sr-users October 2023

sr-users@lists.kamailio.org

47 participants
42 discussions

How to filter out xhttp request in event_route[siptrace:msg]

by Benoit Panizzon

Hi Further trying to eliminate every possible cause of UDP drops... We use Homer as HEP server in conjunction with the siptrace module. So to prevent DQM traffic to be sent to Homer I added: event_route[siptrace:msg] { if(is_method("KDMQ")) { drop(); } } And had a closer look to the traffic sent to homer. I noticed, also xhttp:requests, as far as I see the replies, not the requests, get mirrored to homer. xhttp is not a sip method I could match I guess. The event_route[siptrace:msg] docs state only method can be filtered. So I can not filter requests on the separate http port I listen on for xhttp json-rpc requests. How can I prevent xhttp traffic to be mirrored to homer by the siptrace module? 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 ______________________________________________________

6 months, 2 weeks

STIR/SHAKEN with Kamailio

by Olle E. Johansson

Hello Kamailians! As STIR/SHAKEN seems to cross the ocean and arrive on the European shores, I’m curious on how you’ve implemented it with Kamailio. I asked on our Matrix chat and got two responses that seems to not use the Kamailio STIR/SHAKEN support but rather a 3rd party service that they’ve integrated using restful APIs. Please help me and share how you’re implementing STIR/SHAKEN with Kamailio. I’m not shaken, nor stirred. yet. Regards, /O

6 months, 2 weeks

How to configure ndb_redis sentinel

by mayamatakeshi

Hi, regarding ndb_redis, I have 3 sentinels like this: "192.168.88.155:26379" "192.168.88.156:26379" "192.168.88.157:26379" So was thinking that I would configure ndb_redis this way: modparam("ndb_redis", "server", "name=srvZ;sentinel_group=myredis;sentinel_master=1;sentinel= 192.168.88.155:26379") modparam("ndb_redis", "server", "name=srvZ;sentinel_group=myredis;sentinel_master=0;sentinel= 192.168.88.156:26379") modparam("ndb_redis", "server", "name=srvZ;sentinel_group=myredis;sentinel_master=0;sentinel= 192.168.88.157:26379") But the samples in the documentation show an extra sentinel key in the config lines: # sentinel (for a redis slave) modparam("ndb_redis", "server", "name=srvZ;sentinel_group=group_name;sentinel_master=0;sentinel=1.2.3.4:26379 ;sentinel=1.2.3.5:26379") # sentinel (for a redis master) modparam("ndb_redis", "server", "name=srvZ;sentinel_group=group_name;sentinel_master=1;sentinel=1.2.3.4:26379 ;sentinel=1.2.3.5:26379") So I am not sure if my configuration is correct. Can someone clarify?

6 months, 3 weeks

lookup() and database

by Jawaid Bazyar

Hi, I have an application as follows: Multiple Kamailio Nodes Shared Postgres database Auth information will be stored in DB and accessed with Authdb. Here is the behavior I seek: I would like all the Kamailo nodes to cooperate in updating the database, each updating data as appropriate: When a registration comes in to any Node, and it authenticated, create or update a database record in location table. When a registration expires on that Node, delete database record in location table. When INVITE comes in on any Node, ignore local cache and always lookup() record from database. #1 and #2 are used so that a third SIP system can query the database and look up the Node a NAT’d subscriber is ‘connected’ to. #3 is to let NON-NAT’d subscribers be contacted from any Node. Now I have something close to this now – using following settings: modparam("usrloc", "db_mode", 1) # immediately write registrations to the database. modparam("usrloc", "db_timer_clean", 0) # do not expire DB records separately modparam("usrloc", "db_check_update", 1) This takes care of #1 and #2 above. However, #3 seems to still be relying purely on local cache and does not hit the database. Is there a way to do #3? I can set db_mode to 3 but that would seem to have other effects besides merely making lookup() read from database. Thanks in advance, Jawaid

6 months, 3 weeks

TOPOS + Forcing the send socket

by Marrold

Hi all, I am using Kamailio (5.7.2, Debian 11, Python KEMI) in a mutli-homed environment with the topos module to hide the topology. I have noticed that when I use `set_send_socket` or `$fsn` to force the socket it's sent from, it breaks topos and the ACK is not proxied to the other leg of the call. I am calling record_route() *after* forcing the socket. If I disable TOPOS the ACK works as expected, and the signalling looks correct. Or, with TOPOS enabled, if I comment out set_send_socket TOPOS works as expected. If I disable enable_double_rr, TOPOS works as expected and the ACK is forwarded, but without the double RR subsequent in-dialog requests use the wrong socket. One other observation, when I change the send socket I also see these warnings in the logs: 24(32) WARNING: PY3 {ACK}: dialog [dlg_handlers.c:1348]: dlg_onroute(): tight matching failed for ACK with callid='!!:MByLMlFAM.NfWxFAM.cAMxyfWjyLz.yAO.y6MxF1MxVZWG4ZMy**'/55, ftag='2023101714101800015'/19, ttag='2f55349a-2c59-4e37-bf58-fd84fb69ece9'/36 and direction=0 24(32) WARNING: PY3 {ACK}: dialog [dlg_handlers.c:1355]: dlg_onroute(): dialog identification elements are callid='2023101714101800015@2900-0601-0284-80'/37, caller tag='2023101714101800015'/19, callee tag='2f55349a-2c59-4e37-bf58-fd84fb69ece9'/36 Is anyone using TOPOS + forcing the socket, or could someone advise where I am going wrong? Thanks in advance Matthew

6 months, 3 weeks

Kamailio SIP proxy role

by Ali Taher

Hello, I'm using Kamailio as SIP proxy between Session Manager and SBC So here is the scenario : Session Manager sends a call (Invite) to Kamailio, kamailio needs to relay it to SBC after doing some modification. (-----Session Manager-----) ----------- > (----Kamailio----) ----------- > (-----SBC-----) My routing logic is similar to the below: route{ if ((method==OPTIONS) && (! uri=~"sip:.*[@]+.*")) { options_reply(); } if (is_method("INVITE")) { sql_query("cb", "select number from pool1 order by random() limit 1", "ra"); $var(rand)=$dbr(ra=>[0,0]); uac_replace_from($var(rand),"sip:$var(rand)@192.168.1.1"); xlog("L_INFO","Random: $var(rand)"); $var(ip) = $(ct{s.select,1,(a)}{s.select,0,;}{s.replace,>,}); remove_hf("Contact"); insert_hf("Contact: $var(rand) <sip:$var(rand)@$var(ip)>\r\n"); remove_hf("P-Asserted-Identity"); remove_hf("Route"); insert_hf("Route: <sip: 192.168.1.10>\r\n","Route"); record_route(); t_relay(); $td = "192.168.1.10"; $rd = "192.168.1.10"; t_relay(); } } Where: 192.168.1.1 is Kamailio IP 192.168.1.10 is SBC IP My question here is that Kamailio handle all in dialog messages between Session Manager and SBC? For example, after forwarding the invite packet to SBC, SBC sends 183 Session Progress to Kamailio, which in its turn relay it to the session manager, knowing that this is not mentioned in my routing logic, so how Kamailio knew that it must relay it to session manager in this case? Also I need to add to the 183 Session Progress sent to session manager by Kamailio record route and Via headers , how to do it ? Regards,

6 months, 3 weeks

Dispatcher HA:won't route headers be wrong if in-dialog request is sent to backup?

by Richard Chan

Hi sr-users, In the case of dispatcher-based HA such as the Wazo example ( https://wazo-platform.org/blog/kamailio-ha-dispatcher-and-dmq) won't the Route: headers be wrong for in-dialog requests if the request is sent to the backup proxy? front proxy -> proxy A, B, C, D,... -> network Suppose the INVITE goes through proxy A, then proxy A goes down. Based on the HA logic, an in-dialog request will go to a backup, say, proxy B. However, won't the Route headers be wrong (since they are placed by A) in the original transaction? How do you handle Route:, Record-Route: headers at the HA tier? 1. In the front proxy & HA tier don't use route headers at all ?? 2. Manually check and remove route headers (i.e proxy B treats route headers for proxy A as "myself") ?? I am curious as to what is your practice. Regards Shih-Ping

6 months, 3 weeks

Restarts Kamailio after general protection fault

by Daian Conrad

Hi all, I have started using kamailio as a proxy register for asterisk's and works fine, but in some moment kamailio show WARNINGs, ALERTS and CRITICAL logs and restart. I don't no how debug correctly, but the crash show this log lines *Oct 16 21:34:26 proxy /usr/sbin/kamailio[2148]: CRITICAL: <core> [core/mem/q_malloc.c:123]: qm_debug_check_frag(): BUG: qm: fragm. 0x7f14a27a5ab8 (address 0x7f14a27a5af0) beginning overwritten (7f14a27a8e58)! Memory allocator was called from uac: uac_send.c:860. Fragment marked by UH��AWAVAUATSH��:139726602009368. Exec from core/mem/q_malloc.c:511.* ------------------------------------------------------------- *Oct 16 21:39:46 proxy /usr/sbin/kamailio[2556]: CRITICAL: <core> [core/mem/q_malloc.c:123]: qm_debug_check_frag(): BUG: qm: fragm. 0x7ff7ad456330 (address 0x7ff7ad456368) beginning overwritten (0)! Memory allocator was called from uac: uac_send.c:860. Fragment marked by (null):0. Exec from core/mem/q_malloc.c:511.Oct 16 21:39:46 proxy /usr/sbin/kamailio[2578]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 19Oct 16 21:39:46 proxy /usr/sbin/kamailio[2539]: ALERT: <core> [main.c:774]: handle_sigs(): child process 2556 exited by a signal 6Oct 16 21:39:46 proxy /usr/sbin/kamailio[2539]: ALERT: <core> [main.c:777]: handle_sigs(): core was not generatedOct 16 21:39:46 proxy systemd[1]: kamailio.service: Main process exited, code=exited, status=1/FAILUREOct 16 21:39:46 proxy systemd[1]: kamailio.service: Failed with result 'exit-code'.Oct 16 21:39:46 proxy systemd[1]: kamailio.service: Service RestartSec=100ms expired, scheduling restart.Oct 16 21:39:46 proxy systemd[1]: kamailio.service: Scheduled restart job, restart counter is at 9.Oct 16 21:39:46 proxy systemd[1]: Stopped Kamailio - the Open Source SIP Server.* any advice is apreciated -- ================ **Daian Conrad** E-mail: daian.conrad(a)gmail.com OpenS Team (DaCoD) Linux user: #248912

6 months, 3 weeks

ERROR: run_top_route

by satyaprakash ch

Hi, We are having an error in the Kamailio logs which we need to resolve this issue, ERROR is ::* /usr/local/sbin/kamailio[10149]: ERROR: tm [t_reply.c:1081]: run_failure_handlers(): error running run_top_route for failure handler.* We are getting this error at the time of the 3xx response, Can anyone help me on this? Thank you.

6 months, 3 weeks

Wrong RTP endpoint announced by Kamailio when forwarding calls, but looks correct from RTPEngine

by Leonardo Arena

Hi all, I have the following setup: Phone <-> private net using TCP transport <-> (172.30.0.156) Kamailio v5.6.3 + RTPEngine v9.3.1 (22.33.177.156) <-> public net using UDP transport <-> Asterisk (22.33.178.77) Phone is registered on Kamailio and forward calls on no answer or when ext is unregistered, to Asterisk's custom IVR (nodejs app + CouchDB), using failure_route() [1]. The problem is that when the none answer the phone and the call is redirected to Asterisk, Kamailio announce its private IP address in the SDP connection endpoint, hence there is no audio [2]. From RTPEngine debug logs, it seems that RTPEngine replies with the correct SDP connection endpoint IP [3]. See also RTPEngine interfaces definition [4]. What am I missing? Why Kamailio is offering its private IP address in the SDP connection, instead of its public address when forwarding the call? If comment "route(SDP_MANAGE_IN)" in route[IN], Kamailio uses the correct IP endpoint address in the SDP offer when forwarding the call to Asterisk. But of course this breaks audio when someone answer the phone. It seems that Kamailio is storing the initial SDP offer settings and does not update it upon RTPEngine's request, even though I have added "replace-session-connection" in route[SDP_OFFER_IVR]. Thanks for any help! leo [1] route[IN] { if (!lookup('location')) { t_on_reply("SDP_MANAGE"); route(TOIVR); } else { xlog("L_INFO", "Route(IN) Req $mi $rm From <$fu> To <$tu> RURI <$ru>\n"); record_route(); t_on_reply("SDP_MANAGE"); route(SDP_MANAGE_IN); t_on_failure("TOIVR"); t_relay(); exit; } } route[SDP_MANAGE] { if (has_body("application/sdp")) { rtpengine_manage(); } } route[SDP_MANAGE_IN] { if (has_body("application/sdp")) { rtpengine_manage("direction=public direction=private via-branch=1"); } } ... route[SDP_OFFER_IVR] { if (has_body("application/sdp")) { rtpengine_offer("direction=public direction=public replace-session-connection via-branch=1"); } } failure_route[TOIVR] { if (t_is_canceled()) { exit; } append_branch(); route(SDP_OFFER_IVR); # using t_relay_to so that UDP is used with the correct IP address if (!t_relay_to("xh.voip.net")) { xlog("L_INFO", "No answer - Failure Route(TOIVR) Req $mi $rm From <$fu> To <$tu> RURI <$ru> failed\n"); t_reply("500", "Unable to route"); } else { xlog("L_INFO", "No answer - Failure Route(TOIVR) Req $mi $rm From <$fu> To <$tu> RURI <$ru>\n"); } } [2] U 2023/10/11 09:05:43.797393 22.33.177.156:5060 -> 22.33.176.8:5060 #3 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 22.33.176.8;branch=z9hG4bK2cb2.e559c4a326678f69ae9b4656be7bc36c.0. Via: SIP/2.0/UDP 22.33.176.20;branch=z9hG4bK2cb2.2920561c845ee5279051359f40cb7c0f.0. Via: SIP/2.0/UDP 44.55.73.157:5060;branch=z9hG4bK0cBdccef20c0d8bfefb. Record-Route: <sip:172.30.0.156;transport=TCP;r2=on;lr=on;ftag=gK0c1bb12e>. Record-Route: <sip:22.33.177.156;r2=on;lr=on;ftag=gK0c1bb12e>. Record-Route: <sip:22.33.176.8;lr=on;ftag=gK0c1bb12e;did=7ffe.26c8>. Record-Route: <sip:22.33.176.20;lr=on;ftag=gK0c1bb12e>. From: "Unavailable" <sip:013335553222@44.55.73.157>;tag=gK0c1bb12e. To: <sip:8555000033@22.33.176.20>;tag=660096088. Call-ID: 474758100_116775912(a)44.55.73.157. CSeq: 704680 INVITE. Contact: <sip:+18555000033@172.30.96.2:11815;transport=TCP>. Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE. User-Agent: Yealink SIP-T31P 124.86.0.40. Allow-Events: talk,hold,conference,refer,check-sync. Content-Length: 0. . U 2023/10/11 09:06:03.751585 22.33.177.156:5060 -> 22.33.178.77:5060 #4 INVITE sip:+18555000033@172.30.96.2:11815;transport=TCP SIP/2.0. Record-Route: <sip:22.33.177.156;lr=on;ftag=gK0c1bb12e>. Record-Route: <sip:22.33.176.8;lr=on;ftag=gK0c1bb12e;did=7ffe.26c8>. Record-Route: <sip:22.33.176.20;lr=on;ftag=gK0c1bb12e>. Via: SIP/2.0/UDP 22.33.177.156;branch=z9hG4bK2cb2.afb825e30e294872c72c5e21ef26f3f3.1. Via: SIP/2.0/UDP 22.33.176.8;branch=z9hG4bK2cb2.e559c4a326678f69ae9b4656be7bc36c.0. Via: SIP/2.0/UDP 22.33.176.20;branch=z9hG4bK2cb2.2920561c845ee5279051359f40cb7c0f.0. Via: SIP/2.0/UDP 44.55.73.157:5060;branch=z9hG4bK0cBdccef20c0d8bfefb. f: "Unavailable" <sip:013335553222@44.55.73.157>;tag=gK0c1bb12e. t: <sip:8555000033@22.33.176.20>. i: 474758100_116775912(a)44.55.73.157. CSeq: 704680 INVITE. Max-Forwards: 32. Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH. Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed. m: "Unavailable" <sip:013335553222@44.55.73.157:5060>. Remote-Party-ID: "Unavailable" <sip:013335553222@44.55.73.157:5060>;privacy=off. k: timer,100rel,precondition. Session-Expires: 1800. Min-SE: 90. l: 604. Content-Disposition: session; handling=required. c: application/sdp. . v=0. o=Sonus_UAC 525653 764118 IN IP4 44.55.73.157. s=SIP Media Capabilities. c=IN IP4 172.30.0.156. t=0 0. m=audio 30030 RTP/AVP 0 18 101. a=maxptime:20. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=n [3] Oct 11 09:06:03 voipsipr7 daemon.info rtpengine[35061]: INFO: [474758100_116775912(a)44.55.73.157]: [control] Received command 'offer' from 127.0.0.1:39942 Oct 11 09:06:03 voipsipr7 daemon.debug rtpengine[35061]: DEBUG: [474758100_116775912(a)44.55.73.157]: [control] Dump for 'offer' from 127.0.0.1:39942: { "supports": [ "load limit" ], "sdp": "v=0^M o=Sonus_UAC 5 25653 764118 IN IP4 44.55.73.157^M s=SIP Media Capabilities^M c=IN IP4 44.55.78.60^M t=0 0^M m=audio 61610 RTP/AVP 0 18 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:18 G729/8000^M a=fmtp:18 annexb=no^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-15^M a=sendrecv^M a=maxptime:20^M ", "direction": [ "public", "public" ], "replace": [ "session-connection" ], "call-id": "474758100_116775912(a)44.55.73.157 ... Oct 11 09:06:03 voipsipr7 daemon.debug rtpengine[35061]: DEBUG: [474758100_116775912(a)44.55.73.157]: ... ", "via-branch": "z9hG4bK2cb2.e559c4a326678f69ae9b4656be7bc36c.0", "received-from": [ "IP4", "22.33.176 .8" ], "from-tag": "gK0c1bb12e", "command": "offer" } ... Oct 11 09:06:03 voipsipr7 daemon.info rtpengine[35061]: INFO: [474758100_116775912(a)44.55.73.157]: [control] Replying to 'offer' from 127.0.0.1:39942 (elapsed time 0.000638 sec) Oct 11 09:06:03 voipsipr7 daemon.debug rtpengine[35061]: DEBUG: [474758100_116775912(a)44.55.73.157]: [control] Response dump for 'offer' to 127.0.0.1:39942: { "sdp": "v=0^M o=Sonus_UAC 525653 764118 IN IP4 207 .223.73.157^M s=SIP Media Capabilities^M c=IN IP4 22.33.177.156^M t=0 0^M m=audio 30068 RTP/AVP 0 18 101^M a=maxptime:20^M a=rtpmap:0 PCMU/8000^M a=rtpmap:18 G729/8000^M a=fmtp:18 annexb=no^M a=rtpmap:101 teleph one-event/8000^M a=fmtp:101 0-15^M a=sendrecv^M a=rtcp:30069^M ", "result": "ok" } Oct 11 09:06:03 voipsipr7 daemon.info /usr/sbin/kamailio[35261]: INFO: <script>: No answer - Failure Route(TOIVR) Req 4 INVITE From <sip:013335553222@44.55.73.157> To <sip:8555000033@22.33.176.20> RURI <sip: +18555000033@172.30.96.2:11815;transport=TCP> Oct 11 09:06:03 voipsipr7 daemon.info rtpengine[35061]: INFO: [474758100_116775912(a)44.55.73.157]: [control] Received command 'answer' from 127.0.0.1:53245 Oct 11 09:06:03 voipsipr7 daemon.debug rtpengine[35061]: DEBUG: [474758100_116775912(a)44.55.73.157]: [control] Dump for 'answer' from 127.0.0.1:53245: { "supports": [ "load limit" ], "sdp": "v=0^M o=- 525653 7 64120 IN IP4 22.33.178.77^M s=Asterisk^M c=IN IP4 22.33.178.77^M t=0 0^M m=audio 11264 RTP/AVP 0 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-16^M a=ptime:20^M a=maxptime:150^M a=sendrecv^M m=audio 0 RTP/AVP 0 18 101^M ", "call-id": "474758100_116775912(a)44.55.73.157", "received-from": [ "IP4", "22.33.178.77" ], "from-tag": "gK0c1bb12e", "to-tag": "a21c53b6-ca33-43b ... Oct 11 09:06:03 voipsipr7 daemon.debug rtpengine[35061]: DEBUG: [474758100_116775912(a)44.55.73.157]: ... 4-a585-af9b568a5ecb", "command": "answer" } ... Oct 11 09:06:03 voipsipr7 daemon.info rtpengine[35061]: INFO: [474758100_116775912(a)44.55.73.157]: [control] Replying to 'answer' from 127.0.0.1:53245 (elapsed time 0.000915 sec) Oct 11 09:06:03 voipsipr7 daemon.debug rtpengine[35061]: DEBUG: [474758100_116775912(a)44.55.73.157]: [control] Response dump for 'answer' to 127.0.0.1:53245: { "sdp": "v=0^M o=- 525653 764120 IN IP4 22.33.178 .77^M s=Asterisk^M c=IN IP4 22.33.177.156^M t=0 0^M m=audio 30056 RTP/AVP 0 101^M a=maxptime:150^M a=rtpmap:0 PCMU/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-16^M a=sendrecv^M a=rtcp:30057^M a=ptime :20^M m=audio 0 RTP/AVP 0 18 101^M ", "result": "ok" } Oct 11 09:06:03 voipsipr7 daemon.debug rtpengine[35061]: DEBUG: [474758100_116775912(a)44.55.73.157]: [core] Forward to sink endpoint: 22.33.178.77:11264 (RTP seq 0 TS 0) [4] Interface definition in RTPEngine: interface = public/22.33.177.156;public/22:33:201:105::156;private/172.30.0.156

6 months, 3 weeks

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sr-users October 2023