- sr-users - kamailio.org

Problem with Record-Route in Public to Private Network
by Carlos Escalona 03 May '22

03 May '22

Hello! I wish everyone who reads this message a good day. I would appreciate some guidance with the configuration of my Kamailio. Initially, I have the following architecture: Endpoints <----Public IP----> Kamailio (listen PrivateIP advertise PublicIP) <----Private IP----> Asterisk CONFIG: https://gist.github.com/Carlos-Escalona94/3a289af05b05c69ae563ab29c9ec4710 In general, calls initiated from outside the system work perfectly, but I have a problem with calls initiated from Asterisk. According to what I was able to investigate and understand, the problem arises from the fact that Kamailio uses the public IP for the entire routing system that involves the Record-Route and Route Headers. This in turn causes Asterisk to get lost trying to send messages outside of the INITIAL INVITE transaction, for example, the ACK response to the 200 Ok received from the Endpoint since it tries to use Kamailio's public IP to which Asterisk doesn't have access. I tried to force Asterisk to send all messages to Kamailio's private IP regardless of their nature, but it doesn't seem like an appropriate solution. On the other hand, I tried to modify the architecture a bit so that it was something similar to this: Endpoints <---- Public IP----> Kamailio (listen PrivateIP:P1 advertise PublicIP) <---- PrivateIP-----> Kamailio (listen PrivateIP:P2) <---- PrivateIP ----> Asterisk CONFIG: https://gist.github.com/Carlos-Escalona94/4d681bb189c6190941d291965e123889 It seems to me that this would solve the problem, but I have two doubts about this architecture, the first is that I would like to know if there is an easier way to solve the problem that I am not considering, and on the other hand, I have not found a way to identify from which interface is receiving the message to perform the routing properly. Thanks for the attention. -- *Esta mensagem pode conter informação confidencial ou privilegiada, sendo seu sigilo protegido por lei. Se você não for o destinatário ou a pessoa autorizada a receber esta mensagem, não pode usar, copiar ou divulgar as informações nela contidas ou tomar qualquer ação baseada nessas informações. Se você recebeu esta mensagem por engano, por favor, avise imediatamente ao remetente, respondendo o e-mail e em seguida apague-a. Agradecemos sua cooperação. ** * *This message may contain confidential or privileged information and its confidentiality is protected by law. If you are not the addressed or authorized person to receive this message, you must not use, copy, disclose or take any action based on it or any information herein. If you have received this message by mistake, please advise the sender immediately by replying the e-mail and then deleting it. Thank you for your cooperation. *

3 2

Getting "uac.reg_add" to work (documentation unclarities)
by Rick van Rein 29 Apr '22

29 Apr '22

Hello, I am trying to make UAC register with a SIP telco. But when I run... shell$ kamctl rpc uac.reg_add mydomain.com 0123456789 myteldomain.com 0123456789 sip6.telco.nl telcoreseller.nl 999999 secretpw . udp:sip6.telco.nl 360 0 10 'udp:[fd8:9::10]:5090' ...it gives me the cryptic response... ERROR: jsonrpcs [jsonrpcs_mod.c:666]: jsonrpc_scan(): field is not a string - type 3 The module docs only give "..." as example of attributes. And, according to... http://www.kamailio.net/docs/modules/5.4.x/modules/uac.html#uac.r.uac.reg_a… https://github.com/kamailio/kamailio/blob/5.4.4/src/modules/uac/uac_reg.c#1… ...it also seems to be behind on diff_expires, timer_expires, reg_init. What I am missing is the format, and in some cases the purpose of attributes. Could you help me to an example, perhaps based on the command I am trying? Thanks, -Rick P.S. It's been years since I worked with OpenSIPS. It feels like Kamailio is friendlier, and easier to use.

3 4

Dispatcher Usage and change of $rU
by Benoît Panizzon 29 Apr '22

29 Apr '22

Hi Gang I have a observation which I don't understand... Let's assume we have an inbound $rU = 12345 And I want to change $rU outbound to 0012345 Doing: $rU = "0012345"; ds_select_dst("2000"),"6"); route(RELAY) The call is being sent out with unaltered R-URI Username: 12345 Doing: $rU = "0012345"; ds_select_domain("2000", "6"); route(RELAY); The call is being sent out with the intended new R-URI Username: 0012345 Looking at the docs: https://kamailio.org/docs/modules/5.4.x/modules/dispatcher.html#dispatcher.… I can not figure out, why I see this difference. -- 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 ______________________________________________________

3 3

Incorrect Socket and Tcpconn-Id with Mid-Registrar and WebSockets
by Grant Bagdasarian 29 Apr '22

29 Apr '22

Hi, We're attempting to implement a mid-registrar for websocket connections but are running into some issues where the tcpconn-id and socket attributes are not stored correctly in memory (usrloc). REGISTER Flow: UA1 -> P1 (Mid-Registrar) ~> P2 (Core Proxy/Registrar) INVITE Flow: UA2 -> P2 -> P1 -> UA1 UA1: baresip cli connecting to P1 using websockets UA2: softphone Everything is running locally on a dev machine, and we're using Docker to launch P1 and P2. The below code snippet is what we use on P1, and has been simplified. The general idea is as following: 1) UA1 sets up the websocket connection and sends a REGISTER request to P1 2) P1 checks if this request has already been sent to P2 2.a) If not, then relay the request to P2. Upon receiving a 200 OK from P2 we save the registration in usrloc 2.b) If so, then absorb the request (save) and generate a reply 3) UA2 tries to call UA1 via P1 and P2 The problem occurs at 2.a since the REGISTER request is forwarded using UDP and the 200 OK is also received using UDP. When the registration is stored in reply route, the Socket (udp:x.x.x.x:xxxx) and Tcpconn-Id (-1) are not set to what we would prefer (see Output 1) TCP was also tried as the transport between P1 and P2. Upon receiving the reply from P2 the Tcpconn-Id and Socket of the connection between P1 and P2 was saved in the binding. When 2.b occurs the Socket and Tcpconn-Id are set to the correct values (see Output 2) Making a call when the binding state is at 2.a we receive the following error: "TCP/TLS connection (id: 0) for WebSocket could not be found" Making a call when the binding state is at 2.b works perfectly fine since the correct Tcpconn-Id is set. We did find a solution, or rather a workaround to this issue: 1) Always call the save function in request_route 2) If P2 responds with an error (non-200 OK) use the unregister function to remove the binding We also tried the tcpops tcp_set_otcpid and tcp_set_otcpid_flag functions after lookup, but this didn't do anything. KEMI Script def ksr_request_route(self, msg): if kemi.is_REGISTER(): aor = "aor" if result := kemi.htable.sht_is_null("registrations", aor) == 1: kemi.tm.t_on_reply("ksr_register_reply_route") kemi.dispatcher.ds_select_dst(1, 4) kemi.tm.t_relay() return 1 kemi.registrar.save("registrations", 1) return 1 if kemi.is_INVITE(): aor = "aor" if result := kemi.registrar.lookup_uri("registrations", aor): kemi.tm.t_relay() return 1 def ksr_register_reply_route(self, msg): if kemi.tm.t_check_status("200") == 1: kemi.registrar.save("registrations", 1) aor = "aor" kemi.htable.sht_sets("registrations", aor, "") return 1 Output 1: Binding state when request is being forwarded kamcmd ul.dump { Domains: { Domain: { Domain: registrations Size: 1024 AoRs: { Info: { AoR: grant(a)127.0.0.1 HashID: 409622189 Contacts: { Contact: { Address: sip:grant-0x7fe915a79138@172.16.99.12:9;transport=ws Expires: 13 Q: 1.000000 Call-ID: 7cdfbf93f23bb54f CSeq: 34818 User-Agent: n/a Received: sip:192.168.80.1:62184;transport=ws Path: [not set] State: CS_NEW Flags: 1 CFlags: 20 Socket: udp:10.2.0.15:5060 Methods: -1 Ruid: uloc-626a80d3-46-1 Instance: <urn:uuid:c67df38d-d5e0-e834-1d98-a8df2d3f291b> Reg-Id: 0 Server-Id: 0 Tcpconn-Id: -1 Keepalive: 0 Last-Keepalive: 1651146965 KA-Roundtrip: 0 Last-Modified: 1651146965 } } } } Stats: { Records: 1 Max-Slots: 1 } } } } kamcmd core.tcp_list { id: 1 type: WS state: CONN_OK timeout: 2 lifetime: 120 ref_count: 2 src_ip: 192.168.80.1 src_port: 62184 dst_ip: 192.168.80.15 dst_port: 80 } kamcmd ws.dump { connections: { 1: ws:192.168.80.1:62184 -> ws:192.168.80.15:80 (state: OPEN, last used 3s ago, sub-protocol: sip) } info: { wscounter: 1 truncated: no } } Output 2: Binding state when request is being absorbed kamcmd ul.dump { Domains: { Domain: { Domain: registrations Size: 1024 AoRs: { Info: { AoR: grant(a)127.0.0.1 HashID: 409622189 Contacts: { Contact: { Address: sip:grant-0x7fe915a79138@172.16.99.12:9;transport=ws Expires: 9 Q: 1.000000 Call-ID: 7cdfbf93f23bb54f CSeq: 34819 User-Agent: baresip v2.0.2 (x86_64/darwin) Received: sip:192.168.80.1:62184;transport=ws Path: [not set] State: CS_NEW Flags: 1 CFlags: 20 Socket: tcp:192.168.80.15:80 Methods: 5087 Ruid: uloc-626a80d3-46-1 Instance: <urn:uuid:c67df38d-d5e0-e834-1d98-a8df2d3f291b> Reg-Id: 0 Server-Id: 0 Tcpconn-Id: 1 Keepalive: 0 Last-Keepalive: 1651146979 KA-Roundtrip: 0 Last-Modified: 1651146979 } } } } Stats: { Records: 1 Max-Slots: 1 } } } } kamcmd core.tcp_list { id: 1 type: WS state: CONN_OK timeout: 3 lifetime: 120 ref_count: 2 src_ip: 192.168.80.1 src_port: 62184 dst_ip: 192.168.80.15 dst_port: 80 } kamcmd ws.dump { connections: { 1: ws:192.168.80.1:62184 -> ws:192.168.80.15:80 (state: OPEN, last used 1s ago, sub-protocol: sip) } info: { wscounter: 1 truncated: no } }

1 0

TOPOS, 183 (before a 200OK) and SIP INFO
by Giovanni Tommasini 28 Apr '22

28 Apr '22

Hi kamailiers, I ask your help for a strange situation with topos, 183 (before a 200OK) and SIP INFO My topology is kamailio in front of asterisks; so kamailio has 2 interfaces (1Public to talk with customers and carriers and 1 Private to talk with asterisk). In the image: - the first column is the customer - the second is the public interface of kamailio - the third is the private interface of kamaiio - the fourth is asterisk As you can see in the image, the customer send an INVITE and the carrier give back the IVR messages in a 183 (before a 200OK) asking for an IVR selection. The customer (configured to use SIP INFO) types a key and sends in INFO, but as you can see kamailio sends the INFO to himself to private interface instead of sending it to asterisk as if the contact that handled topos contained incorrect information. [image: Schermata 2022-04-26 alle 19.40.32.png] - If the customer is configured with a RFC2833 - or if the IVR messages is not in a 183 but is sent after a 200OK - or if I disabe TOPOS *all works fine* I used kamailio 5.5.3, I also try to upgrade to 5.5.4 but nothing change Maybe could be connected with https://github.com/kamailio/kamailio/issues/2344 https://github.com/kamailio/kamailio/issues/1077 Could be a kamailio bug for this particular situation or am I getting lost in a glass of water? thanks a lot cheers Giovanni Tommasini *evoseed s.r.l.* phone: +39 340 11 28 800 mail: giovanni.tommasini(a)evoseed.io site: evoseed.io address: Via Lucrezio 13, Trieste Italy

3 4

Kamailio functionality clarification - call forwarding no replay (call is ringing but not answered)
by Pavel Veselovsky 28 Apr '22

28 Apr '22

Hallo all, I would need help with Kamailio. I hope, you can give me some hints. I would like use Kamailio for CFNRy feature call forward no replay, i.e. - call is forked to multiple destination - 180 is received, call is ringing, but not answered, no 200 OK is received from destinations - after predefined time (CFNRy timer), the specific branch has to be rerouted to another destination Expected behaviour is described in this flow: [image: image.png] My current behavior: - Kamailio TM module is involved for parallel forking ( https://www.kamailio.org/docs/modules/stable/modules/tm.html#:~:text=1.-,Ov… ) - all branches are CANCEL-ed, it is not possible send CANCEL just to one specific branch [image: image.png] My questions: - is it possible to CANCEL just specific branch of parallel forked calls? - is it possible start timer just for specific branch? - do you think is it possible implement this feature with Kamailio? Thanks a lot in advance for any comments. Pavel.

3 2

t_check_trans failed on ACK
by Linux Vince 28 Apr '22

28 Apr '22

I am using Kamailio as registrar and proxy for underlying asterisk server. I am able to call but ACK on 200 OK respond from client phones are not validating transaction and thus not relaying correctly. Below are my Invites and 200 OK responses. Invite from Asterisk to Kamailio INVITE sip:31313004@rigel.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 65.20.69.72:5060 ;rport;branch=z9hG4bKd6ffff9d46ceb899c245ece204e51c87 From: "R31313005 N31313005" <sip:31313005@65.20.69.72:5060 ;user=phone>;tag=GR52RWG346-34 To: "31313004(a)rigel.com" <sip:31313004@rigel.com:5060> Call-ID: 1558172022-144148450000000029280449(a)65.20.69.72 CSeq: 1 INVITE Contact: <sip:65.20.69.72:5060> max-forwards: 70 Allow: PUBLISH, SUBSCRIBE, MESSAGE, ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE Content-Type: application/sdp Content-Length: 228 v=0 o=Clarent 155866 155867 IN IP4 65.20.69.72 s=Clarent CallManager c=IN IP4 65.20.69.72 t=0 0 m=audio 45168 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Invite from Kamailio to Phone INVITE sip:31313004@65.20.67.118;lhst=180.211.119.50:25759;lm=midreg SIP/2.0 Record-Route: <sip:65.20.67.118;lr;ftag=GR52RWG346-34> Via: SIP/2.0/UDP 65.20.67.118;branch=z9hG4bK6cc3.dad7580d3e28f55cbb749dbb82f0c9eb.0 Via: SIP/2.0/UDP 65.20.69.72:5060 ;received=65.20.69.72;rport=5060;branch=z9hG4bKd6ffff9d46ceb899c245ece204e51c87 From: "R31313005 N31313005" <sip:31313005@65.20.69.72:5060 ;user=phone>;tag=GR52RWG346-34 To: "31313004(a)rigel.com" <sip:31313004@rigel.com:5060> Call-ID: 1558172022-144148450000000029280449(a)65.20.69.72 CSeq: 1 INVITE Contact: <sip:65.20.69.72:5060> max-forwards: 69 Allow: PUBLISH, SUBSCRIBE, MESSAGE, ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE Content-Type: application/sdp Content-Length: 228 v=0 o=Clarent 155866 155867 IN IP4 65.20.69.72 s=Clarent CallManager c=IN IP4 65.20.69.72 t=0 0 m=audio 45168 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 200 OK from Phone to Kamailio SIP/2.0 200 OK Via: SIP/2.0/UDP 65.20.67.118;received=65.20.67.118;branch=z9hG4bK6cc3.dad7580d3e28f55cbb749dbb82f0c9eb.0 Via: SIP/2.0/UDP 65.20.69.72:5060 ;rport=5060;received=65.20.69.72;branch=z9hG4bKd6ffff9d46ceb899c245ece204e51c87 Record-Route: <sip:65.20.67.118;lr;ftag=GR52RWG346-34> Call-ID: 1558172022-144148450000000029280449(a)65.20.69.72 From: "R31313005 N31313005" <sip:31313005@65.20.69.72 ;user=phone>;tag=GR52RWG346-34 To: "31313004(a)rigel.com" <sip:31313004@rigel.com >;tag=4ecc1f21f1544fd783965f27ff6d95d2 CSeq: 1 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Contact: <sip:31313004@180.211.119.50:57935;ob> Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 322 v=0 o=- 3860150297 3860150298 IN IP4 180.211.119.50 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4016 RTP/AVP 0 101 c=IN IP4 180.211.119.50 b=TIAS:64000 a=rtcp:4017 IN IP4 180.211.119.50 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ssrc:1461220511 cname:3b657c275d2a43f6 200 OK from kamailio to asterisk SIP/2.0 200 OK Via: SIP/2.0/UDP 65.20.69.72:5060 ;rport=5060;received=65.20.69.72;branch=z9hG4bKd6ffff9d46ceb899c245ece204e51c87 Record-Route: <sip:65.20.67.118;lr;ftag=GR52RWG346-34> Call-ID: 1558172022-144148450000000029280449(a)65.20.69.72 From: "R31313005 N31313005" <sip:31313005@65.20.69.72 ;user=phone>;tag=GR52RWG346-34 To: "31313004(a)rigel.com" <sip:31313004@rigel.com >;tag=4ecc1f21f1544fd783965f27ff6d95d2 CSeq: 1 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Contact: <sip:31313004@180.211.119.50:57935;ob> Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 322 v=0 o=- 3860150297 3860150298 IN IP4 180.211.119.50 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4016 RTP/AVP 0 101 c=IN IP4 180.211.119.50 b=TIAS:64000 a=rtcp:4017 IN IP4 180.211.119.50 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ssrc:1461220511 cname:3b657c275d2a43f6 ACK from phone to Kamailio ACK sip:31313004@180.211.119.50:57935;ob SIP/2.0 Via: SIP/2.0/UDP 65.20.69.72:5060 ;rport;branch=z9hG4bK111270de2729ade9e048afa0f9fbb8df Route: <sip:65.20.67.118;lr;ftag=GR52RWG346-34> From: "R31313005 N31313005" <sip:31313005@65.20.69.72:5060 ;user=phone>;tag=GR52RWG346-34 To: "31313004(a)rigel.com" <sip:31313004@rigel.com >;tag=4ecc1f21f1544fd783965f27ff6d95d2 Call-ID: 1558172022-144148450000000029280449(a)65.20.69.72 CSeq: 1 ACK Contact: <sip:65.20.69.72:5060> max-forwards: 70 Allow: PUBLISH, SUBSCRIBE, MESSAGE, ACK, NOTIFY, OPTIONS, REFER, INFO, BYE, CANCEL, INVITE This ACK is failing on transaction check and not routing correctly to phone. Any help? Thanks

2 3

Modify session name in SDP
by ZZ Wave 28 Apr '22

28 Apr '22

Hello. Kamailio 5.2.1, Debian 10. I need to change session name (s=) in SDP before relaying it to next hop, and for unknown reason I can't do it with virtually any TEXTOPS function, everything replace*, subst*, *body*... For example this from TEXTOPS documentation: subst_body('/^s=(.*)/s=root/') just before 'final' t_relay(). Either Kamailio just adds "s=root" to SDP leaving first "s=" intact, or adds just "root", or does nothing. Also I've tried msg_apply_changes() before and after these manipulations, still no result. Is it possible to make this in Kamailio without source code modification?

1 0

High performance
by Muborakshohi Masoud 27 Apr '22

27 Apr '22

Hello! I have kamailio 5.5 configured with digest authentication. DB Mysql, 2 rtpengine one on the same server with kamailio, the other is separate. Rtpengins work in bridge mode, internal/external/external-ipV6. Kamailio listen 3 address: internal/external/external-ipV6 work in mixed mode: ipV4-ipV4, ipV4-ipV6, ipV6-ipV6. OS Debian 10, CPU 1 core, Ram 4gb, Disk not ssd. internal address is used for the gateway model of the connection to the PSTN. Before sending a call to PSTN, a request is sent to Billing by Diameter (request over the network, ims_charging ), Billing responds in the region of 10-17 milliseconds. usrloc db_mode = 0; UAS contacts are received according to the PUSH scheme (we send PUSH via http and get a REGISTER). children = 8 for each; When running synthetic tests with sipp, I get 36 cps as a maximum, but there are a lot of retrans. This is a test environment. Product: I plan to get 100 cps; 20k concurrent calls, hardware will be 128GB RAM ddr4-2400; Intel Xeon Silver 4216, Optane SSD. 2 such servers, kamailio/rtpengine are configured exactly according to the scheme of the test environment. Question: Can you please tell me how many children, shm/pv do I need? Is it worth going in the direction of asynchrony, or is the standard one enough for my purpose? I will be very glad for any help. Thank you! -- Отправлено из Mail.ru для Android

2 1

Intermittent error seen when using a python script in kamailio
by Wojciech Kopciewicz 27 Apr '22

27 Apr '22

Hi, Intermittently we are observing the following error when checking if given header is present in SIP INVITE: SystemError: returned a result with an error set The complete stack trace is (some names redacted): 11(28) ERROR: {1 195 INVITE nMHZDN.10nOIDt4ldOwz5cqz0XnMepA9} app_python3 [python_support.c:156]: python_handle_exception(): apy_exec: modify_headers((null)): Unhandled exception in the Python code: TypeError: an integer is required (got type NoneType) The above exception was the direct cause of the following exception: Traceback (most recent call last): File "/etc/kamailio/kamailio.py", line 25, in modify_headers self.modify_header("SomeHeader") File "/etc/kamailio/kamailio.py", line 29, in modify_header present = KSR.hdr.is_present(header_name) SystemError: returned a result with an error set Kamailio version (kamailio -v output): version: kamailio 5.4.6 (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, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, 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 8.3.0 Operation system: Linux kamailio-5c7484c74d-vhhr8 5.4.0-1056-azure #58~18.04.1-Ubuntu SMP Wed Jul 28 23:14:18 UTC 2021 x86_64 GNU/Linux Are you able to help me find the root cause of this issue? Pozdrawiam / Best regards, Wojtek -- *For more information on how and why we collect your personal information, please visit our Privacy Policy <https://www.motorolasolutions.com/en_us/about/privacy-policy.html?elqTrackI…>.*

3 9

LCR issues (request_uri)
by Antoine DEBARRE 26 Apr '22

26 Apr '22

Hi, I'm trying to use LCR with kamailio v5.5.4 gw, rule and rule_target are stored on mysql. for lcr rules, do you know if prefix is mandatory ? I first wanted to start using request_uri instead and leave the prefix column null/empty because prefix seems not to support regexp, I don't want to add the same rule with different prefixes. but I'm facing issues my database is fulfilled with : lcr_gw +----+--------+--------------+---------------+----------+------+--------+------------+-----------+-------+--------+-------------+-------+---------+ | id | lcr_id | gw_name | ip_addr | hostname | port | params | uri_scheme | transport | strip | prefix | tag | flags | defunct | +----+--------+--------------+---------------+----------+------+--------+------------+-----------+-------+--------+-------------+-------+---------+ | 1 | 1 | carrier1 | 10.78.51.3 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | fd=carrier1 | 0 | NULL | | 2 | 1 | carrier2 | 10.78.51.3 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | fd=carrier2 | 0 | NULL | | 3 | 1 | carrier3 | 10.192.50.212 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | NULL | 0 | NULL | | 6 | 1 | carrier4 | 10.192.50.3 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | fd=carrier4 | 0 | NULL | | 10 | 1 | carrier5 | 10.192.50.3 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | fd=carrier5 | 0 | NULL | | 18 | 1 | carrier6 | 10.192.50.214 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | NULL | 0 | NULL | | 20 | 1 | carrier7 | 10.192.50.210 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | NULL | 0 | NULL | | 30 | 1 | carrier8 | 10.192.50.211 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | NULL | 0 | NULL | | 31 | 1 | carrier9 | 10.192.50.216 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | NULL | 0 | NULL | | 32 | 2 | media1 | 10.192.50.232 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | NULL | 0 | NULL | | 33 | 2 | media2 | 10.192.50.233 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | NULL | 0 | NULL | | 34 | 2 | media3 | 10.192.50.234 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | NULL | 0 | NULL | | 35 | 2 | media4 | 10.192.50.235 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | NULL | 0 | NULL | | 36 | 2 | media5 | 10.192.50.240 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | NULL | 0 | NULL | | 37 | 2 | media6 | 10.192.50.241 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | NULL | 0 | NULL | | 38 | 2 | media7 | 10.192.50.242 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | NULL | 0 | NULL | | 39 | 3 | xivo-231 | 10.192.50.231 | NULL | 5060 | NULL | NULL | 1 | NULL | NULL | cps=5;cc=50 | 0 | NULL | +----+--------+--------------+---------------+----------+------+--------+------------+-----------+-------+--------+-------------+-------+---------+ lcr_rule +----+------------------------+-----------------------------------+------------------+--------+----------+-----------+---------+---------+ | id | desc | request_uri | lcr_id | prefix | from_uri | mt_tvalue | stopper | enabled | +----+------------------------+-----------------------------------+------------------+--------+----------+-----------+---------+---------+ | 1 | Default | .+ | 1 | NULL | NULL | NULL | 0 | 1 | | 2 | France GSM | (0|(|00|\+)33)[67]\d{8} | 1 | NULL | NULL | NULL | 1 | 1 | | 3 | France | (0|(|00|\+)33)[1-59]\d{8} | 1 | NULL | NULL | NULL | 1 | 1 | | 4 | International default | (00|\+)[1-9].+ | 1 | NULL | NULL | NULL | 1 | 1 | | 5 | Italy | (00|\+)39.+ | 1 | NULL | NULL | NULL | 1 | 1 | | 14 | to_oldmedias | .+ | 2 | NULL | NULL | NULL | 0 | 1 | +----+------------------------+-----------------------------------+-------------------+--------+----------+-----------+---------+---------+ lcr_rule_target +----+--------+---------+-------+----------+--------+ | id | lcr_id | rule_id | gw_id | priority | weight | +----+--------+---------+-------+----------+--------+ | 25 | 1 | 3 | 3 | 1 | 1 | | 26 | 1 | 3 | 18 | 2 | 1 | | 28 | 1 | 3 | 30 | 3 | 1 | | 29 | 1 | 2 | 30 | 4 | 1 | | 30 | 1 | 2 | 3 | 2 | 1 | | 31 | 1 | 2 | 18 | 3 | 1 | | 32 | 1 | 2 | 20 | 1 | 1 | | 33 | 1 | 1 | 30 | 1 | 1 | | 34 | 1 | 12 | 1 | 1 | 1 | | 35 | 1 | 12 | 10 | 1 | 1 | | 37 | 2 | 14 | 32 | 1 | 1 | | 38 | 2 | 14 | 33 | 1 | 1 | | 39 | 2 | 14 | 34 | 1 | 1 | | 40 | 2 | 14 | 35 | 1 | 1 | | 41 | 2 | 14 | 36 | 1 | 1 | | 42 | 2 | 14 | 37 | 1 | 1 | | 43 | 2 | 14 | 38 | 1 | 1 | | 45 | 1 | 2 | 10 | 3 | 1 | +----+--------+---------+-------+----------+--------+ to test gws, I use kamcmd for example : kamcmd lcr.load_gws 1 s:0600000001 s:02111111111 s:0600000001 arg 1 : prefix or $rU arg 2 : from_uri arg 3 : request_uri result is OK with 5 gateways selected but in real test situation with the following config file : kamailio.cfg : request_route { ... if(!load_gws(1,$rU,$fU)) { sl_send_reply("500", "Cannot load gateways"); exit; } ## - dumping the list of lcr gateways $var(i) = 0; while( $(avp(lcr_gw_uri)[$var(i)]) != $null) { xlog("L_ERR","loaded gw_uri_avp[$var(i)]=’$(avp(lcr_gw_uri)[$var(i)])’\n"); $var(i) = $var(i) + 1; } if(!next_gw()) { sl_send_reply("503", "No gateways found"); exit; } } ..... => we receive a 503 "No gateways found", which indicate that load_gws() is OK, but wasn't able to parse avp lcr_gw_uri (we have no xlog output) Then I noticed that some calls are OK and others are KO (with same pattern for $rU) => lcr rule id 2 So I with debug=3 *call to 0611111111 => load_gws OK and next_gw OK* lcr [lcr_mod.c:3018]: do_from_gw(): request did not come from gw lcr [lcr_mod.c:3018]: do_from_gw(): request did not come from gw lcr [lcr_mod.c:3004]: do_from_gw(): request game from gw lcr [lcr_mod.c:3009]: do_from_gw(): added tag_avp <cps=5;cc=50> lcr [lcr_mod.c:2264]: ki_load_gws_furi(): load_gws(1, 0611111111, 0211111111) lcr [lcr_mod.c:2313]: ki_load_gws_furi(): from uri <0211111111> did not match to from regex <AMD> lcr [lcr_mod.c:2349]: ki_load_gws_furi(): request uri < sip:0611111111@10.192.50.227:5060> did not match to request regex <(00|\+)39.+> lcr [lcr_mod.c:2349]: ki_load_gws_furi(): request uri < sip:0611111111@10.192.50.227:5060> did not match to request regex <(00|\+)[1-9].+> lcr [lcr_mod.c:2349]: ki_load_gws_furi(): request uri < sip:0611111111@10.192.50.227:5060> did not match to request regex <(0|(|00|\+)33)[1-59]\d{8}> lcr [lcr_mod.c:2369]: ki_load_gws_furi(): added matched_gws[0]=[4, 0, 3, 2329862] lcr [lcr_mod.c:2369]: ki_load_gws_furi(): added matched_gws[1]=[5, 0, 1, 5553909] lcr [lcr_mod.c:2369]: ki_load_gws_furi(): added matched_gws[2]=[8, 0, 3, 7256660] lcr [lcr_mod.c:2369]: ki_load_gws_furi(): added matched_gws[3]=[7, 0, 2, 3129551] lcr [lcr_mod.c:2369]: ki_load_gws_furi(): added matched_gws[4]=[6, 0, 4, 2234577] lcr [lcr_mod.c:2085]: add_gws_into_avps(): added gw_uri_avp <6|sip:|0|||3543318538||5060||;transport=udp|0|2> with weight <2234577> lcr [lcr_mod.c:2085]: add_gws_into_avps(): added gw_uri_avp <4|sip:|0||fd=carrier5|53657610||5060||;transport=udp|0|2> with weight <2329862> lcr [lcr_mod.c:2085]: add_gws_into_avps(): added gw_uri_avp <8|sip:|0|||3593650186||5060||;transport=udp|0|2> with weight <7256660> lcr [lcr_mod.c:2085]: add_gws_into_avps(): added gw_uri_avp <7|sip:|0|||3560095754||5060||;transport=udp|0|2> with weight <3129551> lcr [lcr_mod.c:2085]: add_gws_into_avps(): added gw_uri_avp <5|sip:|0|||3526541322||5060||;transport=udp|0|2> with weight <5553909> lcr [lcr_mod.c:2610]: generate_uris(): r_uri <sip:0611111111@10.192.50.210:5060;transport=udp>, dst_uri <> lcr [lcr_mod.c:2904]: ki_next_gw(): added ruri_user_avp <0611111111> lcr [lcr_mod.c:2950]: ki_next_gw(): added tag_avp <> lcr [lcr_mod.c:2965]: ki_next_gw(): added defunct_gw_avp <3526541322> lcr [lcr_mod.c:2610]: generate_uris(): r_uri <sip:0611111111@10.192.50.212:5060;transport=udp>, dst_uri <> lcr [lcr_mod.c:2950]: ki_next_gw(): added tag_avp <> lcr [lcr_mod.c:2965]: ki_next_gw(): added defunct_gw_avp <3560095754> lcr [lcr_mod.c:2610]: generate_uris(): r_uri <sip:0611111111@10.192.50.214:5060;transport=udp>, dst_uri <> lcr [lcr_mod.c:2950]: ki_next_gw(): added tag_avp <> lcr [lcr_mod.c:2965]: ki_next_gw(): added defunct_gw_avp <3593650186> lcr [lcr_mod.c:2610]: generate_uris(): r_uri <sip:0611111111@10.192.50.3:5060;transport=udp>, dst_uri <> lcr [lcr_mod.c:2950]: ki_next_gw(): added tag_avp <fd=carrier5> lcr [lcr_mod.c:2965]: ki_next_gw(): added defunct_gw_avp <53657610> lcr [lcr_mod.c:2610]: generate_uris(): r_uri <sip:0611111111@10.192.50.211:5060;transport=udp>, dst_uri <> lcr [lcr_mod.c:2950]: ki_next_gw(): added tag_avp <> lcr [lcr_mod.c:2965]: ki_next_gw(): added defunct_gw_avp <3543318538> the right gws are selected and call are routed to them *call to 0600000001 => load_gws OK and next_gw KO* lcr [lcr_mod.c:3018]: do_from_gw(): request did not come from gw lcr [lcr_mod.c:3018]: do_from_gw(): request did not come from gw lcr [lcr_mod.c:3004]: do_from_gw(): request game from gw lcr [lcr_mod.c:3009]: do_from_gw(): added tag_avp <cps=5;cc=50> lcr [lcr_mod.c:2264]: ki_load_gws_furi(): load_gws(1, 0600000001, 0211111111) lcr [lcr_mod.c:2313]: ki_load_gws_furi(): from uri <0170619309> did not match to from regex <AMD> lcr [lcr_mod.c:2349]: ki_load_gws_furi(): request uri < sip:0600000001@10.192.50.227:5060> did not match to request regex <(00|\+)39.+> the process stops after trying to match rule with regexp <(00|\+)39.+> Don't know if it's a bug or if I misunderstood something ! Thanks in advance ! Best regards, -- <https://app.livestorm.co/comdata-group/metaverse-what-will-cx-look-like-in-…>

3 8

Local Registration Cache
by Linux Vince 26 Apr '22

26 Apr '22

Hello, I am trying to implement scenario described in this thread by Daniel-Constantin Mierla. I am able to cache registration using save but i am not able to handle unregistration. Unregister function of registrar module is not available in onreply routes and I cannot unregister until upstream registrar returns 200 OK on Expire 0 message. Is there any workaround to fix this? Thanks,

3 5

SSL errors with websocket clients
by Vinicius Kwiecien Ruoso 22 Apr '22

22 Apr '22

Hi all! How are you? I have a kamailio instance and some users are having issues connecting via websocket using TLS. The logs show SSLv3 errors. Cannot find why that error would show up if SSLv2/3 is not enabled. Double checked it via SSLLabs that only TLSv1.2 is allowed in the service. Error sample: ``` 15(36) ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14094416:SSL routines:ssl3_read_bytes:sslv3 alert certificate unknown 15(36) ERROR: <core> [core/tcp_read.c:1512]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7fafc8768190 r: 0x7fafc8768278 (-1) ``` This instance has close to 400 websocket connections opened during the day. I have a high log level enabled. I was not able to identify any reason why this happens. The certificate is a Letsencrypt certificate. The issue is that sometimes clients have a hard time connecting to the websocket and that maybe related. Below are a few more details about the config. Please let me know if you have any pointers on how to debug those connection issues. 8<----- TLS config: modparam("tls", "tls_method", "TLSv1.2+") modparam("tls", "verify_certificate", 0) modparam("tls", "require_certificate", 0) modparam("tls", "low_mem_threshold1", 0) modparam("tls", "low_mem_threshold2", 0) modparam("tls", "private_key", "/etc/certs/tls.key") modparam("tls", "certificate", "/etc/certs/tls.crt") version: kamailio 5.3.9 (x86_64/linux) 8<----- This is related to the issue https://github.com/kamailio/kamailio/issues/3085 (as pointed out it is not a problem in Kamailio code). Thanks, Vinicius

2 2

How to disable video in invite with both audio and video
by Simone de Blasiis 22 Apr '22

22 Apr '22

Hello to everyone, we have a Kamailio 5.5 + RTPEngine deployment. Some devices will send an invite with both audio and video in the offer and we are trying to disable the video portion of the invite by either setting the video port to 0 or by setting a=inactive in the video portion of the SDP packet. We know that sdpops has a function that would be able to remove the video portion but it violates standards so we can't use it. We need to disable the video in the SDP offers because one of the pstn providers we are using doesn't accept invites that contain a video offer. I've spent days searching but I can't find an obvious way of achieving this. We were originally doing it with textops but we've been having some issues where the sdp wouldn't be modified correctly in reinvites after the original call is in place so we're looking for alternatives. Can someone please help us? Thanks in advance Simone de Blasiis

3 4

DMQ sync issue between 5.4 and 5.5
by David Villasmil 20 Apr '22

20 Apr '22

Hello guys, We have a cluster of about 10 kamilios 5.4 with several million htable records. Restarting any server is not a problem, CPU goes ip for a second but everything comes back down to normal. We added a 5.5 by mistake and it won't start, it'll just eat up memory until it bails out with qm_malloc(): qm_malloc(0x149dd5ebb000, 16016) called from tm: h_table.c: build_cell(334), module: tm; Free fragment not found! Is there any known issue between DMQ 5.4 and 5.5? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

1 1

"Re: TOPOS (5.1.[23]) breaks PRACK"
by Simone de Blasiis 20 Apr '22

20 Apr '22

Hello, we noticed the exact behaviour on Kamailio 5.5.4 and topos. If we disable topos the UPDATE messages are being recognized as loose in the WITHINDLG route and everything works. When topos is enabled the UPDATE messages fail the loose_route() check and even if we force the record_route() the message is still not being forwarded. I had just posted my own thread here <https://lists.kamailio.org/pipermail/sr-users/2022-April/114710.html> because I didn't see this one. I'll proceed opening an issue on GitHub so we can track this. Best regards Simone

1 1

Re: [SR-Users] TOPOS (5.1.[23]) breaks PRACK
by Simone de Blasiis 20 Apr '22

20 Apr '22

Hello, we noticed the exact behaviour on Kamailio 5.5.4 and topos. If we disable topos the UPDATE messages are being recognized as loose in the WITHINDLG route and everything works. When topos is enabled the UPDATE messages fail the loose_route() check and even if we force the record_route() the message is still not being forwarded. I had just posted my own thread here <https://lists.kamailio.org/pipermail/sr-users/2022-April/114710.html> because I didn't see this one. I'll proceed opening an issue on GitHub so we can track this. Best regards Simone

1 0

Problems with topos and UPDATE messages
by Simone de Blasiis 20 Apr '22

20 Apr '22

Hello to everyone, we are experiencing some issues with UPDATE messages being sent by the PSTN after an early media announcement. To reproduce the issue I have to call a specific number that has an early media announcement, since it's a paid phone number it has to announce how much it would cost before picking up the call. In this initial part the call is being negotiated in g711a. As soon as the announcement ends, the PSTN sends an UPDATE message containing the new desidered codec from the callee which is g729. At this moment Kamailio doesn't route the UPDATE message to the destination and it actually replies with a 404 message. I've attached a pcap with anonymized IPs to make it harder for bots. Here is a legend for the IPs: - - 12.34.123.123 is Kamailio - 62.23.19.74 is the PSTN - 1.2.130.40 is the client making the call I found a similar report <https://github.com/kamailio/kamailio/issues/2659> in the GitHub issues of the repository and we tried forcing the record_route() as suggested but, as the OP reported in his last message, it's not working. As soon as we disable topos, the UPDATE message is correctly relayed to the caller. In the documentation we can't find anything that could give us a hint on what's happening. Does anyone have any idea on what could be happening? Thanks in advance Simone

1 0

rcd using secspid
by Jon Bonilla (Manwe) 20 Apr '22

20 Apr '22

Hi all I've been checking for adding rcd info to te stir/shaken stuff using secspid module. I've only found this thread: https://lists.kamailio.org/pipermail/sr-dev/2021-June/063156.html I'd like to know if someone has working code snippet using this function to create/sign the json containing the rcd and creating the Identity header instead of calling the all_in_one secsipid_add_identity function I'm using now. cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

1 0

PostgreSQL - module sqlops - working with prepared statements ?
by Chaigneau, Nicolas 20 Apr '22

20 Apr '22

Hello, I'm using sql_xquery from sqlops module to execute requests on a PostgreSQL database. This work fine. Now, for performance reasons, I would like to use prepared statements. I'm wondering if this is possible using sql_xquery. I've looked at the code and documentation, it seems it is not supported. Can you confirm ? If not possible with sql_xquery, is it possible with another existing function ? In the code, I've searched for < PQprepare > / < PQexecPrepared >. These can be found only in < modules/db_postgres/pg_cmd.c >, in function < pg_cmd_exec > which is exported from < modules/db_postgres/pg_mod.c > (see below). I have no idea how to use these though... :/ Any ideas ? /* * Postgres module interface */ static cmd_export_t cmds[] = { {"db_ctx", (cmd_function)NULL, 0, 0, 0, 0}, {"db_con", (cmd_function)pg_con, 0, 0, 0, 0}, {"db_uri", (cmd_function)pg_uri, 0, 0, 0, 0}, {"db_cmd", (cmd_function)pg_cmd, 0, 0, 0, 0}, {"db_put", (cmd_function)pg_cmd_exec, 0, 0, 0, 0}, {"db_del", (cmd_function)pg_cmd_exec, 0, 0, 0, 0}, {"db_get", (cmd_function)pg_cmd_exec, 0, 0, 0, 0}, {"db_upd", (cmd_function)pg_cmd_exec, 0, 0, 0, 0}, {"db_sql", (cmd_function)pg_cmd_exec, 0, 0, 0, 0}, {"db_res", (cmd_function)pg_res, 0, 0, 0, 0}, {"db_fld", (cmd_function)pg_fld, 0, 0, 0, 0}, {"db_first", (cmd_function)pg_cmd_first, 0, 0, 0, 0}, {"db_next", (cmd_function)pg_cmd_next, 0, 0, 0, 0}, {"db_setopt", (cmd_function)pg_setopt, 0, 0, 0, 0}, {"db_getopt", (cmd_function)pg_getopt, 0, 0, 0, 0}, {"db_bind_api", (cmd_function)db_postgres_bind_api, 0, 0, 0, 0}, {0, 0, 0, 0, 0, 0} }; Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

3 4

Update: Issue when Kamailio is running as a Docker container and the Docker network=overlay
by Steve Wilkins 17 Apr '22

17 Apr '22

If I don't use 'listen', then packets do arrive on 5060.

1 0

Issue when Kamailio is running as a Docker container and the Docker network=overlay
by Steve Wilkins 16 Apr '22

16 Apr '22

Hello, I am attempting to get Kamailio to run as a docker container using a docker network of type network=overlay (everything works great if network=host). The issue is that incoming calls are rejected because of 'connection refused'. I noticed that when running on an overlay network, netstat shows the following => tcp6 0 0 :::5060 :::* LISTEN 10001/docker-proxy-current (Note that I have tcp6 disabled on the server and must leave it disabled) I was attempting to find a workaround for this issue, but have not been successful. Does any one on the board have experience with Docker overlay networks and Kamailio? If so, I would appreciate any suggestions on how to resolve this issue. Thank you,

1 0

BLF: modules parameters for specific SIP phones
by Liviu Andron 14 Apr '22

14 Apr '22

Hello, I am working on a BLF (Busy Lamp Field, dialog presence) integration using Kamailio. I am not sure how certain settings can be used on a single generic (any phone) Kamailio instance. * https://www.kamailio.org/docs/modules/devel/modules/presence_dialoginfo.htm… "If the phone does not support this, you can activate this parameter." modparam("presence_dialoginfo", "force_single_dialog", 1) * https://kamailio.org/docs/modules/devel/modules/pua_dialoginfo.html#idm152 "This is a workaround for the buggy Linksys SPA962 phones. SNOM phones work well with the default setting." modparam("pua_dialoginfo", "caller_confirmed", 1) How can I activate these settings only for the phones that have problems? Do I need a different Kamailio instance for each buggy phone that appears from a client ? For example, a Kamailio instance with modparam("pua_dialoginfo", "caller_confirmed", 1) for some (like Linksys SPA962) and the opposite for all the others (like SNOM) ? Thank you,

1 0

append new a= in SDP (without rtpengine)
by Lewis Hutchinson 14 Apr '22

14 Apr '22

version: kamailio 5.5.3 Using KEMI Python I have read several posts on this and tried several approaches but can not seem to get it to work. As it stands I am trying to add a=record:on | a=record:off | a=record:pause In the SDP because FreeSWITCH creating the INVITE isn't able to do this in the way required. When it is sent from FreeSWITCH a=record:on is send a=record I have looked at SDPOPS, TEXTOPS and a few others but non of the offer the option to just inject and a= into the SDP body. So as a result I tried a couple of options where the INVITE is created with a=record (unsure why FreeSWITCH wont let us just add :off etc to the end but that's another point that if anyone know would be helpful) and then using replace_body or search_append_body but the result is TEXTOPS is unable to action the request because of a SDP parse error DEBUG: <core> [core/parser/sdp/sdp.c:599]: parse_sdp_session(): ignoring unknown type in a= line: `a=record#015#012a=ptime:20...' Does anyone have any thoughts or tricks on this to give me a few options at ways to handle this? Lewis

1 2

UAC parameter for enabling and disabling remote registration
by Ahmed Fouad 14 Apr '22

14 Apr '22

Hi, I was wondering if this patch was ever merged to the current release(s) https://github.com/kamailio/kamailio/commit/49f354276df3cfc2cdb0f85b5a1839f… I'd like to start Kamailio with all remote registration disabled then enable based on the code logic Thanks in advance,

2 9

Roadmap to the next major release - v5.6.0
by Daniel-Constantin Mierla 14 Apr '22

14 Apr '22

Hello, it is the time to plan a bit the road to the next major release, to be versioned 5.6.0. The 5.5.0 was release about one year ago, therefore it is time to set the milestones for getting 5.6.0 out. I would propose to freeze the development on Thursday, April 14, 2020, test till mid of May or so, then release v5.6.0. There is a lot of development to existing components and a couple of new modules. If anyone wants a different time line towards 5.6.0, let's discuss and choose the one that suits most of the developers. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 3

kamailio fail to load sip-router.cfg
by bayu p 14 Apr '22

14 Apr '22

Dear Sir, Could you please help me, please to share how to load sip-router.cfg file inside running kamailio ? the file is necessary to be load during rerouting kamailio gateway. I have 2 platform kamailio , both of them are failed to load sip-router.cfg file. They are apt kamailio installation source and the second is tar.gz kamailio installation source, which reload / restart kamailio process (and installation) were failed to load sip-router.cfg file. Best Regards,

2 5

Re: [SR-Users] Kamailio IMS heavy load
by Henning Westerholt 12 Apr '22

12 Apr '22

Hello, checking CPU/I/O performance of the database during the load test would be a start. You can also monitor database access e.g. with a network debugging tools. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Евгений Мастратов <dgeka25594(a)gmail.com> Sent: Tuesday, April 12, 2022 10:03 AM To: Henning Westerholt <hw(a)gilawa.com> Subject: Re: [SR-Users] Kamailio IMS heavy load Hey Henning! Thanks for the advice, I understand about logging, I will make edits and re-check. And how can I check the fact that what are the gaps when reading / writing the database? I am using MySQL. вт, 12 апр. 2022 г. в 10:52, Henning Westerholt <hw(a)gilawa.com<mailto:hw@gilawa.com>>: Hello, as mentioned, I would start by monitoring my dependencies, in particular related to I/O, like database or monitoring. Just to be sure, you are not running the tests with enabled debug log level? This would certainly cause problems. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of ??????? ????????? Sent: Monday, April 11, 2022 5:23 PM To: sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Subject: [SR-Users] Kamailio IMS heavy load Hi! Previously, I created a ticket on github https://github.com/kamailio/kamailio/issues/3071 and tried to deal with the problem there, but they sent me here for advice. I set up Kamailio IMS , in the end everything turned out great, calls are made, and I decided to check how Kamailio copes with the load. I am creating a load using the sipp application. There is no problem with a low call generation rate, but if you raise the call generation to more than 20 calls per second, some of the calls fail. I compared the calls and saw that the ICSCF does not send a 180 Ringing message towards the S-CSCF [..] As I found out further and suggested, the problem is that some calls are cut on the S-CSCF and this is probably due to some kind of timer, but I could not figure out which one. Full logs and dump are attached to the issue on github, below is a log sample by bad dialog 0x7fc133ce61d8 ``` Apr 5 08:24:33 ims-core /usr/local/sbin/kamailio[26061]: DEBUG: ims_dialog [dlg_handlers.c:284]: dlg_iuid_sfree(): freeing dlg iuid [465:10905] (0x7fc133ce61d8) Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1025]: link_dlg(): ref dlg 0x7fc133ce61d8 with 1 -> 1 Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_cb.c:251]: run_create_callbacks(): dialog=0x7fc133ce61d8 Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7fc133ce61d8 with 1 -> 2 Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_handlers.c:789]: dlg_onreq(): dialog [0x7fc133ce61d8] added to tm callbacks Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fc133ce61d8 with 1 -> 1 Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7fc133ce61d8 with 1 -> 2 Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fc133ce61d8 with 1 -> 1 Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26059]: DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7fc133ce61d8 with 1 -> 2 Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26059]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fc133ce61d8 with 1 -> 1 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7fc133ce61d8 with 1 -> 2 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fc133ce61d8 with 1 -> 1 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7fc133ce61d8 with 1 -> 2 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fc133ce61d8 with 1 -> 1 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7fc133ce61d8 with 1 -> 2 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1293]: next_state_dlg(): dialog 0x7fc133ce61d8 changed from state 1 to state 6, due event 4 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_handlers.c:1489]: dlg_onreply(): dialog 0x7fc133ce61d8 failed (negative reply) Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_cb.c:277]: run_dlg_callbacks(): dialog=0x7fc133ce61d8, type=4 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fc133ce61d8 with 1 -> 1 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fc133ce61d8 with 1 -> 0 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): ref <=0 for dialog 0x7fc133ce61d8 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:196]: destroy_dlg(): destroying dialog 0x7fc133ce61d8 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:207]: destroy_dlg(): removed timer for dlg 0x7fc133ce61d8 [3791:1419] with clid '43-3057631(a)10.2.16.63<mailto:43-3057631@10.2.16.63>' and tags '78885556600' Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_cb.c:277]: run_dlg_callbacks(): dialog=0x7fc133ce61d8, type=8192 ``` What could be the reasons for this behavior and how to diagnose them? I did not observe problems with MEM or CPU during the tests. I will be grateful for any advice.

1 0

Kamailio IMS heavy load
by Евгений Мастратов 12 Apr '22

12 Apr '22

Hi! Previously, I created a ticket on github https://github.com/kamailio/kamailio/issues/3071 and tried to deal with the problem there, but they sent me here for advice. I set up Kamailio IMS , in the end everything turned out great, calls are made, and I decided to check how Kamailio copes with the load. I am creating a load using the sipp application. There is no problem with a low call generation rate, but if you raise the call generation to more than 20 calls per second, some of the calls fail. I compared the calls and saw that the ICSCF does not send a 180 Ringing message towards the S-CSCF [image: image.png] [image: image.png] As I found out further and suggested, the problem is that some calls are cut on the S-CSCF and this is probably due to some kind of timer, but I could not figure out which one. Full logs and dump are attached to the issue on github, below is a log sample by bad dialog 0x7fc133ce61d8 ``` Apr 5 08:24:33 ims-core /usr/local/sbin/kamailio[26061]: DEBUG: ims_dialog [dlg_handlers.c:284]: dlg_iuid_sfree(): freeing dlg iuid [465:10905] (0x7fc133ce61d8) Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1025]: link_dlg(): ref dlg 0x7fc133ce61d8 with 1 -> 1 Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_cb.c:251]: run_create_callbacks(): dialog=0x7fc133ce61d8 Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7fc133ce61d8 with 1 -> 2 Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_handlers.c:789]: dlg_onreq(): dialog [0x7fc133ce61d8] added to tm callbacks Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fc133ce61d8 with 1 -> 1 Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7fc133ce61d8 with 1 -> 2 Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fc133ce61d8 with 1 -> 1 Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26059]: DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7fc133ce61d8 with 1 -> 2 Apr 5 08:24:35 ims-core /usr/local/sbin/kamailio[26059]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fc133ce61d8 with 1 -> 1 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7fc133ce61d8 with 1 -> 2 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fc133ce61d8 with 1 -> 1 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7fc133ce61d8 with 1 -> 2 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fc133ce61d8 with 1 -> 1 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7fc133ce61d8 with 1 -> 2 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1293]: next_state_dlg(): dialog 0x7fc133ce61d8 changed from state 1 to state 6, due event 4 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_handlers.c:1489]: dlg_onreply(): dialog 0x7fc133ce61d8 failed (negative reply) Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_cb.c:277]: run_dlg_callbacks(): dialog=0x7fc133ce61d8, type=4 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fc133ce61d8 with 1 -> 1 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fc133ce61d8 with 1 -> 0 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): ref <=0 for dialog 0x7fc133ce61d8 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:196]: destroy_dlg(): destroying dialog 0x7fc133ce61d8 Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_hash.c:207]: destroy_dlg(): removed timer for dlg 0x7fc133ce61d8 [3791:1419] with clid '43-3057631(a)10.2.16.63' and tags '78885556600' Apr 5 08:24:38 ims-core /usr/local/sbin/kamailio[26056]: DEBUG: ims_dialog [dlg_cb.c:277]: run_dlg_callbacks(): dialog=0x7fc133ce61d8, type=8192 ``` What could be the reasons for this behavior and how to diagnose them? I did not observe problems with MEM or CPU during the tests. I will be grateful for any advice.

2 1

How to submit updates for core documentation?
by Ben Kaufman 11 Apr '22

11 Apr '22

Hello, I had a small patch merged for 5.6 to add preprocessor macros #!trydefenv and #!trydefenvs. (https://github.com/kamailio/kamailio/issues/2967), but they need documentation. I believe that they should ultimately end up on this page: https://www.kamailio.org/wiki/cookbooks/devel/core. I don't see this in the /doc section of the repo. What's the process for me to submit this documentation? Ben Kaufman Sr. VoIP Engineer - BCM One +1.612.735.9309 bkaufman(a)bcmone.com<mailto:bkaufman@bcmone.com> 24 hour client support: +1.855.639.6300 [https://www.bcmone.com/files/bcmone-logo.png]<https://www.bcmone.com/> [https://www.bcmone.com/files/twitter-icon-circle.png] <https://www.facebook.com/bcmonenyc/> [https://www.bcmone.com/files/twitter-icon-circle.png] <https://twitter.com/bcm_one/> [https://www.bcmone.com/files/linkedin-icon-circle.png] <https://www.linkedin.com/company/business-communications-management/> NOTE: This e-mail and any attachments are from BCM One, Inc. and are intended solely for the use and review of the individual(s) to whom it is addressed. If you believe you received this e-mail in error, please notify the sender immediately, delete the e-mail (and any attachments) from your computer and do not copy or disclose it to anyone else.

2 4

reply sent
by Daniel Paape 11 Apr '22

11 Apr '22

Hello, Is there a function that can be called from a module that will return the last response code sent, regardless of whether it was a local or relayed reply? thanks, Daniel

2 1

LCR from_gw function
by Antoine DEBARRE 11 Apr '22

11 Apr '22

Hi, Kamailio 5.5.4 from deb packages I tried to use the lcr function from_gw(lcr_id[, ip_addr, proto]) I've declared several gws with different lcr_id Unfortunately from_gw doesn't work with lcr_id > 1 For example : if (from_gw(2,$si,1)) { .... } ... error message => ki_from_gw(): invalid lcr_id parameter value 2 Thanks in advance ! -- <https://app.livestorm.co/comdata-group/metaverse-what-will-cx-look-like-in-…>

2 1

Kamailio + Asterisk + Voip SIP Trunk = Incorrect IP on ACK
by Luciano Motti 09 Apr '22

09 Apr '22

Hello! I have the following architecture: - Kamailio with RTPEngine (with public access on IP *YY.YY.42.207*) - Asterisk (with no public access, internal IP *172.31.69.198*) I am using a Voip SIP (with public access on IP *54.XXX.XXX.44*) configured as a Trunk on Kamailio using UAC Module. The issue I am facing is that the SIP messages replies I get from the VOIP provider are being destined to Kamailio's public IP (YY.YY.42.207) instead of Asterisk's IP (as sent on the message). For example, this is the 200 message I sent to the VOIP provider (I clearly state that the contact is Asterisk sip:172.31.69.198:5080): 2022/04/07 14:17:19.260864 172.31.32.7:5060 -> 54.XXX.XXX.44:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 54.XXX.XXX.44:5060;rport=5060;branch=z9hG4bK90b6.f7a6b0a4.0 Record-Route: <sip:YY.YY.42.207;lr;ftag=p0BjH21gpa3rc> Call-ID: 464f21c3-3120-123b-91b2-026ef2fede4c From: "419XXXXX998" <sip:38XXXX02@172.31.32.169>;tag=p0BjH21gpa3rc To: <sip:38XXXX02@54.XXX.XXX.44>;tag=8c3bb09e-a909-4cb4-99a7-3c57aa8f946e CSeq: 50116727 INVITE Server: Asterisk PBX 18.11.0 *Contact: <sip:172.31.69.198:5080 <http://172.31.69.198:5080>>* Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub Content-Type: application/sdp Content-Length: 239 v=0 o=- 1649321756 1649321759 IN IP4 172.31.69.198 s=Asterisk c=IN IP4 172.31.69.198 t=0 0 m=audio 10010 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv But the ACK I get from them is destined to Kamailio instead of Asterisk: 2022/04/07 14:17:19.263469 54.XXX.XXX.44:5060 -> 172.31.32.7:5060 *ACK sip:YY.YY.42.207:5060 SIP/2.0* Route: <sip:YY.YY.42.207;lr;ftag=p0BjH21gpa3rc> Via: SIP/2.0/UDP 54.XXX.XXX.44:5060;branch=z9hG4bK90b6.f7a6b0a4.2 Max-Forwards: 69 From: "419XXXXX998" <sip:38XXXX02@172.31.32.169>;tag=p0BjH21gpa3rc To: <sip:38XXXX02@54.XXX.XXX.44>;tag=8c3bb09e-a909-4cb4-99a7-3c57aa8f946e Call-ID: 464f21c3-3120-123b-91b2-026ef2fede4c CSeq: 50116727 ACK Contact: <sip:voip.voip-sbc.6792.fab37901@54.XXX.XXX.44:5060> Content-Length: 0 This causes Kamailio to not know where to forward the message to... Asterisk never gets the reply. This is happening to one of the SIP Trunks I am using, with the other, everything is fine. Is there anything I can do to work around it (other than contacting the provider to fix on their end)? Thanks! ps: I have attached the full SIP messages trail to help.

2 1

Htable reload
by David Villasmil 07 Apr '22

07 Apr '22

Hello all, I’ve been trying to figure out how htable reloads happen. Does it replaces the existing one or creates a new one, populates it from the db and then renames it? The latter would be (I think) ideal, even if it uses more resources. Thanks! David -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

4 6

Kamailio 5.5.4 lcr command disabled
by bayu p 07 Apr '22

07 Apr '22

Hello Expert, We faces problem in utilizing lcr command line in "kamctl lcr show_gws", whereas kamailio version is 5.5.4 which are new kamailio release, "kamctl lcr show_gws" shown result "command disabled". in this case, LCR module is already been uploaded and the systemd has been running well active kamailio. Could you share, what probably wrong in my kamailio configuration? Best Regards,

2 4

Kamailio 5.5.4 lcr command disabled
by bayu p 07 Apr '22

07 Apr '22

Hello Expert, I have problem in utilizing lcr command line in "kamctl lcr show_gws", whereas my kamailio version is 5.5.4 which are new kamailio release, "kamctl lcr show_gws" shown result "command disabled". in this case, LCR module is already been uploaded and the systemd has been running active kamailio. Here detailed screenshot : [image: image.png] Could you share, what probably wrong in my kamailio configuration? Best Regards,

1 1

Kamailio 5.5.4 lcr command disabled
by bayu p 07 Apr '22

07 Apr '22

problem in "kamctl lcr show_gws" resulting "command disabled". Kamailio uses LCR module uploaded and running active kamailio. what probably wrong in kamailio? [image: image.png]

1 0

Kamailio 5.5.4 lcr command disabled
by bayu p 07 Apr '22

07 Apr '22

problem in "kamctl lcr show_gws" (kamailio version 5.5.4 / new release) resulting "command disabled". LCR module had been uploaded on running well active kamailio. Could you share, what probably wrong in my kamailio configuration? [image: image.png] Best regards

1 0

Kamailio 5.5.4 lcr command disabled
by bayu p 07 Apr '22

07 Apr '22

Hello Expert, I have problem in utilize lcr command line "kamctl lcr show_gws", whereas kamailio version is 5.5.4 (new kamailio release), those resulting "command disabled". In this case, LCR module is already been uploaded and the systemd has been running active kamailio. Could you share, what probably wrong in my kamailio configuration? [image: image.png] Best regards

1 0

kamailio 5.5.4 source not found file pcre.h file
by bayu p 07 Apr '22

07 Apr '22

Dear Developer, would you please share which pcre.h file ? due to my compiling source kamailio 5.5.4 found error as captured below. In file included from dialplan.c:59: dialplan.h:33:10: fatal error: pcre.h: No such file or directory 33 | #include <pcre.h> | ^~~~~~~~ compilation terminated. Kind Regards,

2 2

Kamailio preprocessor to get hostname
by David VILLAUME 06 Apr '22

06 Apr '22

Hello all , Do you know any way to get the hostname of my server at kamailio runtime ? The simple idea behind this would be ton import a specific environment file base on hostname. Regards, David

2 1

parse XML nested name-spaces
by Artur Hovsepyan 06 Apr '22

06 Apr '22

good evening everyone, can you please help how to parse following XML body and get the "<gml:pos>31.204359 -44.512358</gml:pos>" in a variable? <?xml version="1.0"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:gp="urn:ietf:params:xml:ns:pidf:geopriv10" xmlns:gml="http://www.opengis.net/gml" xmlns:gs="http://www.opengis.net/pidflo/1.0" xmlns:con="urn:ietf:params:xml:ns:geopriv:conf" entity="sip:+6478897789@ims.mnc0xx.mccxxx.3gppnetwork.org"> <dm:device id="Wifi"> <gp:geopriv> <gp:location-info> <gs:Circle srsName="urn:ogc:def:crs:EPSG::5524"> <gml:pos>31.204359 -44.512358</gml:pos> <gs:radius uom="urn:ogc:def:uom:EPSG::8190">10.578000</gs:radius> </gs:Circle> <con:confidence pdf="normal">92</con:confidence> </gp:location-info> <gp:method>DBH_HELO</gp:method> <gp:usage-rules/> </gp:geopriv> <dm:timestamp>2022-04-05T19:16:34Z</dm:timestamp> </dm:device> </presence> i tried few variants: xinfo("GML_LOC: $xml(x=>xpath:/pidf:presence/pidf:dm:device/pidf:gp:geopriv/pidf:gp:location-info/pidf:gs:Circle/pidf:gml:pos)\n") xinfo("GML_LOC: $xml(x=>xpath:/pidf:presence/pidf:dm\:device/pidf:gp\:geopriv/pidf:gp\:location-info/pidf:gs\:Circle/pidf:gml\:pos)\n"); xinfo("GML_LOC: $xml(x=>xpath:/pidf:presence/pidf:dm:device/pidf:gp:geopriv/pidf:gp:location-info/pidf:gs:Circle/pidf:gml:pos)\n"); xinfo("GML_LOC: $xml(x=>xpath:/pidf:presence/dm:device/gp:geopriv/gp:location-info/gs:Circle/gml:pos)\n"); and end up with error "unable to evaluate xpath expression": ERROR: xmlops [pv_xml.c:308]: pv_get_xml(): unable to evaluate xpath expression [/pidf:presence/dm:device/gp:geopriv/gp:location-info/gs:Circle/gml:pos/70] thanks in advance,Art

2 2

least cost routing module activation error
by bayu p 06 Apr '22

06 Apr '22

Dear Kamailio Expert, I already add module lcr, but found error when searching pvar "ddn" inside source code core/pvapi.c at line 923, as shown error log captured below : Apr 6 10:26:50 kamal /usr/sbin/kamailio[180954]: ERROR: <core> [core/pvapi.c:923]: pv_parse_spec2(): error searching pvar "ddn" Apr 6 10:26:50 kamal /usr/sbin/kamailio[180954]: ERROR: <core> [core/pvapi.c:1126]: pv_parse_spec2(): wrong char [n/110] in [$ddn] at [3 (0)] Apr 6 10:26:50 kamal /usr/sbin/kamailio[180954]: ERROR: xlog [xlog.c:511]: xdbg_fixup_helper(): wrong format[Domain of destination: $ddn] Apr 6 10:26:50 kamal /usr/sbin/kamailio[180954]: ERROR: <core> [core/route.c:1166]: fix_actions(): fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:901 Apr 6 10:26:50 kamal /usr/sbin/kamailio[180954]: ERROR: <core> [core/route.c:1166]: fix_actions(): fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:903 Apr 6 10:26:50 kamal /usr/sbin/kamailio[180954]: ERROR: <core> [core/route.c:1166]: fix_actions(): fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:918 The question are, how to define correct "INVITE" methode (or correct format) inside route[LCR] which is already defined inside kamailio.cfg: route[LCR] { if (status=="200") { xlog("LCR: Inside the LCR routen"); } if(method=="INVITE") { xlog("We got an invite"); if(!load_gws(1, $rU, $var(caller_uri))) { xlog("Couldn't load gateways"); sl_send_reply("500", "Server Internal Error - Cannot load gateways"); exit; } else { xlog("GW Selected '$avp(i:709)'n"); xlog("Domain of destination: $ddn"); xlog("To URI: $tun"); } if(!next_gw()) { xlog("Couldn't proceed to next gateway"); sl_send_reply("503", "Service not available, no gateways found"); exit; } else { xlog("Calling the first matched gatewayn"); xlog("ruri_user_avp: '$avp(i:500)'n"); xlog("To URI after next_gw: $tun"); xlog("Request URI: $rUn"); } } } Kind Regards

1 0

escape firt two lines from get_body_part to get xml only part
by Artur Hovsepyan 06 Apr '22

06 Apr '22

hello everyone, i am trying to parse the xml from the message body: get_body_part("application/pidf+xml", "$var(pbody)"); and i am getting the following in my pbody var: Content-Disposition: render;handling=optionalContent-ID: <sip:+6478897789@ims.mnc0xx.mccxxx.3gppnetwork.org><?xml version="1.0"?><presence xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:gp="urn:ietf:params:xml:ns:pidf:geopriv10" xmlns:gml="http://www.opengis.net/gml" xmlns:gs="http://www.opengis.net/pidflo/1.0" xmlns:con="urn:ietf:params:xml:ns:geopriv:conf" entity="sip:+6478897789@ims.mnc0xx.mccxxx.3gppnetwork.org"><dm:device id="Wifi"><gp:geopriv><gp:location-info><gs:Circle srsName="urn:ogc:def:crs:EPSG::5524"><gml:pos>31.204359 -44.512358</gml:pos><gs:radius uom="urn:ogc:def:uom:EPSG::8190">10.578000</gs:radius></gs:Circle><con:confidence pdf="normal">92</con:confidence></gp:location-info><gp:method>DBH_HELO</gp:method><gp:usage-rules/></gp:geopriv><dm:timestamp>2022-04-05T19:16:34Z</dm:timestamp></dm:device></presence> how can i escape the first two lines in order to get the xml part of the content? thanks in advance,Art

2 6

Jansson question
by Barry Flanagan 06 Apr '22

06 Apr '22

Hi I am trying to use the jannson_get function to pull field "secret" from the below json: {"3":{"options":{"secret":"mybigsecret"}}} This works is I use jansson_get("3.options.secret"..... However, I need to know the value of "3" in order to use it. I have tried all the variations on path I can find, but none apart from using the actual root value work for me. How can I either: 1. Construct a PATH which will pick this up, or 2. Obtain this root value ("3" in my example) I have tried testing using https://jsonpath.com/ and a path of "$..secret" and "$..options.secret" works there, but not in jansson kamailio module. Any assistance much appreciated. Thanks -- -Barry

2 1

escape firt two lines from get_body_part to get xml only part
by Does it really matter? 06 Apr '22

06 Apr '22

1 0

Manual usage and example kamailio kamctl
by bayu p 06 Apr '22

06 Apr '22

Dear Expert Kamailio, I use this tutorial kamctl to make kamailio and other sip server communicated https://manpages.ubuntu.com/manpages/xenial/man8/kamctl.8.html but, i cannot found further tutorial how use kamctl. Could you please share, the detail/further tutorial and example of kamctl usage.. Since i always found error/failure during execute some command from the link, here captured: (1). # kamctl cr dump -e \E[37;31mERROR: Error opening Kamailio's FIFO /var/run/kamailio/kamailio_rpc.fifo -e \E[37;31mERROR: Make sure you have loaded the jsonrpcs module and set FIFO transport parameters (2). # kamctl lcr show_gws -e \E[37;31mERROR: command disabled (3). # kamctl lcr reload -e \E[37;31mERROR: command disabled (4). # kamctl ping 192.168.31.50 { "jsonrpc": "2.0", "error": { "code": 400, "message": "Invalid request uri \"192.168.31.50\"" }, "id": 76608 } (5). # kamctl cr reload { "jsonrpc": "2.0", "error": { "code": 500, "message": "Method Not Found" }, "id": 76659 } Warm Regards, >

5 16

Kamailio stop processing SIP in high call load with dlg replication via DMQ
by José Seabra 05 Apr '22

05 Apr '22

Hi there, I've two kamailio servers sharing dlg vars between them and I've noticed that during high call load (~50-70 CPS) the kamailio that is processing all the calls and sharing the dialogs with the other kamailio, stops processing SIP. If I set the parameter enable_dmq to 0 I cannot replicate the issue so I assume that it's related to dialog replication through DMQ. *Kamailio version: *5.4.7 - I've tested with recent versions but the issue remains *Configurations of Kamailio_1:* # ----- dialog params ----- modparam("dialog", "dlg_flag", FLD_DLG) modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "enable_dmq", 1) modparam("dialog", "h_id_start", 1000) modparam("dialog", "h_id_step", 6000) # ----- dmq params ----- modparam("dmq", "server_address", "sip:INTERNAL_INSTANCE_IP:5060") modparam("dmq", "notification_address", "DMQ_NOTIFICATION_ADDRESS") modparam("dmq", "multi_notify", 1) modparam("dmq", "ping_interval", 10) modparam("dmq", "num_workers",4) *Configurations of Kamailio_2:* # ----- dialog params ----- modparam("dialog", "dlg_flag", FLD_DLG) modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "enable_dmq", 1) modparam("dialog", "h_id_start", 5000) modparam("dialog", "h_id_step", 6000) # ----- dmq params ----- modparam("dmq", "server_address", "sip:INTERNAL_INSTANCE_IP:5060") modparam("dmq", "notification_address", "DMQ_NOTIFICATION_ADDRESS") modparam("dmq", "multi_notify", 1) modparam("dmq", "ping_interval", 10) modparam("dmq", "num_workers",4) In this link you can find the kamctl trap <https://textbin.net/quykw71nyl>output after kamailio stop processing SIP. Thank you Best Regards -- Cumprimentos José Seabra

1 0

Auth_ephemeral Module > username_format not accepted
by Luciano Motti 05 Apr '22

05 Apr '22

Hi! I was able to successfully use the auth_ephemeral module on my environment: Kamailio > Asterisk > ARI I ran into a problem with ARI, where Asterisk wouldn't send the complete from URI to ARI program, just the first part (timestamp), and the most important part (the username) isn't forwarded. For example: Kamailio username: 1649106119:1001 Asterisk only forwards 1649106119 to ARI, which makes it impossible for me to track the user. So I planned on using the deprecated username_format from Auth Ephemeral module: https://kamailio.org/docs/modules/4.1.x/modules/auth_ephemeral.html#auth_ep… By switching the timestamp with the username, my problems would be fixed. After setting the variable, and trying to register on Kamailio with username 1001:1649106119, I get an error message: SIP/2.0 400 Bad From URI Am I doing something wrong? Thanks!

2 2

Re: [SR-Users] More than TCP 800 concurrent calls, slows Kamailio (LUA FILE API REQUESTS)
by Henning Westerholt 05 Apr '22

05 Apr '22

Hello, please keep the list in CC, that others can comment as well. Beside from the obvious (improving the performance of the lua API call), you could use indeed asynchronous processing. If you just use the lua code to emit the REST request, using a native http client module would be probably also faster as doing it over lua. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Giovanni Jose <gio.jose(a)gmail.com> Sent: Wednesday, March 30, 2022 5:17 PM To: Henning Westerholt <hw(a)gilawa.com> Subject: Re: [SR-Users] More than TCP 800 concurrent calls, slows Kamailio (LUA FILE API REQUESTS) Hello Henning, Thanks so much for the fast response I just commented the loop and I was hitting the backend and it's slow anyway maybe it's because, for this type of implementation when the destination of the call is decided by the API backend, I need to use the http_async_client module? I am noticing that for high volume calls kamailio slows the traffic and shows errors because there's a delay that is been added when the backend responds for every call. On Tue, Mar 29, 2022 at 11:53 PM Henning Westerholt <hw(a)gilawa.com<mailto:hw@gilawa.com>> wrote: Hello, why are you sleeping for 1s in the loop? This will cause Kamailio to block processing. I also do not see any stop condition on the loop, so it will loop until the limit. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Giovanni Jose Sent: Tuesday, March 29, 2022 9:46 PM To: sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Subject: [SR-Users] More than TCP 800 concurrent calls, slows Kamailio (LUA FILE API REQUESTS) Hello Guys I am having this issue where kamailio is receiving requests from SIPp (we are currently testing the platform and doing QA) but if we do more than 800 concurrent calls kamailio starts failing and slowing the requests, The scenario is the following: I am sending 5000 calls at 100 CPS on TCP and I am having this route in the configuration file: # Handle the calls to api route[CALL_API] { xlog("L_NOTICE"," Call request $var(call_request) method: $rm \n"); $var(loop_true)=1; while ($var(loop_true)) { if(!lua_run("call_request","$var(call_request)","$sht(token=>new_token)")) { xlog("L_NOTICE", "SCRIPT: failed to execute lua function!\n"); } if ($var(loop_true)){ sleep("1"); } } xlog("L_NOTICE", "SCRIPT: Sucess to execute lua function!\n"); } So I am executing a LUA script for the requests and the API response will tell the destination of the call. At 800 concurrent calls, I can notice SLOWLINESS in the SIPp testing tool, and then I will start receiving 408 timeouts. This sounds like a load issue or something because I believe there's a time that it's been added every time we hit the backend and wait for the response. any help will be appreciated Regards Gio

2 1

Jansson question
by Barry Flanagan 05 Apr '22

05 Apr '22

Hi I am trying to use the jannson_get function to pull field "secret" from the below json: {"3":{"options":{"secret":"mybigsecret"}}} This works is I use jansson_get("3.options.secret"..... However, I need to know the value of "3" in order to use it. I have tried all the variations on path I can find, but none apart from using the actual root value work for me. How can I either: 1. Construct a PATH which will pick this up, or 2. Obtain this root value ("3" in my example) I have tried testing using https://jsonpath.com/ and a path of "$..secret" and "$..options.secret" works there, but not in jansson kamailio module. Any assistance much appreciated. Thanks -- -Barry

1 0

rpm.kamailio.org is down
by Riccardo Villa 05 Apr '22

05 Apr '22

Host is up and responding to ping but not to http requests. Riccardo -- -- NOTICE: This electronic mail transmission may contain confidential information and is intended only for the person(s) named. Any use, copying, or disclosure by any other person is strictly prohibited. If you have received this transmission in error, please notify the sender via e-mail.

2 1

Asterisk, unable to find dialog for ACK with route param and call-id
by Trevor Peirce 05 Apr '22

05 Apr '22

Hello, I'm having some trouble with what I think is the dialog module. I've got Kamailio 5.5.4 running well for some time. Recently I added Asterisk to the mix and have come across an issue. Call to Kamailio. Kamailio sends it to Asterisk. Asterisk may decide to "forward" the call. To do this it will place a new outbound call through Kamailio. From the user perspective, this works absolutely fine (invite, ringing, ok, bye). The acc module handles it fine also. However dialog does not, and the events for ok and bye do not fire on the original call. They do fire for the forwarded call. I've added some logging output to try to isolate the problem, but I'm no further ahead. Here's what I have. Any suggestions? ** Call initiated ** INFO: <script>: onreply_route[MANAGE_REPLY] branch [0] rm=INVITE rs=100 ru=<null> tu=sip:destination@endpoint DLG_status=<null> dlg-h-id=3360 INFO: <script>: onreply_route[MANAGE_REPLY] branch [0] rm=INVITE rs=100 ru=<null> tu=sip:fwddest@0.0.0.1:5060 DLG_status=<null> dlg-h-id=12036 INFO: <script>: onreply_route[MANAGE_REPLY] branch [0] rm=INVITE rs=183 ru=<null> tu=sip:fwddest@0.0.0.1:5060 DLG_status=<null> dlg-h-id=12036 INFO: <script>: onreply_route[MANAGE_REPLY] branch [0] rm=INVITE rs=183 ru=<null> tu=sip:destination@endpoint DLG_status=<null> dlg-h-id=3360 INFO: <script>: onreply_route[MANAGE_REPLY] branch [0] rm=INVITE rs=200 ru=<null> tu=sip:fwddest@0.0.0.1:5060 DLG_status=<null> dlg-h-id=12036 CRITICAL: dialog [dlg_timer.c:200]: update_dlg_timer(): Trying to update a bogus dlg tl=0x7fa0494006e8 tl->next=(nil) tl->prev=(nil) ERROR: dialog [dlg_hash.c:1329]: update_dlg_timeout(): failed to update dialog lifetime CRITICAL: dialog [dlg_timer.c:200]: update_dlg_timer(): Trying to update a bogus dlg tl=0x7fa0494006e8 tl->next=(nil) tl->prev=(nil) ERROR: dialog [dlg_hash.c:1329]: update_dlg_timeout(): failed to update dialog lifetime ERROR: dialog [dlg_profile.c:1028]: dlg_set_timeout_by_profile(): Unable to set timeout on 3306:12036 NOTICE: acc [acc.c:270]: acc_log_request(): ACC: transaction answered: timestamp=1648611824;time_unix=1648611824.346;method=INVITE;from_tag=as292200a1;to_tag=as75834dc4;call_id=7cef233547dca0ac7882ce5249b3b54c;code=200;reason=OK INFO: <script>: onreply_route[MANAGE_REPLY] branch [0] rm=INVITE rs=200 ru=<null> tu=sip:destination@endpoint DLG_status=<null> dlg-h-id=<null> NOTICE: acc [acc.c:270]: acc_log_request(): ACC: transaction answered: timestamp=1648611824;time_unix=1648611824.354;method=INVITE;from_tag=as730369e7;to_tag=as3b06b3e2;call_id=53bad4f0034b61a5745163d82daef402;code=200;reason=OK INFO: <script>: onreply_route[MANAGE_REPLY] branch [0] rm=INVITE rs=200 ru=<null> tu=sip:destination@endpoint DLG_status=<null> dlg-h-id=<null> WARNING: dialog [dlg_handlers.c:1328]: dlg_onroute(): unable to find dialog for ACK with route param '161.02d' [353:3360] and call-id '53bad4f0034b61a5745163d82daef402' WARNING: dialog [dlg_handlers.c:1328]: dlg_onroute(): unable to find dialog for ACK with route param '161.02d' [353:3360] and call-id '53bad4f0034b61a5745163d82daef402' ** The call is answered and endpoints can talk to each other ** ** Now the original caller hangs up ** WARNING: dialog [dlg_handlers.c:1328]: dlg_onroute(): unable to find dialog for BYE with route param '161.02d' [353:3360] and call-id '53bad4f0034b61a5745163d82daef402' NOTICE: acc [acc.c:270]: acc_log_request(): ACC: transaction answered: timestamp=1648611831;time_unix=1648611831.933;method=BYE;from_tag=as730369e7;to_tag=as3b06b3e2;call_id=53bad4f0034b61a5745163d82daef402;code=200;reason=OK INFO: <script>: onreply_route[MANAGE_REPLY] branch [0] rm=INVITE rs=100 ru=<null> tu=sip:fwddest@0.0.0.1:5060 DLG_status=<null> dlg-h-id=12036 INFO: <script>: onreply_route[MANAGE_REPLY] branch [0] rm=INVITE rs=200 ru=<null> tu=sip:fwddest@0.0.0.1:5060 DLG_status=<null> dlg-h-id=12036 ERROR: <script>: got result for bye = Bye!; DLG_lifetime = 8 NOTICE: acc [acc.c:270]: acc_log_request(): ACC: transaction answered: timestamp=1648611832;time_unix=1648611832.115;method=BYE;from_tag=as292200a1;to_tag=as75834dc4;call_id=7cef233547dca0ac7882ce5249b3b54c;code=200;reason=OK ERROR: dialog [dlg_handlers.c:363]: dlg_terminated_confirmed(): failed to get dialog from params! ** Everything ends cleanly on the other SIP endpoints ** What can I do from here to try to isolate this further? Thanks, -- Trevor Peirce AcroVoice Solutions Inc

2 4

Kamailio as a SBC
by Nileshwar Goswami 05 Apr '22

05 Apr '22

Hi There, I'm looking out for an SBC and came across your platform. Can Kamailio work as a SBC and provide all the functionalities of a SBC ? A quick reply will be appreciated. Best Regards, Nileshwar Goswami

2 1

RTPENGINE : Kamailio - Get status if max sessions reached
by Youssef Boujraf 01 Apr '22

01 Apr '22

Dear, QUESTION : Through kamailio,We are using rtpengine_manage and the return value is true or false but don"t know for which reason. * How to be informed if rtpengine reach the maximum of sessions and will not accept any rtp routing ? * We have seen if rtpengine reached the max-session; The call is accept by kamailio and finally no audio because SDP not processed- kamailio.log Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: ALERT: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} <script>: [NATMANAGE] branch_id:0 ruri: sip:9919993@public:32617;transport=UDP;rinstance=f5d124565504bdb6, method:INVITE, status:<null>, extra_id: z9hG4bK-524287-1---290944f5547c740a0, rtpengine_manage: replace-origin replace-session-connection via-branch=extraMar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine_funcs.c:144]: check_content_type(): type <application/sdp> found validMar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:1834]: build_rtpp_socks(): same rtpengines list version: 1 (1648562844)Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:3276]: select_rtpp_node_old(): rtpengine hash table lookup failed to find node for calllen=24 callid=nIdGvZZNiFnt3_r0xSIKpQ.. viabranch=z9hG4bK-524287-1---290944f5547c740a0Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:3167]: select_rtpp_node_new(): sum is = 27Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:2691]: rtpp_function_call(): proxy reply: d6:result10:load limit7:message30:Parallel session limit reachedeMar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: INFO: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:2715]: rtpp_function_call(): proxy udp:192.168.1.33:2223 has reached its load limit (Parallel session limit reached) - trying next oneMar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:1834]: build_rtpp_socks(): same rtpengines list version: 1 (1648562844) Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:3276]: select_rtpp_node_old(): rtpengine hash table lookup failed to find node for calllen=24 callid=nIdGvZZNiFnt3_r0xSIKpQ.. viabranch=z9hG4bK-524287-1---290944f5547c740a0 Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:3167]: select_rtpp_node_new(): sum is = 27 Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:2691]: rtpp_function_call(): proxy reply: d6:result10:load limit7:message30:Parallel session limit reachede Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: INFO: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:2715]: rtpp_function_call(): proxy udp:127.0.0.1:2223 has reached its load limit (Parallel session limit reached) - trying next one Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:1834]: build_rtpp_socks(): same rtpengines list version: 1 (1648562844) Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:3276]: select_rtpp_node_old(): rtpengine hash table lookup failed to find node for calllen=24 callid=nIdGvZZNiFnt3_r0xSIKpQ.. viabranch=z9hG4bK-524287-1---290944f5547c740a0 Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:3167]: select_rtpp_node_new(): sum is = 27 Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: INFO: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:2900]: rtpp_test(): rtpengine instance <udp:127.0.0.1:2223> found, support for it enabled Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: INFO: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:2900]: rtpp_test(): rtpengine instance <udp:192.168.1.33:2223> found, support for it enabled Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:3167]: select_rtpp_node_new(): sum is = 27 Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: ERROR: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:3330]: select_rtpp_node(): rtpengine failed to select new for calllen=24 callid=nIdGvZZNiFnt3_r0xSIKpQ.. Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21065]: ERROR: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:2678]: rtpp_function_call(): no available proxies Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21070]: ALERT: {2 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} <script>: [NATMANAGE] branch_id:0 ruri: <null>, method:INVITE, status:100, extra_id: z9hG4bK-524287-1---290944f5547c740a0, rtpengine_manage: replace-origin replace-session-connection trust-address via-branch=extra Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21069]: ALERT: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} <script>: [NATMANAGE] branch_id:1 ruri: sip:9919993@public:32617;transport=UDP;rinstance=f5d124565504bdb6, method:INVITE, status:<null>, extra_id: z9hG4bK-524287-1---290944f5547c740a0, rtpengine_manage: replace-origin replace-session-connection Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21069]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:1834]: build_rtpp_socks(): same rtpengines list version: 1 (1648562844) Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21069]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:3276]: select_rtpp_node_old(): rtpengine hash table lookup failed to find node for calllen=24 callid=nIdGvZZNiFnt3_r0xSIKpQ.. viabranch= Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21069]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:3167]: select_rtpp_node_new(): sum is = 27 Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21069]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:2691]: rtpp_function_call(): proxy reply: d7:warning38:Call-ID not found or tags didn't match6:result2:oke Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21069]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:2784]: rtpp_function_call(): rtpengine hash table insert node=udp:192.168.1.33:2223 for calllen=24 callid=nIdGvZZNiFnt3_r0xSIKpQ.. viabranch= Mar 29 16:08:14 kamailio /usr/sbin/kamailio[21069]: DEBUG: {1 2 INVITE nIdGvZZNiFnt3_r0xSIKpQ..} rtpengine [rtpengine.c:2801]: rtpp_function_call(): rtpengine hash table remove entry for callen=24 callid=nIdGvZZNiFnt3_r0xSIKpQ.. viabranch= - rtpengine.log : Mar 29 16:08:14 rtp01srv rtpengine[3921]: INFO: [nIdGvZZNiFnt3_r0xSIKpQ..]: [control] Received command 'offer' from 192.168.1.41:42822 Mar 29 16:08:14 rtp01srv rtpengine[3921]: DEBUG: [nIdGvZZNiFnt3_r0xSIKpQ..]: [control] Dump for 'offer' from 192.168.1.41:42822: { "supports": [ "load limit" ], "sdp": "v=0 Mar 29 16:08:14 rtp01srv rtpengine[3921]: o=Zoiper 141319690 1 IN IP4 192.168.30.216 Mar 29 16:08:14 rtp01srv rtpengine[3921]: s=Z Mar 29 16:08:14 rtp01srv rtpengine[3921]: c=IN IP4 192.168.30.216 Mar 29 16:08:14 rtp01srv rtpengine[3921]: t=0 0 Mar 29 16:08:14 rtp01srv rtpengine[3921]: m=audio 8000 RTP/AVP 0 101 8 3 97 Mar 29 16:08:14 rtp01srv rtpengine[3921]: a=rtpmap:101 telephone-event/8000 Mar 29 16:08:14 rtp01srv rtpengine[3921]: a=fmtp:101 0-16 Mar 29 16:08:14 rtp01srv rtpengine[3921]: a=rtpmap:97 iLBC/8000 Mar 29 16:08:14 rtp01srv rtpengine[3921]: a=fmtp:97 mode=30 Mar 29 16:08:14 rtp01srv rtpengine[3921]: a=sendrecv Mar 29 16:08:14 rtp01srv rtpengine[3921]: ", "replace": [ "origin", "session-connection" ], "call-id": "nIdGvZZNiFnt3_r0xSIKpQ..", "via-branch": "z9hG4bK-524287-1---290944f5547c740a0", "received-from": [ "IP4", "public" ], "from-tag": "75983054", "command": "offer" } Mar 29 16:08:14 rtp01srv rtpengine[3921]: ERR: [nIdGvZZNiFnt3_r0xSIKpQ..]: [core] Parallel session limit reached (2) Mar 29 16:08:14 rtp01srv rtpengine[3921]: INFO: [nIdGvZZNiFnt3_r0xSIKpQ..]: [control] Replying to 'offer' from 192.168.1.41:42822 (elapsed time 0.000058 sec) Mar 29 16:08:14 rtp01srv rtpengine[3921]: DEBUG: [nIdGvZZNiFnt3_r0xSIKpQ..]: [control] Response dump for 'offer' to 192.168.1.41:42822: { "result": "load limit", "message": "Parallel session limit reached" } Mar 29 16:08:14 rtp01srv rtpengine[3921]: INFO: [control] Received command 'ping' from 192.168.1.41:42822 Mar 29 16:08:14 rtp01srv rtpengine[3921]: DEBUG: [control] Dump for 'ping' from 192.168.1.41:42822: { "command": "ping" } Mar 29 16:08:14 rtp01srv rtpengine[3921]: INFO: [control] Replying to 'ping' from 192.168.1.41:42822 (elapsed time 0.000001 sec) Mar 29 16:08:14 rtp01srv rtpengine[3921]: DEBUG: [control] Response dump for 'ping' to 192.168.1.41:42822: { "result": "pong" } How to get this information then we could cancel the call or send sip message that the number of max call is reached ? -- Youssef

3 7

Kamailio PCSCF route logic
by Weiqi Yao 30 Mar '22

30 Mar '22

Hello community, We made some modifications to Kamailio PCSCF sample cfg file, so when PCSCF receiving an tcp SUBSCRIBE message it add ;transport=tcp to the end of Contact URI. Below is what we added the route block right after if (is_method("INVITE|SUBSCRIBE")) logic if (is_method("SUBSCRIBE") && proto=='tcp') { $var(original_contact_header) = $ct; $var(new_contact_header) = $(var(original_contact_header){re.subst,/(.*:.*)(>);/\1;transport=tcp>/g}); remove_hf("Contact"); append_hf("Contact: $var(new_contact_header)\r\n"); msg_apply_changes(); } We did see the ;transport=tcp been added but we also seeing a bad line ;alias=[...] is added before the To: line [cid:image001.png@01D84375.8D109B20] Does anyone know what function might cause this ;alias been added before To: line? BR, Weiqi

2 2

Parallel forking call drop scenario
by Koray Vatansever 30 Mar '22

30 Mar '22

Hi everyone, I'm facing the following situation. Clients are connected over TLS. Therefore there is no retransmission in Kamailio. How can I prevent this call drop? +---+ +-----------+ +-----+ +-----+ | A | | Kamailio | | B1 | | B2 | +---+ +-----------+ +-----+ +-----+ | | | | | INVITE | | | |------------------------------------>| | | | | -------------------\ | | | |-| Parallel forking | | | | | |------------------| | | | | | | | | INVITE | | | |---------------------->| | | | | | | | INVITE | | | |--------------------------------------------------->| | | | | | | 180 Ringing | | | |<----------------------| | | | | | | | | 180 Ringing | | |<---------------------------------------------------| | | | ------------------------\ | | | |-| User answers the call | | | | | |-----------------------| | | | | | | | 200 OK | | | |<----------------------| | | | | | | 200 OK | | | |<------------------------------------| | | | | | | | ACK | | | |------------------------------------>| | | | | | | | | ACK | | | |---------------------->| | | ------------------------------\ | | | |-| Session established with B1 | | | | | |-----------------------------| | | | | | | | ----------------------------\ | | | |-| Client changes network | | | | | | LTE to Wifi or vice versa | | | | | | Temporary connection lost | | | CANCEL | | |---------------------------| | |----------------------------------------------X | | | | | -------------------------------------------------\ | | | |-| Reconnected with different IP | | | | | | CANCEL is missed and client continues ringing. | | | | | | User answers the call. | | | | 200 OK | |------------------------------------------------| | |<---------------------------------------------------| | | | | | 200 OK | | | |<------------------------------------| | | | | | | | | | BYE | | |<---------------------------------------------------| | | | | | BYE | | | |<------------------------------------| | | | ---------------------------\ | | | |-| Session terminated by B2 | | | | | | Call Drop | | | | | |--------------------------| | | |

2 2

More than TCP 800 concurrent calls, slows Kamailio (LUA FILE API REQUESTS)
by Giovanni Jose 30 Mar '22

30 Mar '22

Hello Guys I am having this issue where kamailio is receiving requests from SIPp (we are currently testing the platform and doing QA) but if we do more than 800 concurrent calls kamailio starts failing and slowing the requests, The scenario is the following: I am sending 5000 calls at 100 CPS on TCP and I am having this route in the configuration file: # Handle the calls to api route[CALL_API] { xlog("L_NOTICE"," Call request $var(call_request) method: $rm \n"); $var(loop_true)=1; while ($var(loop_true)) { if(!lua_run("call_request","$var(call_request)","$sht(token=>new_token)")) { xlog("L_NOTICE", "SCRIPT: failed to execute lua function!\n"); } if ($var(loop_true)){ sleep("1"); } } xlog("L_NOTICE", "SCRIPT: Sucess to execute lua function!\n"); } So I am executing a LUA script for the requests and the API response will tell the destination of the call. At 800 concurrent calls, I can notice SLOWLINESS in the SIPp testing tool, and then I will start receiving 408 timeouts. This sounds like a load issue or something because I believe there's a time that it's been added every time we hit the backend and wait for the response. any help will be appreciated Regards Gio

2 1

dispatcher socket attrs binding
by David Villasmil 29 Mar '22

29 Mar '22

Hello guys, Im starting kamailio with listen=0.0.0.0:5060 and in my dispatcher list i have in the attrs something like: socket=udp:172.17.0.2:5060 That IP address belongs to kamailio's host and it is binging to it. Is it right i can't set a specific IP in the socket attribute if listen is 0.0.0.0? maybe i should be able to? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 4

kamailio - mariadb hosted on remote server - issue
by Youssef Boujraf 29 Mar '22

29 Mar '22

Dear all, I don't belive to connect Kamailio to a remote Mariadb will take too much time. STEP 1 : KAMAILIO : -------------------------------- $ sudo kamailio -v version: kamailio 5.5.4 (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, 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 STEP 2 : Set the kamctlrc config ------------------------------------------------ KAMAILIO IP : 192.168.1.41 MARIADB IP : 192.168.1.39 - /etc/kamailio/kamctlrc SIP_DOMAIN=example.be DBENGINE=MYSQL DBHOST=192.168.1.39 // where the mariadb is installed DBNAME=kamailio DBRWUSER="kamailio" DBRWPW="XX" DBROUSER="kamailioro" DBROPW="XX" DBACCESSHOST=192.168.1.41 // where kamctl is launched (here kamailio srv) DBROOTUSER="root" DBROOTPW="kamailio" CHARSET="latin1" STEP 3 : MARIADB : hosted on different srv than kamailio -------------------------------------------------------------------------------------- IP : 192.168.1.39 - /etc/mysql/mariadb.conf.d/50-server.cnf # this is only for the mysqld standalone daemon [mysqld] # Instead of skip-networking the default is now to listen only on # localhost which is more compatible and is not less secure. #bind-address = 127.0.0.1 bind-address = 0.0.0.0 STEP 4 : TEST : Reote connection with the DB ---------------------------------------------------------------------- REMOTE IP : 192.168.1.41 DB IP : 192.168.1.39 $ sudo mysql -u root -h 192.168.1.39 -p Enter password: Welcome to the MariaDB monitor. Commands end with ; or \g. Your MariaDB connection id is 298 Server version: 10.5.15-MariaDB-1:10.5.15+maria~bullseye mariadb.org binary distribution Copyright (c) 2000, 2018, Oracle, MariaDB Corporation Ab and others. Type 'help;' or '\h' for help. Type '\c' to clear the current input statement. MariaDB [(none)]>The connection is established with success. STEP 5 : KAMAILIO DB : creation ---------------------------------------------------- $ sudo kamdbctl create -e \E[37;33mINFO: creating database kamailio ... -e \E[37;33mINFO: granting privileges to database kamailio ... -e \E[37;33mINFO: creating standard tables into kamailio ... -e \E[37;33mINFO: Core Kamailio tables successfully created. Create the presence related tables? (y/n): y -e \E[37;33mINFO: creating presence tables into kamailio ... -e \E[37;33mINFO: Presence tables successfully created. Create the tables for imc cpl siptrace domainpolicy carrierroute drouting userblocklist htable purple uac pipelimit mtree sca mohqueue rtpproxy rtpengine secfilter? (y/n): y -e \E[37;33mINFO: creating extra tables into kamailio ... -e \E[37;33mINFO: Extra tables successfully created. Create the tables for uid_auth_db uid_avp_db uid_domain uid_gflags uid_uri_db? (y/n): y -e \E[37;33mINFO: creating uid tables into kamailio ... -e \E[37;33mINFO: UID tables successfully created. STEP 6 : DB : verify the users creation and rights -------------------------------------------------------------------------- MariaDB [(none)]> select user, host from mysql.user; +-------------+--------------+ | User | Host | +-------------+--------------+ | kamailio | 192.168.1.39 | | kamailioro | 192.168.1.39 | | kamailio | 192.168.1.41 | | kamailioro | 192.168.1.41 | | root | 192.168.1.41 | | kamailio | localhost | | kamailioro | localhost | | mariadb.sys | localhost | | mysql | localhost | | root | localhost | +-------------+--------------+ 10 rows in set (0.001 sec) MariaDB [(none)]> SHOW GRANTS FOR kamailio(a)192.168.1.41; +--------------------------------------------------------------------------------------------------------------------+ | Grants for kamailio(a)192.168.1.41 | +--------------------------------------------------------------------------------------------------------------------+ | GRANT USAGE ON *.* TO `kamailio`(a)`192.168.1.41` IDENTIFIED BY PASSWORD '*157PASSWORD' | | GRANT ALL PRIVILEGES ON `kamailio`.* TO `kamailio`(a)`192.168.1.41` | +--------------------------------------------------------------------------------------------------------------------+ 2 rows in set (0.000 sec) MariaDB [(none)]> SHOW GRANTS FOR kamailio(a)192.168.1.39; +--------------------------------------------------------------------------------------------------------------------+ | Grants for kamailio(a)192.168.1.39 | +--------------------------------------------------------------------------------------------------------------------+ | GRANT USAGE ON *.* TO `kamailio`(a)`192.168.1.39` IDENTIFIED BY PASSWORD '*157PASSWORD' | | GRANT ALL PRIVILEGES ON `kamailio`.* TO `kamailio`(a)`192.168.1.39` | +--------------------------------------------------------------------------------------------------------------------+ 2 rows in set (0.000 sec) MariaDB [(none)]> SHOW GRANTS FOR kamailio@localhost; +-----------------------------------------------------------------------------------------------------------------+ | Grants for kamailio@localhost | +-----------------------------------------------------------------------------------------------------------------+ | GRANT USAGE ON *.* TO `kamailio`@`localhost` IDENTIFIED BY PASSWORD '*157PASSWORD' | | GRANT ALL PRIVILEGES ON `kamailio`.* TO `kamailio`@`localhost` | +-----------------------------------------------------------------------------------------------------------------+ 2 rows in set (0.000 sec)STEP 7 : ADDRESS TABLE : add new ip address$ sudo systemctl restart kamailio sipsecure@kamailio:/etc$ sudo kamctl address add 0 192.168.1.26 32 0 HP -e \E[37;33mINFO: execute '/sbin/kamctl address reload' to synchronize cache and database STEP 8 : DB : verify if the new ip address is created ? Yes MariaDB [(none)]> select * from kamailio.address; +----+-----+--------------+------+------+------+ | id | grp | ip_addr | mask | port | tag | +----+-----+--------------+------+------+------+ | 1 | 0 | 192.168.1.26 | 32 | 0 | HP | +----+-----+--------------+------+------+------+ 1 row in set (0.000 sec) STEP 9 : Reload the content of the DB into kamailio. It failed ---------------------------------------------------------------------------------------- The host: '192.168.1.41' is kamailio (remote client)$ sudo kamctl address reload { "jsonrpc": "2.0", "error": { "code": 500, "message": "Reload failed." }, "id": 12397 } In /var/log/daemon.log Mar 28 17:09:49 db04srv mariadbd[2430]: 2022-03-28 17:09:49 281 [Warning] Aborted connection 281 to db: 'kamailio' user: 'kamailio' host: '192.168.1.41' (Got an error reading communication packets) Mar 28 17:09:49 db04srv mariadbd[2430]: 2022-03-28 17:09:49 273 [Warning] Aborted connection 273 to db: 'kamailio' user: 'kamailio' host: '192.168.1.41' (Got an error reading communication packets) Mar 28 17:09:49 db04srv mariadbd[2430]: 2022-03-28 17:09:49 295 [Warning] Aborted connection 295 to db: 'kamailio' user: 'kamailio' host: '192.168.1.41' (Got an error reading communication packets) Mar 28 17:09:49 db04srv mariadbd[2430]: 2022-03-28 17:09:49 264 [Warning] Aborted connection 264 to db: 'kamailio' user: 'kamailio' host: '192.168.1.41' (Got an error reading communication packets) Now, if I am trying to restart "kamailio" service, I got an error : Mar 28 17:36:30 kamailio /usr/sbin/kamailio[14196]: DEBUG: <core> [core/sr_module.c:940]: init_mod(): permissions Mar 28 17:36:30 kamailio /usr/sbin/kamailio[14196]: DEBUG: permissions [address.c:179]: reload_address_db_table(): Number of rows in address table: 1 Mar 28 17:36:30 kamailio /usr/sbin/kamailio[14196]: DEBUG: permissions [address.c:191]: reload_address_db_table(): failure during checks of database value 1 (group) in address table Mar 28 17:36:30 kamailio /usr/sbin/kamailio[14196]: ERROR: permissions [address.c:233]: reload_address_db_table(): database problem - invalid record Mar 28 17:36:30 kamailio /usr/sbin/kamailio[14196]: CRITICAL: permissions [address.c:536]: init_addresses(): reload of address table failed Mar 28 17:36:30 kamailio /usr/sbin/kamailio[14196]: ERROR: permissions [permissions.c:657]: mod_init(): failed to initialize the allow_address function Mar 28 17:36:30 kamailio /usr/sbin/kamailio[14196]: ERROR: <core> [core/sr_module.c:942]: init_mod(): Error while initializing module permissions (/usr/lib/x86_64-linux-gnu/kamailio/modules/permissions.so) Mar 28 17:36:30 kamailio kamailio: DEBUG: <core> [core/cfg.y:1860]: yyparse(): loading module permissions.so Mar 28 17:36:30 kamailio kamailio: DEBUG: <core> [core/sr_module.c:513]: ksr_locate_module(): found module to load </usr/lib/x86_64-linux-gnu/kamailio/modules/permissions.so> Mar 28 17:36:30 kamailio kamailio: DEBUG: <core> [core/sr_module.c:563]: load_module(): trying to load </usr/lib/x86_64-linux-gnu/kamailio/modules/permissions.so> Mar 28 17:36:30 kamailio kamailio: DEBUG: <core> [core/kemi.c:3090]: sr_kemi_modules_add(): adding module: permissions Mar 28 17:36:30 kamailio kamailio: DEBUG: <core> [core/cfg.lex:1963]: pp_define(): defining id: MOD_permissions Mar 28 17:36:30 kamailio kamailio: DEBUG: <core> [core/modparam.c:112]: set_mod_param_regex(): 'permissions' matches module 'permissions' Mar 28 17:36:30 kamailio kamailio: DEBUG: <core> [core/sr_module.c:811]: find_param_export(): found <db_url> in module permissions [/usr/lib/x86_64-linux-gnu/kamailio/modules/permissions.so] Mar 28 17:36:30 kamailio kamailio: DEBUG: <core> [core/modparam.c:128]: set_mod_param_regex(): found <db_url> in module permissions [/usr/lib/x86_64-linux-gnu/kamailio/modules/permissions.so] Mar 28 17:36:30 kamailio kamailio: DEBUG: <core> [core/modparam.c:112]: set_mod_param_regex(): 'permissions' matches module 'permissions' Mar 28 17:36:30 kamailio kamailio: DEBUG: <core> [core/sr_module.c:811]: find_param_export(): found <load_backends> in module permissions [/usr/lib/x86_64-linux-gnu/kamailio/modules/permissions.so] Mar 28 17:36:30 kamailio kamailio: DEBUG: <core> [core/modparam.c:128]: set_mod_param_regex(): found <load_backends> in module permissions [/usr/lib/x86_64-linux-gnu/kamailio/modules/permissions.so] Mar 28 17:36:31 kamailio kamailio: DEBUG: <core> [core/sr_module.c:719]: find_mod_export_record(): found export of <allow_source_address> in module permissions [/usr/lib/x86_64-linux-gnu/kamailio/modules/permissions.so] Mar 28 17:36:31 kamailio kamailio: DEBUG: <core> [core/sr_module.c:719]: find_mod_export_record(): found export of <allow_source_address_group> in module permissions [/usr/lib/x86_64-linux-gnu/kamailio/modules/permissions.soIf I remove the record from the database manually, I can restart kamailio but no way to add any address. Thanks if someone could highlight my path. Best Regards, -- Youssef

2 4

avps on regexps?
by David Villasmil 28 Mar '22

28 Mar '22

Hello guys, Is it possible to use apvs on regexps like: $avp(myvar)=$(re{uri.user}{re.subst,/^\+$avp(some_var)/$avp(some_var)/}); doesn't seem possible Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 3

proxy and forward
by jaflong jaflong 28 Mar '22

28 Mar '22

2 1

nathelper OPTIONS ping
by Xuo Guoto 28 Mar '22

28 Mar '22

Hello all, I am trying to periodically send SIP OPTIONS to all connected WebRTC clients from Kamailio. The functionality is similar to qualify=yes of Asterisk. Following are the configuration changes I have made to get this working. #!define FLB_NATSIPPING 7 <snip/> loadmodule "nathelper.so" <snip/> # ----- nathelper params ----- modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "natping_interval", 20) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") <snip/> # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { xlog("L_INFO", "Processing REGISTER in route[REGISTRAR]\n"); if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # do SIP NAT pinging xlog("L_INFO", "Setting FLB_NATSIPPING\n"); setbflag(FLB_NATSIPPING); } if (!save("location")) { sl_reply_error(); } xlog("L_INFO", "Successfully processed REGISTER in route[REGISTRAR]\n"); exit; } } When the WebRTC client registers, I can see the log: Setting FLB_NATSIPPING, but SIP OPTIONS packets are not seen. I am checking it using the Chrome console, at client side as well as sipdump module in server side. Do I have to do any additional configuration? I am not posting the full config file here so that its easy to focus on the relevant parts, but can do that if needed. Thanks and regards, X.

2 1

Re: [SR-Users] Kamailio : kamctl address reload with HAProxy
by Henning Westerholt 28 Mar '22

28 Mar '22

Hello, please keep the list in CC – that others can comment as well. Can you actually connect from the kamailio system e.g. with mysql client? If yes, as I said – I would try to debug the issue further with network analysis tools. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Paclinio <thomas.leutscher(a)gmail.com> Sent: Monday, March 28, 2022 1:11 PM To: Henning Westerholt <hw(a)gilawa.com> Subject: Re: [SR-Users] Kamailio : kamctl address reload with HAProxy Hello, KAMCTL : I've resolved the problem with kamctl that now can communicate with my remote Cluster by adding this line in kamctlrc. // HAPROXY = 192.168.1.37 DBHOST=192.168.1.37 DBACCESSHOST=192.168.1.37 I can now add and reload new addresses with success. KAMAILIO + HAPROXY: But now, I've this issue when restarting kamailio. Mar 28 10:54:39 kamailio /usr/sbin/kamailio[8416]: ERROR: permissions [address.c:233]: reload_address_db_table(): database problem - invalid record Mar 28 10:54:39 kamailio /usr/sbin/kamailio[8416]: ERROR: permissions [permissions.c:657]: mod_init(): failed to initialize the allow_address function Mar 28 10:54:39 kamailio /usr/sbin/kamailio[8416]: ERROR: <core> [core/sr_module.c:942]: init_mod(): Error while initializing module permissions (/usr/lib/x86_64-linux-gnu/kamailio/modules/permissions.so) My DBURL is #!substdef "!DB_USER!kamailio!g" #!substdef "!DB_PASS!XXXXX!g" #!substdef "!DB_HOST!192.168.1.37!g" // HA PROXY #!substdef "!DB_PORT!MY_PORT_DB!g" #!substdef "!DB_NAME!kamailio!g" #!define DBURL "mysql://DB_USER:DB_PASS@DB_HOST/DB_NAME" KAMAILIO + REMOTE DB : Even I using a remote database I'm getting the same issue. I've discovered the connection between kamailio and remote database is aborded instantly after restarting kamailio. Mar 28 11:47:30 db03srv mariadbd[874]: 2022-03-28 11:47:30 44076 [Warning] Aborted connection 44076 to db: 'kamailio' user: 'kamailio' host: '192.168.1.41' (Got an error reading communication packets) Thank you [Das Bild wurde vom Absender entfernt.]<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> Virus-free. www.avast.com<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> Le dim. 27 mars 2022 à 22:52, Henning Westerholt <hw(a)gilawa.com<mailto:hw@gilawa.com>> a écrit : Hello, it seems the data that gets delivered to the Kamailio is corrupted. Analyse it further e.g. with a network trace, wireshark etc.. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Paclinio Sent: Friday, March 25, 2022 5:22 PM To: sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Subject: [SR-Users] Kamailio : kamctl address reload with HAProxy Dear colleagues, I've installed a Galera cluster with a HAProxy that kamailio interacts with. Kamailio's DBURL is linked to the proxy that will contact the databases. Kamailio works fine except when I need to reload something from the database (Using kamctl). When cache and database are the same I receive sudo kamctl address reload { "jsonrpc": "2.0", "result": "Reload OK", "id": 13111 } Now when I have something using kamctl, it still works and is replaced on all my databases. sipsecure@kamailio:~$ sudo kamctl address add 0 192.168.48.15 32 0 Test -e \E[37;33mINFO: execute '/sbin/kamctl address reload' to synchronize cache and database Then when i try to reload i receive sipsecure@kamailio:~$ sudo kamctl address reload { "jsonrpc": "2.0", "error": { "code": 500, "message": "Reload failed." }, "id": 13504 } And when I'm going to read my logs to see a error, I only see that little sentence Mar 25 17:14:40 kamailio /usr/sbin/kamailio[13343]: ERROR: permissions [address.c:233]: reload_address_db_table(): database problem - invalid record Could I have some explanation ? Thank you. [Das Bild wurde vom Absender entfernt.]<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> Virus-free. www.avast.com<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…>

2 2

t_suspend / t_continue & insert_hf
by Fuad Trle 28 Mar '22

28 Mar '22

Hello, I'm inserting some headers after getting value from http_async_query. If a call is relayed, I see them. But If I t_suspend and t_continue call (before time out), I do not see added headers. Here is flow: route(START) t_newtran(); #.. prep for query http_async_query("url","HTTP"); route(HTTP) #.. collect and using data from query insert_hf("$var(key): $var(value)\r\n"); t_store(); if (registered("location") { route(RELAY); } else { t_suspend(); exit; } route(AWAKE) t_continue(); I see that the branch index ($T_branch_idx) is 0 at start. 1 after http query, +1 if there is forking or location. +1 after t_suspend. It looks like t_continue uses a branch with index after t_suspend. I tried with t_save_lumps() to carry over changes. Any idea how to do this?

3 3

What is the status of nat_traversal.so
by hanxiaofang 28 Mar '22

28 Mar '22

Hello Kamailio community, What is the status of nat_traversal.so? https://www.kamailio.org/wiki/alphaindexes/devel/modfunctions Could not find any functions for this module here. Regards hanxiaofang

2 1

Kamailio : kamctl address reload with HAProxy
by Paclinio 27 Mar '22

27 Mar '22

Dear colleagues, I've installed a Galera cluster with a HAProxy that kamailio interacts with. Kamailio's DBURL is linked to the proxy that will contact the databases. Kamailio works fine except when I need to reload something from the database (Using kamctl). When cache and database are the same I receive sudo kamctl address reload > { > "jsonrpc": "2.0", > "result": "Reload OK", > "id": 13111 > } Now when I have something using kamctl, it still works and is replaced on all my databases. sipsecure@kamailio:~$ sudo kamctl address add 0 192.168.48.15 32 0 Test > -e \E[37;33mINFO: execute '/sbin/kamctl address reload' to synchronize > cache and database Then when i try to reload i receive sipsecure@kamailio:~$ sudo kamctl address reload > { > "jsonrpc": "2.0", > "error": { > "code": 500, > "message": "Reload failed." > }, > "id": 13504 > } And when I'm going to read my logs to see a error, I only see that little sentence Mar 25 17:14:40 kamailio /usr/sbin/kamailio[13343]: ERROR: permissions > [address.c:233]: reload_address_db_table(): database problem - invalid > record Could I have some explanation ? Thank you. <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> Virus-free. www.avast.com <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

2 1

Using DNS log in
by Steve 26 Mar '22

26 Mar '22

Hello, I have a very basic Kamailio 5.4 configuration on Debian 11 with RTPProxy 2.2 behind a firewall. It works fine with the sip client logging into the server public ip address. However, the sip client is not able to log in when using a dns domain name directed to that public ip address. Is there a specific setting in the Kamailio and/or RTPProxy config files for using a dns domain name instead of a numeric ip address? Also, is there a Kamailio module that allows use of encrypted keys instead of passwords when logging in? Thanks for any info, Steve

2 2

Forward and proxy
by jaflong 26 Mar '22

26 Mar '22

Hi Kamailio community Any advise on how to archive this sip.mydomain.com sip.mypbx.com sip_phone =======sip/tls=======> KAMAILIO =========sip/udp=======>PBX sip_phone should register to Kamilio with tls Kamailio should proxy register to PBX with udp sip_phone invites should be proxied to PBX in sip_client setting he will point to sip.mydomain.com as his server Regards

2 1

(no subject)
by jaflong jaflong 26 Mar '22

26 Mar '22

2 1

t_suspend / t_continue & insert_hf
by Fuad Trle 25 Mar '22

25 Mar '22

Hello, I'm inserting some headers after getting value from http_async_query. If a call is relayed, I see them. But If I t_suspend and t_continue call (before time out), I do not see added headers. Here is flow: route(START) t_newtran(); #.. prep for query http_async_query("url","HTTP"); route(HTTP) #.. collect and using data from query insert_hf("$var(key): $var(value)\r\n"); t_store(); if (registered("location") { route(RELAY); } else { t_suspend(); exit; } route(AWAKE) t_continue(); I see that the branch index ($T_branch_idx) is 0 at start. 1 after http query, +1 if there is forking or location. +1 after t_suspend. It looks like t_continue uses a branch with index after t_suspend. I tried with t_save_lumps() to carry over changes. Any idea how to do this?

1 0

Kamailio SCSCF forward SIP INVITE over TCP
by Weiqi Yao 24 Mar '22

24 Mar '22

Hello community, We currently have some issues with Kamailio SCSCF forwarding a large SIP UDP INVITE packet(size around 3000bytes), it got dropped somewhere in the ethernet. I tried increase the MTU size on both sides' interface and even define it on the switch port, but still no luck. So I am thinking to force SCSCF to send this INVITE over TCP, I followed this link: https://andersbrownworth.com/cms/264/, enabled naptr in SCSCF config and configured NAPTR and SRV DNS record, but the SCSCF still stop even forward it out. I think I did the DNS part right cause when I switch the NAPTR record to point to UDP it worked. Anyone have some experience with configure SCSCF forwarding SIP through TCP? BR, Weiqi

1 0

JsSIP:UA Request-URI does not point to us
by Xuo Guoto 24 Mar '22

24 Mar '22

Hello all, I am facing an issue with JsSIP not recognizing replies from Kamailio. the call sequence goes as follows: INVITE -----------------------------> <-------------------------------SIP/2.0 100 Trying <-------------------------------SIP/2.0 180 Ringing <-------------------------------SIP/2.0 200 OK ACK --------------------------------> <-------------------------------SIP/2.0 200 OK ACK --------------------------------> <-------------------------------SIP/2.0 200 OK ACK --------------------------------> <-------------------------------SIP/2.0 200 OK ACK --------------------------------> <-------------------------------SIP/2.0 200 OK ACK --------------------------------> <-------------------------------SIP/2.0 200 OK ACK --------------------------------> <-------------------------------SIP/2.0 200 OK ACK --------------------------------> <-------------------------------BYE 404 Not Found ----------------------> When JsSIP receives ACK it prints an error: JsSIP:UA Request-URI does not point to us From another thread with similar issue at https://groups.google.com/g/sip_js/c/uiaXS_qc2n8 it could be that JsSIP is not recognizing the GRUU is pointing towards it. In the INVITE message I can see a line "Record-Route: > <mailto:sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com> <mailto:sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com> < <mailto:sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com> <mailto:sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com> mailto:;expires=600;received=mailto:;gr=urn:uuid:b8a691ff-f678-4866-8cf0-780d670e7e33 < <mailto:sip:9747815015@erx-staging-q01.mydomain.com> <mailto:sip:Stg-CQD0r2-10020005@10.10.1.9> > <mailto:sip:Stg-CQD0r2-10020005@10.10.1.9> <mailto:sip:9747815015@erx-staging-q01.mydomain.com> > <mailto:sip:Stg-CQD0r2-10020005@10.10.1.9> <mailto:sip:9747815015@erx-staging-q01.mydomain.com> <mailto:sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com> > <mailto:sip:Stg-CQD0r2-10020005@10.10.1.9> <mailto:sip:9747815015@erx-staging-q01.mydomain.com> <mailto:sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com> < mailto:sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com <mailto:sip:9747815015@erx-staging-q01.mydomain.com> <mailto:sip:Stg-CQD0r2-10020005@10.10.1.9> < mailto:sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com <mailto:sip:9747815015@erx-staging-q01.mydomain.com> <mailto:sip:Stg-CQD0r2-10020005@10.10.1.9> > <mailto:sip:Stg-CQD0r2-10020005@10.10.1.9> <mailto:sip:9747815015@erx-staging-q01.mydomain.com>

2 4

Re: [SR-Users] sr-users Digest, Vol 202, Issue 25
by 韩小仿 24 Mar '22

24 Mar '22

Hi, Henning Westerholt Pls close case1 Thanks! hanxiaofang From: sr-users-request Date: 2022-03-24 19:00 To: sr-users Subject: sr-users Digest, Vol 202, Issue 25 Send sr-users mailing list submissions to sr-users(a)lists.kamailio.org To subscribe or unsubscribe via the World Wide Web, visit https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.kamailio.org You can reach the person managing the list at sr-users-owner(a)lists.kamailio.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Re: {param.in, name[, delimiter]} error when delimiter is comma (Henning Westerholt) 2. TOPOH is generating new Call-ID for BYE (Marrold) 3. Re: SIPS Errors on Kamailio (Christopher Vincent) 4. bug in tm module? (Daniel Paape) 5. Re: bug in tm module? (Henning Westerholt) 6. carrierroute with non-numeric characters (Arek ?azarczyk) 7. Re: carrierroute with non-numeric characters (Henning Westerholt) 8. Re: carrierroute with non-numeric characters (Arek ?azarczyk) 9. Re: carrierroute with non-numeric characters (Henning Westerholt) ---------------------------------------------------------------------- Message: 1 Date: Wed, 23 Mar 2022 12:49:40 +0000 From: Henning Westerholt <hw(a)gilawa.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Cc: ??? <13715209697(a)139.com> Subject: Re: [SR-Users] {param.in, name[, delimiter]} error when delimiter is comma Message-ID: <PAXPR05MB855899124FF2FD96E43C7E7FBF189(a)PAXPR05MB8558.eurprd05.prod.outlook.com> Content-Type: text/plain; charset="utf-8" Hello, Please have a look to the documentation, the issue is described there, and a workaround also suggested. http://www.kamailio.org/wiki/cookbooks/devel/transformations#parameters_lis… Cheers, Henning -- Henning Westerholt ? https://skalatan.de/blog/ Kamailio services ? https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of ??? Sent: Wednesday, March 23, 2022 11:12 AM To: sr-users <sr-users(a)lists.kamailio.org> Subject: [SR-Users] {param.in, name[, delimiter]} error when delimiter is comma Hi, all Kamailio5.5.4 $var(s) = "a=1;b=2;c=3"; $var(s1) = $(var(s){param.value,b}); // work well $var(s) = "a=1|b=2|c=3"; $var(s1) = $(var(s){param.value,b,|}); // work well // but $var(s) = "a=1,b=2,c=3"; $var(s1) = $(var(s){param.value,b,,}); ERROR: <core> [core/pvapi.c:1131]: pv_parse_spec2(): invalid parsing in [$(var(s){param.value,b,,})] at (4) CRITICAL: <core> [core/cfg.y:3683]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 366, column 13-38: Can't get from cache: $(var(s){param.value,b,,}) how to fix this? Thanks! ________________________________ hanxiaofang -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20220323/836492af/…> ------------------------------ Message: 2 Date: Wed, 23 Mar 2022 15:05:58 +0000 From: Marrold <kamailio(a)marrold.co.uk> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: [SR-Users] TOPOH is generating new Call-ID for BYE Message-ID: <CAC-Lcd8H5H=V4ua7y7Fv-t=LFoC=JZC9FZaKH5JtO+K1QJNxcw(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" Hi, I have enabled the TOPOH module with mask_callid enabled, but it's generating a new Call-ID for a BYE which gets rejected with a 481 downstream. If I disable the TOPOH module it works as expected. Does anyone know where I should start looking? SIP-wise the initial BYE seems compliant. This is Kamailio 5.5.3 on Debian 11 Thanks Matthew -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20220323/be811b10/…> ------------------------------ Message: 3 Date: Wed, 23 Mar 2022 18:13:54 +0000 From: Christopher Vincent <CDV(a)redwoodtech.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] SIPS Errors on Kamailio Message-ID: <3C0E2FDC7BD2334492F46C0DA6E07428014E8D1BD0(a)RTG-Exchange01.Redwood.com> Content-Type: text/plain; charset="utf-8" Recompiled Kamailio from source kamailio -v version: kamailio 5.5.4 (x86_64/linux) 4c8938 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 4c8938 compiled on 14:21:51 Mar 23 2022 with gcc 4.8.5 Compiled a newer version of OpenSSL openssl version OpenSSL 1.1.1m 14 Dec 2021 Then followed through this http://www.kamailio.org/wiki/tutorials/tls/howto-openssl-1-0 Errors changed slightly, but issue still persists kamailio -c loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(3878) ERROR: <core> [core/sr_module.c:570]: load_module(): could not open module </usr/local/lib64/kamailio/modules/tls.so>: /usr/local/lib64/kamailio/modules/tls.so: undefined symbol: OPENSSL_sk_num 0(3878) CRITICAL: <core> [core/cfg.y:3684]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 106, column 12-19: failed to load module 0(3878) INFO: pv [pv_shv.c:60]: shvar_init_locks(): locks array size 16 0(3878) ERROR: <core> [core/modparam.c:181]: set_mod_param_regex(): No module matching <tls> found 0(3878) CRITICAL: <core> [core/cfg.y:3687]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 253, column 72: Can't set module parameter 0(3878) ERROR: <core> [core/modparam.c:181]: set_mod_param_regex(): No module matching <tls> found 0(3878) CRITICAL: <core> [core/cfg.y:3687]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 254, column 72: Can't set module parameter 0(3878) ERROR: <core> [core/modparam.c:181]: set_mod_param_regex(): No module matching <tls> found 0(3878) CRITICAL: <core> [core/cfg.y:3687]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 255, column 68: Can't set module parameter 0(3878) ERROR: <core> [core/modparam.c:181]: set_mod_param_regex(): No module matching <tls> found 0(3878) CRITICAL: <core> [core/cfg.y:3687]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 260, column 39: Can't set module parameter ERROR: bad config file (5 errors) (parsing code: 0) 0(3878) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized sed /usr/local/etc/kamailio/kamailio.cfg -n -e 106p -e 253p -e 254p -e 255p -e 260p loadmodule "tls.so" modparam("tls", "private_key", "/etc/kamailio/star.redwoodtest.com.pem") modparam("tls", "certificate", "/etc/kamailio/star.redwoodtest.com.pem") modparam("tls", "ca_list", "/etc/kamailio/star.redwoodtest.com.pem") modparam("tls", "tls_method", "TLSv1+") Thanks, Chris From: Christopher Vincent Sent: 21 March 2022 09:41 To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Cc: Christopher Vincent <CDV(a)redwoodtech.com> Subject: RE: [SR-Users] SIPS Errors on Kamailio Hi Karsten, Sergey, Thank you for your responses. Kamailio module appears to be installed rpm -ql kamailio-tls /usr/lib64/kamailio/modules/auth_identity.so /usr/lib64/kamailio/modules/tls.so /usr/lib64/kamailio/openssl_mutex_shared /usr/lib64/kamailio/openssl_mutex_shared/openssl_mutex_shared.so /usr/share/doc/kamailio/modules/README.auth_identity /usr/share/doc/kamailio/modules/README.tl Kamailio was installed from the repository using yum / dnf https://rpm.kamailio.org/centos/kamailio.repo Load module order has been corrected to as below loadmodule "sl.so" loadmodule "tls.so" loadmodule "jsonrpcs.so" loadmodule "db_mysql.so" loadmodule "kex.so" ... Kamailio ?c still showed the same errors Thanks, Chris From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Safarov Sent: 19 March 2022 07:38 To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> Subject: Re: [SR-Users] SIPS Errors on Kamailio SECURITY WARNING: This email is from an external source - do not open or click any attachments if you suspect the email is suspicious. Please report any suspicious emails to Information Security (InformationSecurity(a)redwoodtech.com<mailto:InformationSecurity@redwoodtech.com>) In your config "tls" module after "sl" module. loadmodule "sl.so" loadmodule "tls.so" you need load tls module before "jsonrpcs" module. On Fri, Mar 18, 2022 at 9:59 PM Karsten Horsmann <khorsmann(a)gmail.com<mailto:khorsmann@gmail.com>> wrote: Hi, are you sure the Kamailio tls module is on your system? Check the module path for tls.so like this or if you have build it from source? rpm -ql kamailio-tls /usr/lib64/kamailio/modules/auth_identity.so /usr/lib64/kamailio/modules/tls.so /usr/lib64/kamailio/openssl_mutex_shared /usr/lib64/kamailio/openssl_mutex_shared/openssl_mutex_shared.so /usr/share/doc/kamailio/modules/README.auth_identity /usr/share/doc/kamailio/modules/README.tls Christopher Vincent <CDV(a)redwoodtech.com<mailto:CDV@redwoodtech.com>> schrieb am Fr., 18. M?rz 2022, 12:37: Hi, Kamailio / RTPEngine was set up on CentOS 8 running SIP to SIPS and RTP to SDES SRTP conversion. This worked as expected Attempted to duplicate the setup on RHEL but errors were seen. These errors were present on both RHEL 7 / RHEL 8. The errors seen were as below kamailio -c loading modules under config path: /usr/lib64/kamailio/modules/ 0(9165) ERROR: tls [tls_init.c:611]: tls_pre_init(): Unable to set the memory allocation functions 0(9165) ERROR: tls [tls_init.c:613]: tls_pre_init(): libssl current mem functions - m: 0x7f7a77c367a0 r: 0x7f7a77c367f0 f: 0x7f7a77c36770 0(9165) ERROR: tls [tls_init.c:615]: tls_pre_init(): module mem functions - m: 0x7f7a72db7653 r: 0x7f7a72db769f f: 0x7f7a72db76fc 0(9165) ERROR: tls [tls_init.c:617]: tls_pre_init(): Be sure tls module is loaded before any other module using libssl (can be loaded first to be safe) 0(9165) ERROR: <core> [core/sr_module.c:590]: load_module(): /usr/lib64/kamailio/modules/tls.so: mod_register failed 0(9165) CRITICAL: <core> [core/cfg.y:3683]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 137, column 12-19: failed to load module 0(9165) INFO: pv [pv_shv.c:60]: shvar_init_locks(): locks array size 16 0(9165) ERROR: <core> [core/modparam.c:181]: set_mod_param_regex(): No module matching <tls> found 0(9165) CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 249, column 72: Can't set module parameter 0(9165) ERROR: <core> [core/modparam.c:181]: set_mod_param_regex(): No module matching <tls> found 0(9165) CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 250, column 72: Can't set module parameter 0(9165) ERROR: <core> [core/modparam.c:181]: set_mod_param_regex(): No module matching <tls> found 0(9165) CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 251, column 68: Can't set module parameter 0(9165) ERROR: <core> [core/modparam.c:181]: set_mod_param_regex(): No module matching <tls> found 0(9165) CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 256, column 39: Can't set module parameter ERROR: bad config file (5 errors) (parsing code: 0) 0(9165) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized The kamailio config was exactly the same as on the CentOS systems and started as below /* Server ports: */ #!substdef "!SIP_PORT!5060!g" #!substdef "!SIPS_PORT!5061!g" /* Listen addresses */ #!substdef "!UDP_LOCAL_ADDR!udp:SERVER_IP_ADDR:SIP_PORT!g" #!substdef "!TCP_LOCAL_ADDR!tcp:SERVER_IP_ADDR:SIPS_PORT!g" /* Server connections: */ #!ifndef MAX_CONNECTIONS #!define MAX_CONNECTIONS 8192 #!endif ##!define WITH_DEBUG /* Transaction and branch flags: FLT_ - per transaction (message) flags FLB_ - per branch flags */ #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLT_OUT 8 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 #!define KAMAILIODBURL1 "mysql://kamailio:kamailiorw@localhost/kamailio" #!define WITH_TLS enable_tls=1 listen=tls:<ipaddr>:5062 ####### Global Parameters ######### ### LOG Levels: ALERT=-5, BUG=-4, CRIT=-3, ERR=-1, WARN=0, NOTICE=1, INFO=2, DBG=3 #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 /* display memory usage on exit */ mem_summary=15 /* join free memory fragments */ mem_join=1 /* proxy will fork and run in daemon mode */ /* one process will be created for each network interface the proxy listens to and for each protocol (TCP/UDP), multiplied with the value of 'children' parameter */ fork=yes children=8 listen=TCP_LOCAL_ADDR listen=UDP_LOCAL_ADDR /* life time of TCP connection when there is no traffic - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 /* sip over websockets may not specify a content length header */ tcp_accept_no_cl=yes /* buffer size used for tcp reads, limits the maximum message size (SIP, HTTP) that can be received over tcp */ tcp_rd_buf_size=65536 /* max number of tcp connections */ tcp_max_connections=MAX_CONNECTIONS ####### Modules Section ######## # set paths to location of modules mpath="/usr/lib64/kamailio/modules/" loadmodule "jsonrpcs.so" loadmodule "db_mysql.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "dispatcher.so" loadmodule "cfgutils.so" loadmodule "textopsx.so" loadmodule "nathelper.so" loadmodule "uac.so" loadmodule "ipops.so" loadmodule "debugger.so" loadmodule "exec.so" loadmodule "avpops.so" loadmodule "sqlops.so" loadmodule "rtpengine.so" loadmodule "sl.so" loadmodule "tls.so" # ----------------- setting module-specific parameters --------------- # ----- usrloc params ----- # store contacts in memory only modparam("usrloc", "db_mode", 0) # hash size of 16,384 modparam("usrloc", "hash_size", 14) # removes contact if ws disconnects modparam("usrloc", "handle_lost_tcp", 1) modparam("tm|usrloc", "xavp_contact", "ulattrs") # ----- jsonrpcs params ----- modparam("jsonrpcs", "fifo_name", "/tmp/kamailio_jsonrpc.fifo") modparam("jsonrpcs", "dgram_socket", "/tmp/kamailio_rpc.sock") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) #default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- uac params ----- modparam("uac", "restore_mode", "none") # ----- registrar params ----- modparam("registrar", "method_filtering", 1) modparam("registrar", "max_contacts", 1) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # disable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direction of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) # ----- dispatcher params ----- modparam("dispatcher", "db_url", KAMAILIODBURL1) modparam("dispatcher", "flags", 2) modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "sip:<address>.com<sip:%3caddress%3e.com>") modparam("dispatcher", "ds_ping_interval", 5) modparam("dispatcher", "ds_probing_threshold", 1) modparam("dispatcher", "ds_inactive_threshold", 1) modparam("dispatcher", "ds_probing_mode", 3) # ----- pv params ----- modparam("pv", "shvset", "maintenance=i:0") modparam("pv", "shvset", "virtualIP1=i:0") modparam("pv", "shvset", "virtualIP2=i:0") # ----- nathelper params ----- modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") # Note: leaving NAT pings turned off here as nathelper is only being used for # WebSocket connections. NAT pings are not needed as WebSockets have # their own keep-alives. # ----- rtpengine params ----- modparam("rtpengine", "rtpengine_sock", "udp:localhost:2223") modparam("rtpengine", "rtpengine_sock", "udp:localhost:2223") #modparam("rtpengine", "write_sdp_pv", "$avp(sdp)") #modparam("rtpengine", "force_send_interface", SERVER_IP_ADDR) #modparam("rtpengine", "setid_default", -1) #modparam("rtpengine", "rtp_inst_pvar", "$avp(RTPENGINE)") #modparam("rtpengine", "rtpengine_retr", 5) #modparam("rtpengine", "queried_nodes_limit", 5) #modparam("rtpengine", "rtpengine_allow_op", 1) #modparam("rtpengine", "hash_table_size", MAX_CONNECTIONS) #modparam("rtpengine", "hash_table_tout", 7200) modparam("tls", "private_key", "<cert path>") modparam("tls", "certificate", "<cert path>") modparam("tls", "ca_list", "<cert path>") # modparam("tls", "ca_list", "<cert path>") modparam("tls", "tls_method", "TLSv1+") ####### Routing Logic ######## If load module lines for TLS are move to near the top of the config file, config will parse and non-SIPS calls will work loadmodule "sl.so" loadmodule "tls.so" But logs will show WARNING: <core> [main.c:2985]: main(): tls support enabled, but no tls engine available (forgot to load the tls module?) WARNING: <core> [main.c:2987]: main(): disabling tls... Presumably loading the module before configuring it just gives it default values so the latter config is ignored Any advice on the matter would be appreciated Thanks in advance, Chris __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20220323/4d2d15a7/…> ------------------------------ Message: 4 Date: Wed, 23 Mar 2022 20:16:03 +0000 From: Daniel Paape <dpaape(a)amtelco.com> To: "sr-users(a)lists.kamailio.org" <sr-users(a)lists.kamailio.org> Subject: [SR-Users] bug in tm module? Message-ID: <CO6PR13MB5321D941E99DE2BA6702E24BC9189(a)CO6PR13MB5321.namprd13.prod.outlook.com> Content-Type: text/plain; charset="iso-8859-1" Hello I'm running K5.5.3 and have checked the 'issues' link in github and saw nothing regarding what I am seeing. I think that there is a bug in the tm module with respect to the "tm:local-response" event route. In the "t_reply.c" file, there is a static variable called '_tm_local_response_set_lookup'. This variable is initialized at load time to zero. It is checked in the '_reply_light()' routine and will initiate a local callback from the config script if "armed". The problem as I see it is that if the callback is readied, the variable is set to one. But it is never reset. So the practical outworking is as follows. A register is received and the request_route arms the callback. The register requires an authorization (local database sqlite). After the 401 is sent back, the callback via the module event route is called as expected. All good except that the module event route script fragment is never executed again after the first call. I'm no expert on the tm module but I would think that somewhere in there, the variable should be reset to zero so that subsequent transactions can initiate the event route again. thanks for any help, Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20220323/a0311764/…> ------------------------------ Message: 5 Date: Wed, 23 Mar 2022 20:21:26 +0000 From: Henning Westerholt <hw(a)gilawa.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] bug in tm module? Message-ID: <PAXPR05MB8558B125061A99B7E6A3D352BF189(a)PAXPR05MB8558.eurprd05.prod.outlook.com> Content-Type: text/plain; charset="us-ascii" Hello, If you think there is a bug in the tm module, you could create an issue in our tracker here: https://github.com/kamailio/kamailio/issues/new?assignees=&labels=&template… Please also add some description of the wrong behaviour that you are seeing in the server to it. Thanks, Henning From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Daniel Paape Sent: Wednesday, March 23, 2022 9:16 PM To: sr-users(a)lists.kamailio.org Subject: [SR-Users] bug in tm module? Hello I'm running K5.5.3 and have checked the 'issues' link in github and saw nothing regarding what I am seeing. I think that there is a bug in the tm module with respect to the "tm:local-response" event route. In the "t_reply.c" file, there is a static variable called '_tm_local_response_set_lookup'. This variable is initialized at load time to zero. It is checked in the '_reply_light()' routine and will initiate a local callback from the config script if "armed". The problem as I see it is that if the callback is readied, the variable is set to one. But it is never reset. So the practical outworking is as follows. A register is received and the request_route arms the callback. The register requires an authorization (local database sqlite). After the 401 is sent back, the callback via the module event route is called as expected. All good except that the module event route script fragment is never executed again after the first call. I'm no expert on the tm module but I would think that somewhere in there, the variable should be reset to zero so that subsequent transactions can initiate the event route again. thanks for any help, Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20220323/61051c0f/…> ------------------------------ Message: 6 Date: Thu, 24 Mar 2022 07:39:26 +0100 From: Arek ?azarczyk <arek(a)lazarczyk.org.pl> To: sr-users(a)lists.kamailio.org Subject: [SR-Users] carrierroute with non-numeric characters Message-ID: <a20892cf-fa42-eefe-2bea-b75a74e3196d(a)lazarczyk.org.pl> Content-Type: text/plain; charset=UTF-8; format=flowed Hi, I'm in a process of long overdue update of kamailio from 4.3 to 5.5, after update i can not start kamailio casue: Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: INFO: carrierroute [cr_data.c:456]: add_route(): adding prefix TR, prob 1.000000 Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: INFO: carrierroute [cr_data.c:469]: add_route(): found carrier and domain, now adding route Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: ERROR: carrierroute [cr_domain.c:166]: add_route_to_tree(): cannot insert route flags into d-trie Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: ERROR: <core> [dtrie.c:138]: dtrie_insert(): cannot insert non-numerical character Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: ERROR: carrierroute [cr_data.c:179]: reload_route_data(): could not load routing data Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: ERROR: carrierroute [carrierroute.c:240]: mod_init(): could not prepare route data I know that string TR is non-numeric data ;) but i have thoes routes deep in my infrastructure logic so i just can not change them to some numbers, is there any way i can resolve this without changeing routes in my whole infrastrucure (almost 50 servers)? Please help :) -- Arek ?azarczyk ------------------------------ Message: 7 Date: Thu, 24 Mar 2022 07:07:35 +0000 From: Henning Westerholt <hw(a)gilawa.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] carrierroute with non-numeric characters Message-ID: <PAXPR05MB8558847862B2480F6CC9E2EABF199(a)PAXPR05MB8558.eurprd05.prod.outlook.com> Content-Type: text/plain; charset="utf-8" Hello, give this parameter a try: https://kamailio.org/docs/modules/5.5.x/modules/carrierroute.html#idm173 If it also not works with this parameter set to 128 and its used to work on 4.3, its probably a code regression. Please open an issue on our GitHub tracker about it then. Cheers, Henning -- Henning Westerholt ? https://skalatan.de/blog/ Kamailio services ? https://gilawa.com -----Original Message----- From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Arek Lazarczyk Sent: Thursday, March 24, 2022 7:39 AM To: sr-users(a)lists.kamailio.org Subject: [SR-Users] carrierroute with non-numeric characters Hi, I'm in a process of long overdue update of kamailio from 4.3 to 5.5, after update i can not start kamailio casue: Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: INFO: carrierroute [cr_data.c:456]: add_route(): adding prefix TR, prob 1.000000 Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: INFO: carrierroute [cr_data.c:469]: add_route(): found carrier and domain, now adding route Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: ERROR: carrierroute [cr_domain.c:166]: add_route_to_tree(): cannot insert route flags into d-trie Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: ERROR: <core> [dtrie.c:138]: dtrie_insert(): cannot insert non-numerical character Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: ERROR: carrierroute [cr_data.c:179]: reload_route_data(): could not load routing data Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: ERROR: carrierroute [carrierroute.c:240]: mod_init(): could not prepare route data I know that string TR is non-numeric data ;) but i have thoes routes deep in my infrastructure logic so i just can not change them to some numbers, is there any way i can resolve this without changeing routes in my whole infrastrucure (almost 50 servers)? Please help :) -- Arek ?azarczyk __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(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: * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users ------------------------------ Message: 8 Date: Thu, 24 Mar 2022 08:42:20 +0100 From: Arek ?azarczyk <arek(a)lazarczyk.org.pl> To: Henning Westerholt <hw(a)gilawa.com>, "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] carrierroute with non-numeric characters Message-ID: <97be0b82-e083-673b-1e2b-f11a62692f34(a)lazarczyk.org.pl> Content-Type: text/plain; charset=UTF-8; format=flowed Thanks, that worked :) , i've must missed that parameter during initial config. BR, A?. W dniu 24.03.2022 o?08:07, Henning Westerholt pisze: > Hello, > > give this parameter a try: > > https://kamailio.org/docs/modules/5.5.x/modules/carrierroute.html#idm173 > > If it also not works with this parameter set to 128 and its used to work on 4.3, its probably a code regression. Please open an issue on our GitHub tracker about it then. > > Cheers, > > Henning > -- Arek ?azarczyk ------------------------------ Message: 9 Date: Thu, 24 Mar 2022 07:58:10 +0000 From: Henning Westerholt <hw(a)gilawa.com> To: Arek ?azarczyk <arek(a)lazarczyk.org.pl>, "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] carrierroute with non-numeric characters Message-ID: <PAXPR05MB8558EA2CE9C0F379042C9DBABF199(a)PAXPR05MB8558.eurprd05.prod.outlook.com> Content-Type: text/plain; charset="utf-8" Great, thanks for the feedback. Henning -----Original Message----- From: Arek ?azarczyk <arek(a)lazarczyk.org.pl> Sent: Thursday, March 24, 2022 8:42 AM To: Henning Westerholt <hw(a)gilawa.com>; Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] carrierroute with non-numeric characters Thanks, that worked :) , i've must missed that parameter during initial config. BR, A?. W dniu 24.03.2022 o?08:07, Henning Westerholt pisze: > Hello, > > give this parameter a try: > > https://kamailio.org/docs/modules/5.5.x/modules/carrierroute.html#idm1 > 73 > > If it also not works with this parameter set to 128 and its used to work on 4.3, its probably a code regression. Please open an issue on our GitHub tracker about it then. > > Cheers, > > Henning > -- Arek ?azarczyk ------------------------------ Subject: Digest Footer _______________________________________________ sr-users mailing list sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users ------------------------------ End of sr-users Digest, Vol 202, Issue 25 *****************************************

1 0

TOPOH is generating new Call-ID for BYE
by Marrold 24 Mar '22

24 Mar '22

Hi, I have enabled the TOPOH module with mask_callid enabled, but it's generating a new Call-ID for a BYE which gets rejected with a 481 downstream. If I disable the TOPOH module it works as expected. Does anyone know where I should start looking? SIP-wise the initial BYE seems compliant. This is Kamailio 5.5.3 on Debian 11 Thanks Matthew

2 2

core parsing of SIP message failed
by Seyyed Soroosh Hosseinalipour 24 Mar '22

24 Mar '22

ITNOA Description I try to connect Kamailio and Freeswitch together, and initiate call from Zoiper5 soft phone through Kamailio My Kamailio config is below #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_DEBUG #!define DBGLEVEL 3 #!else #!define DBGLEVEL 2 #!endif #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!trydef DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### /* LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR, ... */ debug=DBGLEVEL /* set to 'yes' to print log messages to terminal or use '-E' cli option */ log_stderror=yes memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes for each UDP socket * - value inherited by tcp_children and sctp_children when not set explicitely */ children=8 /* uncomment the next line to disable TCP (default on) */ disable_tcp=yes /* number of SIP routing processes for all TCP/TLS sockets */ # tcp_children=8 /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases - it can be set many times */ # alias="sip.mydomain.com" /* listen sockets - if none set, Kamailio binds to all local IP addresses * - basic prototype (full prototype can be found in Wiki - Core Cookbook): * listen=[proto]:[localip]:[lport] advertise [publicip]:[pport] * - it can be set many times to add more sockets to listen to */ listen=udp:172.28.72.99:5060 advertise 172.28.72.99:5060 #listen=tcp:172.28.72.99:5060 /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 /* upper limit for TCP connections (it includes the TLS connections) */ tcp_max_connections=2048 #!ifdef WITH_JSONRPC tcp_accept_no_cl=yes #!endif #!ifdef WITH_TLS enable_tls=no /* upper limit for TLS connections */ tls_max_connections=2048 #!endif /* set it to yes to enable sctp and load sctp.so module */ enable_sctp=no ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif #!ifdef WITH_JSONRPC loadmodule "xhttp.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "textopsx.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" #!ifdef WITH_RTPENGINE loadmodule "rtpengine.so" #!else loadmodule "rtpproxy.so" #!endif #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/run/kamailio/kamailio_rpc.sock") #!ifdef WITH_JSONRPC modparam("jsonrpcs", "transport", 7) #!endif # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/run/kamailio/kamailio_ctl") # ----- sanity params ----- modparam("sanity", "autodrop", 0) # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) /* set it to 0 to disable Path handling */ modparam("registrar", "use_path", 1) /* save Path even if not listed in Supported header */ modparam("registrar", "path_mode", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- modparam("usrloc", "timer_interval", 60) modparam("usrloc", "timer_procs", 1) modparam("usrloc", "use_domain", MULTIDOMAIN) /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT #!ifdef WITH_RTPENGINE # ----- rtpengine params ----- modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223") #!else # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") #!endif # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks # route(REQINIT); # NAT detection #route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } if (is_method("REGISTER")) { xdbg("request from gggggggggggggggggggggggggggggggggggggggggg - $rm from $fu (IP:$si:$sp)\n"); } rewritehostport("172.28.69.52:5060"); # handle requests within SIP dialogs # route(WITHINDLG); ### only initial requests (no To tag) # authentication # route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers #remove_hf("Route"); #if (is_method("INVITE|SUBSCRIBE")) { # record_route(); #} # account only INVITEs #if (is_method("INVITE")) { # setflag(FLT_ACC); # do accounting #} # dispatch requests to foreign domains #route(SIPOUT); ### requests for my local domains # handle presence related requests #route(PRESENCE); # handle registrations # route(REGISTRAR); #if ($rU==$null) { # # request with no Username in RURI # sl_send_reply("484","Address Incomplete"); # exit; #} # dispatch destinations to PSTN #route(PSTN); # user location service #route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { # no connect for sending replies set_reply_no_connect(); # enforce symmetric signaling # - send back replies to the source address of request force_rport(); #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if($ua =~ "friendly|scanner|sipcli|sipvicious|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP request from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) { sl_reply_error(); } exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE #!ifdef WITH_MSGREBUILD # apply changes in case the request headers or body were modified msg_apply_changes(); #!endif if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; #!ifdef WITH_RTPENGINE if(nat_uac_test("8")) { rtpengine_manage("replace-origin replace-session-connection"); } else { rtpengine_manage("trust-address replace-origin replace-session-connection"); } #!else if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } #!endif if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } if(isbflagset(FLB_NATB)) { # no connect message in a dialog involving NAT traversal if (is_request()) { if(has_totag()) { set_forward_no_connect(); } } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-Hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # JSONRPC over HTTP(S) routing #!ifdef WITH_JSONRPC event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if(src_ip!=127.0.0.1) { xhttp_reply("403", "Forbidden", "text/html", "<html><body>Not allowed from $si</body></html>"); exit; } if ($hu =~ "^/RPC") { jsonrpc_dispatch(); exit; } xhttp_reply("200", "OK", "text/html", "<html><body>Wrong URL $hu</body></html>"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies reply_route { if(!sanity_check("17604", "6")) { xlog("Malformed SIP response from $si:$sp\n"); drop; } } # Manage incoming replies in transaction context onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } As you can see I just one thing in request_route request_route { # per request initial checks # route(REQINIT); # NAT detection #route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } if (is_method("REGISTER")) { xdbg("request from gggggggggggggggggggggggggggggggggggggggggg - $rm from $fu (IP:$si:$sp)\n"); } rewritehostport("172.28.69.52:5060"); The 172.28.69.52 is ip of my Freeswitch After I try to call from Zoiper5, I get below error in Kamailio log and Zoiper does not work. 4(10) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 4(10) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) Troubleshooting Reproduction * pull official Kamailio docker image kamailio/kamailio 5.4.6-buster * then run below command docker run --net=host --name kamailio \ -v /home/ssoroosh/kamailio:/etc/kamailio \ kamailio/kamailio:5.4.6-buster Debugging Data I do not have any debug data Log Messages Listening on udp: 172.28.72.99 [172.28.72.99]:5060 advertise 172.28.72.99:5060 Aliases: 0(1) INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded 0(1) INFO: rr [rr_mod.c:185]: mod_init(): outbound module not available 0(1) INFO: <core> [main.c:2919]: main(): processes (at least): 16 - shm size: 67108864 - pkg size: 8388608 0(1) INFO: <core> [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(1) INFO: <core> [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 13(19) INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/19 15(21) INFO: ctl [io_listener.c:210]: io_listen_loop(): io_listen_loop: using epoll_lt as the io watch method (auto detected) 2(8) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 2(8) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 2(8) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 2(8) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 4(10) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 4(10) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 4(10) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 4(10) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 1(7) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 1(7) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 1(7) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 1(7) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 3(9) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 3(9) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 3(9) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 3(9) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 5(11) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 5(11) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 5(11) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 5(11) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 8(14) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 8(14) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 8(14) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 8(14) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 1(7) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 1(7) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 1(7) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 1(7) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 5(11) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 5(11) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 5(11) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 5(11) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 4(10) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 4(10) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 4(10) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 4(10) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 5(11) INFO: <core> [core/parser/parse_fline.c:156]: parse_first_line(): ERROR:parse_first_line: method not followed by SP 5(11) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 5(11) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 5(11) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 6(12) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 6(12) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 6(12) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 7(13) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 7(13) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 7(13) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 8(14) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 8(14) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 8(14) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 2(8) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 2(8) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 2(8) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 4(10) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 4(10) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 4(10) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 1(7) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 1(7) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 1(7) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 3(9) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 3(9) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 3(9) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 5(11) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 5(11) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 5(11) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 6(12) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 6(12) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 6(12) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) 7(13) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 41) 7(13) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<��> 7(13) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP message failed (172.28.64.1:51370/1) SIP Traffic My pcap is attached sip.zip<https://github.com/kamailio/kamailio/files/8326814/sip.zip> Possible Solutions I have not any solution yet. Additional Information * Kamailio Version - output of kamailio -v version: kamailio 5.4.6 (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, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, 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 8.3.0 * Operating System: Linux master 5.4.0-104-generic #118-Ubuntu SMP Wed Mar 2 19:02:41 UTC 2022 x86_64 GNU/Linux root@master:/home# lsb_release -a No LSB modules are available. Distributor ID: Debian Description: Debian GNU/Linux 10 (buster) Release: 10 Codename: buster How to resolve this? thanks

2 1

carrierroute with non-numeric characters
by Arek Łazarczyk 24 Mar '22

24 Mar '22

Hi, I'm in a process of long overdue update of kamailio from 4.3 to 5.5, after update i can not start kamailio casue: Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: INFO: carrierroute [cr_data.c:456]: add_route(): adding prefix TR, prob 1.000000 Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: INFO: carrierroute [cr_data.c:469]: add_route(): found carrier and domain, now adding route Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: ERROR: carrierroute [cr_domain.c:166]: add_route_to_tree(): cannot insert route flags into d-trie Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: ERROR: <core> [dtrie.c:138]: dtrie_insert(): cannot insert non-numerical character Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: ERROR: carrierroute [cr_data.c:179]: reload_route_data(): could not load routing data Mar 24 07:30:13 DEV-SBC-FE1 /usr/sbin/kamailio[22768]: ERROR: carrierroute [carrierroute.c:240]: mod_init(): could not prepare route data I know that string TR is non-numeric data ;) but i have thoes routes deep in my infrastructure logic so i just can not change them to some numbers, is there any way i can resolve this without changeing routes in my whole infrastrucure (almost 50 servers)? Please help :) -- Arek Łazarczyk

2 3

bug in tm module?
by Daniel Paape 23 Mar '22

23 Mar '22

Hello I'm running K5.5.3 and have checked the 'issues' link in github and saw nothing regarding what I am seeing. I think that there is a bug in the tm module with respect to the "tm:local-response" event route. In the "t_reply.c" file, there is a static variable called '_tm_local_response_set_lookup'. This variable is initialized at load time to zero. It is checked in the '_reply_light()' routine and will initiate a local callback from the config script if "armed". The problem as I see it is that if the callback is readied, the variable is set to one. But it is never reset. So the practical outworking is as follows. A register is received and the request_route arms the callback. The register requires an authorization (local database sqlite). After the 401 is sent back, the callback via the module event route is called as expected. All good except that the module event route script fragment is never executed again after the first call. I'm no expert on the tm module but I would think that somewhere in there, the variable should be reset to zero so that subsequent transactions can initiate the event route again. thanks for any help, Dan

2 1

SIPS Errors on Kamailio
by Christopher Vincent 23 Mar '22

23 Mar '22

Hi, Kamailio / RTPEngine was set up on CentOS 8 running SIP to SIPS and RTP to SDES SRTP conversion. This worked as expected Attempted to duplicate the setup on RHEL but errors were seen. These errors were present on both RHEL 7 / RHEL 8. The errors seen were as below kamailio -c loading modules under config path: /usr/lib64/kamailio/modules/ 0(9165) ERROR: tls [tls_init.c:611]: tls_pre_init(): Unable to set the memory allocation functions 0(9165) ERROR: tls [tls_init.c:613]: tls_pre_init(): libssl current mem functions - m: 0x7f7a77c367a0 r: 0x7f7a77c367f0 f: 0x7f7a77c36770 0(9165) ERROR: tls [tls_init.c:615]: tls_pre_init(): module mem functions - m: 0x7f7a72db7653 r: 0x7f7a72db769f f: 0x7f7a72db76fc 0(9165) ERROR: tls [tls_init.c:617]: tls_pre_init(): Be sure tls module is loaded before any other module using libssl (can be loaded first to be safe) 0(9165) ERROR: <core> [core/sr_module.c:590]: load_module(): /usr/lib64/kamailio/modules/tls.so: mod_register failed 0(9165) CRITICAL: <core> [core/cfg.y:3683]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 137, column 12-19: failed to load module 0(9165) INFO: pv [pv_shv.c:60]: shvar_init_locks(): locks array size 16 0(9165) ERROR: <core> [core/modparam.c:181]: set_mod_param_regex(): No module matching <tls> found 0(9165) CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 249, column 72: Can't set module parameter 0(9165) ERROR: <core> [core/modparam.c:181]: set_mod_param_regex(): No module matching <tls> found 0(9165) CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 250, column 72: Can't set module parameter 0(9165) ERROR: <core> [core/modparam.c:181]: set_mod_param_regex(): No module matching <tls> found 0(9165) CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 251, column 68: Can't set module parameter 0(9165) ERROR: <core> [core/modparam.c:181]: set_mod_param_regex(): No module matching <tls> found 0(9165) CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 256, column 39: Can't set module parameter ERROR: bad config file (5 errors) (parsing code: 0) 0(9165) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized The kamailio config was exactly the same as on the CentOS systems and started as below /* Server ports: */ #!substdef "!SIP_PORT!5060!g" #!substdef "!SIPS_PORT!5061!g" /* Listen addresses */ #!substdef "!UDP_LOCAL_ADDR!udp:SERVER_IP_ADDR:SIP_PORT!g" #!substdef "!TCP_LOCAL_ADDR!tcp:SERVER_IP_ADDR:SIPS_PORT!g" /* Server connections: */ #!ifndef MAX_CONNECTIONS #!define MAX_CONNECTIONS 8192 #!endif ##!define WITH_DEBUG /* Transaction and branch flags: FLT_ - per transaction (message) flags FLB_ - per branch flags */ #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLT_OUT 8 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 #!define KAMAILIODBURL1 "mysql://kamailio:kamailiorw@localhost/kamailio" #!define WITH_TLS enable_tls=1 listen=tls:<ipaddr>:5062 ####### Global Parameters ######### ### LOG Levels: ALERT=-5, BUG=-4, CRIT=-3, ERR=-1, WARN=0, NOTICE=1, INFO=2, DBG=3 #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 /* display memory usage on exit */ mem_summary=15 /* join free memory fragments */ mem_join=1 /* proxy will fork and run in daemon mode */ /* one process will be created for each network interface the proxy listens to and for each protocol (TCP/UDP), multiplied with the value of 'children' parameter */ fork=yes children=8 listen=TCP_LOCAL_ADDR listen=UDP_LOCAL_ADDR /* life time of TCP connection when there is no traffic - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 /* sip over websockets may not specify a content length header */ tcp_accept_no_cl=yes /* buffer size used for tcp reads, limits the maximum message size (SIP, HTTP) that can be received over tcp */ tcp_rd_buf_size=65536 /* max number of tcp connections */ tcp_max_connections=MAX_CONNECTIONS ####### Modules Section ######## # set paths to location of modules mpath="/usr/lib64/kamailio/modules/" loadmodule "jsonrpcs.so" loadmodule "db_mysql.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "dispatcher.so" loadmodule "cfgutils.so" loadmodule "textopsx.so" loadmodule "nathelper.so" loadmodule "uac.so" loadmodule "ipops.so" loadmodule "debugger.so" loadmodule "exec.so" loadmodule "avpops.so" loadmodule "sqlops.so" loadmodule "rtpengine.so" loadmodule "sl.so" loadmodule "tls.so" # ----------------- setting module-specific parameters --------------- # ----- usrloc params ----- # store contacts in memory only modparam("usrloc", "db_mode", 0) # hash size of 16,384 modparam("usrloc", "hash_size", 14) # removes contact if ws disconnects modparam("usrloc", "handle_lost_tcp", 1) modparam("tm|usrloc", "xavp_contact", "ulattrs") # ----- jsonrpcs params ----- modparam("jsonrpcs", "fifo_name", "/tmp/kamailio_jsonrpc.fifo") modparam("jsonrpcs", "dgram_socket", "/tmp/kamailio_rpc.sock") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) #default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- uac params ----- modparam("uac", "restore_mode", "none") # ----- registrar params ----- modparam("registrar", "method_filtering", 1) modparam("registrar", "max_contacts", 1) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # disable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direction of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) # ----- dispatcher params ----- modparam("dispatcher", "db_url", KAMAILIODBURL1) modparam("dispatcher", "flags", 2) modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "sip:<address>.com") modparam("dispatcher", "ds_ping_interval", 5) modparam("dispatcher", "ds_probing_threshold", 1) modparam("dispatcher", "ds_inactive_threshold", 1) modparam("dispatcher", "ds_probing_mode", 3) # ----- pv params ----- modparam("pv", "shvset", "maintenance=i:0") modparam("pv", "shvset", "virtualIP1=i:0") modparam("pv", "shvset", "virtualIP2=i:0") # ----- nathelper params ----- modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") # Note: leaving NAT pings turned off here as nathelper is only being used for # WebSocket connections. NAT pings are not needed as WebSockets have # their own keep-alives. # ----- rtpengine params ----- modparam("rtpengine", "rtpengine_sock", "udp:localhost:2223") modparam("rtpengine", "rtpengine_sock", "udp:localhost:2223") #modparam("rtpengine", "write_sdp_pv", "$avp(sdp)") #modparam("rtpengine", "force_send_interface", SERVER_IP_ADDR) #modparam("rtpengine", "setid_default", -1) #modparam("rtpengine", "rtp_inst_pvar", "$avp(RTPENGINE)") #modparam("rtpengine", "rtpengine_retr", 5) #modparam("rtpengine", "queried_nodes_limit", 5) #modparam("rtpengine", "rtpengine_allow_op", 1) #modparam("rtpengine", "hash_table_size", MAX_CONNECTIONS) #modparam("rtpengine", "hash_table_tout", 7200) modparam("tls", "private_key", "<cert path>") modparam("tls", "certificate", "<cert path>") modparam("tls", "ca_list", "<cert path>") # modparam("tls", "ca_list", "<cert path>") modparam("tls", "tls_method", "TLSv1+") ####### Routing Logic ######## If load module lines for TLS are move to near the top of the config file, config will parse and non-SIPS calls will work loadmodule "sl.so" loadmodule "tls.so" But logs will show WARNING: <core> [main.c:2985]: main(): tls support enabled, but no tls engine available (forgot to load the tls module?) WARNING: <core> [main.c:2987]: main(): disabling tls... Presumably loading the module before configuring it just gives it default values so the latter config is ignored Any advice on the matter would be appreciated Thanks in advance, Chris

3 4

{param.in, name[, delimiter]} error when delimiter is comma
by 韩小仿 23 Mar '22

23 Mar '22

Hi, all Kamailio5.5.4 $var(s) = "a=1;b=2;c=3"; $var(s1) = $(var(s){param.value,b}); // work well $var(s) = "a=1|b=2|c=3"; $var(s1) = $(var(s){param.value,b,|}); // work well // but $var(s) = "a=1,b=2,c=3"; $var(s1) = $(var(s){param.value,b,,}); ERROR: <core> [core/pvapi.c:1131]: pv_parse_spec2(): invalid parsing in [$(var(s){param.value,b,,})] at (4) CRITICAL: <core> [core/cfg.y:3683]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 366, column 13-38: Can't get from cache: $(var(s){param.value,b,,}) how to fix this? Thanks! hanxiaofang

2 1

bug in tm module?
by dpaape＠frontier.com 22 Mar '22

22 Mar '22

Hello, I'm running K5.5.3 and have checked the 'issues' link in github and saw nothing regarding what I am seeing. I think that there is a bug in the tm module with respect to the "tm:local-response" event route. In the "t_reply.c" file, there is a static variable called '_tm_local_response_set_lookup'. This variable is initialized at load time to zero. It is checked in the '_reply_light()' routine and will initiate a local callback from the config script if "armed". The problem as I see it is that if the callback is readied, the variable is set to one. But it is never reset.So the practical outworking is as follows. A register is received and the request_route arms the callback. The register requires and authorization. After the 401 is sent back, the callback via the module event route is called. All good except that the module event route script fragment is never executed again after the first call.I'm no expert on the tm module but I would think that somewhere in there, the variable should be reset to zero so that subsequent transactions can initiate the event route. thanks for any help,Dan

1 0

SERIAL FORKING : call to a group number
by Youssef Boujraf 22 Mar '22

22 Mar '22

Dear all, My use case is : I defined a username = 112 (for emergency call) This username doesn't exist in subscribers table and unavailable in location table. The idea is when Sopftphone1 call the group = 112 then Kamailio will forward the call to a group using SERIAL FORKING. => Softphone 1 -> Sends INVITE to 112 -> Kamailio -> FORWARD TO SERIAL FORKING (4x softphone all registered and in location table). The q priority could be hybrid that means 2x softphone could have same priority. My issues : - the q priority forced for each softphone is not taking into account. - the first branch created when send INVITE to 112 is still available. My setup Kamailio is behind a NAT. Kamailio version : sipsecure@kamailio:/$ kamailio -version version: kamailio 5.5.4 (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, 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 Kamailio.cfg # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 5sec modparam("tm", "fr_timer", 5000) # default invite retransmission timeout after 1xx: 10sec modparam("tm", "fr_inv_timer", 10000) # This is the name of an XAVP that the t_load_contacts() function uses to store contacts of the destination set and that t_next_contacts() function uses to restore those contacts. modparam("tm", "contacts_avp", "tm_contacts") modparam("tm", "contact_flows_avp", "tm_contact_flows") # modparam("tm|usrloc", "xavp_contact", "tm_contacts") # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) route[INVITE]{ route(THREAD_CHECK); route(AUTH); route(RECORD_ROUTE); setflag(FLT_ACC); switch($tU) { case "112": xlog("MY_SEVERITY"," -- Calling Group Serial"); route(FORK_SERIAL); break; case "113": xlog("MY_SEVERITY"," -- Calling Group Parralelle"); route(FORK_PARALLEL); break; default: xlog("MY_SEVERITY"," -- Calling Someone"); route(LOCATION); route(RELAY); break; } exit; } route[FORK_SERIAL]{ t_on_reply("MANAGE_REPLY"); #!ifndef WITH_BRIDGE_ON_FAIL setbflag(FLB_BRIDGE); #!endif route(IPV4V6); route(NATMANAGE); $var(uri1)="sip:9919995@192.168.1.41"; $var(uri2)="sip:9919994@192.168.1.41"; $var(uri3)="sip:9919991@192.168.1.41"; $var(uri4)="sip:9919997@192.168.1.41"; lookup("location","$var(uri4)"); append_branch("$ru","0.2"); lookup("location","$var(uri1)"); append_branch("$ru","0.8"); lookup("location","$var(uri2)"); append_branch("$ru","0.5"); lookup("location","$var(uri3)"); append_branch("$ru","0.5"); t_load_contacts(); t_next_contacts(); t_on_failure("SERIAL"); #!ifdef WITH_SIPSECURE_DEBUG xlog("MY_SEVERITY"," ROUTE_SERIAL : Relaying \n"); #!endif #Relay (aka Forward) the request t_relay(); break; } failure_route[SERIAL]{ xlog("MY_SEVERITY","At failure route - Trying next destination"); if (!t_next_contacts()) { send_reply("408","Nobody available"); exit; } t_on_failure("SERIAL"); t_relay(); } Any highlight will be welcome. Best Regards, -- Youssef

1 0

$ru vs. $fu as r/w pseudo variables
by Juha Heinanen 22 Mar '22

22 Mar '22

If I set $ru in config file and then print $ru to log, the new value is printed. If I do the same with $fu, the old value is printed. I haven't found in Pseudo-Variables Wiki any documentation about this. Does it exist somewhere else? -- Juha

3 9

MESSAGE METHOD - Send message to a softphone - $uac_req
by Youssef Boujraf 22 Mar '22

22 Mar '22

Dear, I am expecting to send a MESSAGE METHOD to a softphone supporting MESSAGE. route[SEND_MESSAGE_METHOD]{ $uac_req(method)="MESSAGE"; $uac_req(ruri)="sip:192.168.1.29:5066"; $uac_req(furi)="sip:192.168.1.41:5066"; // $uac_req(turi)="sip:9919995@192.168.1.24:5066"; $uac_req(callid)=$(mb{s.md5}); $uac_req(hdrs)="Subject: Emergency Alert\r\n"; $uac_req(hdrs)=$uac_req(hdrs) + "Content-Type: text/plain\r\n"; $uac_req(body)="Emergency Message"; $uac_req(evroute)=1; uac_req_send(); } event_route[uac:reply] { xlog("MY_SEVERITY"," -- MESSAGE PROCESSING - received reply code is: $uac_req(evcode)\n");} My use case is to send message from KAMAILIO -> SOFTPHONE kamailio : sip:192.168.1.41:5066 (physical ip address of the kamailio) softphone : sip:192.168.1.29:5066 (physical ip address of the softphone) $uac_req(ruri)="sip:192.168.1.29:5066"; $uac_req(furi)="sip:192.168.1.41:5066"; // $uac_req(turi)="sip:9919995@192.168.1.24:5066"; The sip capture : 2022/03/18 02:20:58.947603 192.168.1.29:5066 -> 192.168.1.41:5066 MESSAGE sip:192.168.1.41:5066@192.168.1.41:5066 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.29:5066;branch=z9hG4bKPj40bcc72324d647f8b96fc783e181f611 Max-Forwards: 70 From: "codec7-private" <sip:9919997@192.168.1.41>;tag=416038ac55514b439a5cd5894770c3d8 To: <sip:192.168.1.41:5066@192.168.1.41> Call-ID: 9381648157be49ff845b5e413dc9a95c CSeq: 9148 MESSAGE User-Agent: MicroSIP/3.20.7 Content-Type: text/plain Content-Length: 3 ;-) 2022/03/18 02:20:58.948822 192.168.1.41:5066 -> 192.168.1.29:5066 SIP/2.0 200 Stored Via: SIP/2.0/UDP 192.168.1.29:5066;branch=z9hG4bKPj40bcc72324d647f8b96fc783e181f611;rport=5066 From: "codec7-private" <sip:9919997@192.168.1.41>;tag=416038ac55514b439a5cd5894770c3d8 To: <sip:192.168.1.41:5066@192.168.1.41>;tag=cbfe6543b9079987bce3b5b9a0d1195a.b63b503b Call-ID: 9381648157be49ff845b5e413dc9a95c CSeq: 9148 MESSAGE Server: kamailio (5.5.4 (x86_64/linux)) Content-Length: 0 Now, if I want to use his username@domain kamailio : sip:192.168.1.41:5066 (physical ip address of the kamailio) softphone : sip:9919997@192.168.1.41:5066 The message is sent to kamailio itself not to the softphone 2022/03/18 02:30:39.285927 192.168.1.41:5060 -> 192.168.1.41:5066 MESSAGE sip:9919997@192.168.1.41:5066 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.210:5060;branch=z9hG4bK0cb8.a39b5180000000000000000000000000.0 To: <sip:9919997@192.168.1.41:5066> From: <sip:192.168.1.41:5066>;tag=b4685c5f5f5fc30f8dcbb49a48fe9e25-4047894c CSeq: 10 MESSAGE Call-ID: 4cb973823835008ec4b20d1c67d9fb84 Max-Forwards: 70 Content-Length: 17 User-Agent: kamailio (5.5.4 (x86_64/linux)) Subject: Emergency Alert Content-Type: text/plain Emergency Message 2022/03/18 02:30:39.287590 192.168.1.41:5066 -> 192.168.1.41:5060 SIP/2.0 200 Stored Via: SIP/2.0/UDP xxx.xxx.xxx.210:5060;branch=z9hG4bK0cb8.a39b5180000000000000000000000000.0;rport=5060;received=192.168.1.41 To: <sip:9919997@192.168.1.41:5066>;tag=cbfe6543b9079987bce3b5b9a0d1195a.76a88393 From: <sip:192.168.1.41:5066>;tag=b4685c5f5f5fc30f8dcbb49a48fe9e25-4047894c CSeq: 10 MESSAGE Call-ID: 4cb973823835008ec4b20d1c67d9fb84 Server: kamailio (5.5.4 (x86_64/linux)) Content-Length: 0 - finaly, how to relay the message from one softphone to another softphone through kamailio proxy? Best Regards -- Youssef

2 3

Re: [SR-Users] Kamailio : Serial forking
by Paclinio 22 Mar '22

22 Mar '22

Following the example of TM Module. https://www.kamailio.org/docs/modules/devel/modules/tm.html#tm.serial_forki… Code : modparam("tm", "contacts_avp", "tm_contacts"); > > route { > seturi("sip:a@example.com"); > append_branch("sip:b@example.com", "0.5"); > append_branch("sip:c@example.com", "0.5"); > append_branch("sip:d@example.com", "1.0"); > > t_load_contacts(); > > t_next_contacts(); > t_on_failure("serial"); > t_relay(); > break; > } > > failure_route["serial"] > { > if (!t_next_contacts()) { > exit; > } > > t_on_failure("serial"); > t_relay(); > } > My code : failure_route[GroupFailureRoute]{ > t_next_contacts(); > while(!route(GROUP_LOCATION)) > { > if (!t_next_contacts()) { > send_reply("408","Nobody available"); > exit; > } > } > t_on_failure("GroupFailureRoute"); > t_relay(); > } > > route[GROUP_RELAY_SER]{ > append_branch("sip:9919995@192.168.1.41","1"); > append_branch("sip:9919992@192.168.1.41", "0.21"); > append_branch("sip:9919997@192.168.1.41", "0.22"); > append_branch("sip:9919991@192.168.1.41", "0.22"); > t_load_contacts(); > t_next_contacts(); > while(!route(GROUP_LOCATION)) > { > if (!t_next_contacts()) { > send_reply("408","Timeout or nobody available"); > exit; > } > } > t_on_failure("GroupFailureRoute"); > t_relay(); > break; > } > The result : [image: image.png] > > public2_addr > 192.168.1.41 192.168.1.24 > 192.168.1.10 > ──────────┬───────── ──────────┬───────── > ──────────┬───────── ──────────┬───────── > > │ INV (192.168.1.41) │ > | > > │ audio 39776 (g711u) │ > | > > │ > ─────────────────────────────── ──────────────> │ > > │ INV > (192.168.1.41) │ > │ > > │─┐ audio 39776 (g711u) │ > │ > > │<┘ > │ > │ > > │─┐ 407 Proxy Authentication R│ │ > > │<┘ │ > │ > > │ > CANCEL │ > │ > > │ ─────────────────────> │ > │ > > │─┐ ACK > │ > │ > > │<┘ > │ > │ > > │ 487 Request Terminated │ > │ > > │ <────────────────────── │ > │ > > │ > ACK │ > │ > > │ ─────────────────────> │ > │ > > │ 200 OK │ │ > > │ <───────────────────── │ > │ > > │ 100 Trying │ > > │ <────────────────────────────────────────────── │ > > │ 180 Ringing │ > > │ <─────────────────────────────────────────── ── │ > │ 180 Ringing > │ │ > │ > │ <─────────────────── │ > │ > │ > On the Second "wave" of INVITE with the parallel only the first one is correctly changed with the lookup("location") and not the second. Could someone give me some tips? Best regards Thomas <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> Virus-free. www.avast.com <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

1 0

Re: [SR-Users] Kamailio : Parallel Forking
by Paclinio 22 Mar '22

22 Mar '22

Dear, Following the configuration available on the TM Modules documentation. Src : https://www.kamailio.org/docs/modules/devel/modules/tm.html#tm.serial_forki… request_route { > seturi("sip:a@example.com"); > append_branch("sip:b@example.com"); > append_branch("sip:c@example.com"); > append_branch("sip:d@example.com"); > > t_relay(); > break; > } > > My code : request_route{ > $var(uri1)="sip:9919995@192.168.1.41"; > $var(uri2)="sip:9919992@192.168.1.41"; > $var(uri3)="sip:9919991@192.168.1.41"; > $var(uri4)="sip:9919994@192.168.1.41"; > > append_branch ("$var(uri1"); > append_branch ("$var(uri2"); > append_branch ("$var(uri3"); > append_branct_("$var(uri4"); > t_relay(); > break; > } > The server send INVITE to himself. Fixing the problem : request_route{ > $var(uri1)="sip:9919995@192.168.1.41"; > $var(uri2)="sip:9919992@192.168.1.41"; > $var(uri3)="sip:9919991@192.168.1.41"; > $var(uri4)="sip:9919994@192.168.1.41"; > > lookup("location","$var(uri4)"); > append_branch(); > lookup("location","$var(uri1)"); > append_branch(); > lookup("location","$var(uri2)"); > append_branch(); > lookup("location","$var(uri3)"); > t_relay(); > break; > > } > Is this the correct way ? Best regards. Thomas <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> Virus-free. www.avast.com <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

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Re: [SR-Users] MESSAGE METHOD - Send message to a softphone - $uac_req
by Boujraf, Youssef 22 Mar '22

22 Mar '22

Dear, then could I use to retrieve the user from location then set the ruri ? $var(uri1)="sip:9919995@192.168.1.41"; > lookup("location","$var(uri1)"); > > $uac_req(ruri)=$ru; > Best Regards *Youssef *

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Re: [SR-Users] ?==?utf-8?q? Table : address - address_reload?==?utf-8?q? (exported MI function) - expect to call this function in kamailio.cfg to reload in cache the address
by Youssef Boujraf 22 Mar '22

22 Mar '22

Dear, When using : sudo kamctl address reload sipsecure@kamailio:/$ sudo kamctl address reload [sudo] password for sipsecure: { "jsonrpc": "2.0", "result": "Reload OK", "id": 73482 } I looked for the command I could use in kamailio.cfg route[JSON_ADDRESS_RELOAD]{ xlog("MY_SEVERITY"," RPC REQUEST : RELOAD ADDRESS TABLE"); jsonrpc_exec('{"jsonrpc": "2.0","method": "permissions.addressReload"}'); exit; } Now I am able to call a rpc command from the code ;-) Best Regards -- Youssef

1 0

Periodic keepalive for WebRTC clients
by Xuo Guoto 22 Mar '22

22 Mar '22

Hello all, I have a [SIP client] <-> Kamailio <-> Asterisk installation. Sip clients are both WebRTC and Android app. WebRTC connects via 443 while Android app connects VIA TLS at port 5061. WebRTC client uses jsSIP library. My Kamailio version is 5.5.3 (x86_64/linux) I had been using asterisk directly to terminate WebRTC clients previously and was using the qualify feature to periodically send OPTIONS packet to all connected clients as a keep alive mechanism. This allowed the client to detect any loss of connectivity and reconnect back to server if there is any loss of connectivity. After moving to Kamailio I am looking for a similar feature, where Kamailio sends OPTIONS (or any other packets) periodically to the WebRTC clients. My attempts so far hasn't been a success. I am aware of couple of options for keepliave in kamailio: 1. nat_keeplaive() I have some what naively copied the example from nat_keepalive() documentation into route[NATDETECT] but that is not working. route[NATDETECT] { force_rport(); #!ifdef WITH_WEBSOCKETS if (nat_uac_test(64)) { if (is_method("REGISTER")) { fix_nated_register(); } else { if (!add_contact_alias()) { sl_send_reply("400", "Bad Request"); exit; } } setflag(FLT_NATS); return; } #!endif if (nat_uac_test(19)) { if (is_method("REGISTER")) { fix_nated_register(); } fix_contact(); } setflag(FLT_NATS); if ((method=="REGISTER" || method=="SUBSCRIBE" || (method=="INVITE" && !has_totag())) && client_nat_test("3")) { nat_keepalive(); } return; } 2. keepalive module This looks like what I want, but I am not sure how to add a new WebRTC connection to the list of endpoints for pinging. This is the handler for xhttp:request, should the ka_add_destination be added here? event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT #!ifdef WITH_TLS && $Rp != MY_WSS_PORT #!endif ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm=~"GET") { if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xhttp_reply("403", "Forbidden", "", ""); exit; } if (ws_handle_handshake()) { exit; } } xhttp_reply("404", "Not Found", "", ""); } My search so far has not returned any other means to send options packets to WebRTC clients. Been banging my head around this for couple of days, and any help will be much appreciated. X.

1 0

Dialog module errors with dmq
by Olli Attila 22 Mar '22

22 Mar '22

Hello, I have two Kamailio nodes. Couple of days of ago I enabled dmq to replicate dialogs between these two Kamailio nodes. Now that the dmq is enabled, we are seeing lot of warning and notice messages on kamailio logs related to the dialog module. Is there something I have missed when configuring dlg_manage & dmq replication that might cause this? No calls have been affected negatively by this though. Kamailio 1 node prints lot of this: Mar 4 08:20:25 /usr/sbin/kamailio[14316]: WARNING: dialog [dlg_handlers.c:1328]: dlg_onroute(): unable to find dialog for ACK with route param 'a5d.ebf1' [3418:8126] and call-id 'SDiu98001-dd9c12fea650b9d1408dd672a2c51c9f-v300g00020' Mar 4 08:20:28 /usr/sbin/kamailio[14313]: WARNING: dialog [dlg_handlers.c:1328]: dlg_onroute(): unable to find dialog for BYE with route param 'a5d.ebf1' [3418:8126] and call-id 'SDiu98001-dd9c12fea650b9d1408dd672a2c51c9f-v300g00020' Mar 4 08:21:56 /usr/sbin/kamailio[14315]: WARNING: dialog [dlg_handlers.c:1328]: dlg_onroute(): unable to find dialog for ACK with route param '689.f2f1' [2438:7983] and call-id 'SD8o63c01-47972a87ee1fd8791ddf27b650badcba-v300g00010' Kamailio 2 node prints lot of this: Mar 4 08:20:29 /usr/sbin/kamailio[8005]: NOTICE: dialog [dlg_hash.c:242]: dlg_clean_run(): dialog in early state is too old (0x7fe07b84e980 ref 1) Mar 4 08:20:29 /usr/sbin/kamailio[8005]: NOTICE: dialog [dlg_hash.c:242]: dlg_clean_run(): dialog in early state is too old (0x7fe07bb03360 ref 1) Mar 4 08:21:59 /usr/sbin/kamailio[8005]: NOTICE: dialog [dlg_hash.c:242]: dlg_clean_run(): dialog in early state is too old (0x7fe07b528300 ref 1) (Version: kamailio 5.5.3 (x86_64/linux) 473cef on both kamailio nodes) [root@kamailio-1]# kamcmd dmq.list_nodes { host: 172.19.194.140 port: 5060 proto: * resolved_ip: 172.19.194.140 status: active last_notification: 0 local: 0 } { host: 172.19.194.132 port: 5060 proto: * resolved_ip: 172.19.194.132 status: active last_notification: 0 local: 1 } [root@kamailio-2]# kamcmd dmq.list_nodes { host: 172.19.194.132 port: 5060 proto: * resolved_ip: 172.19.194.132 status: active last_notification: 0 local: 0 } { host: 172.19.194.140 port: 5060 proto: * resolved_ip: 172.19.194.140 status: active last_notification: 0 local: 1 } ### On Kamailio 1: ### #!define DMQ_LOCAL_ADDR "sip:172.19.194.132:5060" #!define DMQ_PEER_ADDR "sip:172.19.194.140:5060" ### On Kamailio 2: ### #!define DMQ_LOCAL_ADDR "sip:172.19.194.140:5060" #!define DMQ_PEER_ADDR "sip:172.19.194.132:5060" ### On Kamailio 1 & 2:### modparam("dmq", "server_address", DMQ_LOCAL_ADDR) modparam("dmq", "notification_address", DMQ_PEER_ADDR) modparam("dialog", "profiles_with_value", "total_dialogs") modparam("dialog", "enable_dmq", 1) route[INITIAL] { ...located at the bottom of the initial route... if(is_method("INVITE") && !has_totag()) { dlg_manage(); } } Cheers, Olli

2 2

About SMSC multi-byte support
by shigeru ishida 22 Mar '22

22 Mar '22

Hi all, I have a little question. I tried SMS over IMS on my smartphones with docker_open5gs (using Kamailio and Open5GS), and the multibyte message (Japanese) was garbled at the recipient smartphone. docker_open5gs URL: https://github.com/herlesupreeth/docker_open5gs Also, docker_open5gs uses Kamailio v5.3. I would be very glad if you could tell me about Kamailio's support for multi-byte SMS. Best regards, --Shigeru

2 2

Kamailio : Parallel Forking
by Paclinio 21 Mar '22

21 Mar '22

I've been trying to do parallel forking but i've a problem. It's when the conversation ends that the user that is in a private network can't send a bye to kamailio and i don't know why. But the on in public network (4G) can. I think it's related to record_route() or Via but I'm not sure. I'm also trying to use lookup on users that are in my domain but the problem is that I don't know how to go through all my append_branch() and then t_relay. My Audio with RTPENGINE works correctly, the only problem is really this SIP method that doesn't go to kamailio. Config : request_route { route(INVITE); exit; } route[INVITE]{ route(RECORD_ROUTE); route(GROUP_RELAY_SER); switch($tU) { case "112": xlog("MY_SEVERITY"," -- Calling Group Serial"); route(GROUP_RELAY_SER); break; case "113": // This number does not exist (It's not a subscriber). xlog("MY_SEVERITY"," -- Calling Group Parallel"); route(GROUP_RELAY_PAR); break; default: xlog("MY_SEVERITY"," -- Calling Someone"); route(LOCATION); route(RELAY); } exit; } route[RECORD_ROUTE]{ # - remove preloaded route headers remove_hf("Route"); if(is_method("INVITE|SUBSCRIBE")) { # record routing for dialog forming requests (in case they are routed) record_route(); } } route[GROUP_RELAY_PAR]{ t_on_reply("MANAGE_REPLY"); #!ifndef WITH_BRIDGE_ON_FAIL setbflag(FLB_BRIDGE); #!endif route(IPV4V6); route(NATMANAGE); append_branch("sip:9919995@192.168.1.24:5066"); append_branch("sip:9919992@192.168.1.26:5066"); append_branch("sip:9919991@192.168.1.10:5067"); // I Would like to do an IP in my domain and then do a lookup but I don't know how to do it. t_on_failure("GroupFailureRoute"); #Relay (aka Forward) the request t_relay(); break; } # User location service route[GROUP_LOCATION] { $avp(oexten) = $rU; if (lookup_branches("location")){ return (1); } else { return (-1); } } failure_route[GroupFailureRoute]{ t_next_contacts(); while(!route(GROUP_LOCATION)){ // Check if the next user (in serial) is logged in) if (!t_next_contacts()) { send_reply("408","Nobody available"); exit; } } t_on_failure("GroupFailureRoute"); t_relay(); } # Manage incoming replies in transaction context onreply_route[MANAGE_REPLY] { if(status=~"[12][0-9][0-9]") { if(has_body("application/sdp")){ setbflag(FLB_BRIDGE); } route(NATMANAGE); } } ----------------- Sip Trace : 2022/03/21 18:36:46.469257 149.154.195.121:9665 -> 192.168.1.41:5060 INVITE sip:113@192.168.1.41:5060;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 100.79.133.159:5060 ;branch=z9hG4bK-524287-1---4108e717927214fd;rport Max-Forwards: 70 Contact: <sip:9919993@100.79.133.159:5060;transport=tcp> To: <sip:113@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper v2.10.17.3-mod Allow-Events: presence, kpml, talk Content-Length: 226 v=0 o=Zoiper 53798437 1 IN IP4 100.79.133.159 s=Z c=IN IP4 100.79.133.159 t=0 0 m=audio 8000 RTP/AVP 0 101 8 3 97 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=sendrecv 2022/03/21 18:36:46.470066 192.168.1.41:5060 -> 149.154.195.121:9665 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/TCP 100.79.133.159:5060 ;branch=z9hG4bK-524287-1---4108e717927214fd;rport=9665;received=149.154.195.121 To: <sip:113@192.168.1.41:5060 >;tag=cbfe6543b9079987bce3b5b9a0d1195a.6dfd9117 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 1 INVITE Proxy-Authenticate: Digest realm="192.168.1.41", nonce="Yji42mI4t64hHbET9W5/8CcWSWEhiKD/" Server: kamailio (5.5.4 (x86_64/linux)) Content-Length: 0 2022/03/21 18:36:46.500471 149.154.195.121:9665 -> 192.168.1.41:5060 ACK sip:113@192.168.1.41:5060;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 100.79.133.159:5060 ;branch=z9hG4bK-524287-1---4108e717927214fd;rport Max-Forwards: 70 To: <sip:113@192.168.1.41:5060 >;tag=cbfe6543b9079987bce3b5b9a0d1195a.6dfd9117 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 1 ACK Content-Length: 0 2022/03/21 18:36:46.549993 149.154.195.121:9665 -> 192.168.1.41:5060 INVITE sip:113@192.168.1.41:5060;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 100.79.133.159:5060 ;branch=z9hG4bK-524287-1---75020835629f7238;rport Max-Forwards: 70 Contact: <sip:9919993@100.79.133.159:5060;transport=tcp> To: <sip:113@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Proxy-Authorization: Digest username="9919993",realm="192.168.1.41",nonce="Yji42mI4t64hHbET9W5/8CcWSWEhiKD/",uri="sip:113@Public_IP ;transport=TCP",response="c8654099b37ca083ace2b245f22ebc71",algorithm=MD5 User-Agent: Zoiper v2.10.17.3-mod Allow-Events: presence, kpml, talk Content-Length: 226 v=0 o=Zoiper 53798437 1 IN IP4 100.79.133.159 s=Z c=IN IP4 100.79.133.159 t=0 0 m=audio 8000 RTP/AVP 0 101 8 3 97 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=sendrecv 2022/03/21 18:36:46.556454 192.168.1.41:5060 -> 149.154.195.121:9665 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/TCP 100.79.133.159:5060 ;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665;received=149.154.195.121 To: <sip:113@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Server: kamailio (5.5.4 (x86_64/linux)) Content-Length: 0 2022/03/21 18:36:46.558035 192.168.1.41:55008 -> 192.168.1.41:5060 INVITE sip:113@192.168.1.41:5060;transport=TCP SIP/2.0 Record-Route: <sip:Public_IP;transport=tcp;lr> Via: SIP/2.0/TCP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.0;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 Max-Forwards: 69 Contact: <sip:9919993@100.79.133.159:5060 ;transport=tcp;alias=149.154.195.121~9665~2> To: <sip:113@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper v2.10.17.3-mod Allow-Events: presence, kpml, talk Content-Length: 302 v=0 o=Zoiper 53798437 1 IN IP4 192.168.1.41 s=Z c=IN IP4 192.168.1.41 t=0 0 m=audio 36564 RTP/AVP 0 101 8 3 97 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=sendrecv a=rtcp:36565 2022/03/21 18:36:46.558616 192.168.1.41:5060 -> 192.168.1.41:55008 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/TCP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.0;i=6;rport=55008;received=192.168.1.41 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 To: <sip:113@192.168.1.41:5060 >;tag=cbfe6543b9079987bce3b5b9a0d1195a.4ecd297d From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Proxy-Authenticate: Digest realm="192.168.1.41", nonce="Yji42mI4t64hHbET9W5/8CcWSWEhiKD/" Server: kamailio (5.5.4 (x86_64/linux)) Content-Length: 0 2022/03/21 18:36:46.559281 192.168.1.41:55008 -> 192.168.1.41:5060 ACK sip:113@192.168.1.41:5060;transport=TCP SIP/2.0 Via: SIP/2.0/TCP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.0;i=6 Max-Forwards: 69 To: <sip:113@192.168.1.41:5060 >;tag=cbfe6543b9079987bce3b5b9a0d1195a.4ecd297d From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 ACK Content-Length: 0 2022/03/21 18:36:46.559660 192.168.1.41:5060 -> 192.168.1.24:5066 INVITE sip:9919995@192.168.1.24:5066 SIP/2.0 Record-Route: <sip:Public_IP:5060;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.1;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 Max-Forwards: 69 Contact: <sip:9919993@100.79.133.159:5060 ;transport=tcp;alias=149.154.195.121~9665~2> To: <sip:113@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper v2.10.17.3-mod Allow-Events: presence, kpml, talk Content-Length: 302 v=0 o=Zoiper 53798437 1 IN IP4 192.168.1.41 s=Z c=IN IP4 192.168.1.41 t=0 0 m=audio 36564 RTP/AVP 0 101 8 3 97 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=sendrecv a=rtcp:36565 2022/03/21 18:36:46.559717 192.168.1.41:5060 -> 192.168.1.26:5066 INVITE sip:9919992@192.168.1.26:5066 SIP/2.0 Record-Route: <sip:Public_IP:5060;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.2;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 Max-Forwards: 69 Contact: <sip:9919993@100.79.133.159:5060 ;transport=tcp;alias=149.154.195.121~9665~2> To: <sip:113@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper v2.10.17.3-mod Allow-Events: presence, kpml, talk Content-Length: 302 v=0 o=Zoiper 53798437 1 IN IP4 192.168.1.41 s=Z c=IN IP4 192.168.1.41 t=0 0 m=audio 36564 RTP/AVP 0 101 8 3 97 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=sendrecv a=rtcp:36565 2022/03/21 18:36:46.559744 192.168.1.41:5060 -> 192.168.1.10:5067 INVITE sip:9919991@192.168.1.10:5067 SIP/2.0 Record-Route: <sip:Public_IP:5060;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.3;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 Max-Forwards: 69 Contact: <sip:9919993@100.79.133.159:5060 ;transport=tcp;alias=149.154.195.121~9665~2> To: <sip:113@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper v2.10.17.3-mod Allow-Events: presence, kpml, talk Content-Length: 302 v=0 o=Zoiper 53798437 1 IN IP4 192.168.1.41 s=Z c=IN IP4 192.168.1.41 t=0 0 m=audio 36564 RTP/AVP 0 101 8 3 97 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=sendrecv a=rtcp:36565 2022/03/21 18:36:46.566320 192.168.1.24:5066 -> 192.168.1.41:5060 SIP/2.0 100 Trying To: <sip:113@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.1;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 Record-Route: <sip:Public_IP:5060;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Server: Cisco/SPA508G-7.6.2c Content-Length: 0 2022/03/21 18:36:46.580166 192.168.1.24:5066 -> 192.168.1.41:5060 SIP/2.0 180 Ringing To: <sip:113@192.168.1.41:5060>;tag=57af61cc7a9aaed7i0 From: "codec3-public"<sip:9919993@192.168.1.41:5060>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.1;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 Record-Route: <sip:Public_IP:5060;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: "codec5-private" <sip:9919995@192.168.1.24:5066> Server: Cisco/SPA508G-7.6.2c Content-Length: 0 2022/03/21 18:36:46.580487 192.168.1.41:5060 -> 149.154.195.121:9665 SIP/2.0 180 Ringing To: <sip:113@192.168.1.41:5060>;tag=57af61cc7a9aaed7i0 From: "codec3-public"<sip:9919993@192.168.1.41:5060>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 Record-Route: <sip:Public_IP:5060;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: "codec5-private" <sip:9919995@192.168.1.24:5066> Server: Cisco/SPA508G-7.6.2c Content-Length: 0 2022/03/21 18:36:46.640281 192.168.1.26:5066 -> 192.168.1.41:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.2;received=192.168.1.41;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 To: <sip:113@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Content-Length: 0 2022/03/21 18:36:46.658312 192.168.1.10:5067 -> 192.168.1.41:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.3;received=192.168.1.41;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 To: <sip:113@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Content-Length: 0 2022/03/21 18:36:46.740855 192.168.1.26:5066 -> 192.168.1.41:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.2;received=192.168.1.41;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 Record-Route: <sip:Public_IP:5060;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: <sip:9919992@192.168.1.26:5066> To: <sip:113@192.168.1.41:5060>;tag=28e56d08 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Z 5.5.9 v2.10.17.3 Allow-Events: presence, kpml, talk Content-Length: 0 2022/03/21 18:36:46.741955 192.168.1.41:5060 -> 149.154.195.121:9665 SIP/2.0 180 Ringing Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 Record-Route: <sip:Public_IP:5060;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: <sip:9919992@192.168.1.26:5066> To: <sip:113@192.168.1.41:5060>;tag=28e56d08 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Z 5.5.9 v2.10.17.3 Allow-Events: presence, kpml, talk Content-Length: 0 2022/03/21 18:36:46.769015 192.168.1.10:5067 -> 192.168.1.41:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.3;received=192.168.1.41;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 Record-Route: <sip:Public_IP:5060;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: <sip:9919991@192.168.1.10:50561;transport=TCP> To: <sip:113@192.168.1.41:5060>;tag=be610079 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Z 5.5.10 v2.10.17.3 Allow-Events: presence, kpml, talk Content-Length: 0 2022/03/21 18:36:46.769313 192.168.1.41:5060 -> 149.154.195.121:9665 SIP/2.0 180 Ringing Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 Record-Route: <sip:Public_IP:5060;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: <sip:9919991@192.168.1.10:50561;transport=TCP> To: <sip:113@192.168.1.41:5060>;tag=be610079 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Z 5.5.10 v2.10.17.3 Allow-Events: presence, kpml, talk Content-Length: 0 2022/03/21 18:36:48.945301 192.168.1.26:5066 -> 192.168.1.41:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.2;received=192.168.1.41;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 Record-Route: <sip:Public_IP:5060;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: <sip:9919992@192.168.1.26:5066> To: <sip:113@192.168.1.41:5060>;tag=28e56d08 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Z 5.5.9 v2.10.17.3 Allow-Events: presence, kpml, talk Content-Length: 165 v=0 o=Z 0 1 IN IP4 192.168.1.26 s=Z c=IN IP4 192.168.1.26 t=0 0 m=audio 8000 RTP/AVP 0 8 3 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv 2022/03/21 18:36:48.947154 192.168.1.41:5060 -> 149.154.195.121:9665 SIP/2.0 200 OK Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 Record-Route: <sip:Public_IP:5060;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: <sip:9919992@192.168.1.26:5066> To: <sip:113@192.168.1.41:5060>;tag=28e56d08 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Z 5.5.9 v2.10.17.3 Allow-Events: presence, kpml, talk Content-Length: 245 v=0 o=Z 0 1 IN IP4 192.168.1.41 s=Z c=IN IP4 192.168.1.41 t=0 0 m=audio 36572 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:36573 2022/03/21 18:36:48.953782 192.168.1.41:5060 -> 192.168.1.24:5066 CANCEL sip:9919995@192.168.1.24:5066 SIP/2.0 Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.1;i=6 Max-Forwards: 69 To: <sip:113@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 CANCEL Content-Length: 0 Reason: SIP;cause=200;text="Call completed elsewhere" 2022/03/21 18:36:48.953817 192.168.1.41:5060 -> 192.168.1.10:5067 CANCEL sip:9919991@192.168.1.10:5067 SIP/2.0 Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.3;i=6 Max-Forwards: 69 To: <sip:113@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 CANCEL Content-Length: 0 Reason: SIP;cause=200;text="Call completed elsewhere" 2022/03/21 18:36:48.964605 192.168.1.24:5066 -> 192.168.1.41:5060 SIP/2.0 487 Request Terminated To: <sip:113@192.168.1.41:5060>;tag=57af61cc7a9aaed7i0 From: "codec3-public"<sip:9919993@192.168.1.41:5060>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.1;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 Record-Route: <sip:Public_IP:5060;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Server: Cisco/SPA508G-7.6.2c Content-Length: 0 2022/03/21 18:36:48.964766 192.168.1.41:5060 -> 192.168.1.24:5066 ACK sip:9919995@192.168.1.24:5066 SIP/2.0 Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.1;i=6 Max-Forwards: 69 To: <sip:113@192.168.1.41:5060>;tag=57af61cc7a9aaed7i0 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 ACK Content-Length: 0 2022/03/21 18:36:48.966945 192.168.1.24:5066 -> 192.168.1.41:5060 SIP/2.0 200 OK To: <sip:113@192.168.1.41:5060>;tag=57af61cc7a9aaed7i0 From: "codec3-public"<sip:9919993@192.168.1.41:5060>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 CANCEL Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.1;i=6 Server: Cisco/SPA508G-7.6.2c Content-Length: 0 2022/03/21 18:36:48.989036 192.168.1.10:5067 -> 192.168.1.41:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.3;received=192.168.1.41;i=6 Contact: <sip:9919991@192.168.1.10:50561;transport=TCP> To: <sip:113@192.168.1.41:5060>;tag=be610079 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 CANCEL User-Agent: Z 5.5.10 v2.10.17.3 Content-Length: 0 2022/03/21 18:36:48.989036 192.168.1.10:5067 -> 192.168.1.41:5060 SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.3;received=192.168.1.41;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---75020835629f7238;rport=9665 To: <sip:113@192.168.1.41:5060>;tag=be610079 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 INVITE User-Agent: Z 5.5.10 v2.10.17.3 Content-Length: 0 2022/03/21 18:36:48.989268 192.168.1.41:5060 -> 192.168.1.10:5067 ACK sip:9919991@192.168.1.10:5067 SIP/2.0 Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.08911f0196683c93f4519bca6c923da6.3;i=6 Max-Forwards: 69 To: <sip:113@192.168.1.41:5060>;tag=be610079 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 ACK Content-Length: 0 2022/03/21 18:36:49.021133 149.154.195.121:9665 -> 192.168.1.41:5060 ACK sip:9919992@192.168.1.26:5066 SIP/2.0 Via: SIP/2.0/TCP 100.79.133.159:5060 ;branch=z9hG4bK-524287-1---28ec75a236e47eb1;rport Max-Forwards: 70 Route: <sip:Public_IP;transport=tcp;r2=on;lr> Route: <sip:Public_IP:5060;r2=on;lr> Contact: <sip:9919993@100.79.133.159:5060;transport=tcp> To: <sip:113@192.168.1.41:5060>;tag=28e56d08 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 ACK User-Agent: Zoiper v2.10.17.3-mod Content-Length: 0 2022/03/21 18:36:49.021618 192.168.1.41:5060 -> 192.168.1.26:5066 ACK sip:9919992@192.168.1.26:5066 SIP/2.0 Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK04b4.064a5839a4d237786d792a9b52b0a37c.0;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---28ec75a236e47eb1;rport=9665 Max-Forwards: 69 Contact: <sip:9919993@100.79.133.159:5060 ;transport=tcp;alias=149.154.195.121~9665~2> To: <sip:113@192.168.1.41:5060>;tag=28e56d08 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 2 ACK User-Agent: Zoiper v2.10.17.3-mod Content-Length: 0 2022/03/21 18:37:23.350203 149.154.195.121:9665 -> 192.168.1.41:5060 BYE sip:9919992@192.168.1.26:5066 SIP/2.0 Via: SIP/2.0/TCP 100.79.133.159:5060 ;branch=z9hG4bK-524287-1---e5232f6b2a0b88d6;rport Max-Forwards: 70 Route: <sip:Public_IP;transport=tcp;r2=on;lr> Route: <sip:Public_IP:5060;r2=on;lr> Contact: <sip:9919993@100.79.133.159:5060;transport=tcp> To: <sip:113@192.168.1.41:5060>;tag=28e56d08 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 3 BYE Proxy-Authorization: Digest username="9919993",realm="192.168.1.41",nonce="Yji42mI4t64hHbET9W5/8CcWSWEhiKD/",uri=" sip:9919992@192.168.1.26:5066 ",response="a7440eb01dfb200b92cc1320808251da",algorithm=MD5 User-Agent: Zoiper v2.10.17.3-mod Content-Length: 0 2022/03/21 18:37:23.357139 192.168.1.41:5060 -> 192.168.1.26:5066 BYE sip:9919992@192.168.1.26:5066 SIP/2.0 Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK14b4.ece7c52456dd1d2ba5e46ca9a10b6cee.0;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---e5232f6b2a0b88d6;rport=9665 Max-Forwards: 69 Contact: <sip:9919993@100.79.133.159:5060 ;transport=tcp;alias=149.154.195.121~9665~2> To: <sip:113@192.168.1.41:5060>;tag=28e56d08 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 3 BYE Proxy-Authorization: Digest username="9919993",realm="192.168.1.41",nonce="Yji42mI4t64hHbET9W5/8CcWSWEhiKD/",uri=" sip:9919992@192.168.1.26:5066 ",response="a7440eb01dfb200b92cc1320808251da",algorithm=MD5 User-Agent: Zoiper v2.10.17.3-mod Content-Length: 0 2022/03/21 18:37:23.359207 192.168.1.26:5066 -> 192.168.1.41:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP Public_IP:5060;branch=z9hG4bK14b4.ece7c52456dd1d2ba5e46ca9a10b6cee.0;received=192.168.1.41;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---e5232f6b2a0b88d6;rport=9665 Contact: <sip:9919992@192.168.1.26:5066> To: <sip:113@192.168.1.41:5060>;tag=28e56d08 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 3 BYE User-Agent: Z 5.5.9 v2.10.17.3 Content-Length: 0 2022/03/21 18:37:23.359668 192.168.1.41:5060 -> 149.154.195.121:9665 SIP/2.0 200 OK Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---e5232f6b2a0b88d6;rport=9665 Contact: <sip:9919992@192.168.1.26:5066> To: <sip:113@192.168.1.41:5060>;tag=28e56d08 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=ed51991c Call-ID: GlECSiXh3tJ37DqYLLvm_g.. CSeq: 3 BYE User-Agent: Z 5.5.9 v2.10.17.3 Content-Length: 0 Thank you. <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> Virus-free. www.avast.com <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

1 0

Kamailio : Serial forking
by Paclinio 21 Mar '22

21 Mar '22

I've been trying to do serial forking, I've configured something that is working but when I'm looking on github and other sources it seems like my code is a little too complicated and I would like to know if there is a simpler config that I can use. To have more precision, I'm using RTPENGINE and I need to be able to go through NAT. My Config : request_route { route(INVITE); exit; } route[INVITE]{ route(RECORD_ROUTE); route(GROUP_RELAY_SER); switch($tU) { case "112": // This number does not exist (It's not a subscriber). xlog("MY_SEVERITY"," -- Calling Group Serial"); route(GROUP_RELAY_SER); break; case "113": xlog("MY_SEVERITY"," -- Calling Group Parralelle"); route(GROUP_RELAY_PAR); break; default: xlog("MY_SEVERITY"," -- Calling Someone"); route(LOCATION); route(RELAY); } exit; } route[RECORD_ROUTE]{ # - remove preloaded route headers remove_hf("Route"); if(is_method("INVITE|SUBSCRIBE")) { # record routing for dialog forming requests (in case they are routed) record_route(); } } route[GROUP_RELAY_SER]{ t_on_reply("MANAGE_REPLY"); route(IPV4V6); route(NATMANAGE); (RTP Routing is working correctly) append_branch("sip:9919995@192.168.1.41","1"); append_branch("sip:9919992@192.168.1.41", "0.2"); append_branch("sip:75575757@192.168.1.41", "0.4"); append_branch("sip:9919991@192.168.1.41", "0.5"); t_load_contacts(); t_next_contacts(); while(!route(GROUP_LOCATION)) { if (!t_next_contacts()) { send_reply("408","Timeout or nobody available"); exit; } } t_on_failure("GroupFailureRoute"); #Relay (aka Forward) the request t_relay(); break; } # User location service route[GROUP_LOCATION] { $avp(oexten) = $rU; if (lookup_branches("location")){ return (1); } else { return (-1); } } failure_route[GroupFailureRoute]{ t_next_contacts(); while(!route(GROUP_LOCATION)){ if (!t_next_contacts()) { send_reply("408","Nobody available"); exit; } } t_on_failure("GroupFailureRoute"); t_relay(); } # Manage incoming replies in transaction context onreply_route[MANAGE_REPLY] { if(status=~"[12][0-9][0-9]") { if(has_body("application/sdp")){ setbflag(FLB_BRIDGE); } route(NATMANAGE); } } --------------------------------- Here's The SIP Trace : 2022/03/21 18:28:45.265009 149.154.195.121:9665 -> 192.168.1.41:5060 INVITE sip:112@192.168.1.41:5060;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 100.79.133.159:5060 ;branch=z9hG4bK-524287-1---75083bb7c6ed8a35;rport Max-Forwards: 70 Contact: <sip:9919993@100.79.133.159:5060;transport=tcp> To: <sip:112@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper v2.10.17.3-mod Allow-Events: presence, kpml, talk Content-Length: 226 v=0 o=Zoiper 53721112 1 IN IP4 100.79.133.159 s=Z c=IN IP4 100.79.133.159 t=0 0 m=audio 8000 RTP/AVP 0 101 8 3 97 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=sendrecv 2022/03/21 18:28:45.274652 192.168.1.41:5060 -> 149.154.195.121:9665 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/TCP 100.79.133.159:5060 ;branch=z9hG4bK-524287-1---75083bb7c6ed8a35;rport=9665;received=149.154.195.121 To: <sip:112@192.168.1.41:5060 >;tag=cbfe6543b9079987bce3b5b9a0d1195a.064b0d7d From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 1 INVITE Proxy-Authenticate: Digest realm="192.168.1.41", nonce="Yji2+WI4tc0uQkJ6GbQ4I9VeSUQmR6Gk" Server: kamailio (5.5.4 (x86_64/linux)) Content-Length: 0 2022/03/21 18:28:45.335876 149.154.195.121:9665 -> 192.168.1.41:5060 ACK sip:112@192.168.1.41:5060;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 100.79.133.159:5060 ;branch=z9hG4bK-524287-1---75083bb7c6ed8a35;rport Max-Forwards: 70 To: <sip:112@192.168.1.41:5060 >;tag=cbfe6543b9079987bce3b5b9a0d1195a.064b0d7d From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 1 ACK Content-Length: 0 2022/03/21 18:28:45.385686 149.154.195.121:9665 -> 192.168.1.41:5060 INVITE sip:112@192.168.1.41:5060;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 100.79.133.159:5060 ;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport Max-Forwards: 70 Contact: <sip:9919993@100.79.133.159:5060;transport=tcp> To: <sip:112@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Proxy-Authorization: Digest username="9919993",realm="192.168.1.41",nonce="Yji2+WI4tc0uQkJ6GbQ4I9VeSUQmR6Gk",uri="sip:112@Public_IP ;transport=TCP",response="54837810a66298e4542486bb6ab8ca04",algorithm=MD5 User-Agent: Zoiper v2.10.17.3-mod Allow-Events: presence, kpml, talk Content-Length: 226 v=0 o=Zoiper 53721112 1 IN IP4 100.79.133.159 s=Z c=IN IP4 100.79.133.159 t=0 0 m=audio 8000 RTP/AVP 0 101 8 3 97 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=sendrecv 2022/03/21 18:28:45.418871 192.168.1.41:5060 -> 149.154.195.121:9665 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/TCP 100.79.133.159:5060 ;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665;received=149.154.195.121 To: <sip:112@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Server: kamailio (5.5.4 (x86_64/linux)) Content-Length: 0 2022/03/21 18:28:45.421797 192.168.1.41:5066 -> 192.168.1.24:5066 INVITE sip:9919995@192.168.1.24:5066 SIP/2.0 Record-Route: <sip:192.168.1.41:5066;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Via: SIP/2.0/UDP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.0;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 Max-Forwards: 69 Contact: <sip:9919993@100.79.133.159:5060 ;transport=tcp;alias=149.154.195.121~9665~2> To: <sip:112@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper v2.10.17.3-mod Allow-Events: presence, kpml, talk Content-Length: 302 v=0 o=Zoiper 53721112 1 IN IP4 192.168.1.41 s=Z c=IN IP4 192.168.1.41 t=0 0 m=audio 36502 RTP/AVP 0 101 8 3 97 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=sendrecv a=rtcp:36503 2022/03/21 18:28:45.435659 192.168.1.24:5066 -> 192.168.1.41:5066 SIP/2.0 100 Trying To: <sip:112@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Via: SIP/2.0/UDP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.0;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 Record-Route: <sip:192.168.1.41:5066;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Server: Cisco/SPA508G-7.6.2c Content-Length: 0 2022/03/21 18:28:45.449138 192.168.1.24:5066 -> 192.168.1.41:5066 SIP/2.0 180 Ringing To: <sip:112@192.168.1.41:5060>;tag=694833b3bcc5722bi0 From: "codec3-public"<sip:9919993@192.168.1.41:5060>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Via: SIP/2.0/UDP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.0;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 Record-Route: <sip:192.168.1.41:5066;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: "codec5-private" <sip:9919995@192.168.1.24:5066> Server: Cisco/SPA508G-7.6.2c Content-Length: 0 2022/03/21 18:28:45.455211 192.168.1.41:5060 -> 149.154.195.121:9665 SIP/2.0 180 Ringing To: <sip:112@192.168.1.41:5060>;tag=694833b3bcc5722bi0 From: "codec3-public"<sip:9919993@192.168.1.41:5060>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 Record-Route: <sip:192.168.1.41:5066;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: "codec5-private" <sip:9919995@192.168.1.24:5066 ;alias=192.168.1.24~5066~1> Server: Cisco/SPA508G-7.6.2c Content-Length: 0 2022/03/21 18:28:55.445916 192.168.1.41:5066 -> 192.168.1.10:50561 INVITE sip:9919991@192.168.1.10:50561;transport=TCP;rinstance=034d5c22d5466df2 SIP/2.0 Record-Route: <sip:192.168.1.41:5066;transport=tcp;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Via: SIP/2.0/TCP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.1;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 Max-Forwards: 69 Contact: <sip:9919993@100.79.133.159:5060 ;transport=tcp;alias=149.154.195.121~9665~2> To: <sip:112@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper v2.10.17.3-mod Allow-Events: presence, kpml, talk Content-Length: 302 v=0 o=Zoiper 53721112 1 IN IP4 192.168.1.41 s=Z c=IN IP4 192.168.1.41 t=0 0 m=audio 36502 RTP/AVP 0 101 8 3 97 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=sendrecv a=rtcp:36503 2022/03/21 18:28:55.446096 192.168.1.41:5066 -> 192.168.1.24:5066 CANCEL sip:9919995@192.168.1.24:5066 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.0;i=6 Max-Forwards: 69 To: <sip:112@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 CANCEL Content-Length: 0 2022/03/21 18:28:55.457941 192.168.1.24:5066 -> 192.168.1.41:5066 SIP/2.0 487 Request Terminated To: <sip:112@192.168.1.41:5060>;tag=694833b3bcc5722bi0 From: "codec3-public"<sip:9919993@192.168.1.41:5060>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Via: SIP/2.0/UDP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.0;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 Record-Route: <sip:192.168.1.41:5066;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Server: Cisco/SPA508G-7.6.2c Content-Length: 0 2022/03/21 18:28:55.458874 192.168.1.41:5066 -> 192.168.1.24:5066 ACK sip:9919995@192.168.1.24:5066 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.0;i=6 Max-Forwards: 69 To: <sip:112@192.168.1.41:5060>;tag=694833b3bcc5722bi0 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 ACK Content-Length: 0 2022/03/21 18:28:55.459394 192.168.1.24:5066 -> 192.168.1.41:5066 SIP/2.0 200 OK To: <sip:112@192.168.1.41:5060>;tag=694833b3bcc5722bi0 From: "codec3-public"<sip:9919993@192.168.1.41:5060>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 CANCEL Via: SIP/2.0/UDP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.0;i=6 Server: Cisco/SPA508G-7.6.2c Content-Length: 0 2022/03/21 18:28:55.536475 192.168.1.10:50561 -> 192.168.1.41:5066 SIP/2.0 100 Trying Via: SIP/2.0/TCP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.1;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 To: <sip:112@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Content-Length: 0 2022/03/21 18:28:55.758738 192.168.1.10:50561 -> 192.168.1.41:5066 SIP/2.0 180 Ringing Via: SIP/2.0/TCP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.1;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 Record-Route: <sip:192.168.1.41:5066;transport=tcp;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: <sip:9919991@192.168.1.10:50561;transport=TCP> To: <sip:112@192.168.1.41:5060>;tag=575a7d2e From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Z 5.5.10 v2.10.17.3 Allow-Events: presence, kpml, talk Content-Length: 0 2022/03/21 18:28:55.763664 192.168.1.41:5060 -> 149.154.195.121:9665 SIP/2.0 180 Ringing Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 Record-Route: <sip:192.168.1.41:5066;transport=tcp;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: <sip:9919991@192.168.1.10:50561 ;transport=TCP;alias=192.168.1.10~50561~2> To: <sip:112@192.168.1.41:5060>;tag=575a7d2e From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Z 5.5.10 v2.10.17.3 Allow-Events: presence, kpml, talk Content-Length: 0 2022/03/21 18:29:05.761477 192.168.1.41:5066 -> 192.168.1.26:32821 INVITE sip:9919992@192.168.1.26:5066;rinstance=791ce762698e7422;transport=tcp SIP/2.0 Record-Route: <sip:192.168.1.41:5066;transport=tcp;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Via: SIP/2.0/TCP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.2;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 Max-Forwards: 69 Contact: <sip:9919993@100.79.133.159:5060 ;transport=tcp;alias=149.154.195.121~9665~2> To: <sip:112@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper v2.10.17.3-mod Allow-Events: presence, kpml, talk Content-Length: 302 v=0 o=Zoiper 53721112 1 IN IP4 192.168.1.41 s=Z c=IN IP4 192.168.1.41 t=0 0 m=audio 36502 RTP/AVP 0 101 8 3 97 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=sendrecv a=rtcp:36503 2022/03/21 18:29:05.761547 192.168.1.41:5066 -> 192.168.1.10:50561 CANCEL sip:9919991@192.168.1.10:50561;transport=TCP;rinstance=034d5c22d5466df2 SIP/2.0 Via: SIP/2.0/TCP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.1;i=6 Max-Forwards: 69 To: <sip:112@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 CANCEL Content-Length: 0 2022/03/21 18:29:05.799795 192.168.1.10:50561 -> 192.168.1.41:5066 SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.1;i=6 Contact: <sip:9919991@192.168.1.10:50561;transport=TCP> To: <sip:112@192.168.1.41:5060>;tag=575a7d2e From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 CANCEL User-Agent: Z 5.5.10 v2.10.17.3 Content-Length: 0 2022/03/21 18:29:05.800776 192.168.1.10:50561 -> 192.168.1.41:5066 SIP/2.0 487 Request Terminated Via: SIP/2.0/TCP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.1;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 To: <sip:112@192.168.1.41:5060>;tag=575a7d2e From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE User-Agent: Z 5.5.10 v2.10.17.3 Content-Length: 0 2022/03/21 18:29:05.801104 192.168.1.41:5066 -> 192.168.1.10:50561 ACK sip:9919991@192.168.1.10:50561;transport=TCP;rinstance=034d5c22d5466df2 SIP/2.0 Via: SIP/2.0/TCP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.1;i=6 Max-Forwards: 69 To: <sip:112@192.168.1.41:5060>;tag=575a7d2e From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 ACK Content-Length: 0 2022/03/21 18:29:05.844860 192.168.1.26:32821 -> 192.168.1.41:5066 SIP/2.0 100 Trying Via: SIP/2.0/TCP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.2;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 To: <sip:112@192.168.1.41:5060> From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Content-Length: 0 2022/03/21 18:29:06.045373 192.168.1.26:32821 -> 192.168.1.41:5066 SIP/2.0 180 Ringing Via: SIP/2.0/TCP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.2;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 Record-Route: <sip:192.168.1.41:5066;transport=tcp;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: <sip:9919992@192.168.1.26:5066;transport=tcp> To: <sip:112@192.168.1.41:5060>;tag=dd58ff56 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Z 5.5.9 v2.10.17.3 Allow-Events: presence, kpml, talk Content-Length: 0 2022/03/21 18:29:06.046528 192.168.1.41:5060 -> 149.154.195.121:9665 SIP/2.0 180 Ringing Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 Record-Route: <sip:192.168.1.41:5066;transport=tcp;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: <sip:9919992@192.168.1.26:5066 ;transport=tcp;alias=192.168.1.26~32821~2> To: <sip:112@192.168.1.41:5060>;tag=dd58ff56 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Z 5.5.9 v2.10.17.3 Allow-Events: presence, kpml, talk Content-Length: 0 2022/03/21 18:29:11.055497 192.168.1.26:32821 -> 192.168.1.41:5066 SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.1.41:5066 ;branch=z9hG4bKf831.6cdc4420824fc17383578fd933a4963b.2;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 Record-Route: <sip:192.168.1.41:5066;transport=tcp;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: <sip:9919992@192.168.1.26:5066;transport=tcp> To: <sip:112@192.168.1.41:5060>;tag=dd58ff56 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Z 5.5.9 v2.10.17.3 Allow-Events: presence, kpml, talk Content-Length: 165 v=0 o=Z 0 1 IN IP4 192.168.1.26 s=Z c=IN IP4 192.168.1.26 t=0 0 m=audio 8000 RTP/AVP 0 8 3 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv 2022/03/21 18:29:11.061482 192.168.1.41:5060 -> 149.154.195.121:9665 SIP/2.0 200 OK Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---ba46a700c4dfa090;rport=9665 Record-Route: <sip:192.168.1.41:5066;transport=tcp;r2=on;lr> Record-Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: <sip:9919992@192.168.1.26:5066 ;transport=tcp;alias=192.168.1.26~32821~2> To: <sip:112@192.168.1.41:5060>;tag=dd58ff56 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Z 5.5.9 v2.10.17.3 Allow-Events: presence, kpml, talk Content-Length: 245 v=0 o=Z 0 1 IN IP4 192.168.1.41 s=Z c=IN IP4 192.168.1.41 t=0 0 m=audio 36518 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:36519 2022/03/21 18:29:11.144532 149.154.195.121:9665 -> 192.168.1.41:5060 ACK sip:9919992@192.168.1.26:5066;transport=tcp;alias=192.168.1.26~32821~2 SIP/2.0 Via: SIP/2.0/TCP 100.79.133.159:5060 ;branch=z9hG4bK-524287-1---f49bbd15731e8f8d;rport Max-Forwards: 70 Route: <sip:Public_IP;transport=tcp;r2=on;lr> Route: <sip:192.168.1.41:5066;transport=tcp;r2=on;lr> Contact: <sip:9919993@100.79.133.159:5060;transport=tcp> To: <sip:112@192.168.1.41:5060>;tag=dd58ff56 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 ACK User-Agent: Zoiper v2.10.17.3-mod Content-Length: 0 2022/03/21 18:29:11.152633 192.168.1.41:5066 -> 192.168.1.26:32821 ACK sip:9919992@192.168.1.26:5066;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 192.168.1.41:5066 ;branch=z9hG4bKf831.422a25f74a8ad6c743d75a28764bc8b7.0;i=6 Via: SIP/2.0/TCP 100.79.133.159:5060 ;received=149.154.195.121;branch=z9hG4bK-524287-1---f49bbd15731e8f8d;rport=9665 Max-Forwards: 69 Contact: <sip:9919993@100.79.133.159:5060 ;transport=tcp;alias=149.154.195.121~9665~2> To: <sip:112@192.168.1.41:5060>;tag=dd58ff56 From: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 ACK User-Agent: Zoiper v2.10.17.3-mod Content-Length: 0 2022/03/21 18:29:20.514914 192.168.1.26:32821 -> 192.168.1.41:5066 BYE sip:9919993@100.79.133.159:5060;transport=tcp;alias=149.154.195.121~9665~2 SIP/2.0 Via: SIP/2.0/TCP 192.168.1.26:5066 ;branch=z9hG4bK-524287-1---404c84aa134e2a04 Max-Forwards: 70 Route: <sip:192.168.1.41:5066;transport=tcp;r2=on;lr> Route: <sip:Public_IP;transport=tcp;r2=on;lr> Contact: <sip:9919992@192.168.1.26:5066;transport=tcp> To: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 From: <sip:112@192.168.1.41:5060>;tag=dd58ff56 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 BYE User-Agent: Z 5.5.9 v2.10.17.3 Content-Length: 0 2022/03/21 18:29:20.517617 192.168.1.41:5060 -> 149.154.195.121:9665 BYE sip:9919993@100.79.133.159:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP Public_IP:5060;branch=z9hG4bKf831.2578b48bc3463ab1aed37eba97036001.0;i=5 Via: SIP/2.0/TCP 192.168.1.26:5066 ;rport=32821;branch=z9hG4bK-524287-1---404c84aa134e2a04 Max-Forwards: 69 Contact: <sip:9919992@192.168.1.26:5066 ;transport=tcp;alias=192.168.1.26~32821~2> To: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 From: <sip:112@192.168.1.41:5060>;tag=dd58ff56 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 BYE User-Agent: Z 5.5.9 v2.10.17.3 Content-Length: 0 2022/03/21 18:29:20.584230 149.154.195.121:9665 -> 192.168.1.41:5060 SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.1.41:5060 ;branch=z9hG4bKf831.2578b48bc3463ab1aed37eba97036001.0;i=5 Via: SIP/2.0/TCP 192.168.1.26:5066 ;rport=32821;branch=z9hG4bK-524287-1---404c84aa134e2a04 Contact: <sip:9919993@100.79.133.159:5060;transport=tcp> To: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 From: <sip:112@192.168.1.41:5060>;tag=dd58ff56 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 BYE User-Agent: Zoiper v2.10.17.3-mod Content-Length: 0 2022/03/21 18:29:20.584431 192.168.1.41:5066 -> 192.168.1.26:32821 SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.1.26:5066 ;rport=32821;branch=z9hG4bK-524287-1---404c84aa134e2a04 Contact: <sip:9919993@100.79.133.159:5060;transport=tcp> To: "codec3-public"<sip:9919993@192.168.1.41:5060 ;transport=TCP>;tag=35ce2334 From: <sip:112@192.168.1.41:5060>;tag=dd58ff56 Call-ID: RtH8fBIrIgUYPAkxlxjI4Q.. CSeq: 2 BYE User-Agent: Zoiper v2.10.17.3-mod Content-Length: 0 Thank you. Thomas <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> Virus-free. www.avast.com <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

1 0

Can Kamailio send an ACK to a final response?
by Marrold 21 Mar '22

21 Mar '22

Hello, I'm exploring using Kamailio as a dumb edge proxy, and as such I want to keep things as lean as possible. In some scenarios I would like the "core" proxies to signal to the edge that a request should be dropped without a response. This could be with a custom header in the reply or perhaps a 444 response ( nginx uses this internally to drop a request without responding ) I have discovered that if I drop the response in the reply_route, the edge proxy doesn't send an ACK and keeps retransmitting the request until it times out, at which point it sends a 408 back to the UA. Is there any way to ACK the response from the core and destroy the transaction? Thanks Matthew

3 2

Kamailio WebRTC Call Not Working
by Ankit Jayswal 21 Mar '22

21 Mar '22

Hi, I have configured Kamailio to register a SIP user over websocket. SIP users are getting registered over webphone like JSSIP fine but not able to establish the call scenario below. >From Twilio Trunk - INVITE - KAMAILIO - ASTERISK - play hello world - Dial SIP user 101@Kamailio_IP - Rings 101 and Answer - Hangup. Here, 101 is registered with Kamailio IP, and I am sending a call from Twilio SIP trunk to Kamailio server IP. Asterisk is on port 5080 and Kamailio is on port 5090 both are on the same host 45.118.163.244. This above call scenario is NOT working with Webphones registered via JSSIP. I think the second call leg from Asterisk -> Kamailio is not getting set up properly and ACK is not getting confirmed here, call connects for some time and no voice transmits and then disconnects. The same call scenario is working with users registered at Softphone like Zoiper perfectly fine. No idea why it is not working with a webphone. I have attached kamailio.cfg, Kamailio log and Asterisk's pjsip logs here. So, please guide me on this as soon as possible. -- Thanks & Regards, *Ankit Jayswal* | Specialist - Software Development

1 0

pseudo var for From and To sip uri port
by Patrick Karton 20 Mar '22

20 Mar '22

Hello, is there a pseudo variable for setting only sip uri port for From and To Headers like $rp for R-URI ?

1 0

dialog module issues (BYE Request issued) when topos is enabled
by Patrick Karton 19 Mar '22

19 Mar '22

i am using dialog and topos module. but i have an issue i think with topos module. When topos module is disabled all the call flow works well but when i enable topos module i have this weird behaviour : after Caller sends BYE and complete the call , i receive later a BYE Request issued by the dialog module towards Caller and Callee. Does anyone already had that issue ? modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 1024) modparam("dialog", "rr_param", "oovdlg") modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "db_mode", 0) modparam("dialog", "db_update_period", 30) modparam("dialog", "wait_ack", 1) modparam("dialog", "ka_timer", 10) modparam("dialog", "ka_interval", 30) modparam("dialog", "profiles_with_value", "concurrent_calls") modparam("dialog", "end_timeout", 180) modparam("dialog", "default_timeout", 3600) modparam("dialog", "noack_timeout", 90) modparam("dialog", "early_timeout", 300) modparam("dialog", "dlg_extra_hdrs", "X-OOV-Hint: Dlg-hit\r\n") modparam("db_text", "db_mode", 0) modparam("db_text", "emptystring", 0) modparam("db_text", "file_buffer_size", 8192) modparam("db_text", "max_result_rows", 1000) modparam("topos", "storage", "db") modparam("topos", "db_url", "text:///home/sbc/database") modparam("topos", "mask_callid", 0) modparam("topos", "sanity_checks", 0) modparam("topos", "contact_mode", 0)

1 0

DMQ version compatability
by Ben Kaufman 19 Mar '22

19 Mar '22

Daniel, In a recent thread you stated the following regarding DMQ compatibility between versions: "if you do dmq replication between kamailio systems running different major versions, then it is likely to get memory leaks due to replication of data and most probably cannot be fixed. This is because internal structures of modules (also dmq commands) can change, practically what an instance does is not ensured to happen on the other instance. Just for example, from my mind, htable got some changes during past releases, dmq also has significant enhancements by getting support for more transport protocols." Is it possible to get a bit more clarification on this, particularly as it relates to DMQ_USRLOC? I'm hoping to do rolling updates, but don't want to end up in a situation where I've caused myself more problems than I've solved. Ben Kaufman

3 2

help to configure topos module configuration
by Patrick Karton 18 Mar '22

18 Mar '22

Caller <--> (internal_IP) Kamailio (external_IP) <---> Callee I have this set up of kamailio with 2 interfaces. i use topos module for topology hiding and what i want to do is to send external_IP in Contact Header of request relayed to Callee and internal_IP in Contact Header of response relayed to Caller. i noticed that : when i use topos module with contact_mode to 0 all Record-Route and Caller Via Header are removed and thats great. but i dont find way to put 2 different ip addresses in Request and Response Contact Header? when i use contact_mode to 2 Record-Route and Caller Via Header are not anymore removed thats bad for topology hiding. And its seems we can not use $xavu(_tps_=>contact_host) parameter of module to put 2 different ip addresses in Request and Response Contact Header. What i want is : - remove all Record-Route and Caller Via Header for request like when contact_mode to 0 - and put 2 different ip addresses in Request and Response Contact Header is it possible to achieve it ? Thanks.

2 2

Upgrading from version 5.2 to 5.4 - any unpublished words of wisdom?
by Kevin Greene 18 Mar '22

18 Mar '22

Hi: I have a 5.2 version of Kamailio running and I was thinking to get it upgraded to the 5.4 branch. The 5.2 is working well but I have heard the 5.4 is a bit more stable. Thus the question becomes, is it worth the risk to upgrade and what should I look out for. Thanks. Kevin

2 1

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