLEREDIRECT] { xlog("L_INFO", "DBG: HANDLEREDIRECT"); msg_apply_changes(); $var(test) = $hdr(Contact); xlog("L_INFO","DBG: redirect headers [$var(test)]: from=[$fu] from_user=[$fU], to=[$tu] to_user=[$tU]"); # send to gateway $var(new_uri) = "sip:" + $tU + "@" + "gateway"; if ($var(new_uri) != "") { xlog("DBG: Redirecting to: $var(new_uri)\n"); $ru = $var(new_uri); $tu = $ru; record_route(); route(RELAY); exit; # i have also tried using #t_newtran(); # and #t_relay(); } else { xlog("DBG: No contact header found in 300 response\n"); } } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("DBG: new branch [$T_branch_idx] to $ru\n"); xlog("L_INFO", "DBG: manage branch"); route(NATMANAGE); return; } onreply_route[MANAGE_REPLY] { xdbg("DBG: incoming reply\n"); xlog("L_INFO", "DBG: manage reply"); if (t_check_status("3[0-9][0-9]")) { xlog("L_INFO", "DBG: got 3xx redirect"); route(HANDLEREDIRECT); drop; } return; } I also tried using uac_redirect module, but it stalls the incoming invite also, and i cant figure out how to modify the uri from the contact before sending (to add params) # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; if (t_check_status("3[0-9][0-9]")) { xlog("L_INFO", "DBG: following 3xx redirect"); get_redirects("3:1"); t_relay(); exit; } } Rob Green Chief Information Officer (CIO) Tel: 1-352-218-1490 Cell: 1-407-620-0932 Email: rob.green(a)nuwave.com Web: www.nuwave.com This communication is the Property of NUWAVE Communications and may contain confidential or privileged information. Unauthorized use of this communication is strictly prohibited and may be unlawful. If you have received this communication in Error, Please immediately notify the sender by reply e-mail and destroy all copies of the communication and any attachments.

2 2

Setting Key/Value Pairs at Dialogue Level
by Pavan Kumar 08 Jun '24

08 Jun '24

Hi everyone, I am creating a dialogue when I receive INVITE. Is there a way to set key/value pair like attributes at dialog level, so that, when I receive BYE message I can retrieve that information and do some processing? Thank you, Pavan Kumar

2 2

Recommendation for a signaling B2BUA to work with Kamailio on Ubuntu
by Benoît Panizzon 07 Jun '24

07 Jun '24

Hi I wonder if anyone could maybe recommend what to use as signaling B2BUA to work with Kamailio to handle spiralling / looping calls and to reduce the number of Route, Via Header and message size, which cause issue with a large number of CPE. Requirements: * New CallID needs to be generated * Handling Route and Via * ~ 2000 concurrent calls. * Preserving custom SIP X- header * Routing NOTIFY for MWI etc. Handling RTP is not required! What I have been looking at more closely: = FreeSwitch = Got stuck trying to solve the dependencies to get it to run/build on ubuntu jammy. Has anyone experience in running it in a docker container? Will it perform? = Sippy = Very small, very simple, unfortunately removes customer SIP headers and does not forward NOTIFY. (It's python, so maybe this could be easily implemented/fixed). = SEMS = Does not seem to be actively developed in the last, couple of years. Looks quite bloated to me for what I need. = Asterisk = Each customer header has to be copied, wants to handle RTP and bridge by issuing Re-Invites. Not sure how to route stuff like NOTIFY. Also from experience I fear asterisk would translate sip reply codes (like 608 and 607) it does not know to some 4XX default. -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

4 3

Filtering codes in db acc
by Jon Bonilla (Manwe) 06 Jun '24

06 Jun '24

Hi all I'm trying to make kamailio not to insert a row in missed_calls table when reply code is 407. But even if I set the "failed_filter" modparam the 407 row is inserted. The acc mmodparams are: modparam("acc", "db_table_acc", "kam_acc") modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACC_MISS) modparam("acc", "db_url",DBURL) modparam("acc", "db_insert_mode", 1) modparam("acc", "time_mode", 2) modparam("acc", "time_attr", "time_hires") modparam("acc", "failed_filter", "401,407") modparam("acc", "db_extra", "whatever" db is mysql. Kamailio version is 5.7.4 When I send a call that receives a 407 and after auth a 480 y get both rows in missed_calls table. Log: DEBUG: acc [acc_mod.c:404]: mod_init(): acc db table initialized to: kam_acc DEBUG: acc [acc_mod.c:362]: parse_failed_filter(): failed_filter 0 = 401 DEBUG: acc [acc_mod.c:362]: parse_failed_filter(): failed_filter 1 = 407 DEBUG: acc [acc_logic.c:721]: tmcb_func(): acc callback called for t(0x7f7cec807480) event type 512, reply code 407 DEBUG: acc [acc_logic.c:443]: should_acc_reply(): probing acc state - code: 407 flags: 0x220 DEBUG: acc [acc_logic.c:449]: should_acc_reply(): failed acc is off DEBUG: acc [acc_logic.c:721]: tmcb_func(): acc callback called for t(0x7f7cebf59bc0) event type 512, reply code 486 DEBUG: acc [acc_logic.c:443]: should_acc_reply(): probing acc state - code: 486 flags: 0x220 DEBUG: acc [acc_logic.c:449]: should_acc_reply(): failed acc is off am I configuring this wrong? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 3

CentOS 7 and 8 deprecated
by Sergey Safarov 06 Jun '24

06 Jun '24

The CentOS project page shows that the 7 and 8 releases are deprecated. https://www.centos.org/ Does the Kamailio community use these releases for prod? If yes then I will keep the packaging for this release. If not, I will remove the packaging for 7 and 8 releases, and in the future Kamailio will be packaged for 9 and 10 releases (planned). https://www.phoronix.com/news/CentOS-Stream-10-Start https://composes.stream.centos.org/stream-10/development/latest-CentOS-Stre… https://composes.stream.centos.org/stream-10/development/latest-CentOS-Stre…

2 1

Assistance Needed: Converting 11-Digit US Numbers to E.164 in Kamailio
by Pavan Kumar 04 Jun '24

04 Jun '24

Hey everyone, I have the following setup: External User/Trunk <===> Kamailio <===> FreeSWITCH I have configured FreeSWITCH and Kamailio to work only with phone numbers in E.164 format. Recently, I needed to integrate a new SIP trunk that sends phone numbers in an 11-digit US format. My initial idea to solve this is to use Kamailio as a translator that converts 11-digit numbers to E.164 when sending to FreeSWITCH, and back to 11-digit when sending to the trunk. Before writing the code, I want to know if there is already a solution to this problem in one of the modules, so that I won't reinvent the wheel. Thanks, Pavan Kumar

7 11

DMQ_USRLOC and multiple domains
by Lợi Đặng 03 Jun '24

03 Jun '24

Hi, I'm setting up shared user location for 2 instances of kamailio via dmq_usrloc. Everything seems fine except my setup is multi-domains. The issue arises when the producer node produces the Contact of domain X, Contact of domain Y separately, but the consumer node stores both as the Contact of domain Y (seen in kamcmd ul.dump), in which Y is set in usrloc_domain. modparam("dmq_usrloc", "usrloc_domain", "{{domain_Y}}") modparam("usrloc", "use_domain", 1) ==== I had a quick ngrep and see the domain was put in the "aor" key of the body in KMQ message for usrloc {"action":1,"aor":"test_username_0dppxcl@$domain_X"} But I'm not sure how to make my consumer node to store the contact in appropriate domain. Any help will be appreciated. Thanks and regards, Loi Dang Thanh Phone : +84. 774.735.448 Email : loi.dangthanh(a)gmail.com

1 1

ACK and Bye not relaying when record_route is not used
by Arun K R 03 Jun '24

03 Jun '24

Issue : Not getting relay of ACK and BYE to the next hop after the call is answered my Scenario : Asterisk ------->kamailio sip proxy-------------------> carrier (outgoing call) My carrier is not allowed to send the SIP packet with Record-Route header. So that I have removed record_route(). After that the call is getting connected. I am getting 200 OK (SDP) from carrier side and forwarded that to the Asterisk on the other side. As a response I am getting ACK from asterisk. But the kamailio is not forwarding the ACK to the carrier side. I understood this is because the record-route is not there. The same thing is happening for BYE also. The Bye is not forwarding to carrier side. Kindly suggest me a solution for this for relaying ACK and bye without Record-Route in kamailio Bellow is the 200 OK SDP I am sending back to asterisk 2024/06/02 10:27:04.756610 103.155.114.101:5060 -> 103.182.153.113:5060 SIP/2.0 200 OK Call-ID: 1d2897fc-8ae2-4835-b7ac-a96cc28adcc2 Via: SIP/2.0/UDP 103.182.153.113:5060 ;received=103.182.153.113;rport=5060;branch=z9hG4bKPj463fedcf-6258-4655-a53d-58ea57f144af To: <sip:09496381412@103.155.114.101 >;tag=6541fd97-665bfb9a2aea9a78-gm-po-lucentPCSF-109109 From: <sip:917946357720@gaesip.teleforce.in >;tag=8dba28f5-d250-48c6-99fc-35d84590cfc1 CSeq: 22823 INVITE Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,INFO,REFER,UPDATE Contact: <sip:lucentNGFS-115804@10.5.110.117:5060 ;alias=10.5.110.117~5060~1;x-afi=11> Content-Type: application/sdp Session-Expires: 7200;refresher=uas Supported: timer Content-Length: 248 v=0 o=LucentPCSF 130227946 130227946 IN IP4 103.155.114.101 s=- c=IN IP4 103.155.114.101 t=0 0 m=audio 12806 RTP/AVP 8 101 ------------------------------------------------------------------------------------------------------- The ACK I am getting back from asterisk is 2024/06/02 10:27:04.760392 103.182.153.113:5060 -> 103.155.114.101:5060 ACK sip:lucentNGFS-115804@103.155.114.101:5060;alias=10.5.110.117~5060~1;x-afi=11 SIP/2.0 Via: SIP/2.0/UDP 103.182.153.113:5060 ;rport;branch=z9hG4bKPjb7299c25-13d3-484e-8e78-e7c83620edce From: <sip:917946357720@gaesip.teleforce.in >;tag=8dba28f5-d250-48c6-99fc-35d84590cfc1 To: <sip:09496381412@103.155.114.101 >;tag=6541fd97-665bfb9a2aea9a78-gm-po-lucentPCSF-109109 Call-ID: 1d2897fc-8ae2-4835-b7ac-a96cc28adcc2 CSeq: 22823 ACK Max-Forwards: 70 User-Agent: Asterisk PBX 18.13.0 Content-Length: 0 Thanks Arun

3 3

StriShaken module compilation
by Sergio Charrua 01 Jun '24

01 Jun '24

Hello all! I am struggling on compiling StriShaken module on RHEL 9.2. So far, the module was compiled as well as libstirshaken. Below are the steps used to compile (as sudo) both library and module (I hope this may help someone) and, of course, correct me if I am wrong in any step or if there is a better way. Also, note that the RHEL 9,2 is in a VM environment without access to the WWW, instead it uses Red Hat Satellite to download packages (via Yum or DNF), if available. Lib LibKS download libks from https://github.com/signalwire/libks unzip move to dir created by Unzip run yum groupinstall "Development Tools" dnf install libuuid-devel libatomic openssl-devel cmake . make make install cp /usr/lib/pkgconfig/libks2.pc /usr/lib64/pkgconfig/. For module LibStirShaken: download https://github.com/signalwire/libstirshaken unzip move to dir if using OpenSSL3.0 or + edit file configure.ac and add after line 28: if test x$HAVE_OPENSSL = x1; then openssl_CFLAGS="$openssl_CFLAGS -DHAVE_OPENSSL"; /* PATCH FOR OPENSSL3 */ AC_MSG_CHECKING([for OpenSSL >= 3.0.0]) AC_COMPILE_IFELSE([AC_LANG_PROGRAM([[ #include <openssl/opensslv.h> #if OPENSSL_VERSION_PREREQ(3,0) #error "you_have_version_3" #endif ]], [[]])], [ AC_MSG_RESULT([no]) ], [ AC_MSG_RESULT([yes]); AC_DEFINE(OPENSSL_SUPPRESS_DEPRECATED, 1, [disable openssl deprecated-function warnings]) ]) /* END OF PATCH */ else AC_MSG_ERROR([OpenSSL >= 1.0.1e and associated developement headers required]) fi run configure make sudo make install next, move to dir /root/kamailio/kamailio-5.7.4/src/modules/stirshaken cd /root/kamailio/kamailio-5.7.4/src/modules/stirshaken ln -s /root/stirshakenlib/libks-master/src/include/libks libks make make install All seem to compile OK , *but* when starting Kamailio I get the following errors: May 29 17:42:02 kamailio1 kamailio[248991]: ERROR: <core> [core/sr_module.c:599]: ksr_load_module(): could not open module </usr/local/lib64/kamailio/modules/stirshaken.so>: libstirshaken.so.1: cannot open shared object file: No such file or directory May 29 17:42:02 kamailio1 kamailio[248991]: CRITICAL: <core> [core/cfg.y:3915]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio_stir_shaken_mod.cfg, line 106, column 12-26: failed to load module I haven't found a solution for this. Can anyone help? Thanks in advance! *Sérgio Charrua*

2 1

Parsing Error with SIP 607 Response
by Ghulam Mustafa 31 May '24

31 May '24

Hello Kamailio Community, I'm encountering a parsing error in Kamailio when receiving a SIP 607 "Unwanted" response. The relevant log entries are: May 31 12:32:09 k-proxy1 /usr/sbin/kamailio[320898]: ERROR: <core> [core/receive.c:378]: receive_msg(): core parsing of SIP message failed (36.53.xx.xx:5060/1) May 31 12:32:10 k-proxy1 /usr/sbin/kamailio[320895]: ERROR: <core> [core/parser/parse_fline.c:272]: parse_first_line(): parse_first_line: bad message (offset: 11) May 31 12:32:10 k-proxy1 /usr/sbin/kamailio[320895]: ERROR: <core> [core/parser/msg_parser.c:790]: parse_msg(): ERROR: parse_msg: message=<SIP/2.0 607#015#012Via: SIP/2.0/UDP 134.195.195.40:5060;branch=z9hG4bKc87a.ed6403ae1529f36b65171bf964604be0.0;i=2#015#012Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-Sz6QPbTDsrqSQc7ys87Ds87VP37Ds87YG870sXCWo1G6ujo6oXzisr0isX7Dsr0isXCSZXcFujwtH.KOH.QqGXs09XNtw.dEJ5GhvjhauNHzwFIzl5c9wbfNG3oaHkl3#015#012From: "15516756486" <sip:15516756486@10.x.x.10>;tag=KXtSBZ1at71Hc#015#012To: <sip:99904441743709685@36.53.xx.xx;transport=udp>;tag=sbcsipuas_1_C35684_20240531083209909_sbc06#015#012Call-ID: a40d965d-99ec-123d-6f92-ecf4bbd64f6c#015#012Contact: <sip:36.53.xx.xx:5060>#015#012CSeq: 84025572 INVITE#015#012Server: sbc_5#015#012Content-Length: 0#015#012#015#012> And the actual SIP message is U 36.53.xx.xx:5060 -> 10.x.x.30:5060 #850 SIP/2.0 607. Via: SIP/2.0/UDP 134.195.195.40:5060 ;branch=z9hG4bKbdbd.95060a558d66b77d258b8d0799be4cf0.0;i=2. Via: SIP/2.0/UDP 127.0.0.8 ;branch=z9hG4bKsr-Sz6QPbTDsrqSQc7ys87Ds87VP37Ds87YG870sXCWo1G6ujo6oXzisr0isX7Dsr0isXCSZXcFujwtH.KOH.QqGXs09XNtw.dEJ5GhvjhauNHzwFCaHXyLA3G.SjwaAmdn. From: "13xxxxxxxxxx" <sip:13xxxxxxxxxx@10.x.x.10>;tag=pFFatBpUc75BH. To: <sip:441xxxxxxxxx@36.53.xx.xx ;transport=udp>;tag=sbcsipuas_1_C56639_20240531081932874_sbc04. Call-ID: e0d2f5fa-99ea-123d-6f92-ecf4bbd64f6c. Contact: <sip:36.53.xx.xx:5060>. CSeq: 84025194 INVITE. Server: sbc_5. Content-Length: 0. Any insights or suggestions would be greatly appreciated. Thank you! Mustafa

5 9

Access avp vars in a cancel? How to pass information from an INVITE to a CANCEL?
by Benoit Panizzon 31 May '24

31 May '24

Hi To determine, if rtpengine was engaged during an invite or not, I make use of rr params for all messages in following transactions and I set an avp for all messages in the current transaction. When the call is canceled, I would need to call rtpengine_delete to make sure rtpengine stops listening for rtp packets. Unfortunately, the avp I set during the INVITE, does not seem to be present while the CANCEL message is processed, despite to my understanding being in the same transaction. How can I pass some information from an INVITE to the corresponding CANCEL? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

3 10

core dump on AWS with ~4000 TCP sockets
by David Villasmil 31 May '24

31 May '24

Hello all, I'm seeing core dumps (can't find them!) on AWS with lots of TCP sockets open (not related to load), kamailio dies randomly with: *May 30 14:31:03 sip-proxy /usr/sbin/kamailio[743739]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 142 May 30 14:31:03 sip-proxy /usr/sbin/kamailio[743542]: ALERT: <core> [main.c:788]: handle_sigs(): child process 743672 exited by a signal 11 May 30 14:31:03 sip-proxy /usr/sbin/kamailio[743542]: ALERT: <core> [main.c:791]: handle_sigs(): core was generated May 30 14:32:03 sip-proxy /usr/sbin/kamailio[743542]: CRITICAL: <core> [main.c:698]: sig_alarm_abort(): shutdown timeout triggered, dying...* before this happens, there's a ton of May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 172.56.51.197:34846 failed (timeout) May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 98.196.165.187:38593 failed (timeout) May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 10.241.49.185:49372 failed (timeout) May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 97.101.162.40:45248 failed (timeout) May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 172.59.213.36:57074 failed (timeout) May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 207.195.63.14:2521 failed (timeout) May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 10.241.49.185:22155 failed (timeout) May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 10.241.49.185:10339 failed (timeout) Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 1

Re: Via header force change protocol to TLS?
by David Villasmil 31 May '24

31 May '24

trying to use msg_apply_changes(); subst_hf("Via","/TCP/TLS/","f"); But as i don't seem to have access the top-most via, i can'it doesn't work... i cam change the incoming Via, but not the outgoing... I need this urgently.. any help anyone? > Sergey, > > Thanks for pointing me to the PR but that's not exactly what i need, let > me explain: > > We have an AWS an NLB on which our clients connect to and terminate TLS > connections and from then on into our proxies it is TCP. Now when clients > send invites (with TLS in their VIA as protocol) and we reply to them > (because the socket is TCP) I need to set the protocol on the Via to be > TLS. Otherwise, the client would not understand. This message or will > believe that it's not. about the same connection > > I haven't figured out a way of doing this since it can't get the actual > VIA that will be sent out. I've tried on the onsend_route (enabling > onsend_route_reply) > > Help is greatly appreciated! > > > Regards, > > David Villasmil > email: david.villasmil.work(a)gmail.com > phone: +34669448337 > > > On Thu, May 30, 2024 at 11:44 PM David Villasmil < > david.villasmil.work(a)gmail.com> wrote: > >> HEllo Sergey, >> >> i can send one. yes. >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work(a)gmail.com >> phone: +34669448337 >> >> >> On Thu, May 23, 2024 at 5:14 PM Sergey Safarov <s.safarov(a)gmail.com> >> wrote: >> >>> Hi David >>> Could you send PCAP for an inbound call via TCP connection? >>> >>> Sergey >>> >>> On Thu, May 23, 2024 at 5:53 PM David Villasmil < >>> david.villasmil.work(a)gmail.com> wrote: >>> >>>> it's still in progress though. >>>> Regards, >>>> >>>> David Villasmil >>>> email: david.villasmil.work(a)gmail.com >>>> phone: +34669448337 >>>> >>>> >>>> On Thu, May 23, 2024 at 4:51 PM David Villasmil < >>>> david.villasmil.work(a)gmail.com> wrote: >>>> >>>>> Thanks, I'll check it out! >>>>> Regards, >>>>> >>>>> David Villasmil >>>>> email: david.villasmil.work(a)gmail.com >>>>> phone: +34669448337 >>>>> >>>>> >>>>> On Thu, May 23, 2024 at 4:16 PM Sergey Safarov <s.safarov(a)gmail.com> >>>>> wrote: >>>>> >>>>>> We have tested this PR using the Linphone app. >>>>>> So your case will be resolved using this PR. >>>>>> Need to enable HAproxy protocol headers. >>>>>> >>>>>> On Wed, May 22, 2024 at 4:36 PM Sergey Safarov <s.safarov(a)gmail.com> >>>>>> wrote: >>>>>> >>>>>>> Please try Kamailio PR >>>>>>> https://github.com/kamailio/kamailio/pull/3731 >>>>>>> >>>>>>> We have developed this PR for use case you have described. >>>>>>> We have tested Route and Record-Route headers not Via. >>>>>>> So will provide some review for this PR then will be fine. >>>>>>> >>>>>>> On Wed, May 22, 2024 at 2:22 PM David Villasmil < >>>>>>> david.villasmil.work(a)gmail.com> wrote: >>>>>>> >>>>>>>> Hello Sergey, >>>>>>>> >>>>>>>> Thanks for the suggestion. Not sure if his is what i'm looking for, >>>>>>>> allow me to explain further: >>>>>>>> We set up an NetworkLoadBalancer on AWS to offload tls on it. This >>>>>>>> Load balancer is a TLS listener on the outside and a TCP connection to the >>>>>>>> proxy inside. >>>>>>>> So when sending an INVITE to the connected client, the via has a >>>>>>>> TCP protocol like >>>>>>>> >>>>>>>> Via: SIP/2.0/TCP >>>>>>>> mydomain:port;branch=z9hG4bKf176.53ac8af0d7090a31e44548f15ea420ff.0 >>>>>>>> >>>>>>>> and the client (linphone) disconnects and tries to contact the >>>>>>>> proxy on that address on a TCP socket, which doesn't exist. I tried many >>>>>>>> solutions none of which actually work... last one setting $du =$du + >>>>>>>> ";transport=tls" and forcing the socket to the TCP socket to the load >>>>>>>> balancer, but of course i'm getting warnings about this. >>>>>>>> >>>>>>>> is this something that PR (not merged) would be addressing, i >>>>>>>> didn't see that. >>>>>>>> If not, is there a way of doing this without any trickery? >>>>>>>> >>>>>>>> Thanks! >>>>>>>> >>>>>>>> Regards, >>>>>>>> >>>>>>>> David Villasmil >>>>>>>> email: david.villasmil.work(a)gmail.com >>>>>>>> phone: +34669448337 >>>>>>>> >>>>>>>> >>>>>>>> On Wed, May 22, 2024 at 12:16 PM Sergey Safarov < >>>>>>>> s.safarov(a)gmail.com> wrote: >>>>>>>> >>>>>>>>> Probable you need this PR >>>>>>>>> https://github.com/kamailio/kamailio/pull/3810 >>>>>>>>> >>>>>>>>> Or you can try >>>>>>>>> https://github.com/kamailio/kamailio/pull/3731 >>>>>>>>> In this PR we faced the same issue and solved this. >>>>>>>>> >>>>>>>>> >>>>>>>>> On Wed, May 22, 2024 at 3:43 AM David Villasmil via sr-users < >>>>>>>>> sr-users(a)lists.kamailio.org> wrote: >>>>>>>>> >>>>>>>>>> Hello Anthony, did you solve this problem? I'm facing the same >>>>>>>>>> problem >>>>>>>>>> >>>>>>>>>> Thanks! >>>>>>>>>> Regards, >>>>>>>>>> >>>>>>>>>> David Villasmil >>>>>>>>>> email: david.villasmil.work(a)gmail.com >>>>>>>>>> phone: +34669448337 >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> On Mon, Feb 5, 2018 at 5:57 AM Anthony Alba < >>>>>>>>>> ascanio.alba7(a)gmail.com> wrote: >>>>>>>>>> >>>>>>>>>>> I have kamailio behind a TLS termination proxy so the sockets >>>>>>>>>>> are correctly deduced to be TCP. However the clients only talk TLS to the >>>>>>>>>>> proxy and are confused when the top Via header added by Kamailio is TCP. Is >>>>>>>>>>> there a way for Kamailio to forcibly pretend its protocol is TLS? Like >>>>>>>>>>> advertised_address but "advertised_protocol" instead. >>>>>>>>>>> >>>>>>>>>>> (With pjsip testing: it has a flag use_tls which ignores TCP >>>>>>>>>>> from Kamailio and continues to use the persistent TLS transport to proxy. >>>>>>>>>>> Linphone fails because it tries to honor TCP in Via and is unable to >>>>>>>>>>> establish TCP transport). >>>>>>>>>>> >>>>>>>>>>> BTW I am using t_relay_to_tcp so Kamailio will return traffic to >>>>>>>>>>> the proxy as TCP even though the contact addresses specify transport=TLS. >>>>>>>>>>> _______________________________________________ >>>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>>> >>>>>>>>>> __________________________________________________________ >>>>>>>>>> Kamailio - Users Mailing List - Non Commercial Discussions >>>>>>>>>> To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org >>>>>>>>>> Important: keep the mailing list in the recipients, do not reply >>>>>>>>>> only to the sender! >>>>>>>>>> Edit mailing list options or unsubscribe: >>>>>>>>>> >>>>>>>>>

1 0

Inbound Identity Verification Stir/Shaken
by tfayomi＠untangledtechnology.com 30 May '24

30 May '24

Hi All, I have a quick question regarding verification on incoming calls regarding the stir/shaken policy. I'm using the secsipid library to check the identity header, but the problem I'm running into is that there is no identity header in the invite from the incoming call (It's showing up as null). Have most carriers enabled passing along an identity header, or is this something that is still in the works for some of them. Would I need to request and get the certificate from the terminating carrier? I've tried using the auth_identity module to see if i need to use the verification functions to get/request the certificate, but I have had no luck so far. Would anyone be able to provide direction on how to request/get/pass along the identity header from an incoming call with stir/shaken. (I am using kamailio v5.6 and debian 12) Thanks in advance, Temi

4 6

Via header force change protocol to TLS?
by Anthony Alba 30 May '24

30 May '24

I have kamailio behind a TLS termination proxy so the sockets are correctly deduced to be TCP. However the clients only talk TLS to the proxy and are confused when the top Via header added by Kamailio is TCP. Is there a way for Kamailio to forcibly pretend its protocol is TLS? Like advertised_address but "advertised_protocol" instead. (With pjsip testing: it has a flag use_tls which ignores TCP from Kamailio and continues to use the persistent TLS transport to proxy. Linphone fails because it tries to honor TCP in Via and is unable to establish TCP transport). BTW I am using t_relay_to_tcp so Kamailio will return traffic to the proxy as TCP even though the contact addresses specify transport=TLS.

4 10

libstirshaken installation issues
by tfayomi＠untangledtechnology.com 30 May '24

30 May '24

Hi All, I am attempting to install the stirshaken module for kamailio and I ran into a couple issues installing the open source c library for libstirshaken. When I use the make command, I'm getting this error: src/stir_shaken.c: In function 'stir_shaken_is_key_trusted': src/stir_shaken.c:726:9: error: 'EVP_PKEY_cmp' is deprecated: Since OpenSSL 3.0 [-Werror=deprecated-declarations] 726 | if (!EVP_PKEY_cmp(pkey, candidate_pkey)) { | ^~ In file included from /usr/include/openssl/x509.h:29, from /usr/include/openssl/ssl.h:31, from /usr/include/libks2/libks/ks_ssl.h:25, from /usr/include/libks2/libks/ks.h:80, from include/stir_shaken.h:15, from src/stir_shaken.c:1: /usr/include/openssl/evp.h:1418:5: note: declared here 1418 | int EVP_PKEY_cmp(const EVP_PKEY *a, const EVP_PKEY *b); | ^~~~~~~~~~~~ At top level: cc1: note: unrecognized command-line option '-Wno-gnu-zero-variadic-macro-arguments' may have been intended to silence earlier diagnostics cc1: all warnings being treated as errors make: *** [Makefile:1337: src/stir_shaken.lo] Error 1 I tried going into the stir_shaken file and changing EVP_PKEY_cmp to EVP_PKEY_eq, but it just brings up more deprecation errors. I also tried rolling back the openssl version on my machine to openssl version 1.1 and changing EVP_PKEY_eq back to EVP_PKEY_cmp, but the same error persists. Is there any way to resolve this error, or is there another library that can be used to get the stirshaken module for kamailio? Thanks in advance, Temi

6 7

Is udp_mtu and udp_mtu_try_proto broken in kamailio v5.8
by Supreeth Herle 30 May '24

30 May '24

Hello, I have a dockerized IMS setup using kamailio from git commit id 4fb8accc6747ad56fec3dc84d70cb2b8bbd7316e (tag 5.8.x), where I am seeing an issue faced by one of the users of the dockerized IMS. The issue is that the SIP request (in UDP) even though it exceed the MTU of 1300 (udp_mtu=1300 and udp_mtu_try_proto=TCP) the SIP request from P-CSCF is not sent out as a TCP. The pcap and P-CSCF logs in debug can be found in the github issue - https://github.com/herlesupreeth/docker_open5gs/issues/316#issuecomment-213… IP of the containers are follows: P-CSCF - 172.22.0.21 S-CSCF - 172.22.0.20 SIP Client - 192.168.101.x P-CSCF is configured with udp_mtu=1300 and udp_mtu_try_proto=TCP In the pcap you can notice that the SIP NOTIFY originating from S-CSCF is forwarded to P-CSCF (packet no. 698) and packet length is 2716. At this point P-CSCF should have tried to convey the SIP packet to SIP client via TCP. But I dont see any TCP connection establishment either and eventually request times out (packet 1147). [image: image.png] Any help or insights would be highly appreciated. Thanks and Regards, Supreeth

1 0

Re: Failover with dialog dmq while dialog state 2
by Björn Klasen 27 May '24

27 May '24

Hi Henning, thank you for your reply. I did a lot testings during the past days and I think I found a solution that is working, although it's very dirty... I found out it is possible to send a manual KDMQ-Message to all Proxy-Nodes including the one that it sending die KDMQ-Message. So I create a dialog state 4 message on ACK and indeed the Dialog on all Peers are Updated correctly to state 4. I also tried this for state 3 on 200 OK, but this does not work. I haven't had a look into the code yet but my experiments shows, that there is never is state 3 update via DMQ, so I think it's not implemented. Interestingly the TM-Timer still times out after 300 seconds but in normal circumstances this doesn't bother, because either the failed Kamailio will not trigger this, because the Software or the Server crashed, or it tries to send a CANCEL that is denied because we are in dialog state 4. The only thing is your really have to be careful about your hash tables. You have to delete them at the correct point. I really need to test a lot at this point, but till now, billing seems to be OK even on failover scenarios. This is why I really like Kamailio :) BR, Björn Am 09.05.24 um 20:57 schrieb Henning Westerholt: Hello, thanks for the detailed e-mail. As also indicated in the module documentation, the dialog module DMQ replication will not replicate everything, its main use-case is for profile data sharing. https://kamailio.org/docs/modules/5.8.x/modules/dialog.html#dialog.p.enable… In the past months there have been some other discussions on the users lists about similar scenarios (I think related to billing/accounting) and dialog with DMQ, which might be interesting for you in this regard. If you find issues where the DMQ synchronisation is lacking some functionality in the dialog module, you can create a feature request in our issue tracker. There is of course no guarantee that this limitation is also timely addressed. Regarding the INVITE and CANCEL scenario, this is usually not related to dialog but to the tm module. As you also mentioned, there is no replication of transaction state in tm. Cheers, Henning -- Björn Klasen, Senior Specialist (VoIP) TNG Stadtnetz GmbH, TNG-Technik Gerhard-Fröhler-Straße 12 24106 Kiel・Deutschland T +49 431 7097-10 F +49 431 7097-555 bklasen(a)tng.de<mailto:bklasen@tng.de> https://www.tng.de Executive board (Geschäftsführer): Dr. Sven Willert (CEO/Vorsitz), Gunnar Peter, Sven Schade, Carsten Tolkmit, Bernd Sontheimer Amtsgericht Kiel HRB 6002 KI USt-ID: DE225201428 Die Information über die Verarbeitung Ihrer Daten gemäß Artikel 12 DSGVO können Sie unter https://www.tng.de/datenschutz/ abrufen. ______________________________________________________________________

2 3

I'm looking for an get_rr_param() usage example
by Benoit Panizzon 27 May '24

27 May '24

Hi Gang To process spiraling calls without the dialog module. I'm trying to use a rr param to identify in which iteration I am. So when I have a call spiraling through the same instance 3 times, every time I process an invite I would do something like: add_rr_param(";sp-count=$avp(sp-count)"); But to initialize that counter, I need to be able to read what is in there beforehand. if (check_route_param("sp-count=") { get_route_param(msg, "sp-count", $avp(sp-count)) $avp(sp-count) = $avp(sp-count) + 1; add_rr_param(";sp-count=$avp(sp-count)"); } else { add_rr_param(";sp-count=1"); } But what do I pass as msg parameter? Is it the PV $msg(hdrs) ? PS: I need this to append this to the call ID for rtpengine so I hopefully can match new transactions no matter from which side they are initiated to the correct rtp stream. Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

1 0

Is there any funtion to route based on source phone number
by SAMUEL MOYA TINOCO 27 May '24

27 May '24

Hi everyone, I have a kamailio configured to send invites from different phone numbers registered in kamailio to different endpoints. The configuration that I have right now is a big if chain with the phone numbers that then sends the invite using dispatcher module. Example: If (phone number in invite = phone number X) { dispatcher sends to endpoint X} If (phone number in invite = phone number Y) { dispatcher sends to endpoint Y} If (phone number in invite = phone number Z) { dispatcher sends to endpoint Z} Im not sure if this is the correct way of doing this or if there is any module that can do this allowing me to configure it using BBDD instead of kamailio configuration. I started doing it like this on our test environment while I was learning how to use kamailio and I'd like to know if this is the right way to do it or if I should do it with other methods Thank you in advance

7 7

Fwd: rtpengine timestamp jumps
by Jeff Brower 26 May '24

26 May '24

Rtpengine support- I am re-posting again, as I noticed on the SR-USERs web page no HTML or inline images are showing (https://lists.kamailio.org/mailman3/hyperkitty/list/sr-users@lists.kamailio…). The Wireshark screencap that demonstrates the issue is uploaded at: https://www.signalogic.com/images/rtpengine_wireshark_capture_timestamp_jum… all other info remains the same. Thanks, Jeff ----- Forwarded message from Jeff Brower <jbrower(a)signalogic.com> ----- Date: Fri, 17 May 2024 17:10:10 +0000 From: Jeff Brower <jbrower(a)signalogic.com> Subject: Fwd: [SR-Users] rtpengine timestamp jumps To: sr-users(a)lists.kamailio.org Reposting this. If there is an issue with HTML format and/or wireshark screen cap and I need to upload that separately somewhere else, please let me know. Thanks. -Jeff ----- Forwarded message from Jeff Brower <jbrower(a)signalogic.com> ----- Date: Thu, 09 May 2024 05:32:27 +0000 From: Jeff Brower <jbrower(a)signalogic.com> Subject: [SR-Users] rtpengine timestamp jumps To: sr-users(a)lists.kamailio.org Hi rtpengine experts, We have some customers processing long multi-party call pcaps using mediaMin who are reporting large amounts of packets with timestamp jumps but no packet loss (for instance 10% of packets over a 1 hr 45 min call). For example, in the Wireshark excerpt shown below, packets 6 and 8 sent by rtpengine show a timestamp increment of 640, but sequence number increment of 1: [screencap link at https://www.signalogic.com/images/wireshark_capture_timestamp_jump.png] In the mediaMin output packet log we typically see sections similar to: : : Seq num 98584 timestamp = 3902252372, rtp pyld len = 33 media-R Seq num 98585 timestamp = 3902252692, rtp pyld len = 33 media Seq num 98586 timestamp = 3902253012, rtp pyld len = 33 media-R Seq num 98587 timestamp = 3902253332, rtp pyld len = 33 media Seq num 98588 timestamp = 3902253652, rtp pyld len = 33 media-R Seq num 98589 timestamp = 3902253972, rtp pyld len = 33 media Seq num 98590 timestamp = 3902254292, rtp pyld len = 33 media Seq num 98591 timestamp = 3902254612, rtp pyld len = 33 media-R Seq num 98592 timestamp = 3902254932, rtp pyld len = 33 media Seq num 98593 timestamp = 3902255252, rtp pyld len = 33 media Seq num 98594 timestamp = 3902255572, rtp pyld len = 33 media-R Seq num 98595 timestamp = 3902255892, rtp pyld len = 33 media Seq num 98596 timestamp = 3902256212, rtp pyld len = 33 media Seq num 98597 timestamp = 3902256532, rtp pyld len = 33 media Seq num 98598 timestamp = 3902256852, rtp pyld len = 33 media-R Seq num 98599 timestamp = 3902257172, rtp pyld len = 33 media Seq num 98600 timestamp = 3902257492, rtp pyld len = 33 media-R Seq num 98601 timestamp = 3902257812, rtp pyld len = 33 media Seq num 98602 timestamp = 3902258132, rtp pyld len = 33 media Seq num 98603 timestamp = 3902258452, rtp pyld len = 33 media Seq num 98604 timestamp = 3902258772, rtp pyld len = 33 media-R Seq num 98605 timestamp = 3902259092, rtp pyld len = 33 media : : where media-R packets are timestamp gap repairs (i.e. frame loss concealment). The behavior tends to be bursty, but once it gets going it goes for a while and seems relatively consistent. Is this expected behavior for rtpengine ? If so is rptengine in turn dealing with some type of "slow packet rate" issue from a remote sender ? Thanks, Jeff ----- End forwarded message ----- ----- End forwarded message -----

2 4

core children setting
by Alexis Fidalgo 24 May '24

24 May '24

Hello all, im not clear on the children setting, currently we’re listening in a single UDP port, i do understand that children are how many forks per listener ip:port, but i have this question, is this value tied on how many requests are received? In our case every requests takes an average of 80ms to be processed (there are a few external queries executed per request), so, example which value for children should i use if i have 200req/s, 300, 400? Right now im using 16 but i can see some cpu peaks in high traffic hours. thanks

2 2

"Contact does not exist in PCSCF usrloc" error in log
by Shantanu K Roy 23 May '24

23 May '24

Hello, We are trying to integrate Kamailio as the IMS server with our 5G core emulator by following the steps listed out at https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/. In our case, we have our own 5G core emulator instead of the Open5GS core. We are using sipp scripts to generate the SIP calls. The PCRF component of the 5G core is emulated by us and is capable of handling Diamater calls from Kamailio over the Rx interface. We have got the setup working to the point where the SIP Register calls are landing on the PCSCF which in turn is sending out AA Requests to PCRF and the PCRF is responding. However there is an internal error in the Kamailio log "4(3682133) ERROR: ims_qos [ims_qos_mod.c:1305]: w_rx_aar_register(): This contact does not exist in PCSCF usrloc - error in cfg file" Please help us in identifying if we are missing anything in the Kamailio set up. The packet capture and syslog excerpts from the VM where Kamailio is running are attached; here we are trying to perform SIP Registration for three SIP endpoints 001010123456791, 001010123456792 and 001010123456793. If there is any additional information required, please let us know. Regards, Shantanu

1 0

VoLTE/IMS third party registration issues
by ruben.iniguez＠vodafone.com 23 May '24

23 May '24

Checking Kamailio in a VoLTE environment we are finding issues to make it work. First issues arise when getting AS to extract the required info from third party registration, primarily IMPUs. Third party registration is sent to AS with IMPI in To header, preventing the AS from extracting the IMPU from the most obvious place, the To header, according to 3GPP 24.229: "Notification of Application Servers about registration status [...] the To header field, which shall contain a non-barred public user identity belonging to the service profile of the processed Filter Criteria. It may be either a public user identity as contained in the REGISTER request received from the UE or one of the implicitly registered public user identities in the service profile, as configured by the operator;" On the other hand, the third party registration can only embed in its body either the original request or the original response but not both (notice mutually exclusive "else if" statements in third_party_reg.c). Both are needed to extract Feature-Caps (from original request) and IMPUs (for instance from P-Associated-URI in original response). Therefore to extract that info the AS should subscribe to the reg info in S-CSCF, but such SUBSCRIBE is rejected with 403. From Registrar IMS SCSCF documenation it looks like this is not supported: "This function checks to see that a SUBSCRIBE request is authorised to subscribe to the particular identity. Only 3 entities can subscribe: The user agent to its own state The P-CSCF specified in the path header for that user Application Server (AS) not yet implemented" Therefore question to the community is how these limitations are overcome in existing VoLTE deployments with Kamailio? Any advice?

1 0

Kamailio IPSec module
by H Yavari 23 May '24

23 May '24

Hi all, I was reviewing the `ims_ipsec_pcscf` code and noticed that this module creates a pool of sockets using different ports (ipsec_max_connections). I'm unclear on the necessity of this approach. Can't we simply create one server listener and one client listener to handle all UE connections? If this is feasible, is there still a need to create the pool at startup? I'm curious if this is due to an architectural limitation or if the IPSec module can be modified to replace the current implementation with a more efficient one. Thank you for your insights. Best regards,Hossein

3 5

S-CSCF SiFC support
by ruben.iniguez＠vodafone.com 22 May '24

22 May '24

Hello In the context of an IMS deployment, does anybody know if Shared iFC mechanism is supported in Kamailio? I don't see configuration options for it, and just wanted to confirm if only regular iFCs (residing on HSS) are supported. Thank you

1 0

Assistance Required for VoNR Setup with Version 5.8.1
by venkateshcoff＠gmail.com 22 May '24

22 May '24

Hi, I am trying to set up VoNR with version 5.8.1, but I cannot find the configuration information and documentation anywhere. Has anyone tested a VoNR call? Thanks.

1 0

usrloc: keepalives not being sent to UDP endpoints for a very long time
by luke＠primevox.net 21 May '24

21 May '24

We've noticed an issue with UDP endpoints and the usrloc module not sending keepalive packets to them for a long time. Eventually, usrloc will begin doing it, but sometimes it's anywhere between 10 minutes and multiple hours before the endpoint starts receiving an OPTIONS packet. TCP and TLS endpoints do not have this issue - they being receiving OPTIONS after registration + ka_timeout elapsed. Has anyone else seen this? It can be very easily seen with a pcap on a device registering via UDP.

1 0

A question about Db_flatstore Module in Kamailio
by فرزانه سلطان زاده 20 May '24

20 May '24

Hi, Hope to find you well, I have a question about the 'db_flatstore' module in Kamailio. Here's my script for log rotation: for f in acc_cdrs_*.log; do mv -- "$f" "${f%.log}.temp" done docker exec -it pcscf kamcmd flatstore.k_rotate rm *.temp I want to know if the kamcmd command is blocking and the files can be removed after that or if I should wait for it to be done before removing the renamed files. Regards, Farzaneh

3 2

Kamailio multidomain IMS
by Valentin Christoph 20 May '24

20 May '24

Dear IMS enthusiasts :-) Has anyone experience with Kamailio IMS (CSCF) setups, where users with different domain names can be registered at one and the same S-CSCF? Is this feasible at all? I mean one IMS for users <user>@ims1.example.com, <user>@ims2.example.com, <user>@ims3.example.com,........ in one and the same S-CSCF, I-CSCF, P-CSCF. Thanks, Christoph

2 1

q_malloc.c:520 WARNINGs
by Joel Serrano 20 May '24

20 May '24

Hi, Does anyone know what this WARNING means or how to fix it? 2024-03-09T00:39:15.406674-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.406823-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.406973-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.407089-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.407178-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.407278-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.407375-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) It happens after running dp_translate(). These are my settings: debug=L_DBG debug=L_NOTICE memdbg=L_INFO memlog=L_INFO corelog=L_ERR mem_summary=12 log_stderror=no log_facility=LOG_LOCAL0 log_name="csbc" log_prefix="{$mt $hdr(CSeq) $ci} " mem_join=1 Any ideas? Thanks, Joel.

2 5

tm.t_uac_wait ACK does not have Max-Forwards header
by Juha Heinanen 20 May '24

20 May '24

Kamailio sends INVITE by tm.t_uac_wait RPC command: INVITE sip:test-0x55e7f1f78e70@192.168.187.215:5022 SIP/2.0 Via: SIP/2.0/UDP 192.168.187.215;branch=z9hG4bK6cf1.f1d087d6000000000000000000000000.0 To: <sip:test@test.tutpro.com> From: <sip:click2dial@test.tutpro.com>;tag=2dd3d4389fe3d66c5cb97dc370ab7d7b-9c5a2a58 CSeq: 1 INVITE Call-ID: 6649e78355ac2(a)test.tutpro.com Content-Length: 131 User-Agent: OpenSIPg SIP Proxy (5.9.0-0b01 (x86_64/linux)) Max-Forwards: 10 Contact: <sip:click2dial@192.168.187.215:5060> Content-Type: application/sdp .. sdp omitted UAS accepts the INVITE: SIP/2.0 200 Answering Via: SIP/2.0/UDP 192.168.187.215;branch=z9hG4bK6cf1.f1d087d6000000000000000000000000.0 To: <sip:test@test.tutpro.com>;tag=a3302e2a0898de3e From: <sip:click2dial@test.tutpro.com>;tag=2dd3d4389fe3d66c5cb97dc370ab7d7b-9c5a2a58 CSeq: 1 INVITE Call-ID: 6649e78355ac2(a)test.tutpro.com Server: baresip v3.12.0 (x86_64/Linux) Contact: <sip:test-0x55e7f1f78e70@192.168.187.215:5022> Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY,SUBSCRIBE,INFO,MESSAGE,UPDATE,REFER Supported: gruu,replaces,norefersub Content-Type: application/sdp Content-Length: 239 .. sdp omitted After receiving the response, Kamailio automatically sends ACK: ACK sip:test-0x55e7f1f78e70@192.168.187.215:5022 SIP/2.0 Via: SIP/2.0/UDP 192.168.187.215;branch=z9hG4bK6cf1.f1d087d6000000000000000000000000.0 From: <sip:click2dial@test.tutpro.com>;tag=2dd3d4389fe3d66c5cb97dc370ab7d7b-9c5a2a58 Call-ID: 6649e78355ac2(a)test.tutpro.com To: <sip:test@test.tutpro.com>;tag=a3302e2a0898de3e CSeq: 1 ACK User-Agent: OpenSIPg SIP Proxy (5.9.0-0b01 (x86_64/linux)) Content-Length: 0 The problem is that Max-Forwards header is missing from the ACK. Should I have something in config file to add it or is this a bug? -- Juha

1 3

Segfault on startup when using TLS in kamailio 5.7.4
by stefanr＠wave.com 17 May '24

17 May '24

Hi there, We are encountering consistent segfaults after rebooting our Kamailio instance with incoming traffic, specifically when using Kamailio 5.7.4. We think this issue did not occur with version 5.7.2, so it seems to have been introduced in either 5.7.3 or 5.7.4. Due to team bandwidth constraints and the potential impact on production traffic, we don't want to spend time on trying to reproduce the issue. So we have decided to downgrade to 5.6.4, which we confirmed to be stable. (Probably 5.7.2 would be too - but we didn't try). Unfortunately, our logging was only set to WARNING level, and we did not capture a core dump, so we cannot provide additional details beyond the following logs: This was with tcp_reuse_ports=yes: 2024-05-17T15:42:55.582475541Z Listening on 2024-05-17T15:42:55.582512370Z [redacted] 2024-05-17T15:42:55.582538161Z tls: 10.X.X.X:5061 advertise Y.Y.Y:5061 2024-05-17T15:42:55.582543750Z Aliases: 2024-05-17T15:42:55.582549081Z tls: [redacted]:5061 2024-05-17T15:42:55.582574890Z 2024-05-17T15:42:55.587876630Z 0(1) WARNING: tls [tls_init.c:978]: tls_h_mod_init_f(): openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 18874368 and 9437184 bytes 2024-05-17T15:42:55.703927049Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 23 2024-05-17T15:42:55.703972029Z 0(1) ALERT: <core> [main.c:791]: handle_sigs(): child process 15 exited by a signal 11 2024-05-17T15:42:55.703978409Z 0(1) ALERT: <core> [main.c:795]: handle_sigs(): core was generated 2024-05-17T15:42:55.705049839Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 17 2024-05-17T15:42:55.705074209Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 21 2024-05-17T15:42:55.705081209Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 22 2024-05-17T15:42:55.705085879Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 20 2024-05-17T15:42:55.705090319Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 18 2024-05-17T15:42:55.705094649Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 19 2024-05-17T15:42:55.705098879Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 16 2024-05-17T15:42:55.705207399Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 15 2024-05-17T15:42:55.705459439Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 27 Without tcp_reuse_ports=yes, the segfault was always preceded by the following line if any existing TLS connections were stuck in TIME_WAIT: 2024-05-16T19:18:51.654447639Z 9(14) WARNING: {1 1 INVITE XXX(a)0.0.0.0} <core> [core/tcp_main.c:1301]: find_listening_sock_info(): binding to source address 10.X.X>X:5061 failed: Address already in use [98] 2024-05-16T19:18:51.746994728Z 0(1) ALERT: <core> [main.c:791]: handle_sigs(): child process 14 exited by a signal 11 When the server wasn't handling any traffic, the issue didn't occur even in 5.7.4. Does anyone have any insights or suggestions on how to address this issue? Kind regards Stefan

2 2

Assistance Needed to Resolve Duplicate Contact Headers in 200 OK Response
by satyaprakash ch 17 May '24

17 May '24

Hi We are receiving 2 to 3 contact headers in the 200 OK response of the registration request. It appears that the `user_location` is updating multiple contact headers for the same extension due to network disconnections. Please suggest a solution to avoid duplicates in the `user_location` module, or how to ensure that only the latest single contact header is included in the 200 OK response. Your suggestion would be appreciated. Thanks and regards, Satyaprakash.

2 1

Fwd: rtpengine timestamp jumps
by Jeff Brower 17 May '24

17 May '24

Reposting this. If there is an issue with HTML format and/or wireshark screen cap and I need to upload that separately somewhere else, please let me know. Thanks. -Jeff ----- Forwarded message from Jeff Brower <jbrower(a)signalogic.com> ----- Date: Thu, 09 May 2024 05:32:27 +0000 From: Jeff Brower <jbrower(a)signalogic.com> Subject: [SR-Users] rtpengine timestamp jumps To: sr-users(a)lists.kamailio.org Hi rtpengine experts, We have some customers processing long multi-party call pcaps using mediaMin who are reporting large amounts of packets with timestamp jumps but no packet loss (for instance 10% of packets over a 1 hr 45 min call). For example, in the Wireshark excerpt shown below, packets 6 and 8 sent by rtpengine show a timestamp increment of 640, but sequence number increment of 1: In the mediaMin output packet log we typically see sections similar to: : : Seq num 98584 timestamp = 3902252372, rtp pyld len = 33 media-R Seq num 98585 timestamp = 3902252692, rtp pyld len = 33 media Seq num 98586 timestamp = 3902253012, rtp pyld len = 33 media-R Seq num 98587 timestamp = 3902253332, rtp pyld len = 33 media Seq num 98588 timestamp = 3902253652, rtp pyld len = 33 media-R Seq num 98589 timestamp = 3902253972, rtp pyld len = 33 media Seq num 98590 timestamp = 3902254292, rtp pyld len = 33 media Seq num 98591 timestamp = 3902254612, rtp pyld len = 33 media-R Seq num 98592 timestamp = 3902254932, rtp pyld len = 33 media Seq num 98593 timestamp = 3902255252, rtp pyld len = 33 media Seq num 98594 timestamp = 3902255572, rtp pyld len = 33 media-R Seq num 98595 timestamp = 3902255892, rtp pyld len = 33 media Seq num 98596 timestamp = 3902256212, rtp pyld len = 33 media Seq num 98597 timestamp = 3902256532, rtp pyld len = 33 media Seq num 98598 timestamp = 3902256852, rtp pyld len = 33 media-R Seq num 98599 timestamp = 3902257172, rtp pyld len = 33 media Seq num 98600 timestamp = 3902257492, rtp pyld len = 33 media-R Seq num 98601 timestamp = 3902257812, rtp pyld len = 33 media Seq num 98602 timestamp = 3902258132, rtp pyld len = 33 media Seq num 98603 timestamp = 3902258452, rtp pyld len = 33 media Seq num 98604 timestamp = 3902258772, rtp pyld len = 33 media-R Seq num 98605 timestamp = 3902259092, rtp pyld len = 33 media : : where media-R packets are timestamp gap repairs (i.e. frame loss concealment). The behavior tends to be bursty, but once it gets going it goes for a while and seems relatively consistent. Is this expected behavior for rtpengine ? If so is rptengine in turn dealing with some type of "slow packet rate" issue from a remote sender ? Thanks, Jeff ----- End forwarded message -----

1 0

$dlg_var() not implemented in ims_dialog module
by Valentin Christoph 17 May '24

17 May '24

Dear all, I would probably need to store some dialog scoped information in an IMS environment, but the ims_dialog module does not implement the $dlg_var() pseudo variable. Do you know other, maybe better, possibilities to store dialog scoped information in an IMS (CSCF) environment? Thank, Christoph

3 2

making HTTP Async request in stateless configuration
by Sergio Charrua 17 May '24

17 May '24

Hi all! Just wondering what would be you opinion on the following. We are integrating Kamailio as a stateless redirect server, using SIP 300 Multiple Choices response. The script requires that prior to the redirection, a HTTP request must be made. We thought about using HTTP Async Client to make the requests (POST of JSON content), which is our prefered way of sending HTTP requests, and depending on the response from the REST API, reply back to UAC a SIP 300 Multiple Choices message and terminate the call flow on Kamailio. From what I know, HTTP Async Client requires that the TM module be used, which (it is my understanding) will imply that the whole Kamailio script to be stateful, which I think is useless as the SIP flow will be really simple: INVITE -----> <------- 100 Trying <------- 300 Multiple Choices ACK ------> There is a catch, though, the CPS is very high, above 1000 CPS.... Could anyone share their thoughts on this? Should we go stateful and use the HTTP Async module? Should we go stateless and use the HTTP client module? What would be the best approach to implementing this? Thanks guys! *Sérgio Charrua*

1 0

WebRTC-UDP voicemail and call transfer
by Social Boh 16 May '24

16 May '24

Hello list, I'm trying implementing a solution like: https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg The 2 problems I have a this moment are: Call from UDP phone to WebRTC phone, the WebRTC phone reject the call; the call go to voicemail but in the SDP still have WebRTC SDP During a SIP UDP - WebRTC Call tranfer the call using phone button... the transfer don't work. Any hint Regards -- --- I'm SoCIaL, MayBe

1 0

dynamic selection of authentication algorithm
by Juha Heinanen 14 May '24

14 May '24

How about this new function that sets the algorithm? -- Juha iff --git a/src/modules/auth/auth_mod.c b/src/modules/auth/auth_mod.c index 534ad9e20f..4e3a584d22 100644 --- a/src/modules/auth/auth_mod.c +++ b/src/modules/auth/auth_mod.c @@ -74,6 +74,10 @@ int w_consume_credentials(struct sip_msg *msg, char *s1, char *s2); * Check for credentials with given realm */ int w_has_credentials(struct sip_msg *msg, char *s1, char *s2); +/* + * Set authentication algorithm + */ +int w_auth_algorithm(struct sip_msg *msg, char *alg, char* s2); static int pv_proxy_authenticate( struct sip_msg *msg, char *realm, char *passwd, char *flags); @@ -170,6 +174,8 @@ static cmd_export_t cmds[] = { REQUEST_ROUTE}, {"pv_auth_check", (cmd_function)w_pv_auth_check, 4, fixup_pv_auth_check, 0, REQUEST_ROUTE}, + {"auth_algorithm", w_auth_algorithm, 1, fixup_spve_null, 0, + REQUEST_ROUTE}, {"bind_auth_s", (cmd_function)bind_auth_s, 0, 0, 0}, {0, 0, 0, 0, 0, 0} @@ -475,6 +481,35 @@ int w_has_credentials(sip_msg_t *msg, char *realm, char *s2) return -1; } return ki_has_credentials(msg, &srealm); + +} +/** + * + */ +int w_auth_algorithm(sip_msg_t *msg, char* alg, char* s2) +{ + if (fixup_get_svalue(msg, (gparam_t*)alg, &auth_algorithm) < 0) { + LM_ERR("failed to get algorithm value\n"); + return -1; + } + + if (strcmp(auth_algorithm.s, "MD5") == 0) { + hash_hex_len = HASHHEXLEN; + calc_HA1 = calc_HA1_md5; + calc_response = calc_response_md5; + } + else if (strcmp(auth_algorithm.s, "SHA-256") == 0) { + hash_hex_len = HASHHEXLEN_SHA256; + calc_HA1 = calc_HA1_sha256; + calc_response = calc_response_sha256; + } + else { + LM_ERR("Invalid algorithm provided." + " Possible values are \"\", \"MD5\" or \"SHA-256\"\n"); + return -1; + } + + return 1; } #ifdef USE_NC

3 6

302 Message Handling
by tfayomi＠untangledtechnology.com 14 May '24

14 May '24

Hi All, I'm having trouble understanding/finding resources on how to handle a 302 message I receive. I'm working on making an inbound sbc with kamailio with clearip integration. I'm successfully sending to kamailio, but am having trouble retrieving the response from clearip in kamailio. For some background, the intended call flow should be this: first it should send from caller to kamailio (caller > kamailio), then over tcp kamailio sends to clearip (kamailio > clearip) [This is what is working as expected] Then I should be able to signal to kamailio to continue the call to the intended destination. When clearip sends a code back (in this case 302), I want to be able to see that code and send the caller to the correct destination. The problem I keep running into is that I can't seem to see the 302 code in a reply, branch or failure root or checking the status. I'm just lost on what I am misunderstanding and would appreciate if anyone could point me in the right direction on how to approach retrieving the 302 message. I'm using kamailio version 5.6.3 and debian 12.1 Thanks in advance, Temi

2 1

Prefix match in htable
by Antonio Reale 13 May '24

13 May '24

Hello, I'm testing Kamailio 5.x as a redirecting server based on calling number. The calling numbers are stored on a DB and loaded in htable at runtime. When the INVITE or REGISTER arrive Kamailio checks if there is a match on $fU in htable (KEY_NAME) and if there is one, it replies with 302 adding the Contact HF with $fU@$sht(cli=>$fU)(with the domain part found in KEY_VALUE): append_to_reply("<Contact: sip:$fU@$sht(cli=>$fU)>\r\n"); If I use the full number as KEY_NAME it works as expected, but how can I match on a prefix instead of the whole $fU? For example, If I receive a call form 0655123456 I want to match it also if I have the prefix 0655 in htable. Do you have some advice to achieve this goal with htable or in a similar way? Thank you. Best regards. -- Antonio Reale

1 0

Kamailio and RTPengine and no voice on calls
by palany 10 May '24

10 May '24

I have an AWS ec2 instance with Kamailio and rtpengine configured. I am having a challenge of no voice on calls. I have sip client on my mobile phone and soft phone on my laptop and the two are on different networks behind nat. Signaling is working fine but there is no media and when I check on on the SDP the call logs are showing private ips instead of public ips : o=- 3 2 IN IP4 192.168.1.33 and c=IN IP4 172.31.36.50. I understand this is a NAT issue but I am failing to see where is the challenge. My rtpengine seems to be working fine . Below is my Kamailio.cfg file link : https://docs.google.com/document/d/1MKZJGlQjxYbdtw6QcCm9_EmvVPOkMs-YXjlq1ZM2 mmY/edit?usp=drive_link PCAP file for the call: https://docs.google.com/document/d/12gofLYhN0-lYIZORngIeuaYxLudLjiNdZgY504Ei XFs/edit?usp=drive_link rtpengine.conf https://docs.google.com/document/d/1QPxey49UzuN3Wd8jp8jc0mFhh31rr1alIO-gl4gP JEI/edit?usp=drive_link Please help.

2 3

base64 decoding issue
by Sergio Charrua 10 May '24

10 May '24

Hi all! I have been dealing with STIR/SHAKEN for a few weeks now, and while doing some tests I have found an issue which I can't find the reason for. The script has the following route logic: route[HANDLE_STIRSHAKEN] { xlog("L_INFO", "HANDLE_STIRSHAKEN - STIR/SHAKEN Logic"); if ($hdrc(Identity) == 0 ){ xlog("L_INFO", "HANDLE_STIRSHAKEN - No Identity Header -> Skipping this logic"); return; } #Verify Call Identity xlog("L_INFO", "HANDLE_STIRSHAKEN - Identity Header: $hdr(Identity)"); $var(header_1)= $(hdr(Identity){s.select,0,.}); $var(header_2)= $(hdr(Identity){s.select,1,.}); xlog("L_INFO", "HANDLE_STIRSHAKEN - Encoded Header 1: $var(header_1)"); xlog("L_INFO", "HANDLE_STIRSHAKEN - Encoded Header 2: $var(header_2)"); xlog("L_INFO", "HANDLE_STIRSHAKEN - Decoded Header 1: $(var(header_1){s.decode.base64})"); xlog("L_INFO", "HANDLE_STIRSHAKEN - Decoded Header 2: $(var(header_2){s.decode.base64})"); xlog("L_INFO", "HANDLE_STIRSHAKEN - Parsing!"); $var(hdr_1_json_payload) = $(var(header_1){s.decode.base64}); # Header 1 xlog("L_INFO", "HANDLE_STIRSHAKEN - Parsed!"); xlog("L_INFO", "HANDLE_STIRSHAKEN - Header 1 payload = $var(hdr_1_json_payload) "); $var(hdr_2_json_payload) = $(var(header_2){s.decode.base64}); # Header 2 jansson_get_field($var(hdr_2_json_payload), "attest", "$var(attest_value)"); jansson_get_field($var(hdr_2_json_payload), "ppt", "$var(ppt_value)"); jansson_get_field($var(hdr_2_json_payload), "typ", "$var(typ_value)"); jansson_get_field($var(hdr_2_json_payload), "x5u", "$var(x5u_value)"); xlog("L_INFO", "HANDLE_STIRSHAKEN - $var(attest_value) $var(ppt_value) $var(typ_value) $var(x5u_value) "); } All it does is parse the Identity header received on the initial INVITE, decodes it (it is encoded in base64) and writes some JSON values on log. Nothing serious.... As the Identity header is divided into multiple parts, separated by a ".", I get the first 2 parts in 2 private variables, as per lines: $var(header_1)= $(hdr(Identity){s.select,0,.}); $var(header_2)= $(hdr(Identity){s.select,1,.}); Logs show correctly that both variables are populated: May 10 09:09:42 kamailio1 kamailio[153201]: INFO: {1 10 INVITE 1-1966540(a)10.20.0.1} <script>: HANDLE_STIRSHAKEN - Encoded Header 1: eyJhbGciOiAiRVMyNTYiLCJwcHQiOiAic2hha2VuIiwidHlwIjogInBhc3Nwb3J0IiwieDV1IjogImh0dHBzOi8vY2VydGlmaWNh dGVzLmNsZWFyaXAuY29tL2IxNWQ3Y2M5LTBmMjYtNDZjMi04M2VhLWEzZTYzYTgyZWMzYS83Y2M0ZGI2OTVkMTNlZGFkYTRkMWY5ODYxYjliODBmZS5jcnQifQ May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 230 May 10 09:09:42 kamailio1 kamailio[153201]: INFO: {1 10 INVITE 1-1966540(a)10.20.0.1} <script>: HANDLE_STIRSHAKEN - Encoded Header 2: eyJhdHRlc3QiOiAiQSIsImRlc3QiOiB7InRuIjogWyIxNDA0NTI2NjA2MCJdfSwiaWF0IjogMTU0ODg1OTk4Miwib3JpZyI6IHsi dG4iOiAiMTgwMDEyMzQ1NjcifSwib3JpZ2lkIjogIjNhNDdjYTIzLWQ3YWItNDQ2Yi04MjFkLTMzZDVkZWVkYmVkNCJ9 However, when decoding both values: xlog("L_INFO", "HANDLE_STIRSHAKEN - Decoded Header 1: $(var(header_1){s.decode.base64})"); xlog("L_INFO", "HANDLE_STIRSHAKEN - Decoded Header 2: $(var(header_2){s.decode.base64})"); only $var(header_2) gets decoded: May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 38 May 10 09:09:42 kamailio1 kamailio[153201]: INFO: {1 10 INVITE 1-1966540(a)10.20.0.1} <script>: *HANDLE_STIRSHAKEN - Decoded Header 1:* May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 182 May 10 09:09:42 kamailio1 kamailio[153201]: INFO: {1 10 INVITE 1-1966540(a)10.20.0.1} <script>: *HANDLE_STIRSHAKEN - Decoded Header 2: {"attest": "A","dest": {"tn": ["14045266060"]},"iat": 1548859982,"orig": {"tn": "18001234567"},"origid": "3a47ca23-d7ab-446b-821d-33d5deedbed4"}* As you can veri And to ease debugging, I have isolated the line that generates the issue: May 10 09:09:42 kamailio1 kamailio[153201]: INFO: {1 10 INVITE 1-1966540(a)10.20.0.1} <script>: HANDLE_STIRSHAKEN - Parsing! May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 *May 10 09:09:42 kamailio1 kamailio[153201]: ERROR: {1 10 INVITE 1-1966540(a)10.20.0.1 <1-1966540(a)10.20.0.1>} <core> [core/lvalue.c:351]: lval_pvar_assign(): non existing right pvar* *May 10 09:09:42 kamailio1 kamailio[153201]: ERROR: {1 10 INVITE 1-1966540(a)10.20.0.1 <1-1966540(a)10.20.0.1>} <core> [core/lvalue.c:411]: lval_assign(): assignment failed at pos: (21,32-21,64)* May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 27 May 10 09:09:42 kamailio1 kamailio[153201]: INFO: {1 10 INVITE 1-1966540(a)10.20.0.1} <script>: HANDLE_STIRSHAKEN - Parsed! Even though the $var(header_1) contains a base64 encoded string, the decoder seems to fail. If I copy the value contained in $var(header_1) to an online decoder like https://www.base64decode.org/ the string gets decoded correctly. I have even modified the line : $var(header_1)= $(hdr(Identity){s.select,0,.}); to get the reverse index from the Identity string by doing $var(header_1)= $(hdr(Identity){s.select,-6,.}); It gets the same encoded string, same value (as expected), but the result is the same: Kamailio's s.decode.base64 fails with the same error. i'm not sure what assignment failed at pos: (21,32-21,64) could mean.... I don't think my code is wrong, as the same function is used on $var(header_2) value and it works, but maybe someone can correct me. Greatly appreciate your help. Cheers, *Sérgio Charrua*

1 1

Failover with dialog dmq while dialog state 2
by Björn Klasen 09 May '24

09 May '24

Hello @all as we want to enhance our VoIP-Plattform with DMQ functionalities for replication of Userlocation, Dialog and Hash Tables I made a lot of tests durings the last week. For testing I use two Debian 12 Server with Kamailio 5.7.4 (latest git) Both server have their own floating IP so in case one server fails the floating changes to the working server. This is done via keepalived On both servers Kamailio is active and use ip_free_bind=1 so both Kamailio instances can listen to all necessary IPs. So we have following network configuration: Server1: static IP: 192.168.253.201 floating IP: 192.168.253.200 Server2: static IP: 192.168.253.202 floating IP: 192.168.253.210 Now I test some calls while I trigger a failover, so Server1 takes over 192.168.253.10 Message Test Result CustomerA -> Proxy -> CustomerB Failover while Dialogstate 4 BYE is relayed correctly. Dialog is changing to state 5 CDR needs to be written manually -> no problem at all CustomerA -> Proxy -> CustomerB Failover while Dialogstate 2 200 OK is relayed correctly although no transaction exist an therefore no t_reply route is triggered. This is not a big deal because i can use default onreply_route so all necessary variables can be used via hash table. BUT Big Problem: Dialog state stays 2, so a following ACK is a bogus event and the dialog state is never changed to 4. Even worse: after 300 seconds the dialog is cleaned away by the timer, so I have no chance to do any kind of accounting. Test 2 is a real mess, because for me there is only one solution. I have to check after a failover on incoming replies to an INVITE that was handled on Server2 if these current dialog state is < 4 and cancel them. But I can't see how to do it? Are there other ways than using t_cancel? Or is it even some kind of a bug that a 200 OK does not trigger state 3 on dialog (although I have in mind that it has to be something to do with missing transaction data, that are not replicated because of missing replication capabilities of tm module)? If none of the above is possible I really asking myself of a real use case of dialog dmq replication. -- Björn Klasen, Teamleitung NGN VoIP-Backbone TNG Stadtnetz GmbH, TNG-Technik Gerhard-Fröhler-Straße 12 24106 Kiel・Deutschland T +49 431 7097-10 F +49 431 7097-555 bklasen(a)tng.de https://www.tng.de Executive board (Geschäftsführer): Dr. Sven Willert (CEO/Vorsitz), Gunnar Peter, Sven Schade, Carsten Tolkmit, Bernd Sontheimer Amtsgericht Kiel HRB 6002 KI USt-ID: DE225201428 Die Information über die Verarbeitung Ihrer Daten gemäß Artikel 12 DSGVO können Sie unter https://www.tng.de/datenschutz/ abrufen. ______________________________________________________________________

2 1

How to set "set_dialog_profile" value for Leg-a and leg-b.
by satyaprakash ch 09 May '24

09 May '24

Hi, We are using kamailio and freeswitch, So when we make a call from A to B the call is going via "kamailio -> freeswitch" and "freeswitch -> kamailio". Please suggest me how i can set "set_dialog_profile" for both legs with unique value in kamailio config file. Thanks.

1 0

Is there any way to modify the Refer-To header username?
by Shah Hussain Khattak 09 May '24

09 May '24

Hello Experts, I wanted to check, is there any way to modify the username part of the Refer-To header in Kamailio? For example, I am getting the following header: REFER-TO: <sip:sip.abc.domain.com:5061;transport=tls; ..... ; .... ; ...> the username part is missing. Can I add a username inside it? For example, change it to REFER-TO: <sip:123@sip.abc.domain.com:5061;transport=tls; ..... ; .... ; ...> REFER-TO: <sip:test-kam@sip.abc.domain.com:5061;transport=tls; ..... ; .... ; ...> I could not find any solution in the mailing list, the only way was to remove the header and then re-add it. Thank you! Regards, Shah Hussain

2 2

rtpengine timestamp jumps
by Jeff Brower 09 May '24

09 May '24

Hi rtpengine experts, We have some customers processing long multi-party call pcaps using mediaMin who are reporting large amounts of packets with timestamp jumps but no packet loss (for instance 10% of packets over a 1 hr 45 min call). For example, in the Wireshark excerpt shown below, packets 6 and 8 sent by rtpengine show a timestamp increment of 640, but sequence number increment of 1: In the mediaMin output packet log we typically see sections similar to: : : Seq num 98584 timestamp = 3902252372, rtp pyld len = 33 media-R Seq num 98585 timestamp = 3902252692, rtp pyld len = 33 media Seq num 98586 timestamp = 3902253012, rtp pyld len = 33 media-R Seq num 98587 timestamp = 3902253332, rtp pyld len = 33 media Seq num 98588 timestamp = 3902253652, rtp pyld len = 33 media-R Seq num 98589 timestamp = 3902253972, rtp pyld len = 33 media Seq num 98590 timestamp = 3902254292, rtp pyld len = 33 media Seq num 98591 timestamp = 3902254612, rtp pyld len = 33 media-R Seq num 98592 timestamp = 3902254932, rtp pyld len = 33 media Seq num 98593 timestamp = 3902255252, rtp pyld len = 33 media Seq num 98594 timestamp = 3902255572, rtp pyld len = 33 media-R Seq num 98595 timestamp = 3902255892, rtp pyld len = 33 media Seq num 98596 timestamp = 3902256212, rtp pyld len = 33 media Seq num 98597 timestamp = 3902256532, rtp pyld len = 33 media Seq num 98598 timestamp = 3902256852, rtp pyld len = 33 media-R Seq num 98599 timestamp = 3902257172, rtp pyld len = 33 media Seq num 98600 timestamp = 3902257492, rtp pyld len = 33 media-R Seq num 98601 timestamp = 3902257812, rtp pyld len = 33 media Seq num 98602 timestamp = 3902258132, rtp pyld len = 33 media Seq num 98603 timestamp = 3902258452, rtp pyld len = 33 media Seq num 98604 timestamp = 3902258772, rtp pyld len = 33 media-R Seq num 98605 timestamp = 3902259092, rtp pyld len = 33 media : : where media-R packets are timestamp gap repairs (i.e. frame loss concealment). The behavior tends to be bursty, but once it gets going it goes for a while and seems relatively consistent. Is this expected behavior for rtpengine ? If so is rptengine in turn dealing with some type of "slow packet rate" issue from a remote sender ? Thanks, Jeff

1 0

Kamailio v5.7.5 Released
by Daniel-Constantin Mierla 08 May '24

08 May '24

Hello, Kamailio SIP Server v5.7.5 stable release is out. This is a maintenance release of the latest stable branch, 5.7, that includes fixes since the release of v5.7.4. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.7.x. Deployments running previous v5.7.x versions are strongly recommended to be upgraded to v5.7.5. For more details about version 5.7.5 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/05/kamailio-v5-7-5-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note that 5.7.x is currently the previous stable release series, the latest is 5.8.x and v5.8.1 was released on April 3, 2024. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 0

Releasing v5.7.5
by Daniel-Constantin Mierla 08 May '24

08 May '24

Hello, I am considering to release Kamailio v5.7.5 (out of branch 5.7) on Wednesday, May 8, 2024. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 1

sql_query fails from event_route
by Juha Heinanen 07 May '24

07 May '24

I noticed that sql_query does not (anymore) work when called from event route: event_route [tcp:closed] { # Handle TCP connection close # Delete possible leftover registration if ($conid) sql_query("sip_proxy_cache", "DELETE FROM location WHERE server_id=0 AND connection_id=$conid", "result"); } The query produces errors: 2024-05-07T11:30:14.785915+03:00 siika /usr/bin/sip-proxy[2848078]: CRITICAL: sqlops [sqlops.c:213]: sql_check_connection(): no database handle with reconnect disabled 2024-05-07T11:30:14.785978+03:00 siika /usr/bin/sip-proxy[2848078]: ERROR: sqlops [sqlops.c:232]: sql_query(): invalid connection to database 2024-05-07T11:30:14.786013+03:00 siika /usr/bin/sip-proxy[2848078]: ERROR: sqlops [sqlops.c:237]: sql_query(): with query [DELETE FROM location WHERE server_id=0 AND connection_id=7] I added the last debug message in order to find out where the error comes from. The same query works fine when executed from regular route. Any ideas why the query fails from event_route? -- Juha

2 3

How to use environment variable
by Pavan Kumar 07 May '24

07 May '24

Hi everyone, I am trying to assign environment variable as follows listen=udp:0.0.0.0:5060 advertise $env(MY_IP):5060 Looks like using the environment variable as above is an invalid configuration. Is there a way to use IP from env var to advertise. Even better, is there a way to use result in a stun query as an advertised address? Thank you, Pavan Kumar

2 3

Retransmission behaviour after ACK
by Sergio Charrua 06 May '24

06 May '24

Hi all! I'm testing some redirect scenarios with Kamailio 5.7.4 and using Astterisk 13 and SIPp for unit testing. The kamailio's script is simple and uses TM, HTTP_ASYNC_MODULE, JSONRPC, JANSSON, RTJSON and XLog modules. Request route is simple: request_route { route(HANDLE_OPTIONS); route(HANDLE_RETRANS); route(HANDLE_CANCEL); route(REQINIT); route(HANDLE_INVITE); } where HANDLE_RETRANS is: route[HANDLE_RETRANS]{ # handle retransmissions xlog("L_INFO","HANDLE_RETRANS - message type is $mt for method $rm"); if (!is_method("ACK")) { if(t_precheck_trans()) { xlog("L_INFO","HANDLE_RETRANS - Precheck Trans"); t_check_trans(); xlog("L_INFO", "HANDLE_RETRANS - Exiting after Retransmission check - method $rm"); exit; } xlog("L_INFO","HANDLE_RETRANS - Check Trans"); t_check_trans(); } } and HANDLE INVITE is: route[HANDLE_INVITE]{ if (is_method("INVITE")) { xlog("L_INFO","MAIN - calling TO_CARRIER"); setflag(FLT_ACC); route(TO_CARRIER); exit; } } The HTTP requests call a REST API returning a JSON object, that is iterated and produces a CONTACT header containing multiple choices using the append_branch() command. The RELAY route is : route[RELAY] { # Sends a 300 Multiple Choices back to the proxy that requested the routing lookup xlog("L_INFO","REPLY_302 - send reply"); send_reply("300", "Multiple Choices"); xlog("L_INFO","REPLY_302 - exit"); exit; } Which returns indeed a Contact header with multiple contact choices. When using Asterisk to originate a call to Kamailio, the SIP call flow is as follows and terminates (SNGrep and TCPDump do not detect anything else): SRC DST INVITE (SDP) ──────────────────────────> 100 trying -- your call is <────────────────────────── 300 Multiple Choices <────────────────────────── ACK ──────────────────────────> However when using SIPp I get some retransmissions after the final ACK. SRC DST INVITE (SDP) ──────────────────────────> 100 trying -- your call is <────────────────────────── 300 Multiple Choices <────────────────────────── ACK ──────────────────────────> 300 Multiple Choices <<<──────────────────────── 300 Multiple Choices <<<──────────────────────── 300 Multiple Choices <<<──────────────────────── SIPp scenario is : <?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd"> <scenario name="Basic Sipstone UAC">   <send retrans="500" start_rtd="true" rrs="true"> <![CDATA[ INVITE sip:[call_number]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] Max-Forwards: 70 From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number] To: <sip:[call_number]@[remote_ip]:[remote_port]> Contact: sip:sipp@[local_ip]:[local_port];transport=[transport]> Call-ID: [call_id] CSeq: 102 INVITE User-Agent: SIPp Supported: replaces, timer Content-Type: application/sdp Subject: Performance Test Session-Expires: 3600;refresher=uas Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=SIPp c=IN IP[media_ip_type] [media_ip] t=0 0 m=audio [media_port] RTP/AVP 0 a=rtpmap:0 PCMU/8000 ]]> </send> <recv response="100" rss="true"> </recv> <recv response="300" rtd="true" rss="true"> <action> <ereg regexp="\<([a-zA-Z0-9@.=;]+)[^>]+" search_in="hdr" header="Contact: " assign_to="1"/> <log message="INFO: 300 Contact Header: [$1]" /> <exec command="echo [$1] >> ./300_ContactHeader.txt" /> </action> <action> <ereg regexp="sip:.*@([0-9A-Za-z\.]+):([0-9]+);transport=([A-Z]+)" search_in="hdr" header="Contact:" check_it="true" assign_to="dummy,host,port,transport" /> </action> </recv>   <send rrs="true"> <![CDATA[ ACK sip:[call_number]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] Max-Forwards: 70 From: <sip:sipp@[local_ip]:[local_port]>;tag=[call_number] To: <sip:[call_number]@[remote_ip]:[remote_port]>[peer_tag_param] Contact: <sip:sipp@[local_ip]:[local_port]> Call-ID: [call_id] CSeq: 102 ACK User-Agent: SIPp Content-Length: 0 ]]> </send>   </scenario> The question is : is there anything wrong with my kamailio script? or with SIPp scenario? why does kamailio have different behaviours between Asterisk and SIPp? Any option to disable retransmissions on Kamailio ? (using UDP) Thanks in advance, *Sérgio Charrua*

2 3

topos with mask_callid
by Marrold 05 May '24

05 May '24

Hi all, I am using Kamailio 5.8.1 on a Debian 12 machine with topos and topoh hiding the topology. I have observed with modparam("topos", "mask_callid", 1), in-dialog requests are relayed with the incorrect caller-id. In the scenario, A calls B. The call is set up and the call-id for the B-Leg is successfully masked. Then, when A sends a BYE, the BYE is incorrectly relayed to B with the call-id of the A leg, and B replies with a 481 If I disable masking the call-id, or disable topos/topoh entirely, everything looks and works as it should. I have tested with 5.7.4 and the behaviour is the same. Does anyone have any ideas for where I should start looking? Thanks Matthew

1 0

Re: IMS SCSCF redundancy
by Oleg Belousov 03 May '24

03 May '24

Hi. I have a question re: IMS SCSCF redundancy.:There are several SCSCF instances, registration happens via SCSCF1, it goes down after that. UE make MO call.. PCSCF resolves the common SCSCF fqdn and sends an invite to available SCSCF2 which does not have UE profile as well as AS address (which is received during registration SAR/SAA in iFC). There are 2 options here, either to release the call (as per 3gpp spec upon specific release code, eg 408/503 UE should perform re-registration - not happen in our case unfortunately), or to send request for iFC (suspend the call, fire SAR, receive SAA, check available services, etc) and upon success continue the call (to AS first)/ Lke IMS Registrar SCSCF save function, but for the invite. To continue the call without interaction with AS is not a good idea.. Any advice or workaround on how to achieve desired SCSCF redundancy ? -- obelousov.tel

1 0

http_async_client and sl_send_reply()
by Masud Muborakshohi 02 May '24

02 May '24

Hello! I have a problem when sending a response sl_send_reply("415", "The other party does not support video calling!"); in route[HTTP_REPLY]. When I make a request using http_async_query() to send a push notification to a mobile device, I need to process the response from the push server and respond with a SIP message to the client. Before sending the request I do $http_req(suspend) = 0; What am I doing wrong? Why are $fU and $ru equal to null ? #processing HTTP responses route[HTTP_REPLY] { if ($http_ok && $http_rs == 200) { xlog("L_INFO", "route[HTTP_REPLY]: status $http_rs\n"); xlog("L_INFO", "route[HTTP_REPLY]: body $http_rb\n"); } else if($http_rs == 415 || $http_rs == "415") { xlog("L_ERR", "route[HTTP_REPLY]: caller: $fU phone: $ru error: $http_err)\n"); send_reply("415", "The other party does not support video calling!"); } else { xlog("L_ALERT", "route[HTTP_REPLY]: error $http_err)\n"); } #exit; }

3 5

No ACK from Kamailio
by Arseniy Moskvich 02 May '24

02 May '24

Hi everyone. We are new to Kamailio and using Kamailio 5.8 with FreeSwitch. We have the issue with the incoming call from our provider. They are dropping after 30 seconds. The client doesn't receive ACK from Kamailio. But FreeSwitch sends ACK to Kamailio. The outbound call is working fine. At the moment flow is our provider -> FreeSwitch -> Kamailio -> our web client. Could someone give us some directions please? Please find attached Kemi.py. Thank you so much in advance

2 2

Kamailio works but voice is not present during the calls!
by christian.marinelli＠hotmail.it 02 May '24

02 May '24

Hi all, i have a problem with my Kamailio SIP Server. I set up a Kamailio SIP server in a virtual machine on a private network and i connect to this with a WireGuard VPN. The problem is that i can connect to the SIP server throught different clients and users and i can call the other users, the devices ring and i can answer to the calls but unfortunatly there is no audio during the call :( I can't understand why, it seems all si OK, a year ago i set up another SIP server with the same configuration and all works correctly with it. Can anyone help me to understand why? I can copy and paste here all the config files if you need it. Thank you so much in advance Christian

8 29

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