- sr-users - kamailio.org

Handling of packet loss in the Kamailio to RTPEngine control protocol
by Patrick Wakano 03 Sep '20

03 Sep '20

Hello list, Hope you are all well. Under some load test simulations I've been facing cases where the command Kamailio sends to RTPEngine times out with such message: send_rtpp_command(): timeout waiting reply for command "" from RTP proxy After checking RTPEngine logs, I can see the command was received and a reply was sent, so I am thinking the reply packet could have been lost somewhere in the network (they are in different servers). So my question is, how resilient is the RTPEngine NG protocol to handle packet loss situations? I saw TCP is not supported, so are there UDP retransmissions in place to guarantee packet delivery? Any ideas to make this connection more reliable? Kind regards, Thank you, Patrick Wakano

2 2

Send_notify_request within cluster scenario(path usrloc)
by Zhan Bazarov 03 Sep '20

03 Sep '20

Hello everyone. I have an issue with presence server within kamailio cluster(round-robin DNS, I'm using registrar Path scenario) So, when both of users are located at the same server - everything works fine. But when user which is SUBSCRIBED through another one server - presence module is ignoring path column into usrloc - that leads NAT traversal impossible... So, I've tried using DMQ for handle this: I executed dmq_t_replicate()before each PUBLISH message - in that case I was able to get PUBLISH on each of server in cluster. But BLF lamps are confusing in that case - because a lot of notifies coming to the device at the same time with different dialog states.... Guys, I need a help. Any Ideas how I can handle this? Best regards, Zhan -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

2 1

Info: Kamailio World Updates - 2nd day starting now
by Daniel-Constantin Mierla 03 Sep '20

03 Sep '20

Hello, we are just about to start the 2nd day of Kamailio World Updates - free online event, no registration required, just come and join us. More details at: * https://www.kamailioworld.com/k08-online/ Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 1

Kamailio IMS as kubernetes pods
by Pavithra Mohanraja 02 Sep '20

02 Sep '20

Hi, I am configuring kamailio IMS as kubernetes pods . so each component will be configured as separate pod . In order to establish a connection between pods , I am configuring bind9 dns server in pcscf pod itself. Communication is happening via Pod IP address. Is there any possibility to have the communication using cluster IP . If anybody is aware of kubernetes, Please help me in this regard. I have my /etc/resolv.conf file in pods with nameserver <clusterIP> search <cluster-dns-domain-name> Please help. Thanks, Pavithra

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error playing media with rtpproxy
by Amit Pal 02 Sep '20

02 Sep '20

Dear Team, I am unable to play media file on caller/callee. I am using rtpproxy_stream2uas/rtproxy_stream2uac to play media file. RTPProxy put on remote machine. Kamailio server are on another machine. When do call media relay by rtpproxy well. Now it is fine. But issue when hold the call by caller try to play moh to callee by rtpproxy_stream2uas("/media/moh.wav" ,"-1") getting following error by RTPProxy. DBUG:GLOBAL:rtpp_command_split:396: received command "6920_4 P-1 f954ba62f373b86cMTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI. /media/moh.wav session aee3e27066a043e19f73cbff37e50689;1 904c5107;1" rtpproxy[10858]: DBUG:f954ba62f373b86cMTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.:rtpp_strea m_handle_play:376: rtpp_server_ctor("/media/moh.wav", 8, -1) failed rtpproxy[10858]: DBUG:f954ba62f373b86cMTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.:rtpp_strea m_handle_play:376: rtpp_server_ctor("/media/moh.wav", 101, -1) failed rtpproxy[10858]: ERR:f954ba62f373b86cMTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.:rtpp_stream _handle_play:405: can't create player: rtpp_server_ctor() failed rtpproxy[10858]: DBUG:GLOBAL:rtpc_doreply:192: sending reply "E60\n" rtpproxy[10858]: DBUG:GLOBAL:rtpp_command_split:396: received command "6920_5 P-1 f954ba62f373b86cMTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI. /media/moh.wav session 904c5107;1 aee3e27066a043e19f73cbff37e50689;1" rtpproxy[10858]: DBUG:f954ba62f373b86cMTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.:rtpp_strea m_handle_play:376: rtpp_server_ctor("/media/moh.wav", 0, -1) failed rtpproxy[10858]: DBUG:f954ba62f373b86cMTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.:rtpp_strea m_handle_play:376: rtpp_server_ctor("/media/moh.wav", 8, -1) failed rtpproxy[10858]: DBUG:f954ba62f373b86cMTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.:rtpp_strea m_handle_play:376: rtpp_server_ctor("/media/moh.wav", 101, -1) failed rtpproxy[10858]: ERR:f954ba62f373b86cMTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.:rtpp_stream _handle_play:405: can't create player: rtpp_server_ctor() failed rtpproxy[10858]: DBUG:GLOBAL:rtpc_doreply:192: sending reply "E60\n" Kamailio version:5.4 and rtpproxy version : 2.2.alpha.029f88a4 I am not understand what is being wrong here. Where is fault.? This issue taken my lot of time. I am very much worried. any one please tell what is solution? how can I overcome this problem. Regards Amit

1 0

Kamailio vulnerable to header smuggling possible due to bypass of remove_hf
by Yufei Tao 02 Sep '20

02 Sep '20

Hi, The security tests were done to find theoretically possible flaws and help make Kamailio "bullet proof". Well it's already a lot more robust than most others. I think Daniel and Henning have made it very clear about the scope of the bug. For me if it is something that's been there for so many years without being noticed, it would be a bit surprising if declared as a high risk problem. Plus isn't this something you should find out if you do your testing properly? If it were to create big troubles for anyone, that means they had never tested their deployment properly in the past 18 years?? That's where I get confused. Of course anyone can fork and build Kamailio themselves if they really need something urgently since it's open source. Cheers, Yufei

1 0

Weird config parsing issues - 5.4
by Noah Mehl 02 Sep '20

02 Sep '20

All, I have built branch 5.4 from source, and I’m working with the default config. However, from time to time I get this error when launching kamailio: root@inbound-kamailio-test-02:~# /usr/local/sbin/kamailio -P /run/kamailio/kamailio.pid -f /usr/local/etc/kamailio/kamailio.cfg -m 128 -M 64 -E 0(1296) CRITICAL: <core> [core/cfg.y:3588]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 410, column 30-34: syntax error 0(1296) CRITICAL: <core> [core/cfg.y:3588]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 410, column 30-34: Invalid arguments 0(1296) CRITICAL: <core> [core/cfg.y:3591]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 410, column 35: ERROR: bad config file (3 errors) 0(1296) ERROR: <core> [core/ppcfg.c:234]: pp_ifdef_level_error(): different number of preprocessor directives: 1 more #!if[n]def as #!endif However, if I run a “make install” from the source directory, this error goes away. Has anyone run into this issue before? Configs attached. Thanks! ~Noah

3 5

Fwd: Keep evapi bind address alive on kamailio restart
by Mehrdad EsmaeilPour 02 Sep '20

02 Sep '20

Hi all, I have a question about evapi module. Is it possible to keep the bind address of evapi alive on kamailio restarts? How can I achieve high availability for evpai's bind address? Many thnaks -- *Kind Regards, Mehrdad Esmaeilpour <https://www.linkedin.com/in/mehrdadep/>*

3 2

worker processes
by Ali Taher 01 Sep '20

01 Sep '20

Hi, I just want to ask regarding the children number and listening interfaces set in Kamailio configuration. I am given to understand that the total number of worker processes there will be (children_setting * listeners) , so assuming I have set children=12 and listen=udp:172.16.1.12:5060 (which is the interface IP where the SIP packets as received), in this case there will be 12 worker processes and 12 database connections to postgresql (I'm using SQLOPS). My question is if I set listen=udp:0.0.0.0:5060 , knowing that I have 3 interfaces, in this case the number of worker processes will be 36. Is this good configuration ? will the other 24 processes be used or they are reserved to the 2 other interfaces which are not receiving any SIP packets and then will be useless ? Regards, Ali Taher

3 9

kamailio coring ver 5.1.2
by Andrew Chen 01 Sep '20

01 Sep '20

Hey guys, Our monitoring system caught kamailio coring. I did a dmesg and gave me this: [Tue Sep 1 09:12:34 2020] kamailio[1375]: segfault at 1 ip 0000000000000001 sp 00007fff4203f098 error 14 in kamailio[563969b5a000+4a4000] [Tue Sep 1 09:12:34 2020] Code: Bad RIP value. [Tue Sep 1 09:12:34 2020] kamailio[1363]: segfault at 0 ip 0000000000000000 sp 00007fff4203f368 error 14 in kamailio[563969b5a000+4a4000] [Tue Sep 1 09:12:34 2020] Code: Bad RIP value. [Tue Sep 1 09:12:34 2020] kamailio[1377]: segfault at 1 ip 0000000000000001 sp 00007fff4203e308 error 14 in kamailio[563969b5a000+4a4000] [Tue Sep 1 09:12:34 2020] Code: Bad RIP value. After doing gdb bt, didn't tell me much except it can't access some parts of the memory: warning: Unexpected size of section `.reg-xstate/1624' in core file. Cannot access memory at address 0x7fb23433e148 Cannot access memory at address 0x7fb23433e140 Failed to read a valid object file image from memory. Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.'. Program terminated with signal SIGSEGV, Segmentation fault. warning: Unexpected size of section `.reg-xstate/1624' in core file. #0 0x0000000000000001 in ?? () (gdb) bt full #0 0x0000000000000001 in ?? () No symbol table info available. Backtrace stopped: Cannot access memory at address 0x7fff4203e1f8 (gdb) quit Any idea on how to troubleshoot this further? Thanks. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 2

Releasing Kamailio v5.4.1
by Daniel-Constantin Mierla 01 Sep '20

01 Sep '20

Hello, I am considering to release Kamailio v5.4.1 as the first patch release in the 5.4 series, likely to be done next week, on the 1st of September, 2020. There are a few issues reported on the tracker, hopefully they will be tackled till then. If you are aware of something else not working as expected, report them to issue tracker asap to give a better chance to be fixed till the release day. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

2 5

Kamailio v5.4.1 Released
by Daniel-Constantin Mierla 01 Sep '20

01 Sep '20

Hello, Kamailio SIP Server v5.4.1 stable release is out. This is a maintenance release of the latest stable branch, 5.4, that includes fixes since the release of v5.4.1. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.4.x. Deployments running previous v5.4.x versions are strongly recommended to be upgraded to v5.4.1. For more details about version 5.4.1 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/09/kamailio-v5-4-1-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Kamailio World Online Event - Schedule and access details
by Daniel-Constantin Mierla 31 Aug '20

31 Aug '20

Hello, the schedule for Kamailio World Updates has been published - it is an online event happening this week, on Wed-Thu, Sep 2-3, 2020, from 13:00UTC to 17:00UTC, more details are available at: * https://www.kamailioworld.com/k08-online/ The agenda includes presentations, live demos and open discussion sessions, as well as an online party to celebrate 19 years of development for Kamailio project. Being shorter than the usual Kamailio World, the event has an obvious focus on Kamailio only, with sessions about what is new in Kamailio ecosystem. The sessions are done live, in a self-hosted Jitsi Meet video conferencing room (hosting sponsored by lod.com), with live streaming to YouTube KamailioWorld channel: * https://www.youtube.com/c/KamailioWorld/ If streaming goes without problems, the recordings of presentations and live demos will be made available on YouTube channel after the event. If you want to join the video conferencing room, be sure you are muted and with video turned off, in order not to disturb the sessions with unexpected noise. Un-mute/turn video on only when you want to participate in a discussion. Asking questions and text discussions can be done on our self hosted Matrix server, in the room "kamailio" (full room id: #kamailio:matrix.kamailio.dev): * https://riot.kamailio.dev/#/room/#kamailio:matrix.kamailio.dev The server is federated, so if you have a Matrix account on another server, you can join easily. During the event, we will allow guests to join the room, so it is not really required to have a Matrix account. Should anyone have issues joining the chat room during the event, reply on sr-users and we will try to assist you. As an alternative for posting questions, I will keep an eye on YouTube comments during the live streaming and try to pass them to the speaker, if something pops up there. The last session on the 2nd day is the online celebration party for 19 years of development for Kamailio project. Grab a drink, prepare your birthday speech and say it live, tell an interesting story about your experiences in RTC or just enjoy open discussions with the other participants! It is the first time we are experimenting with a live online event, testing looked good so far and we hope everything will run smooth during the sessions! Anyhow, I am confident everyone will enjoy the event and be able to (re-)connect with people from Kamailio community, no matter where they are located around the world! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

new compiler warning in 5.4 stable release
by Juha Heinanen 30 Aug '20

30 Aug '20

tls_locking.c: In function 'tls_init_locks': tls_locking.c:191:1: warning: label 'error' defined but not used [-Wunused-label] error: ^~~~~

2 2

RTP PROXY (rtpproxy_stream2uas getting error)
by Amit Pal 27 Aug '20

27 Aug '20

Dear Team , I am using Kamailio and RTPProxy for media. I am trying to play caller/callee side media by rtpproxy_stream2uas/ rtpproxy_stream2uac. But I am getting error secondary rtpproxy[27007]: DBUG:handle_command: received command "29329_6 P-1 2c60634699018317MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI. /home/coralswitch/queuesounds/moh2.wav session b2f563cf067c489ca7102e96b622f2f1;1 9f10411f;1" rtpproxy[27007]: ERR:handle_play: can't create player rtpproxy[27007]: DBUG:doreply: sending reply "29329_6 E6#012" rtpproxy[27007]: INFO:process_rtp: session timeout rtpproxy[27007]: INFO:remove_session: RTP stats: 1083 in from callee, 936 in from caller, 2019 relayed, 0 dropped rtpproxy[27007]: INFO:remove_session: RTCP stats: 30 in from callee, 41 in from caller, 71 relayed, 0 dropped rtpproxy[27007]: INFO:remove_session: session on ports 36556/60108 is cleaned up why this error are coming . Even some time with change in script following error "handle_play: can't create player" not showing but even not media playing. Log are attached with mail. Please do the need full. Regards Amit Pal

2 5

Kamailio routing incoming phone call ranges to different dispatch groups
by Patrik Nilsson 25 Aug '20

25 Aug '20

Hi, Although what I'm trying to achieve seems like an easy task, I have been tearing my hair for the past two days getting Kamailio to dispatch incoming phone calls to two different groups of Asterisk servers in my dispatcher list. I have 30 dedicated phone numbers from Telco: 020131600 - 020131629. I want 00-19 to be routed to Group 1, and 20-29 to be routed to Group 2, with a Round-robin algorithm. I'm aware that I can set up these incoming phone numbers in each Asterisk server's extension dial plans; nevertheless, I want Kamailio to do the initial incoming routing to a specific group (as Group 2, for instance, will never receive calls from for instance 02013161). Example dispatcher.list: #Company A (Group 1) 1 sip:10.50.0.1 0 0 maxload=20 1 sip:10.50.0.2 0 0 maxload=20 1 sip:10.50.0.3 0 0 maxload=20 #Company B (Group 2) 2 sip: 10.60.0.1 0 0 maxload=20 2 sip: 10.60.0.2 0 0 maxload=20 2 sip: 10.60.0.3 0 0 maxload=20 I assume this is done in the route[DISPATCH] of my kamailio.cfg, using ds_select_dst, but my previous attempts to get this routing to work just breaks the config file. I appreciate any help that I can get! Best regards, Patrik

3 2

Is the "openssl bug #1491" relevant to openssl v.1.1.1?
by Leonid Fainshtein 25 Aug '20

25 Aug '20

Hi, Is it still necessary to care about the minimal free memory amount in the TLS module configuration (e.g., the low_mem_threshold1/2 parameters) in Ubuntu 18.04 with openssl v.1.1.1? I tried to find some more information about the "openssl bug #1491" but the 1491 issue in the current Openssl bug tracking system seems to be not related to the memory problem: https://github.com/openssl/openssl/issues/1491 Best regards, Leonid Fainshtein Xorcom Ltd

2 1

Setup Push notifications from SIP-PBX to Linphone Android -
by Johnny Ritzer 25 Aug '20

25 Aug '20

I have a setup where we wanted to just build a android app linphone to SIP-PBX. But come to find out upon further research due to potential battery drain/background app usage. The device isn’t registered all the time and need some sort of push notification to google firebase back to android app to re-setup the register / INVITE call. Is there a module/config of this kind of setup. (now could I use cloud kamailio or would this have to be onsite local to the PBX).

4 3

Call forking and pua_dialog problem
by Denys Pozniak 24 Aug '20

24 Aug '20

Hello! We have a next call-flow topology: User A -> FreeSWITCH -> Kamailio Fork -> Kamailio Edge => User(s) B, C, D, ... - FreeSWITCH adds X- header with a list of users to fork call (B, C, D legs). - Kamailio Fork generates branches via append_branch function towards Kamailio Edge. - Kamailio Edge keeps customer registration and sends BLF statuses (pua_dialoginfo). The problem (or normal behavior by RFC) is that the dialog module on Kamailio Edge does not split these incoming legs into separated dialogs (incoming SIP Call-ID and from tag are the same). And as per my checking *kamcmd dlg.list_ctx* shows only the structure from the first leg and finally, it leads to a problem with BLF (as pua_diloginfo loses some SIP PUBLISHes). Any advice on how to fix this is appreciated! -- BR, Denys Pozniak

1 1

KAMAILIO/RTPENGINE - Log
by Amit Pal 22 Aug '20

22 Aug '20

Dear Team, I am using kamailio with rtpengine for media. When I make call between two sipUA then media flow is manage by rtpengine offer/answer. The issue is when hold the call then following some error are coming . Pls find the full log in attached file. rtpengine[14099]: DEBUG: [43402b3cc9425239MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.]: av_log: parser not found for codec pcm_s16le, packets or times may be invalid. rtpengine[14099]: DEBUG: [43402b3cc9425239MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.]: av_log: max_analyze_duration 5000000 reached at 5120000 microseconds st:0 rtpengine[14099]: DEBUG: [43402b3cc9425239MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.]: av_log: stream 0: start_time: -1152921504606847.000 duration: 27.034 rtpengine[14099]: DEBUG: [43402b3cc9425239MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.]: av_log: format: start_time: -9223372036854.775 duration: 27.034 bitrate=128 kb/s rtpengine[14099]: DEBUG: [43402b3cc9425239MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.]: av_log: After avformat_find_stream_info() pos: 131130 bytes read:196666 seeks:1 frames:22 rtpengine[14099]: ERR: [43402b3cc9425239MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.]: No supported output codec found in SDP rtpengine[14099]: INFO: [43402b3cc9425239MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.]: Replying to 'play media' from 127.0.0.1:48757 (elapsed time 0.121111 sec) rtpengine[14099]: DEBUG: [43402b3cc9425239MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.]: Response dump for 'play media' to 127.0.0.1:48757: { "result": "ok" } I am unable to understand this issue. Is this related to rtpenging, which is not supported given codec or file codec, which is used to play after hold call or anything else in script issue. Can you please guide me where is the issue. Regards Amit Pal

2 1

how to check path vector after lookup("location")
by Sergey Safarov 21 Aug '20

21 Aug '20

I use this construction to check the presence of path in device registration if(reg_fetch_contacts("location", "$fu", "caller") && $(ulc(caller=>path)[0]) == $null) { handle_ruri_alias(); } Is possible to check path presence after lookup("location") function call? Sergey

2 1

Bug in Postgres SQL Script?
by Noah Mehl 20 Aug '20

20 Aug '20

Hey all, I’m trying to understand the following lines in the kamdbctl.pgsql script: https://github.com/kamailio/kamailio/blob/bf38c7b04171e6f410ff885f10abe0f81… <https://github.com/kamailio/kamailio/blob/bf38c7b04171e6f410ff885f10abe0f81…> https://github.com/kamailio/kamailio/blob/bf38c7b04171e6f410ff885f10abe0f81… <https://github.com/kamailio/kamailio/blob/bf38c7b04171e6f410ff885f10abe0f81…> https://github.com/kamailio/kamailio/blob/bf38c7b04171e6f410ff885f10abe0f81… <https://github.com/kamailio/kamailio/blob/bf38c7b04171e6f410ff885f10abe0f81…> Basically, there are no tables created for Postgres that end in _id_seq. Am I missing something here? Thanks! ~Noah

2 2

kamailio+rtpengine for media not playing media file while hold the call
by Amit Pal 20 Aug '20

20 Aug '20

Dear Team, I am using kamailio with rtpengine for media. when call getting hold no media sound play (configure hold.mp3). >From rtpengine gets following error. rtpengine[624]: ERR: No suitable SDP section for media playback rtpengine[624]: WARNING: Protocol error in packet from 127.0.0.1:46333: Failed to start media playback from file [d8:supportsl10:load limite4:file38:/home/coralswitch/queuesounds/hold.mp37:call-id60:89162207232 fae50MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.13:received-froml3:IP414:19 2.168.20.254e8:from-tag32:9f5288f14b704b28a1c2dfc2112250eb6:to-tag8:1e05e757 7:command10:play mediae] rtpengine[624]: INFO: Received command 'offer' from 127.0.0.1:46333 rtpengine[624]: INFO: Replying to 'offer' from 127.0.0.1:46333 (elapsed time 0.000106 sec) /usr/local/sbin/kamailio[9094]: ERROR:rtpengine [rtpengine.c:2619]: rtpp_function_call(): proxy replied with error: Failed to start media playback from file Attached file show complete log of one call with hold /unhold actions. Please advise what I miss to configure here. Regards Amit Pal

2 2

Module IMS_QOS does not support Re-Auth Request (RAR)
by Hamid Hashmi 20 Aug '20

20 Aug '20

Kamailio User Does module ims_qos support Re-Auth Request? [cid:282322de-506e-4b6e-a731-24841b517573] I observed that the QoS module is not responding to RAR. Kamailio version is version: kamailio 5.2.0-dev6 (x86_64/linux) 48ed62-dirty flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 48ed62 -dirty compiled with gcc 4.8.4 Regards Hamid R. Hashmi

2 1

t_load_contacts in 5.4.0
by Cindy Leung 20 Aug '20

20 Aug '20

Hello all, Just started using v5.4.0 on our system and I noticed a change in behavior when doing the append_branch, t_load_contacts, and t_next_contacts combo. Previously using v5.3.4 and it appears to be fine. Here's the call scenario: Kamailio receives a call to sip:1001@carrierB. Kamailio sees carrierB and appends 2 contacts: gateway1.carrierB.com;q=0.3 and gateway2.carrierB.com;q=0.2. After t_load_branches(0), I expect to see $rd to be changed to gateway1.carrierB.com, but it's not. This is the piece of config I'm trying to debug: xlog ("=== branch 0: $(branch(uri)[0]), $(branch(q)[0])\n"); xlog ("=== branch 1: $(branch(uri)[1]), $(branch(q)[1])\n"); t_load_contacts(0); while (t_next_contacts()) { xlog ("=== rd = $rd\n"); } This is the log I get. It appears to use the backup contact first. ERROR: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: === branch 0: sip:1001@gateway2.carrierB.com;transport=udp, 200 ERROR: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: === branch 1: sip:1001@gateway1.carrierB.com;transport=tcp, 300 DEBUG: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: tm [t_serial.c:522]: t_load_contacts(): load_contact mode selected: 0 DEBUG: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: tm [t_serial.c:340]: ki_t_load_contacts_mode(): nr_branches is 2 DEBUG: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: <core> [core/xavp.c:539]: xavp_destroy_list(): destroying xavp list 0x7f983cb66608 DEBUG: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: tm [t_serial.c:890]: ki_t_next_contacts(): Appending branch uri-'sip:1001@gateway1.carrierB.com;transport=tcp' dst-'' path-'' inst-'' ruid-'' location_ua-'' DEBUG: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: <core> [core/xavp.c:539]: xavp_destroy_list(): destroying xavp list 0x7f983cb66350 ERROR: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: === rd = carrierB DEBUG: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: <core> [core/xavp.c:539]: xavp_destroy_list(): destroying xavp list 0x7f983cb66078 ERROR: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: === rd = gateway2.carrierB.com DEBUG: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: tm [t_serial.c:627]: ki_t_next_contacts(): no contacts in contacts_avp - we are done! t_load_contacts(1) works a little better. But we don't need the probability feature. ERROR: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: === branch 0: sip:1001@gateway2.carrierB.com;transport=udp, 200 ERROR: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: === branch 1: sip:1001@gateway1.carrierB.com;transport=tcp, 300 DEBUG: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: tm [t_serial.c:522]: t_load_contacts(): load_contact mode selected: 1 DEBUG: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: tm [t_serial.c:340]: ki_t_load_contacts_mode(): nr_branches is 2 DEBUG: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: tm [t_serial.c:280]: t_load_contacts_proportional(): proportionally selected contact with uri: sip:1001@gateway1.carrierB.com;transport=tcp (q: 300, random: 287, q_index: 500, q_total: 500) DEBUG: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: tm [t_serial.c:280]: t_load_contacts_proportional(): proportionally selected contact with uri: sip:1001@gateway2.carrierB.com;transport=udp (q: 200, random: 157, q_index: 200, q_total: 200) DEBUG: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: tm [t_serial.c:303]: t_load_contacts_proportional(): proportionally added backup contact with uri: sip:1001@carrierB SIP/2.0#015#012Via: SIP/2.0/UDP 172.18.1.21:5060;branch=z9hG4bK-21-1-0 blah blah blah (q: -1) DEBUG: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: <core> [core/xavp.c:539]: xavp_destroy_list(): destroying xavp list 0x7f6e457ec078 ERROR: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: === rd = gateway1.carrierB.com DEBUG: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: <core> [core/xavp.c:539]: xavp_destroy_list(): destroying xavp list 0x7f6e457ec350 ERROR: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: === rd = gateway2.carrierB.com DEBUG: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: <core> [core/xavp.c:539]: xavp_destroy_list(): destroying xavp list 0x7f6e457ec608 ERROR: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: === rd = carrierB DEBUG: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: tm [t_serial.c:627]: ki_t_next_contacts(): no contacts in contacts_avp - we are done! As a comparison, this is what we've been getting in 5.3.4 ERROR: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: === branch 0: sip:1001@gateway2.carrierB.com;transport=udp, 200 ERROR: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: === branch 1: sip:1001@gateway1.carrierB.com;transport=tcp, 300 DEBUG: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: tm [t_serial.c:191]: ki_t_load_contacts(): nr_branches is 2 DEBUG: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: <core> [core/xavp.c:529]: xavp_destroy_list(): destroying xavp list 0x7fe13146a680 ERROR: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: === rd = gateway1.carrierB.com DEBUG: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: <core> [core/xavp.c:529]: xavp_destroy_list(): destroying xavp list 0x7fe13146a3a8 ERROR: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: === rd = gateway2.carrierB.com DEBUG: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: <core> [core/xavp.c:529]: xavp_destroy_list(): destroying xavp list 0x7fe13146a0d0 ERROR: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: === rd = carrierB DEBUG: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: tm [t_serial.c:460]: ki_t_next_contacts(): no contacts in contacts_avp - we are done! Is there anything else that's been changed in branch building that we should pay attention to? Thanks. Cindy

2 4

Domain Name changes
by Pavithra Mohanraja 19 Aug '20

19 Aug '20

Hi, Is there any way to change the domain name at run-time Kindly help. Thanks, Pavithra

2 4

dlg_refer of callee leg issue
by Amit Pal 19 Aug '20

19 Aug '20

Dear kamailio Team, I using dlg_refer for refer callee leg on specific condition to Freeswitch conference. But I got decline message. in reverse caller leg refer to freeswitch conference. can pls tell how to achieve it. Regards Amit Pal

2 4

Force registrations on Kamailio save()
by Duarte Rocha 19 Aug '20

19 Aug '20

Greetings, I'm currently creating a Kamailio with registrations and i'm unsure how to use the save() function. I want to limit the number of contacts for the AOR to 2. I also want to force the registration of new contacts even if the maximum number is reached, removing the older ones. I saw that on opensips i could use the flags "fc2" for this but they aren't available on Kamailio's save(). I'm using the max_contacts parameter but in the docs it states the once the limit is reached, it will reject new Registers. How can i implement the force registration behaviour here? Best Regards, Duarte Rocha

2 1

Decrypting SRTP with Kamailio + rtpengine
by Ildar Mukhametzhanov 18 Aug '20

18 Aug '20

Hello! Decrypting SRTP with Kamailio + rtpengine. Hello! I have a task to decrypt SRTP for a legacy switch which doesn’t support it. Simply adding rtpengine_manage("RTP/AVP") didn’t help. SDP converted from SRTP to RTP while proxying INVITE to legacy switch, but 180 Ringing wasn’t converted back to SRTP. Little more detailed config snippet: route[RELAY] { ... if (is_method("INVITE")) { if ($rd=~$var(switch_ip_mask)) { rtpengine_manage("RTP/AVP"); } else { rtpengine_manage(); } ... } Could someone please advise typical scenarios to achieve it, I suppose the task is pretty standard. Thank you in advance!

4 3

Kamailio World Updates Online - Celebration Party - Sep 2-3, 2020
by Daniel-Constantin Mierla 18 Aug '20

18 Aug '20

Hello, we are considering to organize a small online event to compensate a bit the cancellation of Kamailio World 2020, but mainly to discuss about what is new around Kamailio project and RTC space. The event was discussed among a couple developers several weeks ago, when it was clear that the pandemic will last longer, preventing any on-site event, and the intention to do it was also announced during the recording of last ClueCon Weekly about Kamailio project at the beginning of July (https://www.kamailio.org/w/2020/07/cluecon-weekly-kamailio-updates-july-202…). The first commit of Kamailio code was done on the 3rd of September 2001, thus this is like the project's birthday, so a good opportunity to celebrate that as well. The event is planned to be 2 sessions of about 4 hours on each day, with a few presentations per day, but try focus on open discussions about. The goal is to keep in very informal, without any significant burden for organizers or participants, not to add to the daily stress with work and family in the pandemic time. It should be more like meetup style, with no registration for participants, sessions may not be recorded and can be done without slides. Likely we will use a (self-hosted) video conferencing system (Jitsi) for presentations and the IRC or Matrix channels for chatting. The time frames we look at are 15:00-19:00 Berlin time zone (13:00-17:00 UTC) on the 2nd and the 3rd of September 2020. The purpose of this email for now is to see the interest of the community for such event, reply if you think it is useful to organize it and you plan to participate. If there is interest, then I will follow up with more details about speakers and the structure of the event. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

9 10

Do I need RTPEngine at every location onsite for webRTC
by Johnny Ritzer 18 Aug '20

18 Aug '20

We have 100 customer locations and I was hoping to no deploy a kamabox at each location if traffic is mostly just local. But we are building a WEBRTC. and hoping our architecture can use kamailio in the cloud. Each location has a PBX onsite. We are deploying webRTC so need that websocket setup jssip/sip.js. SIP SIGNALING (relay) () UA(webphone) --> Cloud Kamailio --> PBX 10.0.1.100 --> cloud.kama.com --> 10.0.1.1 But then translate uac and get RTP to pass locally without a device onsite at each location. RTP Flow (not going through kamailio) UA(webphone) --> PBX 10.0.1.100 --> 10.0.1.1

2 1

TCP/TLS CRITICAL error. send_fd(): sendmsg failed sending : Too many references: cannot splice
by Dmytro M 18 Aug '20

18 Aug '20

While executing this piece of config script, after a few days of run time, Kamailio starts experiences some issue that leads to instability and unresponsiveness of the proxy via TCP with TLS. UDP workers seems to be working fine... TCP clients can't connect. route[RELAY] { ... if (!t_relay()) { sl_reply_error(); } exit; } Please advise how we can troubleshoot this issue. Are there any immediate (or obvious) TCP/TLS/FD Kamailio/Linux parameters that should be increased to allocate more resources for the Proxy? Or may be any other parameters that may lead to a quicker resources release. Please see the errors bellow... Environment: Ubuntu 16 x86_64, Kamailio 5.3.5. Thanks, Dmytro Aug 15 04:48:19 CRITICAL: <core> [core/pass_fd.c:191]: send_fd(): sendmsg failed sending 164 on 16: Too many references: cannot splice (109) Aug 15 04:48:19 ERROR: <core> [core/tcp_main.c:3908]: handle_ser_child(): CONN_GET_FD: send_fd failed Aug 15 04:48:19 ERROR: tm [../../core/forward.h:292]: msg_send_buffer(): tcp_send failed Aug 15 04:48:19 WARNING: tm [t_fwd.c:1573]: t_send_branch(): sending request on branch 0 failed Aug 15 04:48:19 ERROR: sl [sl_funcs.c:392]: sl_reply_error(): stateless error reply used: Unfortunately error on sending to next hop occurred (477/SL) .... Aug 15 05:52:04 ERROR: <core> [core/tcp_main.c:4225]: send2child(): send_fd failed for 0x7f92e91c31e0 (flags 0x4018), fd 74 Aug 15 05:52:04 CRITICAL: <core> [core/pass_fd.c:191]: send_fd(): sendmsg failed sending 75 on 43: Too many references: cannot splice (109) Aug 15 05:52:04 ERROR: <core> [core/tcp_main.c:4225]: send2child(): send_fd failed for 0x7f92e9d727b0 (flags 0x4018), fd 75 Aug 15 05:52:04 CRITICAL: <core> [core/pass_fd.c:191]: send_fd(): sendmsg failed sending 75 on 73: Too many references: cannot splice (109) Aug 15 05:52:04 ERROR: <core> [core/tcp_main.c:4225]: send2child(): send_fd failed for 0x7f92e9d727b0 (flags 0x4018), fd 75 Aug 15 05:52:04 CRITICAL: <core> [core/pass_fd.c:191]: send_fd(): sendmsg failed sending 75 on 45: Too many references: cannot splice (109) Aug 15 05:52:04 ERROR: <core> [core/tcp_main.c:4225]: send2child(): send_fd failed for 0x7f92e9d727b0 (flags 0x4018), fd 75 Aug 15 05:52:04 CRITICAL: <core> [core/pass_fd.c:191]: send_fd(): sendmsg failed sending 74 on 43: Too many references: cannot splice (109) Aug 15 05:52:04 ERROR: <core> [core/tcp_main.c:4225]: send2child(): send_fd failed for 0x7f92e91c31e0 (flags 0x4018), fd 74 Aug 15 05:52:04 CRITICAL: <core> [core/pass_fd.c:191]: send_fd(): sendmsg failed sending 76 on 43: Too many references: cannot splice (109) Aug 15 05:52:04 ERROR: <core> [core/tcp_main.c:4225]: send2child(): send_fd failed for 0x7f92e7856fa8 (flags 0x4018), fd 76 Aug 15 05:52:04 CRITICAL: <core> [core/pass_fd.c:191]: send_fd(): sendmsg failed sending 74 on 47: Too many references: cannot splice (109) Aug 15 05:52:04 ERROR: <core> [core/tcp_main.c:4225]: send2child(): send_fd failed for 0x7f92e7856fa8 (flags 0x4018), fd 74 .....

2 2

What is a good strategy to build Kamailio <--> pbx
by Johnny Ritzer 16 Aug '20

16 Aug '20

What are your thoughts on architecture build of the following scenarios. The PBX doesn’t support websockets and we want to use sip.js / jsSip. (use Kama as wss:// SIP proxy) PBX has direct PRI/sip Trunks/ DID control/ Voicemail. PBX hosts SIP extensions. UA --> Kamailio --> PBX --------SCENARIO A.) ------- Host SIP[xxxx] extensions on Kama box but same SIP extensions are also on PBX [xxxx] 1. Build some sort of control to register handle kama to pbx …. (but reg webclients) Kama [1100] SIP-PBX [1100] Does kama act as the UA 1100 / registered to PBX What about UA to kamailio (do we do 1100a so UA-kama is 1100a) but kama is maintain REG/status/ but passing that IF client is online ? MWI (voicemail is hosted on PBX) so how does that move through can it. ------- SCENARIO B.) ------ Proxy all/ everything . use route[REGFW] Forward all REG to PBX.. Issues ---- MWI (SIP notify ?) SIP Incoming CALL incoming calls to work. (if UA holds REG to outbound proxy on PBX or ) DTMF (sipINFO) doesn’t work (I see it sending) I noticed I crashed the PBX because all the SIP scanners started pounding my Kamailio box after 2 days when I added route[REGFW] it starting eating up SIP trunk channels on PBX cause started forwarding everything. Like it starting 4241(a)xxxxx.com 4242(a)xxxxx.com . Or might need to look at fail2ban pike module to oget this but still had issues I just don’t know where to look or code to fix on SCENARIO B Snippets on my testing . route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.PBX.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.PBX.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + "kamaproxy"; # $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } #!endif --------------- So this led me to extra security should I host SIP extensions on kamailio as an extra security but then what about MWI/inbound then) or do I have 1100 on kama which mitel is 1100 but I pass all info from inbound.

2 1

Kamailio Query
by Amit Pal 15 Aug '20

15 Aug '20

Dear Denial/Team, I am glad to work with kamailio Team, I am using kamailio freeswitch integration. I want to work like this. My all subscriber register at kamailio. My application server is freeswitch. MY first question: When subscriber kamuser1 and kamuser2 are talking then I want to transfer these two party in freeswitch conference. I am unable to generate re-invite these call . here I was also try with rewritehost(like) but not success. how can we achieve this. My second question : Even MOH not playing when any one party hold the call. I already setup KAMUser1Kamailio(RTPProxy)KAMUser2. I have completely setup the config of rtpproxy and mohqueues. log shows all signal handle by kamailio and RTPProxy but not getting moh. I am attaching my config file for reference. can we do this by mediaproxy with freeswitch ,all thing are configured as document but no success. My third question: Kamailio relay dtmf on sipinfo. How can we get dtmf in between the call and taking actions on it. I am highly thankful for your great support . Thanks Regards

1 0

Kamailio Team Attentions
by Amit Pal 15 Aug '20

15 Aug '20

Dear Team, “I am unable to reach you, my previous mail get bounce. I received bounce mail from sr-users. Below is my previous mail contains.” Last Sent Mail Contents I am glad to work with kamailio Team, I am using kamailio freeswitch integration. I want to work like this. My all subscriber register at kamailio. My application server is freeswitch. MY first question: When subscriber kamuser1 and kamuser2 are talking then I want to transfer these two party in freeswitch conference. I am unable to generate re-invite these call . here I was also try with rewritehost(like) but not success. how can we achieve this. My second question : Even MOH not playing when any one party hold the call. I already setup KAMUser1Kamailio(RTPProxy)KAMUser2. I have completely setup the config of rtpproxy and mohqueues. log shows all signal handle by kamailio and RTPProxy but not getting moh. I am attaching my config file for reference. can we do this by mediaproxy with freeswitch ,all thing are configured as document but no success. My third question: Kamailio relay dtmf on sipinfo. How can we get dtmf in between the call and taking actions on it. I am highly thankful for your great support . Thanks Regards Neol

1 0

Required Kamailio clarification!!!
by Amit Pal 15 Aug '20

15 Aug '20

Dear Team, I am glad to work with kamailio Team, I am using kamailio freeswitch integration. I want to work like this. My all subscriber register at kamailio. My application server is freeswitch. MY first question: When subscriber kamuser1 and kamuser2 are talking then I want to transfer these two party in freeswitch conference. I am unable to generate re-invite these call . here I was also try with rewritehost(like) but not success. how can we achieve this. My second question : Even MOH not playing when any one party hold the call. I already setup KAMUser1==>Kamailio(RTPProxy)==>KAMUser2. I have completely setup the config of rtpproxy and mohqueues. log shows all signal handle by kamailio and RTPProxy but not getting moh. I am attaching my config file for reference. can we do this by mediaproxy with freeswitch ,all thing are configured as document but no success. My third question: Kamailio relay dtmf on sipinfo. How can we get dtmf in between the call and taking actions on it. I am highly thankful for your great support . Thanks Regards Neol

1 0

Can the callid be modified/mapped when branching a call?
by Magnus Kelly 14 Aug '20

14 Aug '20

Good evening all, Can I ask what might be a basic question, but I would like to confirm if the CallID field can be modified/mapped when branching a call ? Thanks Magnus

2 2

How to check TLS versions available
by David Cunningham 14 Aug '20

14 Aug '20

Hello, Does anyone know of a method to check what TLS versions are available from Kamailio for clients to use? For example, is TLS 1.0 available, TLS 1.1, etc. Thanks in advance, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

4 5

Segfault starting kamailio, function "registered"
by Jon Bonilla (Manwe) 14 Aug '20

14 Aug '20

Hi when I start kamailio in my dev system it fails to start and segfaults. Version: 5.4.0 CFG Line: if (registered("kam_location",$tu)) { Log: Aug 14 10:20:29 pekedev2 kamailio: CRITICAL: <core> [core/cfg.y:3588]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 382, column 34-36: function registered: parameter 2 is not constant#012 Aug 14 10:20:29 pekedev2 kernel: [1501045.504630] kamailio[21276]: segfault at 6e6f6974 ip 00005566607d8887 sp 00007ffd6761b240 error 4 in kamailio[55666079d000+431000] Aug 14 10:20:29 pekedev2 kernel: [1501045.504655] Code: 89 bd 68 ff ff ff 64 48 8b 04 25 28 00 00 00 48 89 45 c8 31 c0 48 83 bd 68 ff ff ff 00 0f 84 f5 09 00 00 48 8b 85 68 ff ff ff <8b> 00 83 f8 01 0f 85 65 09 00 00 48 8b 85 68 ff ff ff 8b 40 0c 85 The coredump is not clear to me. I extracted using coredumpctl (fuck you systemd) and running gdb. Is there a new method now that systemd has taken control of my systems? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 1

conditional on secfilter seems does not work or how to do it
by PICCORO McKAY Lenz 14 Aug '20

14 Aug '20

I implemented secfilter in a simple way, in first step routing put that conditional for black list check that already works: ``` secf_check_ip(); if ($? == -2) { xlog("L_ALERT", "$rm from $si is blacklisted"); drop(); } ``` BUT NOW i want to change to whitelick checks, so i reviewed the docs and "2" is resulting for whitelist, so then i said "if not whitelisted so block" using "!= 2" as on https://www.kamailio.org/wiki/cookbooks/5.3.x/core#operators BUT SEEMS DOES NOT WORK: that is the code: ``` secf_check_ip(); if ($? != 2) { xlog("L_ALERT", "$rm from $si is not in whitelist, block"); drop(); } ``` when i tested all the calls passed not matter if are in the table or not as whitelist! I want able to call only if are present and whitelisted the ip address.. so i also tested with: ``` secf_check_ip(); if ($? == 0) { xlog("L_ALERT", "$rm from $si is not present, so block"); drop(); } ``` to check if the ip address are presented in the table but call also passed and must not cos i not put never a entry in the table! Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

3 3

Kamailio drop call with Teams
by sip user 13 Aug '20

13 Aug '20

Hi, I'm connecting Teams with kamailio server. From Kamailio to teams I have no problems, but from teams to Kamailio yes. Drop the call.. With ngrep I see that: INVITE: [image: imagen.png] 200 OK: [image: imagen.png] I never received ACK.. In my configuration: Kamailio.cfg: #!KAMAILIO #!define WITH_TLS event_route[tm:local-request] { if(is_method("OPTIONS") && $ru =~ "pstnhub.microsoft.com") { append_hf("Contact: <sip:FQND:5061;transport=tls>\r\n"); } xlog("L_INFO", "Sent out tm request: $mb\n"); } request_route{ remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { xlog("L_INFO","$fU is trying to call to $rU con valores $tu\n"); $ru = "sip:" + $rU + "@ip asterisk:port"; route(RELAY); } } What I'm doing wrong? I don't understand why not received ACK.. Could anyone help me? Thanks

1 0

Kamailio lookup via Rest Interface
by krishdinesh＠yahoo.com 13 Aug '20

13 Aug '20

Hi, I am trying to do the following and wanted to seek help on how this can be done in Kamailio and if it feasible to do this. If not, whether there are other ways to do this. I would like to do the following .. If there is an incoming call, kamailio should be able to send the incoming calling number to an external application via Rest Interface, the external application will return a response on what to do with this call, it can be simple as sending a call to the called extension or a media server like asterisk for further treatment. I have a kamailio version 5.3.5 running on CentOS 7. I have a few end points registered to kamailio and they are able to call each other Any help would be appreciated. Thank you. Dinesh

1 0

Kamailio v5.4.0 RPM Availability
by Asgaroth 13 Aug '20

13 Aug '20

Hi all, I would just like to check in and find out if/when the Kamailio v5.4.0 rpm's will be available on the OpenSUSE build service. Thanks to all involved for all the hard work in this new release! Thanks There are two major products that came out of Berkeley: LSD and UNIX. We don't believe this to be a coincidence. -- Jeremy S. Anderson 00asgaroth00(a)gmail.com (https://link.getmailspring.com/link/146DE51A-E8C2-4126-B8D5-EA45E62CBA11@ge…)

4 14

smsops module - status report?
by Pafel 13 Aug '20

13 Aug '20

Hello, I am trying to generate sms status report using smsops module but without success. I couldn't find any example on the net. Can somebody help me? According to https://en.wikipedia.org/wiki/GSM_03.40 TP-MTI (tpdu(type) in kamailio) should be "1 0" for status report. Delivery is "0 0" and submit "0 1". According to the provided examples in smsops module: tpdu(type) = 4 is delivery tpdu(type) = 1 is submit So what's the number for status? I could't find the mapping in the code. Only this: // Types of the PDU-Message typedef enum _pdu_message_type { DELIVER = 0x00, SUBMIT = 0x01, COMMAND = 0x02, ANY = 0x03, } pdu_message_type_t; So any hint or example is appreciated. Thanks, Pavel Siderov

1 0

Log-level Kamailio IMS
by Pavithra Mohanraja 13 Aug '20

13 Aug '20

Hi, Is there any way to have global log level changes without restart of kamailio. If there kindly let me know. Thanks, Pavithra

3 6

Mysql lastinsertid
by Kjeld Flarup 12 Aug '20

12 Aug '20

Hello I wonder if there is some way to get the lastinsertid from an "INSERT INTO" sql_query -- -------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

4 6

Best Module for CPS per IP - PIKE or Ratelimit modules ?
by Ameed Jamous 12 Aug '20

12 Aug '20

Dear all, I wanted to confirm, which is the best way to implement CPS limits per source IP in kamilio in 2020? PIKE module or Rate Limit module or both combined. Thanks in advance for your support. -- AJ

3 5

smsops module - status report?
by Pavel Siderov 12 Aug '20

12 Aug '20

Hello, I am trying to generate sms status report using smsops module but without success. I couldn't find any example on the net. Can somebody help me? According to https://en.wikipedia.org/wiki/GSM_03.40 TP-MTI (tpdu(type) in kamailio) should be "1 0" for status report. Delivery is "0 0" and submit "0 1". According to the provided examples in smsops module: tpdu(type) = 4 is delivery tpdu(type) = 1 is submit So what's the number for status? I could't find the mapping in the code. Only this: // Types of the PDU-Message typedef enum _pdu_message_type { DELIVER = 0x00, SUBMIT = 0x01, COMMAND = 0x02, ANY = 0x03, } pdu_message_type_t; So any hint or example is appreciated. Thanks, Pavel Siderov

1 0

Kamalio 5.4.0 should support ka_add_destination function of KeepAlive module ?
by sagar malam 12 Aug '20

12 Aug '20

Hello I want to use ka_add_destination function of KA module but i am getting this error : [root@SBC-4-1] kamailio -c -f /usr/local/etc/kamailio/kamailio.cfg loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(48016) ERROR: <core> [core/cfg.y:3451]: yyparse(): cfg. parser: failed to find command ka_add_destination (params 1) 0(48016) CRITICAL: <core> [core/cfg.y:3592]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 1449, column 58: unknown command, missing loadmodule? I know that 5.3.3 version does not support this function but 5.4.0 should as per documentation : https://kamailio.org/docs/modules/5.4.x/modules/keepalive.html#keepalive.ka… Keepalive module is compiled and loaded in cfg. -- Thanks, Sagar

3 5

Listen IP changes in IMS kamailio
by Pavithra Mohanraja 12 Aug '20

12 Aug '20

Hi, Is it possible to make it work without giving listen IP in pcscf.cfg, icscf.cfg, scscf.cfg files since i am configuring it as kubernetes pods and automating all configurations. Due to this, I am unable to start the kamailio service without giving the proper listen IP. Is there any alternative way. Kindly Help Thanks, Pavithra

1 0

Example config for webRTC-SIP kamailio.cfg for sip.js
by Johnny Ritzer 12 Aug '20

12 Aug '20

Anyone know where to grab an example config for getting web socket-sip working. I tried this one but its failing on kamailio restart it might be some very small thing. https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg wss://sip.kamailio-box.com:8080 Client = sip.js (webrtc) SIP server = (mitel) PBX (straight sip)

3 2

Call in Progress Recovery in Redundancy
by amitsharma＠coraltele.com 12 Aug '20

12 Aug '20

Dear Daniel/Team, I had raised one question in "Workshop 3 - Kamailio" at Cluecon 2020(Last Night), i.e. Can Progress Call(Ringing Calls) be recovered in case of redundancy with Kamailio. You were told me that straight way it is not possible but try with hash table once. I had tried following link https://wazo-platform.org/blog/kamailio-ha-dispatcher-and-dmq and able to recover Call in progress within 2-3 nodes. 1. My one question is that either this approach will work in production or not. 2. I have been using Freeswitch for last 6-7 years but "Call in Progress Recovery in Redundancy" is not possible there in "Freeswitch", So I tried Kamailio and got success. My Second question is that can it be possible that Call established on Kamailio and after call set up Kamailio leave that call and handed over it to Freeswitch for further processing(Like Re-homing available in OpenSIPS). This will save years of time that I have invested building features around Freeswitch. Please suggest me the best way possible to achieve this. Thanks & Regards, Amit Sharma (Sr. Team Leader) (An ISO 9001:2008 company) Mobile: <tel:9891612004> tel:9891612004 PH: +91 120 2595870 Ext.: <tel:870> tel:870 Email : <mailto:amitsharma@coraltele.com> amitsharma(a)coraltele.com Web : <blocked::http://www.coraltele.com> www.coraltele.com

3 7

Kamailio configuration ??
by Amit Pal 12 Aug '20

12 Aug '20

Dear Team, I am glad to work with kamailio Team, I am using kamailio freeswitch integration. I want to work like this. My all subscriber register at kamailio. My application server is freeswitch. Now I want ask is this possible at kamailio? "when subscriber establish their call then kamailio handed over its call control to freeswitch" i.e. when kamailio got "ACK" (call patch signal) of two subscriber then call control will be transferred to freeswitch. Here I am Using Kamailio for its excellent feature recovery of progress call (by DMQ and HTable) in Highly Availability(HA) . Freeswitch doesn't have this feature. But freeswitch we are using as application server. I have configure Kamailio like "KMUser1==>FS==> KAMUser2" . but in that case progress call (KMUser1 Get alert of KMUser2) has not recover on HA. So call will be on kamailio till alerting after connect call will be transferred to Freeswitch. I want to know by which module we can achieve this. It will be great help for me if provided sample . Thanks Amit Pal Coral Telecom Ltd

1 0

ACK no transaction found if re-invite
by Techinnovation 12 Aug '20

12 Aug '20

Hello, Please help. There was issue topic #2436. I have answer: There is some problem with handling Record-Route in client side, maybe not supporting it, because the ACK has no Route header. Is there any hotfix for kamailio ?,As I not sure that customer can change ACK on customers side . ======================== I use kamailio 5.2.7 with Asterisk 16.1.1 . Call Flow: Customer(212.25.64.8) -> Kamailio(48.146.161.152) -> Asterisk(48.146.161.153) ->Kamailio(48.146.161.152) -> Provider(89.210.243.1) I have one issue with call when I receive re-invite in dialog from Provider. My Asterisk not receive ACK from kamailio. I see error on log like "ACK no transaction found" With this I have difference with 30 seconds. Session Customer(212.25.64.8) -> Kamailio(48.146.161.152) -> Asterisk(48.146.161.153) - has 2 seconds Asterisk(48.146.161.153) ->Kamailio(48.146.161.152) -> Provider(89.210.243.1) - has 32 seconds I have this WITHINDLG settings: route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; # must be ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard. exit; } } t_relay(); } exit; } } =================================== In kamailio logs I see ACK no transaction found: Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: [core/receive.c:246]: receive_msg(): --- received sip message - reply - call-id: [4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060] - cseq: [12375830 INVITE] Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: [core/parser/msg_parser.c:185]: get_hdr_field(): content_length=270 Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} tm [t_lookup.c:1043]: t_check_msg(): msg (0x7f330aed6050) id=296/1926 global id=295/1926 T start=0xffffffffffffffff Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} tm [t_lookup.c:920]: t_reply_matching(): t_reply_matching: hash 11128 label 0 branch 0 Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} tm [t_lookup.c:1013]: t_reply_matching(): no matching transaction exists Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} tm [t_lookup.c:1016]: t_reply_matching(): failure to match a transaction Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} tm [t_lookup.c:1113]: t_check_msg(): msg (0x7f330aed6050) id=296/1926 global id=296/1926 T end=(nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} tm [t_reply.c:2230]: reply_received(): transaction not found - (branch -1) Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/msg_translator.c:2290]: generate_res_buf_from_sip_res(): old size: 932, new size: 821 Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/msg_translator.c:2308]: generate_res_buf_from_sip_res(): copied size: orig:127, new: 16, rest: 805 msg= Aug 11 00:06:11 c4-proxy-1 kamailio: SIP/2.0 200 OK Aug 11 00:06:11 c4-proxy-1 kamailio: Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199 Aug 11 00:06:11 c4-proxy-1 kamailio: From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03 Aug 11 00:06:11 c4-proxy-1 kamailio: To: sip:393817097132@48.146.161.153;tag=as73e66fe7 Aug 11 00:06:11 c4-proxy-1 kamailio: Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060 Aug 11 00:06:11 c4-proxy-1 kamailio: CSeq: 12375830 INVITE Aug 11 00:06:11 c4-proxy-1 kamailio: Server: Asterisk PBX 16.1.1 Aug 11 00:06:11 c4-proxy-1 kamailio: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Aug 11 00:06:11 c4-proxy-1 kamailio: Supported: replaces, timer Aug 11 00:06:11 c4-proxy-1 kamailio: Contact: sip:393817097132@48.146.161.153:5060 Aug 11 00:06:11 c4-proxy-1 kamailio: Content-Type: application/sdp Aug 11 00:06:11 c4-proxy-1 kamailio: Content-Length: 270 Aug 11 00:06:11 c4-proxy-1 kamailio: v=0 Aug 11 00:06:11 c4-proxy-1 kamailio: o=root 1995384227 1995384228 IN IP4 48.146.161.153 Aug 11 00:06:11 c4-proxy-1 kamailio: s=Asterisk PBX 16.1.1 Aug 11 00:06:11 c4-proxy-1 kamailio: c=IN IP4 48.146.161.153 Aug 11 00:06:11 c4-proxy-1 kamailio: t=0 0 Aug 11 00:06:11 c4-proxy-1 kamailio: m=audio 18584 RTP/AVP 0 8 18 Aug 11 00:06:11 c4-proxy-1 kamailio: a=rtpmap:0 PCMU/8000 Aug 11 00:06:11 c4-proxy-1 kamailio: a=rtpmap:8 PCMA/8000 Aug 11 00:06:11 c4-proxy-1 kamailio: a=rtpmap:18 G729/8000 Aug 11 00:06:11 c4-proxy-1 kamailio: a=fmtp:18 annexb=no Aug 11 00:06:11 c4-proxy-1 kamailio: a=ptime:20 Aug 11 00:06:11 c4-proxy-1 kamailio: a=maxptime:150 Aug 11 00:06:11 c4-proxy-1 kamailio: a=sendrecv Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/forward.c:691]: update_sock_struct_from_via(): using via host Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/forward.c:700]: update_sock_struct_from_via(): check if dns lookup is needed for [89.210.243.1] Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/forward.c:710]: update_sock_struct_from_via(): address is [89.210.243.1] port 5060 proto 1 Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/forward.c:830]: do_forward_reply(): reply forwarded (2nd via address: 89.210.243.1 port 5060) Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/forward.c:839]: do_forward_reply(): reply forwarding finished (2nd via address: 89.210.243.1 port 5060) Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/receive.c:442]: receive_msg(): reply-route executed in: 115 usec Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/xavp.c:507]: xavp_destroy_list(): destroying xavp list (nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 20(1926) DEBUG: {2 12375830 INVITE 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/receive.c:458]: receive_msg(): cleaning up Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/udp_server.c:492]: udp_rcv_loop(): received on udp socket: (106/100/405) [[ACK sip:393817097132@48.146.161.153:5060 SIP/2.0 0D 0A Call-ID: 4a681d8626b801ce1a8185472fda8e86(a)37.148.1]] Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/msg_parser.c:610]: parse_msg(): SIP Request: Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/msg_parser.c:612]: parse_msg(): method: Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/msg_parser.c:614]: parse_msg(): uri: sip:393817097132@48.146.161.153:5060 Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/msg_parser.c:616]: parse_msg(): version: <SIP/2.0> Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/msg_parser.c:152]: get_hdr_field(): cseq : <12375830> Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=as73e66fe7 Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/msg_parser.c:171]: get_hdr_field(): [50]; uri=[sip:393817097132@48.146.161.153] Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/msg_parser.c:174]: get_hdr_field(): to body [sip:393817097132@48.146.161.153], to tag [as73e66fe7] Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, = ; state=16 Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/msg_parser.c:498]: parse_headers(): Via found, flags=2 Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/msg_parser.c:500]: parse_headers(): this is the first via Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/receive.c:246]: receive_msg(): --- received sip message - request - call-id: [4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060] - cseq: [12375830 ACK] Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/msg_parser.c:185]: get_hdr_field(): content_length=0 Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/receive.c:296]: receive_msg(): preparing to run routing scripts... Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} maxfwd [mf_funcs.c:74]: is_maxfwd_present(): value = 70 Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:247]: check_ruri_sip_version(): check_ruri_sip_version entered Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:270]: check_ruri_sip_version(): check_ruri_sip_version passed Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:277]: check_ruri_scheme(): check_ruri_scheme entered Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:297]: check_ruri_scheme(): check_ruri_scheme passed Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:305]: check_required_headers(): check_required_headers entered Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:313]: check_required_headers(): check_required_headers passed Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:479]: check_cseq_method(): check_cseq_method entered Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:512]: check_cseq_method(): check_cseq_method passed Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:521]: check_cseq_value(): check_cseq_value entered Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:549]: check_cseq_value(): check_cseq_value passed Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:558]: check_cl(): check_cl entered Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:583]: check_cl(): check_cl passed Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:595]: check_expires_value(): check_expires_value entered Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:632]: check_expires_value(): no expires header found Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:732]: check_parse_uris(): check_parse_uris entered Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:736]: check_parse_uris(): parsing ruri Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:750]: check_parse_uris(): looking up From header Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:764]: check_parse_uris(): parsing From header Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=00-07989-012374ff-364731f03 Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:790]: check_parse_uris(): parsing From URI Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:808]: check_parse_uris(): looking up To header Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:830]: check_parse_uris(): check_parse_uris(): parsing To URI Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity.c:875]: check_parse_uris(): check_parse_uris passed Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} sanity [sanity_mod.c:254]: w_sanity_check(): sanity checks result: 1 Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} siputils [checks.c:123]: has_totag(): totag found Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} rr [loose.c:114]: find_first_route(): No Route headers found Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} rr [loose.c:961]: loose_route(): There is no Route HF Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} tm [t_lookup.c:1043]: t_check_msg(): msg (0x7f330aed6050) id=300/1925 global id=299/1925 T start=(nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} tm [t_lookup.c:500]: t_lookup_request(): start searching: hash=11128, isACK=1 Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} tm [t_lookup.c:458]: matching_3261(): RFC3261 transaction matching failed - via branch [z9hG4bK-173F80F1-114A19A] Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} tm [t_lookup.c:682]: t_lookup_request(): no transaction found Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} tm [t_lookup.c:1113]: t_check_msg(): msg (0x7f330aed6050) id=300/1925 global id=300/1925 T end=(nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/receive.c:354]: receive_msg(): request-route executed in: 155 usec Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/xavp.c:507]: xavp_destroy_list(): destroying xavp list (nil) Aug 11 00:06:11 c4-proxy-1 kamailio: 19(1925) DEBUG: {1 12375830 ACK 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060} [core/receive.c:458]: receive_msg(): cleaning up ============================================================================================================= PCAP trace U 2020/08/11 00:05:51.498554 48.146.161.153:5060 -> 48.146.161.152:5060 #96 INVITE sip:79409098828@212.25.64.8:5060 SIP/2.0. Via: SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK01203950. Max-Forwards: 70. From: sip:393817097132@48.146.161.153;tag=as670e72d5. To: sip:79409098828@212.25.64.8:5060. Contact: sip:393817097132@48.146.161.153:5060. Call-ID: 274087066134728e4aa01a5356fde6a8@48.146.161.153:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 16.1.1. Date: Mon, 10 Aug 2020 21:05:51 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 324. . v=0. o=root 926780413 926780413 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 16980 RTP/AVP 8 0 18 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=maxptime:150. a=sendrecv. ............1_H. U 2020/08/11 00:05:51.499026 48.146.161.152:5060 -> 48.146.161.153:5060 #97 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK01203950. From: sip:393817097132@48.146.161.153;tag=as670e72d5. To: sip:79409098828@212.25.64.8:5060. Call-ID: 274087066134728e4aa01a5356fde6a8@48.146.161.153:5060. CSeq: 102 INVITE. Server: kamailio (5.2.7 (x86_64/linux)). Content-Length: 0. . x.....1_........ U 2020/08/11 00:05:51.499105 48.146.161.152:5060 -> 212.25.64.8:5060 #98 INVITE sip:79409098828@212.25.64.8:5060 SIP/2.0. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK415f.07dfd4858a8d7af10f14a65a40c09040.0. Via: SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK01203950. Max-Forwards: 69. From: sip:393817097132@48.146.161.153;tag=as670e72d5. To: sip:79409098828@212.25.64.8:5060. Contact: sip:393817097132@48.146.161.153:5060. Call-ID: 274087066134728e4aa01a5356fde6a8@48.146.161.153:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 16.1.1. Date: Mon, 10 Aug 2020 21:05:51 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 324. . v=0. o=root 926780413 926780413 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 16980 RTP/AVP 8 0 18 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=maxptime:150. a=sendrecv. .............1_. U 2020/08/11 00:05:51.523721 212.25.64.8:5060 -> 48.146.161.152:5060 #99 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK415f.07dfd4858a8d7af10f14a65a40c09040.0. Via: SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK01203950. From: sip:393817097132@48.146.161.153;tag=as670e72d5. To: sip:79409098828@212.25.64.8:5060. Call-ID: 274087066134728e4aa01a5356fde6a8@48.146.161.153:5060. CSeq: 102 INVITE. . ................ U 2020/08/11 00:05:51.614735 212.25.64.8:5060 -> 48.146.161.152:5060 #104 INVITE sip:79409098828@48.146.161.152:5060;user=phone SIP/2.0. Via: SIP/2.0/UDP 212.25.64.8:5060;branch=z9hG4bKtd41as3018urve5udtj0.1. Call-ID: 4ffhw785wd5ah9bba94kke5921674b98@UAC. From: sip:393817097132@212.25.64.8:5060;user=phone;tag=2f6fdw29-CC-1021. To: sip:79409098828@48.146.161.152:5060;user=phone. CSeq: 1 INVITE. P-Asserted-Identity: sip:393817097132@10.96.16.6;user=phone. Max-Forwards: 66. Contact: sip:393817097132@212.25.64.8:5060;transport=udp. Min-SE: 90. Session-Expires: 1800. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER. User-Agent: Huawei SoftX3000 V600R013. Supported: timer. Content-Length: 288. Content-Type: application/sdp. . v=0. o=HuaweiSoftX3000 36222851 36222852 IN IP4 212.25.64.8. s=Sip Call. c=IN IP4 212.25.64.8. t=0 0. m=audio 10892 RTP/AVP 8 0 18 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=ptime:20. a=fmtp:101 0-15. a=fmtp:18 annexb=no. ........1_...$.. U 2020/08/11 00:05:51.615233 48.146.161.152:5060 -> 212.25.64.8:5060 #105 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 212.25.64.8:5060;branch=z9hG4bKtd41as3018urve5udtj0.1. Call-ID: 4ffhw785wd5ah9bba94kke5921674b98@UAC. From: sip:393817097132@212.25.64.8:5060;user=phone;tag=2f6fdw29-CC-1021. To: sip:79409098828@48.146.161.152:5060;user=phone. CSeq: 1 INVITE. Server: kamailio (5.2.7 (x86_64/linux)). Content-Length: 0. . ............1_.@ U 2020/08/11 00:05:51.615334 48.146.161.152:5060 -> 48.146.161.153:5060 #106 INVITE sip:79409098828@48.146.161.152:5060;user=phone SIP/2.0. Record-Route: sip:48.146.161.152;lr. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bKfbce.f74d055cfd186a85dacedcc9f4d282ce.0. Via: SIP/2.0/UDP 212.25.64.8:5060;branch=z9hG4bKtd41as3018urve5udtj0.1. Call-ID: 4ffhw785wd5ah9bba94kke5921674b98@UAC. From: sip:393817097132@212.25.64.8:5060;user=phone;tag=2f6fdw29-CC-1021. To: sip:79409098828@48.146.161.152:5060;user=phone. CSeq: 1 INVITE. P-Asserted-Identity: sip:393817097132@10.96.16.6;user=phone. Max-Forwards: 65. Contact: sip:393817097132@212.25.64.8:5060;transport=udp. Min-SE: 90. Session-Expires: 1800. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER. User-Agent: Huawei SoftX3000 V600R013. Supported: timer. Content-Length: 288. Content-Type: application/sdp. . v=0. o=HuaweiSoftX3000 36222851 36222852 IN IP4 212.25.64.8. s=Sip Call. c=IN IP4 212.25.64.8. t=0 0. m=audio 10892 RTP/AVP 8 0 18 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=ptime:20. a=fmtp:101 0-15. a=fmtp:18 annexb=no. ..........1_*..$ U 2020/08/11 00:05:51.616177 48.146.161.153:5060 -> 48.146.161.152:5060 #107 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bKfbce.f74d055cfd186a85dacedcc9f4d282ce.0;received=48.146.161.152;rport=5060. Via: SIP/2.0/UDP 212.25.64.8:5060;branch=z9hG4bKtd41as3018urve5udtj0.1. Record-Route: sip:48.146.161.152;lr. From: sip:393817097132@212.25.64.8:5060;user=phone;tag=2f6fdw29-CC-1021. To: sip:79409098828@48.146.161.152:5060;user=phone. Call-ID: 4ffhw785wd5ah9bba94kke5921674b98@UAC. CSeq: 1 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Session-Expires: 1800;refresher=uas. Contact: sip:79409098828@48.146.161.153:5060. Content-Length: 0. . ................ U 2020/08/11 00:05:51.690457 48.146.161.153:5060 -> 48.146.161.152:5060 #112 INVITE sip:79409098828@89.210.243.1:5060 SIP/2.0. Via: SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK1b39be5a. Max-Forwards: 70. From: sip:393817097132@48.146.161.153;tag=as73e66fe7. To: sip:79409098828@89.210.243.1:5060. Contact: sip:393817097132@48.146.161.153:5060. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 16.1.1. Date: Mon, 10 Aug 2020 21:05:51 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. P-Asserted-Identity: sip:393817097132@48.146.161.152. Content-Type: application/sdp. Content-Length: 326. . v=0. o=root 1995384227 1995384227 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 8 18 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=maxptime:150. a=sendrecv. ........1_K.-)}. U 2020/08/11 00:05:51.690857 48.146.161.152:5060 -> 48.146.161.153:5060 #113 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK1b39be5a. From: sip:393817097132@48.146.161.153;tag=as73e66fe7. To: sip:79409098828@89.210.243.1:5060. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 102 INVITE. Server: kamailio (5.2.7 (x86_64/linux)). Content-Length: 0. . .............1_D U 2020/08/11 00:05:51.690934 48.146.161.152:5060 -> 89.210.243.1:5060 #114 INVITE sip:79409098828@89.210.243.1:5060 SIP/2.0. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK8b4f.c29b6618fd22db1ce4b8aec0d0dcf809.0. Via: SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK1b39be5a. Max-Forwards: 69. From: sip:393817097132@48.146.161.153;tag=as73e66fe7. To: sip:79409098828@89.210.243.1:5060. Contact: sip:393817097132@48.146.161.153:5060. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 16.1.1. Date: Mon, 10 Aug 2020 21:05:51 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. P-Asserted-Identity: sip:393817097132@48.146.161.152. Content-Type: application/sdp. Content-Length: 326. . v=0. o=root 1995384227 1995384227 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 8 18 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=maxptime:150. a=sendrecv. .........1_...). U 2020/08/11 00:05:51.697205 89.210.243.1:5060 -> 48.146.161.152:5060 #115 SIP/2.0 100 Trying. Allow: UPDATE,REFER,INFO. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. Contact: sip:89.210.243.1:5060. CSeq: 102 INVITE. From: sip:393817097132@48.146.161.153;tag=as73e66fe7. Server: Qirpack/v4.88 (gw_sip). To: sip:79409098828@89.210.243.1:5060. Via: SIP/2.0/UDP 48.146.161.152;received=48.146.161.152;rport=5060;branch=z9hG4bK8b4f.c29b6618fd22db1ce4b8aec0d0dcf809.0,SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK1b39be5a. Content-Length: 0. . ................ U 2020/08/11 00:05:51.890363 89.210.243.1:5060 -> 48.146.161.152:5060 #120 SIP/2.0 183 In band info available. Allow: UPDATE,REFER,INFO. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. Contact: sip:89.210.243.1:5060. Content-Type: application/sdp. CSeq: 102 INVITE. From: sip:393817097132@48.146.161.153;tag=as73e66fe7. Server: Qirpack/v4.88 (gw_sip). To: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Via: SIP/2.0/UDP 48.146.161.152;received=48.146.161.152;rport=5060;branch=z9hG4bK8b4f.c29b6618fd22db1ce4b8aec0d0dcf809.0,SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK1b39be5a. Content-Length: 185. . v=0. o=cp10 159709355110 159709355111 IN IP4 89.210.243.21. s=SIP Call. c=IN IP4 89.210.243.21. t=0 0. m=audio 36322 RTP/AVP 0. b=AS:80. a=rtpmap:0 PCMU/8000/1. a=ptime:20. a=sendrecv. sip+3+......1_.. U 2020/08/11 00:05:51.890412 89.210.243.1:5060 -> 48.146.161.152:5060 #121 SIP/2.0 200 OK. Allow: UPDATE,REFER,INFO. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. Contact: sip:89.210.243.1:5060. Content-Type: application/sdp. CSeq: 102 INVITE. From: sip:393817097132@48.146.161.153;tag=as73e66fe7. P-Asserted-Identity: sip:79409098828@89.210.243.1;user=phone. Server: Qirpack/v4.88 (gw_sip). To: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Via: SIP/2.0/UDP 48.146.161.152;received=48.146.161.152;rport=5060;branch=z9hG4bK8b4f.c29b6618fd22db1ce4b8aec0d0dcf809.0,SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK1b39be5a. Content-Length: 185. . v=0. o=cp10 159709355110 159709355111 IN IP4 89.210.243.21. s=SIP Call. c=IN IP4 89.210.243.21. t=0 0. m=audio 36322 RTP/AVP 0. b=AS:80. a=rtpmap:0 PCMU/8000/1. a=ptime:20. a=sendrecv. ........1_...5.. U 2020/08/11 00:05:51.890666 48.146.161.152:5060 -> 48.146.161.153:5060 #122 SIP/2.0 183 In band info available. Allow: UPDATE,REFER,INFO. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. Contact: sip:89.210.243.1:5060. Content-Type: application/sdp. CSeq: 102 INVITE. From: sip:393817097132@48.146.161.153;tag=as73e66fe7. Server: Qirpack/v4.88 (gw_sip). To: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Via: SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK1b39be5a. Content-Length: 185. . v=0. o=cp10 159709355110 159709355111 IN IP4 89.210.243.21. s=SIP Call. c=IN IP4 89.210.243.21. t=0 0. m=audio 36322 RTP/AVP 0. b=AS:80. a=rtpmap:0 PCMU/8000/1. a=ptime:20. a=sendrecv. ..0.....1_.].5.. U 2020/08/11 00:05:51.890854 48.146.161.152:5060 -> 48.146.161.153:5060 #123 SIP/2.0 200 OK. Allow: UPDATE,REFER,INFO. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. Contact: sip:89.210.243.1:5060. Content-Type: application/sdp. CSeq: 102 INVITE. From: sip:393817097132@48.146.161.153;tag=as73e66fe7. P-Asserted-Identity: sip:79409098828@89.210.243.1;user=phone. Server: Qirpack/v4.88 (gw_sip). To: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Via: SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK1b39be5a. Content-Length: 185. . v=0. o=cp10 159709355110 159709355111 IN IP4 89.210.243.21. s=SIP Call. c=IN IP4 89.210.243.21. t=0 0. m=audio 36322 RTP/AVP 0. b=AS:80. a=rtpmap:0 PCMU/8000/1. a=ptime:20. a=sendrecv. ............1_iA U 2020/08/11 00:05:51.891240 48.146.161.153:5060 -> 48.146.161.152:5060 #124 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bKfbce.f74d055cfd186a85dacedcc9f4d282ce.0;received=48.146.161.152;rport=5060. Via: SIP/2.0/UDP 212.25.64.8:5060;branch=z9hG4bKtd41as3018urve5udtj0.1. Record-Route: sip:48.146.161.152;lr. From: sip:393817097132@212.25.64.8:5060;user=phone;tag=2f6fdw29-CC-1021. To: sip:79409098828@48.146.161.152:5060;user=phone;tag=as49ab4db4. Call-ID: 4ffhw785wd5ah9bba94kke5921674b98@UAC. CSeq: 1 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Session-Expires: 1800;refresher=uas. Contact: sip:79409098828@48.146.161.153:5060. Content-Type: application/sdp. Require: timer. Content-Length: 324. . v=0. o=root 792201108 792201108 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 17696 RTP/AVP 8 18 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=maxptime:150. a=sendrecv. ......P.....1_.. U 2020/08/11 00:05:51.891455 48.146.161.152:5060 -> 212.25.64.8:5060 #125 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 212.25.64.8:5060;branch=z9hG4bKtd41as3018urve5udtj0.1. Record-Route: sip:48.146.161.152;lr. From: sip:393817097132@212.25.64.8:5060;user=phone;tag=2f6fdw29-CC-1021. To: sip:79409098828@48.146.161.152:5060;user=phone;tag=as49ab4db4. Call-ID: 4ffhw785wd5ah9bba94kke5921674b98@UAC. CSeq: 1 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Session-Expires: 1800;refresher=uas. Contact: sip:79409098828@48.146.161.153:5060. Content-Type: application/sdp. Require: timer. Content-Length: 324. . v=0. o=root 792201108 792201108 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 17696 RTP/AVP 8 18 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=maxptime:150. a=sendrecv. ......1_..#5X... U 2020/08/11 00:05:51.891522 48.146.161.153:5060 -> 48.146.161.152:5060 #126 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bKfbce.f74d055cfd186a85dacedcc9f4d282ce.0;received=48.146.161.152;rport=5060. Via: SIP/2.0/UDP 212.25.64.8:5060;branch=z9hG4bKtd41as3018urve5udtj0.1. Record-Route: sip:48.146.161.152;lr. From: sip:393817097132@212.25.64.8:5060;user=phone;tag=2f6fdw29-CC-1021. To: sip:79409098828@48.146.161.152:5060;user=phone;tag=as49ab4db4. Call-ID: 4ffhw785wd5ah9bba94kke5921674b98@UAC. CSeq: 1 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Session-Expires: 1800;refresher=uas. Contact: sip:79409098828@48.146.161.153:5060. Content-Type: application/sdp. Require: timer. Content-Length: 324. . v=0. o=root 792201108 792201108 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 17696 RTP/AVP 8 18 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=maxptime:150. a=sendrecv. ....H.....1_..%5 U 2020/08/11 00:05:51.891651 48.146.161.152:5060 -> 212.25.64.8:5060 #127 SIP/2.0 200 OK. Via: SIP/2.0/UDP 212.25.64.8:5060;branch=z9hG4bKtd41as3018urve5udtj0.1. Record-Route: sip:48.146.161.152;lr. From: sip:393817097132@212.25.64.8:5060;user=phone;tag=2f6fdw29-CC-1021. To: sip:79409098828@48.146.161.152:5060;user=phone;tag=as49ab4db4. Call-ID: 4ffhw785wd5ah9bba94kke5921674b98@UAC. CSeq: 1 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Session-Expires: 1800;refresher=uas. Contact: sip:79409098828@48.146.161.153:5060. Content-Type: application/sdp. Require: timer. Content-Length: 324. . v=0. o=root 792201108 792201108 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 17696 RTP/AVP 8 18 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=maxptime:150. a=sendrecv. ......8.....1_.. U 2020/08/11 00:05:51.896994 89.210.243.1:5060 -> 48.146.161.152:5060 #128 INVITE sip:393817097132@48.146.161.153:5060 SIP/2.0. Allow: UPDATE,REFER,INFO. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. Contact: sip:89.210.243.1:5060. Content-Type: application/sdp. CSeq: 12375830 INVITE. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Max-Forwards: 70. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. User-Agent: Qirpack/v4.88 (gw_sip). Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. Content-Length: 185. . v=0. o=cp10 159709355110 159709355112 IN IP4 89.210.243.21. s=SIP Call. c=IN IP4 89.210.243.21. t=0 0. m=audio 36322 RTP/AVP 0. b=AS:80. a=rtpmap:0 PCMU/8000/1. a=ptime:20. a=sendrecv. .............1_. U 2020/08/11 00:05:51.897340 48.146.161.152:5060 -> 89.210.243.1:5060 #129 SIP/2.0 100 trying -- your call is important to us. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. Server: kamailio (5.2.7 (x86_64/linux)). Content-Length: 0. . .........1_FO}5$ U 2020/08/11 00:05:51.897404 48.146.161.152:5060 -> 48.146.161.153:5060 #130 INVITE sip:393817097132@48.146.161.153:5060 SIP/2.0. Allow: UPDATE,REFER,INFO. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. Contact: sip:89.210.243.1:5060. Content-Type: application/sdp. CSeq: 12375830 INVITE. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Max-Forwards: 69. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. User-Agent: Qirpack/v4.88 (gw_sip). Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK87b2.263f4b70fad82e9db02d9173ce3f65cb.0. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. Content-Length: 185. . v=0. o=cp10 159709355110 159709355112 IN IP4 89.210.243.21. s=SIP Call. c=IN IP4 89.210.243.21. t=0 0. m=audio 36322 RTP/AVP 0. b=AS:80. a=rtpmap:0 PCMU/8000/1. a=ptime:20. a=sendrecv. ......1_F..5.... U 2020/08/11 00:05:51.897746 48.146.161.153:5060 -> 48.146.161.152:5060 #131 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK87b2.263f4b70fad82e9db02d9173ce3f65cb.0;received=48.146.161.152. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Length: 0. . .......(.....1_I U 2020/08/11 00:05:51.897836 48.146.161.153:5060 -> 48.146.161.152:5060 #132 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK87b2.263f4b70fad82e9db02d9173ce3f65cb.0;received=48.146.161.152. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. ..........1_U@.5 U 2020/08/11 00:05:51.898121 48.146.161.152:5060 -> 89.210.243.1:5060 #133 SIP/2.0 200 OK. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. .........1_..5. U 2020/08/11 00:05:51.903991 89.210.243.1:5060 -> 48.146.161.152:5060 #134 ACK sip:393817097132@48.146.161.153:5060 SIP/2.0. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 ACK. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Max-Forwards: 70. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. User-Agent: Qirpack/v4.88 (gw_sip). Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-173F80F1-114A19A. Content-Length: 0. . ................ U 2020/08/11 00:05:51.947711 212.25.64.8:5060 -> 48.146.161.152:5060 #146 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK415f.07dfd4858a8d7af10f14a65a40c09040.0. Via: SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK01203950. From: sip:393817097132@48.146.161.153;tag=as670e72d5. To: sip:79409098828@212.25.64.8:5060;tag=f5d48kwf-CC-1021. Call-ID: 274087066134728e4aa01a5356fde6a8@48.146.161.153:5060. CSeq: 102 INVITE. Contact: sip:79409098828@212.25.64.8:5060;transport=udp. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER. Content-Length: 288. Content-Type: application/sdp. . v=0. o=HuaweiSoftX3000 36222850 36222851 IN IP4 212.25.64.8. s=Sip Call. c=IN IP4 212.25.64.8. t=0 0. m=audio 10052 RTP/AVP 8 18 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=ptime:20. a=fmtp:101 0-15. a=fmtp:18 annexb=no. .............1_v U 2020/08/11 00:05:51.947724 212.25.64.8:5060 -> 48.146.161.152:5060 #147 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK415f.07dfd4858a8d7af10f14a65a40c09040.0. Via: SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK01203950. From: sip:393817097132@48.146.161.153;tag=as670e72d5. To: sip:79409098828@212.25.64.8:5060;tag=f5d48kwf-CC-1021. Call-ID: 274087066134728e4aa01a5356fde6a8@48.146.161.153:5060. CSeq: 102 INVITE. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER. Session-Expires: 1800;refresher=uac. Require: timer. Contact: sip:79409098828@212.25.64.8:5060;transport=udp. Content-Length: 288. Content-Type: application/sdp. . v=0. o=HuaweiSoftX3000 36222850 36222851 IN IP4 212.25.64.8. s=Sip Call. c=IN IP4 212.25.64.8. t=0 0. m=audio 10052 RTP/AVP 8 18 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=ptime:20. a=fmtp:101 0-15. a=fmtp:18 annexb=no. ......1_X..8U... U 2020/08/11 00:05:51.947972 48.146.161.152:5060 -> 48.146.161.153:5060 #148 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.153:5060;branch=z9hG4bK01203950. From: sip:393817097132@48.146.161.153;tag=as670e72d5. To: sip:79409098828@212.25.64.8:5060;tag=f5d48kwf-CC-1021. Call-ID: 274087066134728e4aa01a5356fde6a8@48.146.161.153:5060. CSeq: 102 INVITE. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER. Session-Expires: 1800;refresher=uac. Require: timer. Contact: sip:79409098828@212.25.64.8:5060;transport=udp. Content-Length: 288. Content-Type: application/sdp. . v=0. o=HuaweiSoftX3000 36222850 36222851 IN IP4 212.25.64.8. s=Sip Call. c=IN IP4 212.25.64.8. t=0 0. m=audio 10052 RTP/AVP 8 18 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=ptime:20. a=fmtp:101 0-15. a=fmtp:18 annexb=no. .............1_. U 2020/08/11 00:05:51.987695 212.25.64.8:5060 -> 48.146.161.152:5060 #151 ACK sip:79409098828@48.146.161.153:5060 SIP/2.0. Via: SIP/2.0/UDP 212.25.64.8:5060;branch=z9hG4bKs6oh5g207gbfihgcs6g0.1. Call-ID: 4ffhw785wd5ah9bba94kke5921674b98@UAC. From: sip:393817097132@48.146.161.152:5060;user=phone;tag=2f6fdw29-CC-1021. To: sip:79409098828@212.25.64.8:5060;user=phone;tag=as49ab4db4. CSeq: 1 ACK. Max-Forwards: 69. Content-Length: 0. Route: sip:48.146.161.152;lr. . ..........1_...: U 2020/08/11 00:05:51.988125 48.146.161.152:5060 -> 48.146.161.153:5060 #152 ACK sip:79409098828@48.146.161.153:5060 SIP/2.0. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bKfbce.1f078d36629b5ee0771933dbd92008e5.0. Via: SIP/2.0/UDP 212.25.64.8:5060;branch=z9hG4bKs6oh5g207gbfihgcs6g0.1. Call-ID: 4ffhw785wd5ah9bba94kke5921674b98@UAC. From: sip:393817097132@48.146.161.152:5060;user=phone;tag=2f6fdw29-CC-1021. To: sip:79409098828@212.25.64.8:5060;user=phone;tag=as49ab4db4. CSeq: 1 ACK. Max-Forwards: 68. Content-Length: 0. . ................ U 2020/08/11 00:05:52.397845 48.146.161.153:5060 -> 48.146.161.152:5060 #157 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK87b2.263f4b70fad82e9db02d9173ce3f65cb.0;received=48.146.161.152. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. ..........1_2... U 2020/08/11 00:05:52.398105 48.146.161.152:5060 -> 89.210.243.1:5060 #158 SIP/2.0 200 OK. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. .........1_A.... U 2020/08/11 00:05:52.403706 89.210.243.1:5060 -> 48.146.161.152:5060 #159 ACK sip:393817097132@48.146.161.153:5060 SIP/2.0. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 ACK. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Max-Forwards: 70. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. User-Agent: Qirpack/v4.88 (gw_sip). Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-173F80F1-114A19A. Content-Length: 0. . ................ U 2020/08/11 00:05:53.397379 48.146.161.153:5060 -> 48.146.161.152:5060 #168 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK87b2.263f4b70fad82e9db02d9173ce3f65cb.0;received=48.146.161.152. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. ..........1_.... U 2020/08/11 00:05:53.397668 48.146.161.152:5060 -> 89.210.243.1:5060 #169 SIP/2.0 200 OK. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. .........1_.#... U 2020/08/11 00:05:53.403448 89.210.243.1:5060 -> 48.146.161.152:5060 #170 ACK sip:393817097132@48.146.161.153:5060 SIP/2.0. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 ACK. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Max-Forwards: 70. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. User-Agent: Qirpack/v4.88 (gw_sip). Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-173F80F1-114A19A. Content-Length: 0. . ................ U 2020/08/11 00:05:54.201755 212.25.64.8:5060 -> 48.146.161.152:5060 #177 BYE sip:79409098828@48.146.161.153:5060 SIP/2.0. Via: SIP/2.0/UDP 212.25.64.8:5060;branch=z9hG4bKs6oh5g207gbfihgcs6g0cd0000010.1. Call-ID: 4ffhw785wd5ah9bba94kke5921674b98@UAC. From: sip:393817097132@48.146.161.152:5060;user=phone;tag=2f6fdw29-CC-1021. To: sip:79409098828@212.25.64.8:5060;user=phone;tag=as49ab4db4. CSeq: 2 BYE. Reason: Q.850;cause=16;text="normal call clearing". Max-Forwards: 69. Content-Length: 0. Route: sip:48.146.161.152;lr. . .d...........1_. U 2020/08/11 00:05:54.202256 48.146.161.152:5060 -> 48.146.161.153:5060 #178 BYE sip:79409098828@48.146.161.153:5060 SIP/2.0. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bKcbce.1e200d33771522a7baea07b052e62613.0. Via: SIP/2.0/UDP 212.25.64.8:5060;branch=z9hG4bKs6oh5g207gbfihgcs6g0cd0000010.1. Call-ID: 4ffhw785wd5ah9bba94kke5921674b98@UAC. From: sip:393817097132@48.146.161.152:5060;user=phone;tag=2f6fdw29-CC-1021. To: sip:79409098828@212.25.64.8:5060;user=phone;tag=as49ab4db4. CSeq: 2 BYE. Reason: Q.850;cause=16;text="normal call clearing". Max-Forwards: 68. Content-Length: 0. . ......1_....m... U 2020/08/11 00:05:54.203015 48.146.161.153:5060 -> 48.146.161.152:5060 #179 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bKcbce.1e200d33771522a7baea07b052e62613.0;received=48.146.161.152;rport=5060. Via: SIP/2.0/UDP 212.25.64.8:5060;branch=z9hG4bKs6oh5g207gbfihgcs6g0cd0000010.1. From: sip:393817097132@48.146.161.152:5060;user=phone;tag=2f6fdw29-CC-1021. To: sip:79409098828@212.25.64.8:5060;user=phone;tag=as49ab4db4. Call-ID: 4ffhw785wd5ah9bba94kke5921674b98@UAC. CSeq: 2 BYE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Content-Length: 0. . .............1_. U 2020/08/11 00:05:54.203229 48.146.161.152:5060 -> 212.25.64.8:5060 #180 SIP/2.0 200 OK. Via: SIP/2.0/UDP 212.25.64.8:5060;branch=z9hG4bKs6oh5g207gbfihgcs6g0cd0000010.1. From: sip:393817097132@48.146.161.152:5060;user=phone;tag=2f6fdw29-CC-1021. To: sip:79409098828@212.25.64.8:5060;user=phone;tag=as49ab4db4. Call-ID: 4ffhw785wd5ah9bba94kke5921674b98@UAC. CSeq: 2 BYE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Content-Length: 0. . ................ U 2020/08/11 00:05:55.397519 48.146.161.153:5060 -> 48.146.161.152:5060 #189 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK87b2.263f4b70fad82e9db02d9173ce3f65cb.0;received=48.146.161.152. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. RE.G......1_.... U 2020/08/11 00:05:55.397793 48.146.161.152:5060 -> 89.210.243.1:5060 #190 SIP/2.0 200 OK. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. .;.......1_..... U 2020/08/11 00:05:55.403541 89.210.243.1:5060 -> 48.146.161.152:5060 #191 ACK sip:393817097132@48.146.161.153:5060 SIP/2.0. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 ACK. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Max-Forwards: 70. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. User-Agent: Qirpack/v4.88 (gw_sip). Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-173F80F1-114A19A. Content-Length: 0. . .....m.JXH._.`.5 I 2020/08/11 00:05:58.447029 37.148.171.161 -> 48.146.161.152 3:1 #198 ....E....w..?...%....DB.......)0OPTIONS sip:193.68.66.131 SIP/2.0. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK137c.07826ed187328c4cea4207d10acdeae3.0. Via: SIP/2.0/UDP 37.148.171.164:5060;branch=z9hG4bK13044a4f. Max-Forwards: 69. From: "asterisk" sip:asterisk@37.148.171.164;tag=as4239ef1a. To: sip:193.68.66.131. Contact: sip:asterisk@37.148.171.164:5060. Call-ID: 5acd62973c560c4c7544f5c67e551e86@37.148.171.164:5060. CSeq: 102 OPTIONS. User-Agent: Asterisk PBX GIT-master-c83a44c. Date: Mon, 10 Aug 2020 21:05:37 GMT. Allow: INVITE, ACK, CAsip:393817097132 U 2020/08/11 00:05:59.398803 48.146.161.153:5060 -> 48.146.161.152:5060 #203 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK87b2.263f4b70fad82e9db02d9173ce3f65cb.0;received=48.146.161.152. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. nten......1_.... U 2020/08/11 00:05:59.399150 48.146.161.152:5060 -> 89.210.243.1:5060 #204 SIP/2.0 200 OK. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. .........1_..".. U 2020/08/11 00:05:59.404928 89.210.243.1:5060 -> 48.146.161.152:5060 #205 ACK sip:393817097132@48.146.161.153:5060 SIP/2.0. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 ACK. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Max-Forwards: 70. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. User-Agent: Qirpack/v4.88 (gw_sip). Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-173F80F1-114A19A. Content-Length: 0. . ................ U 2020/08/11 00:06:03.398077 48.146.161.153:5060 -> 48.146.161.152:5060 #214 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK87b2.263f4b70fad82e9db02d9173ce3f65cb.0;received=48.146.161.152. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. .70.......1_8T.. U 2020/08/11 00:06:03.398349 48.146.161.152:5060 -> 89.210.243.1:5060 #215 SIP/2.0 200 OK. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. ..%......1_..... U 2020/08/11 00:06:03.404194 89.210.243.1:5060 -> 48.146.161.152:5060 #216 ACK sip:393817097132@48.146.161.153:5060 SIP/2.0. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 ACK. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Max-Forwards: 70. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. User-Agent: Qirpack/v4.88 (gw_sip). Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-173F80F1-114A19A. Content-Length: 0. . .pBk.^.<.[..P..w U 2020/08/11 00:06:07.398394 48.146.161.153:5060 -> 48.146.161.152:5060 #230 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK87b2.263f4b70fad82e9db02d9173ce3f65cb.0;received=48.146.161.152. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. ster......1_.... U 2020/08/11 00:06:07.398701 48.146.161.152:5060 -> 89.210.243.1:5060 #231 SIP/2.0 200 OK. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. .........1_..... U 2020/08/11 00:06:07.404413 89.210.243.1:5060 -> 48.146.161.152:5060 #232 ACK sip:393817097132@48.146.161.153:5060 SIP/2.0. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 ACK. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Max-Forwards: 70. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. User-Agent: Qirpack/v4.88 (gw_sip). Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-173F80F1-114A19A. Content-Length: 0. . ................ U 2020/08/11 00:06:11.397466 48.146.161.153:5060 -> 48.146.161.152:5060 #249 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK87b2.263f4b70fad82e9db02d9173ce3f65cb.0;received=48.146.161.152. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. ster......1_n... U 2020/08/11 00:06:11.397713 48.146.161.152:5060 -> 89.210.243.1:5060 #250 SIP/2.0 200 OK. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. .........1_j.... U 2020/08/11 00:06:11.403609 89.210.243.1:5060 -> 48.146.161.152:5060 #251 ACK sip:393817097132@48.146.161.153:5060 SIP/2.0. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 ACK. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Max-Forwards: 70. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. User-Agent: Qirpack/v4.88 (gw_sip). Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-173F80F1-114A19A. Content-Length: 0. . gth......1_....b U 2020/08/11 00:06:15.398243 48.146.161.153:5060 -> 48.146.161.152:5060 #266 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK87b2.263f4b70fad82e9db02d9173ce3f65cb.0;received=48.146.161.152. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. $..;......1_.... U 2020/08/11 00:06:15.398528 48.146.161.152:5060 -> 89.210.243.1:5060 #267 SIP/2.0 200 OK. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. t........1_..... U 2020/08/11 00:06:15.404350 89.210.243.1:5060 -> 48.146.161.152:5060 #268 ACK sip:393817097132@48.146.161.153:5060 SIP/2.0. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 ACK. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Max-Forwards: 70. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. User-Agent: Qirpack/v4.88 (gw_sip). Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-173F80F1-114A19A. Content-Length: 0. . .zD8...#:g..!.. U 2020/08/11 00:06:19.397384 48.146.161.153:5060 -> 48.146.161.152:5060 #293 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK87b2.263f4b70fad82e9db02d9173ce3f65cb.0;received=48.146.161.152. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. L, O......1_lc.. U 2020/08/11 00:06:19.397632 48.146.161.152:5060 -> 89.210.243.1:5060 #294 SIP/2.0 200 OK. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. .........1_..... U 2020/08/11 00:06:19.403487 89.210.243.1:5060 -> 48.146.161.152:5060 #295 ACK sip:393817097132@48.146.161.153:5060 SIP/2.0. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 ACK. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Max-Forwards: 70. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. User-Agent: Qirpack/v4.88 (gw_sip). Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-173F80F1-114A19A. Content-Length: 0. . ................ U 2020/08/11 00:06:23.398710 48.146.161.153:5060 -> 48.146.161.152:5060 #313 SIP/2.0 200 OK. Via: SIP/2.0/UDP 48.146.161.152;branch=z9hG4bK87b2.263f4b70fad82e9db02d9173ce3f65cb.0;received=48.146.161.152. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. ._|r......1_f=.. U 2020/08/11 00:06:23.398998 48.146.161.152:5060 -> 89.210.243.1:5060 #314 SIP/2.0 200 OK. Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-5F68B615-114A199. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 INVITE. Server: Asterisk PBX 16.1.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: sip:393817097132@48.146.161.153:5060. Content-Type: application/sdp. Content-Length: 270. . v=0. o=root 1995384227 1995384228 IN IP4 48.146.161.153. s=Asterisk PBX 16.1.1. c=IN IP4 48.146.161.153. t=0 0. m=audio 18584 RTP/AVP 0 8 18. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. a=maxptime:150. a=sendrecv. rQ.......1_..... U 2020/08/11 00:06:23.404721 89.210.243.1:5060 -> 48.146.161.152:5060 #315 ACK sip:393817097132@48.146.161.153:5060 SIP/2.0. Call-ID: 4a681d8626b801ce1a8185472fda8e86@48.146.161.153:5060. CSeq: 12375830 ACK. From: sip:79409098828@89.210.243.1:5060;tag=00-07989-012374ff-364731f03. Max-Forwards: 70. To: sip:393817097132@48.146.161.153;tag=as73e66fe7. User-Agent: Qirpack/v4.88 (gw_sip). Via: SIP/2.0/UDP 89.210.243.1:5060;branch=z9hG4bK-173F80F1-114A19A. Content-Length: 0. . ...@...X.3.....W BR, Alex

2 3

Change Record Route onreply
by Aidar Kamalov 11 Aug '20

11 Aug '20

Hello, I have kamailio with only one interface. Is it possible to set one RR - lan ip address to INVITEs (to dispatcher list members) and 127.0.0.1 to replies 200OK (to softphone)? I can't use record_route() function in onreply route because: Command cannot be used in the block. -- Aydar A. Kamalov

2 2

dns_naptr_ignore_rfc default value
by Juha Heinanen 10 Aug '20

10 Aug '20

What is the point of dns_naptr_ignore_rfc default value: dns_naptr_ignore_rfc If the DNS lookup should ignore the remote side's protocol preferences, as indicated by the Order field in the NAPTR records and mandated by RFC 2915. dns_naptr_ignore_rfc = yes | no (default yes) In my (and RFC 2915's) opinion the default should be 'no'. -- Juha

2 3

CallerID - multiple ISP Prefix
by Ilie Soltanici 10 Aug '20

10 Aug '20

Hello, We are using a few carries where we are sending calls to, each of them is using a special prefix in the From header in order to manage the callerid. Some of them are not recognizing headers like P-Asserted-Identity or Remote-Party-ID, so for them we have to replace the From header with the corresponding prefix. The way we do - it's by using the uac_replace_from function, which works fine. The problem is when we are sending the same call to the failover trunk, and that trunk is using a different prefix. We have to call again the uac_replace_from function which is appending the new prefix to the existing one, something like: From: <sip:12345@domain.localsip:+12345@domain.local>;tag=as404cb50c I was trying to use uac_restore_from() function and call again uac_replace_from, but the result is the same. Is there any way to call the uac_restore_from function twice for the same call? Thank you.

3 3

Preprocessor ERROR! on upgrading to 5.4.0
by sagar malam 08 Aug '20

08 Aug '20

Hello , I was using Kamailio 5.3.3 . After upgrading to 5.4.0 i am getting below error with same cfg file which worked well with 5.3.3. 0(102523) ERROR: <core> [core/ppcfg.c:235]: pp_ifdef_level_error(): different number of preprocessor directives: 2 more #!if[n]def as #!endif What does this mean? Any clue to solve this? -- Thanks, Sagar

2 2

In memory ISP Info
by Ilie Soltanici 07 Aug '20

07 Aug '20

Hello, What will be the best way to save ISP Information in Kamailio memory? I have the following information for each ISP into the DB: *ISP Name | ISP IP | ISP Username | ISP Password | ISP Prefix* ISP1 | 1.1.1.1 | user | pass | 1234 This info it's not changed/updated too often, so I want to avoid querying the DB on every call and get this info directly from the memory. I was thinking to use htable for that, so I created a htable as an array for that and loaded all the information there manually, something like that: *kamcmd htable.sets isp ISP1[address] 1.1.1.1* * kamcmd htable.sets isp ISP1[username] user* * kamcmd htable.sets isp ISP1[password] pass* * kamcmd htable.sets isp ISP1[prefix] 1234 * *$sht( isp => ISP1[address]) == "1.1.1.1"* * $sht( isp => ISP1[username]) == "user" * * $sht( isp => ISP1[password]) == "pass" * * $sht( isp => ISP1[prefix]) == "1234" * So far so good, I'm able to get this information from the htable and use it whenever I need it. Now, the problem comes when I want to load all this information automatically from the DB. How can I setup htable module to get the array key (address, username...) from the database automatically on server restart, and append this array_key to the column value? I've tried this way: modparam("htable", "htable", "isp=>size=8;dbtable=isp;cols='trunk_name[address],trunk_address'") But it's not working unfortunately, getting back this error: You have an error in your SQL syntax; check the manual that corresponds to your MariaDB server version for the right syntax to use near '[address]' If I'm setting this way: modparam("htable", "htable", "isp=>size=8;dbtable=isp;cols='trunk_name,trunk_address'") and setting trunk_name value to: ISP1[address], then it's working fine, but in this case I will have to create by one column for each array key, which may be confusing and it's a lot of unnecessarily information into the DB. Is it possible to do that, or better to trigger a script on each restart which will load this information from the DB into htable? Thanks

2 2

Using wildcard certificates for Kamailio server
by Leonid Fainshtein 07 Aug '20

07 Aug '20

Hello, Is it permitted to use the wildcard TLS certificates for Kamailio server? In reality, it works (tested with v.5.4) but the RFC-5922 disables the wildcard certificates usage: "Implementations MUST match the values in their entirety: Implementations MUST NOT match suffixes. For example, "foo.example.com" does not match "example.com". Implementations MUST NOT match any form of wildcard, such as a leading "." or "*." with any other DNS label or sequence of labels. For example, "*.example.com" matches only "*.example.com" but not "foo.example.com". Similarly, ".example.com" matches only ".example.com", and does not match "foo.example.com". (Ref.:https://tools.ietf.org/html/rfc5922#section-7.2) To be honest, I don't understand why this restriction is good for... Is somebody aware of a newer RFC that removes this limitation? Best regards, Leonid Fainshtein

4 4

Deletion of Prometheus metrics
by Marat Gareev 07 Aug '20

07 Aug '20

Hello, I have confuse about this timeout: https://www.kamailio.org/docs/modules/devel/modules/xhttp_prom.html#xhttp_p… Is there reason for forced deletion of unused metrics? Let's imagine a situation: Zabbix reads metrics at a certain interval, unused metrics are deleted if they are not used for a period of time (xhttp_prom_timeout). I drew 2 cases when Zabbix will receive the same value, but we cannot say unambiguously whether these are new values or not. [image: xhttp_prom_timeout.jpg] Regards, Marat

2 4

Route to PSTN no works
by sip user 07 Aug '20

07 Aug '20

Hi, I have kamailio connect to Teams, and works form Asterisk -> Teams calls. For Teams -> Asterisk calls I'd worked using extension and register Asterisk with that extension. But I'd like to use direct routing with IP. In kamailio.cfg I activate define WITH_PSTN. I configured the IP and PORT for my PSTN. I'm using the default route[PSTN]: route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined xlog("L_INFO","PSTN ACTIVADO"); 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}$")){ xlog("L_INFO", "Error en el formato numerico!!"); return; } # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + #if (starts_with("$rU", "00")) { # strip(2); # prefix("+"); #} if (strempty($sel(cfg_get.pstn.gw_port))) { #$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); xlog("L_INFO","SELECCION CON PUERTO"); $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } else { xlog("L_INFO","SELECCION CON PUERTO"); $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } And in my request_route: remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { if($src_ip != "IP ASTERISK"){ xlog("L_INFO", "***********ROUTE PSTN***********"); route(PSTN); } else { xlog("L_INFO","LLamada desde $si con puerto $sp"); record_route_preset("FQND:5061;transport=tls", "IP KAMAILIO:5060"); add_rr_param(";r2=on"); route(DISPATCH); route(RELAY); } } But never see that the call go to PSTN route.. I'd made any wrong?? Thanks

3 2

Info: 2 Kamailio presentations soon live at ClueCon - 1:30pm CST (20:30 Berlin)
by Daniel-Constantin Mierla 06 Aug '20

06 Aug '20

Hello, quick info for those that want to watch the 2 presentations about Kamailio at ClueCon Deconstructed, starting in about 1 hour 40 minutes from now (or 13:30CST, 20:30Berlin/Paris/Madrid/...). One will be done by me and the 2nd by Fred Posner. A few more details are available at: - https://www.kamailio.org/w/2020/07/cluecon-deconstructed-2020/ The sessions should be live streamed on youtube - should appear in the freeswitch channel there at the time of start: - https://www.youtube.com/c/FreeSWITCH/ Of you can join the video conference room for a more interactive participation -- broadcast room listed at: - https://cluecon.cantina.video/#/home Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

2 2

Audio on Calls Stopping After 15 Minutes
by admin＠conradit.com 06 Aug '20

06 Aug '20

All, hi: I'm a Kamailio newbie! I've managed to set up a proxy server with the minimal configuration file found here: https://github.com/kamailio/kamailio/blob/master/misc/examples/mixed/kamail…. The issue that I'm having is that the audio on calls simply stop precisely after 15 minutes. It's always precisely at 15 minutes. The calls continue connected but without any audio. A summary about my set up: 1. This is an Amazon AWS EC2 server so it doesn't have a public IP directly attached to the interface. This makes me believe that the issue might be related to NAT. 2. I use Flowroute as the SIP trunk for these tests. 3. My phone system, for testing purposes, works on UDP port 6060 an the RTP ports are 20,000 to 30,000. 4. I am not using, for the time being, TLS nor SRTP. 5. My phone system is inside the same subnet as the Kamailio server. Questions: 1) Am I missing the NATHELPER module and if so how could we combine it to the minimal proxy configuration described above? Assuming that my Kamailio server uses local IP 10.80.10.5 and the PBX 10.80.10.6, both on UDP port 6060 for SIP. 2) May I simply combine the example file for nathelper.cfg as per this URL: https://github.com/kamailio/kamailio/blob/master/misc/examples/mixed/nathel… to the proxy example files, change IP addresses, ports, etc.? If not, how would one go about adding the NATHELPER module to the minimal proxy configuration? 3) Could it be some other issue that is causing the 15-minute interruption on audio? Thank you so much for your help!

2 1

Use-case does Kamailio support PBX - webrtc (web sockets)
by Johnny Ritzer 06 Aug '20

06 Aug '20

We have a PBX that doesn’t support websockets . But we want a PWA webaapp ctxSIP that uses sip.js to work . Due to its not SIP generic do we setup like this PBX <----> Kamailio <---->webapp Im looking for developers for android build via react native or ionic framework but not sure what solution/archtectiure Question. Does Kamailio register the SIP user/pass on the Kamailio server meaning it keeps it registered. And thus the ctxSIP client app just handles the pass thru for that connection. What about RTP do I need a separate handler for that. Tha nks

2 1

Info: Kamailio Getting Started Workshop at ClueCon - Live in 1 hour
by Daniel-Constantin Mierla 05 Aug '20

05 Aug '20

Hello, in about 1 hour I will do a quick (1 hour long) workshop at ClueCon deconstructed. It can be watched live on youtube -- should appear in their youtube channel: * https://www.youtube.com/channel/UC0i2xbniNHJQNfPPs_v4T0Q Or if you want to join the live video conference, which allows you to ask questions and interact with the participants, use an webrtc capable browser and go to: * https://cluecon.cantina.video/ Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 2

kamailio and rtpengine_manage() / rtpengine_delete()
by Karsten Horsmann 05 Aug '20

05 Aug '20

Hi list, I use rtpengine 7.2.1 branch with kamailio 5.2.x and 5.3.x in different setups. Due to some issues with my 5.2.x config I saw the "Total timed-out sessions via TIMEOUT" grow but "Total regular terminated sessions never" goes up from 0. That was an indicator for an non-existing rtpengine_manage() or rtpengine_delete() at the end of dialogs. Okay. Now I want to observe that a bit more. I use parallel forking to webrtc clients, that i guess will be the "more Offers" than Answers. Any hints on what numbers are important? rtpengine-ctl -ip 127.0.0.1:4444 list totals Total statistics (does not include current running sessions): Uptime of rtpengine :27726 seconds Total managed sessions :3435 Total rejected sessions :0 Total timed-out sessions via TIMEOUT :0 Total timed-out sessions via SILENT_TIMEOUT :0 Total timed-out sessions via FINAL_TIMEOUT :0 Total timed-out sessions via OFFER_TIMEOUT :0 Total regular terminated sessions :3435 Total forced terminated sessions :0 Total relayed packets :69127802 Total relayed packet errors :5063 Total number of streams with no relayed packets :128 Total number of 1-way streams :8 Average call duration :233.321876 Graphite interval statistics (last reported values to graphite): Total calls duration :0.000000 Min managed sessions :0 Max managed sessions :0 Min/Max/Avg offer processing delay :0.000000/0.000000/0.000000 sec Min/Max/Avg answer processing delay :0.000000/0.000000/0.000000 sec Min/Max/Avg delete processing delay :0.000000/0.000000/0.000000 sec Min/Max/Avg offer requests per second :0/0/0 per sec Min/Max/Avg answer requests per second :0/0/0 per sec Min/Max/Avg delete requests per second :0/0/0 per sec Control statistics: Proxy | Offer | Answer | Delete | Ping | List | Query | StartRec | StopRec | Errors | BlkDTMF | UnblkDTMF 127.0.0.1 | 3896 | 3757 | 3499 | 0 | 0 | 0 | 0 | 0 | 0 | 0 | 0 Kind Regards -- Mit freundlichen Grüßen *Karsten Horsmann*

2 1

Redundancy of Progress Call using Kamailio
by Amit Pal 04 Aug '20

04 Aug '20

Hi Team , Hope all is fine. My requirement : I have a customer who wants high availability for ringing calls and calls that in process of being set up. Can you suggest an alternate mechanism to achieve replication of transaction state. Can you guide how it can be achieved in Kamailio or is there a mechanism to get commercial support to achieve the desired objective. I am waiting your response. Thanks Regards Amit Pal

2 3

Kamailio Kubernetes AWS NLB UDP
by Ewgeny Berladin 04 Aug '20

04 Aug '20

Hi ! Does anyone have success Kamailio deployments via service on AWS EKS with UDP port exposing ? apiVersion: v1 kind: Service metadata: name: sip-proxy annotations: service.beta.kubernetes.io/aws-load-balancer-type: "nlb" spec: selector: app: kamailio-sip externalTrafficPolicy: Local type: LoadBalancer ports: - name: sip-up protocol: UDP port: 5060 As result we have this ERROR: ---- ------ ---- ---- ------- Normal EnsuringLoadBalancer 3m20s (x7 over 8m35s) service-controller Ensuring load balancer Warning SyncLoadBalancerFailed 3m20s (x7 over 8m35s) service-controller Error syncing load balancer: failed to ensure load balancer: Only TCP LoadBalancer is supported for AWS ELB As I know this is a BUG in AWS EKS. Maybe there’s some workarounds ? Kubernetes v 1.17 Thanks

2 1

ACC events missing to-tag when doing t_reply
by Julien Chavanton 04 Aug '20

04 Aug '20

Hi, considering this example : modparam("acc_json", "acc_missed_flag", 6) failure_route[test] { setflag(6); // MISSED_ACC, the failure will be recorded by ACC t_flush_flags(); // This will set the flags in the newly created transaction t_reply("500", "Service Unavailable"); } This does generate an ACC event, however the to-tag is not populated, my guess this is because the TM callback is done before the to-tag is generated. I wonder if there is a way to populate the to-tag immediately ? I will dig it further, but maybe there is a trick I am missing

2 2

Kamailio Dockerization
by Edward Romanenco 04 Aug '20

04 Aug '20

Hi, I am working on a project involving Kamailio dockerezation, which is meant to run alongside Freeswitch and RTPEngine containers, on the basis of a Docker-Compose file which is launched on top of a CentOS 7.7 host system. I was able to create and run the containers successfully, they are starting and listening to the correct ports, but for some unexplained reason - the incoming SIP traffic is not getting picked up by Kamailio. I can easily trace the traffic from the host, but when SSHing the container and running a test from within, no traffic goes by. I've used netcat to generate plain UDP traffic to the container, and it was logged into the Kamailio log files, but real-life traffic doesn't seem to work. I've tried moving to host mode (from bridge), but it didn't make any difference. All required firewall rules were opened obviously, I've also tried shutting the firewall off completely but it didn't help. Does anyone experienced anything similar while running Kamailio in Dockers, and could provide me a go-through on what steps did he take to fix it? EXCERPT FROM MY DOCKERFILE # Getting Kamailio source code from GIT RUN mkdir -p /usr/local/src/kamailio-5.3 WORKDIR /usr/local/src/kamailio-5.3 RUN git clone --depth 1 --no-single-branch https://github.com/kamailio/kamailio WORKDIR /usr/local/src/kamailio-5.3/kamailio RUN git checkout -b 5.3.2 # Compile the source code and install Kamailio RUN make include_modules="phonenum db_mysql xmlrpc http_async_client jansson auth_db nathelper websocket tls outbound topoh http_client" cfg && \ make all && make install # Default setting is to run Kamailio as user “kamailio” and group “kamailio” RUN adduser --quiet --system --group --disabled-password \ --shell /bin/false --gecos "Kamailio" \ --home /var/run/kamailio kamailio # To use init.d script for starting/stopping the Kamailio server COPY Init/kamailio /etc/init.d/ RUN chmod 755 /etc/init.d/kamailio COPY Default/kamailio /etc/default/ COPY kamailio.service /etc/systemd/system/ RUN mkdir -p /var/run/kamailio RUN chown kamailio:kamailio /var/run/kamailio COMPOSE (BRIDGE NETWORK VERSION) kamailioegress: build: kamailio_egress image: kamailioegress:latest container_name: kamailioegress restart: always environment: - DATABASE=kamailioe - SIP_DOMAIN=XXX - DBHOST=kamailiodb - DBROOTUSER=root - DBROOTPASS=XXX - PUBLIC_IPV4=XXX depends_on: - Kamailio-Base - kmdb - freeswitch - rtpengine expose: - "5060/udp" - "5060/tcp" ports: - "XXX:5060:5060/udp" networks: private-net: ipv4_address: "172.18.0.30" deploy: mode: replicated replicas: 1 restart_policy: condition: always delay: 5s max_attempts: 3 window: 120s networks: private-net: driver: bridge ipam: config: - subnet: 172.18.0.0/16 driver_opts: com.docker.network.bridge.name: wrtcpriv public-net: external: name: host COMPOSE (HOST MODE VERSION) kamailioegress: build: kamailio_egress image: kamailioegress:latest container_name: kamailioegress network_mode: host restart: always environment: - DATABASE=kamailioe - SIP_DOMAIN=XXX - DBHOST=172.18.0.10 - DBROOTUSER=root - DBROOTPASS=XXX - PUBLIC_IPV4=XXX - EGPORT=5060 - LINTE=ens224 - LINTI=ens192 - RTPENGINE=localhost depends_on: - Kamailio-Base - kmdb - freeswitch - rtpengine expose: - "5060/udp" ports: - "213.8.76.13:5060:5060/udp" deploy: mode: replicated replicas: 1 restart_policy: condition: always delay: 5s max_attempts: 3 window: 120s CONFIG FILE /* uncomment and configure the following line if you want Kamailio to * bind on a specific interface/port/proto (default bind on all available) */ listen=udp:0.0.0.0:LPORT advertise PUBLIC_IP:LPORT KAMAILIO-LOCALE #!define DBURL "mysql://root:XXX@DBHOST/kamailioe" #!substdef "!MY_DBURL!mysql://root:XXX@DBHOST/kamailioe!g" #!substdef "!RTPENGINE!MY_RTPENGINE!g" #!substdef "!SIP_DOMAIN!MY_SIP_DOMAIN!g" #!substdef "!PUBLIC_IP!MY_PUBLIC_IP!g" #!substdef "!PRIVATE_IP!MY_PRIVATE_IP!g" #!substdef "!LPORT!MY_LPORT!g" #!substdef "!LINT!MY_LINT!g" #!substdef "!HOMER_IP!10.1.0.100!g" #!substdef "!API_URL!http://localhost:3000/v1/mock!g" #!substdef "/CCODES/972|380/" #!substdef "/NUM_TRANSLATE_OUT_RE/\\\\+?(CCODES)([0-9]+)/" #!substdef "/NUM_TRANSLATE_IN_RE/0([0-9]+)/" FIREWALL RULES -bash-4.2# firewall-cmd --list-all public (active) target: default icmp-block-inversion: no interfaces: ens192 ens224 sources: 192.168.1.39 services: dhcpv6-client http https sip ssh ports: 9323/tcp 9323/udp protocols: masquerade: no forward-ports: source-ports: icmp-blocks: rich rules: rule family="ipv4" destination address="XXX" port port="5060" protocol="udp" accept rule family="ipv4" destination address="XXX" port port="5060" protocol="tcp" accept NETSTAT OUTPUT Active Internet connections (only servers) Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name udp 0 0 XXX:5060 0.0.0.0:* 22479/kamailio Edward

4 7

No more updates on Official Precise repositories
by Victor Seva 04 Aug '20

04 Aug '20

Hello, Ubuntu Precise is EOL since April 28, 2017 [0] and in order to support newer Ubuntu releases our deb build system will no longer produce newer packages for precise. Repositories will still be there but don't expect any updates. Cheers, Victor [0] https://wiki.ubuntu.com/Releases

2 3

[Help] Kamailio not receive bye requests
by Duy Phan 04 Aug '20

04 Aug '20

Hi guy! I use ec2 ubuntu on aws. But kamailio not receive bye requests. But when I install Kmailio in vultr is OK. can you help me?

4 3

search_append_body?
by Steve Wilkins 04 Aug '20

04 Aug '20

Hello, Does search_append_body require an "\r\n" in the attribute that is being added? If I don't use a "\r\n", it appears to look correct in the trace, however one provider says the attribute that was added was just a continuation of the prior attribute, which Wireshark does not show to be the case. Thank you,

2 1

topoh problem with bye from trunk
by mohsen khashei 04 Aug '20

04 Aug '20

Hi all. I use topoh module for topology hiding and this is my config for topoh: modparam("topoh", "mask_key", "my key") modparam("topoh", "mask_callid", 1) modparam("topoh", "callid_prefix", "!!:") modparam("topoh", "sanity_checks", 1) modparam("topoh", "mask_ip", "x.x.x.x") modparam("topoh", "uri_prefix_checks", 1) modparam("topoh", "uparam_name", "line") modparam("topoh", "uparam_prefix", "sr-") modparam("topoh", "vparam_name", "branch") modparam("topoh", "vparam_prefix", "z9hG4bKsr-") and also i use: modparam("dialog", "lreq_callee_headers", "TH: dlh\r\n") for dialog module. my calls come from asterisk and routed to the carrier(sip trunk) and vice versa, when i make a call from asterisk every thing works fine until the call hanged up from the carrier. at that modemt the kamailio send an 481 call/leg transaction does not exist. if call dropped bye the asterisk user everything works fine. how can i solve this?

3 3

Kamailio-IMS - Public Network Issue
by Pavithra Mohanraja 04 Aug '20

04 Aug '20

Hi, I have deployed kamailio-ims-5.3.5 as docker containers. I had private IP for containers. So when i try to establish a call from the public network , The via header is taken as container Private IP even if i use advertise address as public IP. So ACK is not reaching the pcscf from ue. Any idea can be provided for this issue. Kindly help. Thanks, Pavithra

2 1

Setting up MSFT Teams SBC <> SIP trunk but forwarding ACK and CANCEL messages
by Daniel van der Tang 02 Aug '20

02 Aug '20

Hello, I'm setting up a Kamailio MSFT Teams SBC that is connected to our SIP trunk provider. This server is directly connected to the Internet and not behind a NAT routing. It is successfully processing invites from both sides (MSFT Teams and SIP trunk). Unfortunately the ACKs and CANCEL messages are not relaying to Teams. In the sipdump i can't find these messages. Does anyone see what I am doing wrong? Below you can find my config #!KAMAILIO ####### Defined Values ######### #!define MULTIDOMAIN 0 # - 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 #!define FROM_TEAMS 11 #!define FROM_PBX 12 ######## Define Modules ########### #!define WITH_RTPENGINE #!define WITH_TLS #!define WITH_SIPDUMP #!define WITH_DISPATCH ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR debug=2 log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes */ children=2 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes alias=SBC_FQDN /* listen addresses */ listen=udp:PUBLIC_IP:5060 advertise SBC_FQDN:5060 #!ifdef WITH_TLS listen=tls:PUBLIC_IP:5061 advertise SBC_FQDN:5061 #!endif server_header= "Server: ABC SBC" user_agent_header= "User-Agent: ABC SBC" ###### TLS Enable ###### #!ifdef WITH_TLS enable_tls=yes #tcp_connect_timeout=1000 tcp_accept_no_cl=yes tcp_async = yes tcp_connection_lifetime=600 /* upper limit for TLS connections */ tls_max_connections=2048 #!endif ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ ####### Modules Section ######## /* set paths to location of modules */ 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 "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" #!ifdef WITH_RTPENGINE loadmodule "rtpengine.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_SIPDUMP loadmodule "sipdump.so" #!endif #!ifdef WITH_DISPATCH loadmodule "dispatcher.so" #!endif # ----------------- setting module-specific parameters --------------- #!ifdef WITH_RTPENGINE # ----- rtpengine params -----+ modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:12222") #!endif #!ifdef WITH_TLS # ----- tls params -----+ modparam("tls", "xavp_cfg", "tls") modparam("tls", "config", "/etc/kamailio/tls.cfg") modparam("tls", "connection_timeout", 10) modparam("tls", "ssl_release_buffers", 1) modparam("tls", "send_close_notify", 1) modparam("tls", "session_cache", 0) #!endif #!ifdef WITH_SIPDUMP modparam("sipdump", "enable", 1) #!endif #!ifdef WITH_DISPATCH #---------- dispatch modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_interval", 300) #!endif # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/run/kamailio/kamailio_ctl") # ----- 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) # ----- 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=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks route(REQINIT); #check who is the sender route(INITCHECK); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # update $du to set the destination address for proxying #$du = "sip:" + $rd + ":9"; route(RELAY); exit; } route[INITCHECK] { if(from_uri =~ ".*microsoft.com") { setflag(FROM_TEAMS); $du = "sip:" + "PBX_IP"; route(HANDLE_RTP_FROM_TEAMS); } else if(from_uri =~ ".*" + "PBX_IP") { setflag(FROM_PBX); $du="sip:sip.pstnhub.microsoft.com;transport=tls"; route(HANDLE_RTP_FROM_PBX); } else { exit; } } #Manage RTP & transcoding comming from Teams to PBX route[HANDLE_RTP_FROM_TEAMS] { if (has_body("application/sdp")) { t_on_reply("PBX_REPLY_TO_TEAMS"); rtpengine_manage("RTP codec-mask=all codec-transcode=PCMA replace-origin replace-session-connection ICE=remove"); record_route(); t_relay_to_udp("PBX_IP","5060"); } } #Manage RTP & transcoding comming from PBX to Teams route[HANDLE_RTP_FROM_PBX] { if (has_body("application/sdp")) { t_on_reply("TEAMS_REPLY_TO_PBX"); rtpengine_manage("SRTP codec-mask=all ICE=force codec-transcode=PCMA replace-origin replace-session-connection"); record_route_preset("SBC_FQDN:5061;transport=tls"); add_rr_param(";r2=on"); $rd = "sip.pstnhub.microsoft.com"; $td = "SBC_FQDN"; $fd = "SBC_FQDN"; #Set TLS SNI (server name & server id) $xavp(tls=>server_name) = "SBC_FQDN"; $xavp(tls=>server_id) = "SBC_FQDN"; t_relay(); } } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS")) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; #Teams reINVITEs if(isflagset(FROM_TEAMS)) { t_relay_to_udp("PBX_IP","5060"); exit; } # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails #set coresponding cert on transactions if($fd == "SBC_FQDN") { $xavp(tls=>server_name) = "SBC_FQDN"; $xavp(tls=>server_id) = "SBC_FQDN"; } } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); } #PBX On Reply onreply_route[PBX_REPLY_TO_TEAMS] { if (has_body("application/sdp")) { rtpengine_manage("SRTP codec-mask=all codec-transcode=PCMA replace-origin replace-session-connection media-address=PUBLIC_IP"); } } #From Teams On Reply onreply_route[TEAMS_REPLY_TO_PBX] { if (has_body("application/sdp")) { rtpengine_manage("RTP codec-mask=all codec-transcode=PCMA replace-origin replace-session-connection media-address=PUBLIC_IP"); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { if (t_is_canceled()) exit; } event_route[tm:local-request] { if(is_method("OPTIONS") && $ru =~ "pstnhub.microsoft.com") { append_hf("Contact: <sip:SBC_FQDN:5061;transport=tls>\r\n"); } xlog("L_INFO", "Sent out tm request: $mb\n"); }

6 5

Manipulating SDP IP for Inbound Calls
by Edward Romanenco 02 Aug '20

02 Aug '20

Hey guys, I am working on a project involving Kamailio dockerezation, which is meant to run alongside Freeswitch and RTPEngine containers, on the basis of a Docker-Compose file which is launched on top of a CentOS 7.7 host system. Anyway, I would love to know if there is any way to manipulate / mask the IP addresses that are being appended to a status 183 response for an incoming invite. For some reason which I am trying to figure out in parallel, Freeswitch uses the local network bridge subnet instead of the defined external RTP IPs, and I was wondering - Can I manipulate them using Kamailio? I know that Mangler module can do it for outbound calls, but can I do the same for inbound? v=0. o=FreeSWITCH 1595974788 1595974789 IN IP4 172.18.0.40. s=FreeSWITCH. c=IN IP4 172.18.0.40. t=0 0. m=audio 45878 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16 Edward

2 1

lua_dostring sr vs KSR
by Julien Chavanton 31 Jul '20

31 Jul '20

lua_dostring("sr.pv.seti(\"$$var(var1)\", $avp(var1) < $avp(var3) and 1 or 0)"); results in error : kamailio app lual_dostring(): error from Lua: attempt to index global 'sr' (a nil value) replacing sr with KSR is working : lua_dostring("KSR.pv.seti(\"$$var(var1)\", $avp(var1) < $avp(var3) and 1 or 0)"); Shall we update the docs for https://kamailio.org/docs/modules/devel/modules/app_lua.html#app_lua.f.lua_… ?

2 4

Issue with kamctl after upgrading to 5.3.4
by Igor Potjevlesh 31 Jul '20

31 Jul '20

Hello there! I'm having trouble with kamctl since I upgraded to 5.3.4. The command is working fine with root user but not with another user. The problem seems to be that the command try to access to /var/run/kamailio/kamailio_rpc.fifo even if the file exists. I guess it's a permission problem. But I was thinking that FIFO is no more used. Only jsonrpcs. How can I fix that? Regards, Igor. -- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus

3 11

publishing hash values for download files of releases
by Daniel-Constantin Mierla 30 Jul '20

30 Jul '20

Hello, being discussed during the last devel meetings, I published the md5, sha1 and sha256 hash values for the tarballs with sources and i386 binaries we make available for download on kamailio.org on each release -- e.g., for 5.4.0: * https://www.kamailio.org/pub/kamailio/5.4.0/src/ * https://www.kamailio.org/pub/kamailio/5.4.0/bin/ Before making a more official announcement about it and adding to the download/install docs, I want to discuss a little bit here and get to the right solution to publish these hash values. For the moment I put them in a single file, adding -checksums.txt to the tarball name, listing inside all 3 hashes as computed by md5sum, sha1sum and sha256sum. That because I couldn't decide alone if there is sort of a standard on how to do it. Couple of projects I checked they just list the hash values on the html page with the link to download file. Others have dedicated files per hashing type, named like MD5SUMS, SHA1SUMS and SHA256SUMS, containing hash values for all downloadable files in the folder. Then, asterisk projects publishes 3 files, asterisk-VERSION.{md5,sha1,sha256}, corresponding to the tar.gz file they made available. Freeswitch publishes more than one archive file type, so it makes available files like freeswitch-VERSION.EXT.{md5,sha1,sha256}, where EXT can be tar.gz, tar.xz, zip ... My questions now. What kind of files with hash values people here are used with? Any variants that tends to be (or become the standard)? Any tools you are aware of for automatically checking the integrity with one of these specific hash files (like, if I have the tarball and the hashes file in the same folder and run it, it gives the ok/not-ok, without me having to do md5/sha1/sha256 manually and check "by eye" the values)? Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

2 2

Kamailio v5.4.0 Released - new major version is out
by Daniel-Constantin Mierla 29 Jul '20

29 Jul '20

Kamailio v5.4.0 is out – it comes with 5 new modules and a considerable set of improvements touching again more than 100 existing modules. You can read detailed release notes at: * https://www.kamailio.org/w/kamailio-v5-4-0-release-notes/ Many thanks to all developers and community members that made possible this release. v5.4.0 brings more flexibility and optimizations across many components, enhancements to load balancer, stateless active calls tracking, STIR/SHAKEN support, Kafka connector, plenty of new variables and transformations, ability do print JSON-structured log messages ... just to list only a few here. Enjoy Kamailio v5.4.0! Thank you for flying Kamailio! Stay safe and healthy! Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

The plan for releasing Kamailio v5.4.0
by Daniel-Constantin Mierla 29 Jul '20

29 Jul '20

Hello, we are considering to release the next major stable release, respectively 5.4.0, on Wednesday, July 29, 2020. The branch 5.4 was already created, several doc resources were already added (wiki pages with what is new, core/variables/transformations/rpcs/stats cookbooks, alphabetic indexes, ...), at least debian/ubuntu packages should be built nightly, ... If anyone is testing an upgrade from branch 5.3 to 5.4 and encounters changes that should be done in Kamailio config, add notes about at: * https://www.kamailio.org/wiki/install/upgrade/5.3.x-to-5.4.0 Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 2

PHONENUM Module on Centos7
by Duarte Rocha 28 Jul '20

28 Jul '20

Greetings, I've trying to use the module PHONENUM on my Kamailio on CentOS 7. However, getting the required libphonenumber library hasn't been possible. Is there any CentOs7 repository for this ? What's the best way to work with this module on CentOS ? Best Regards,

2 1

Kamailio IMS with custom AVPs
by Gaston Siffert 28 Jul '20

28 Jul '20

Hello all, I am investigating on how to integrate few IMS modules in the Platform I am working on. At the moment the main drawback for me is to find a way to export data from the Diameter Answers. In the RFC 7155, it’s written that the AA-Answer could have a list of customisable fields. While looking at the implementation of the Rx_AAR (IMS QOS) in Kamailio, the module does not seems to export any of the fields from the AA-Answer except the return code. The same applies for the implementation of I_perform_location_information_request (IMS ICSCF) in Kamailio. Do you have any recommendation on how to export the other AVPs from the Diameter Answers ? Instead of IMS QOS and IMS ICSCF, I could use diameter_request_async (IMS Diameter Server) to make my own AAR and LIR, but is it the recommended practice ? https://tools.ietf.org/html/rfc7155#section-3.2 <https://tools.ietf.org/html/rfc7155#section-3.2> https://kamailio.org/docs/modules/5.0.x/modules/ims_qos.html#idp18760852 <https://kamailio.org/docs/modules/5.0.x/modules/ims_qos.html#idp18760852> https://kamailio.org/docs/modules/4.4.x/modules/ims_icscf.html#idp39432196 <https://kamailio.org/docs/modules/4.4.x/modules/ims_icscf.html#idp39432196> https://www.kamailio.org/docs/modules/devel/modules/ims_diameter_server.htm… <https://www.kamailio.org/docs/modules/devel/modules/ims_diameter_server.htm…> Best regards, Gaston Siffert

1 0

PHONENUM Module on Centos7
by Ilie Soltanici 28 Jul '20

28 Jul '20

Hello, If you want to use c++ library, then you can try to compile it by yourself: yum install cmake boost boost-devel boost-doc gtest gtest-devel protobuf protobuf-devel libicu libicu-devel $ cd libphonenumber/cpp $ mkdir build $ cd build $ cmake .. $ make $ ./libphonenumber_test cp -r libphonenumber.so* /usr/lib64/ cp -r libgeocoding.so* /usr/lib64/

1 0

ERROR: kazoo [kz_pua.c:131]: kz_pua_update_presentity(): in start_transaction
by Denys Pozniak 28 Jul '20

28 Jul '20

Hello! Sometimes presence server stops sending SIP NOTIFY events with syslog output below (after Kamailio restarting everything is restored): Jul 27 17:07:28 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: db_mysql [km_dbase.c:602]: db_mysql_start_transaction(): transaction already started Jul 27 17:07:28 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: kazoo [kz_pua.c:131]: kz_pua_update_presentity(): in start_transaction Jul 27 17:07:29 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: db_mysql [km_dbase.c:602]: db_mysql_start_transaction(): transaction already started Jul 27 17:07:29 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: kazoo [kz_pua.c:131]: kz_pua_update_presentity(): in start_transaction Jul 27 17:07:37 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: db_mysql [km_dbase.c:602]: db_mysql_start_transaction(): transaction already started Jul 27 17:07:37 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: kazoo [kz_pua.c:131]: kz_pua_update_presentity(): in start_transaction Jul 27 17:07:37 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: db_mysql [km_dbase.c:602]: db_mysql_start_transaction(): transaction already started Jul 27 17:07:37 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: kazoo [kz_pua.c:131]: kz_pua_update_presentity(): in start_transaction Jul 27 17:07:41 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: db_mysql [km_dbase.c:602]: db_mysql_start_transaction(): transaction already started Jul 27 17:07:41 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: kazoo [kz_pua.c:131]: kz_pua_update_presentity(): in start_transaction Jul 27 17:07:41 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: db_mysql [km_dbase.c:602]: db_mysql_start_transaction(): transaction already started Jul 27 17:07:41 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: kazoo [kz_pua.c:131]: kz_pua_update_presentity(): in start_transaction Where is a possible source of the problem? Kamailio 5.1.2 with MySQL modparam("presence", "db_url", DBURL) modparam("presence", "subs_db_mode", 3) modparam("presence", "timeout_rm_subs", 0) modparam("presence", "expires_offset", 0) modparam("presence", "max_expires", 1800) modparam("presence", "db_update_period", 30) modparam("presence", "clean_period", 180) modparam("presence", "send_fast_notify", 1) modparam("presence", "pres_htable_size", 32) modparam("presence", "subs_htable_size", 32) modparam("presence", "publ_cache", 0) modparam("presence", "notifier_processes", 0) modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) modparam("presence_xml", "integrated_xcap_server", 0) modparam("presence_xml", "disable_presence", 1) modparam("presence_xml", "disable_winfo", 1) modparam("presence_dialoginfo", "force_single_dialog", 1) modparam("presence_dialoginfo", "force_dummy_dialog", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_match_mode", 1) modparam("kazoo", "node_hostname", NODE_HOSTNAME) modparam("kazoo", "amqp_connection", "AMQP_URL") modparam("kazoo", "pua_mode", 1) modparam("kazoo", "db_url", DBURL) modparam("kazoo", "amqp_consumer_processes", 10) modparam("kazoo", "amqp_max_channels", 50) -- BR, Denys Pozniak

2 1

Kamailio CentOS 7 repos broken
by Patrick Wakano 28 Jul '20

28 Jul '20

Hi list, Hope you are all good! I've been trying to install kamailio 5.2.7 from the centos 7 repo, but it is failing in both repos..... With the http://rpm.kamailio.org/centos/ repo it fails with: Error: Package: kamailio-sipcapture-daemon-config-5.2.7-0.el7.centos.x86_64 (kamailio-5.2) Requires: kamailio-sipcapture = 5.2.7 With the http://download.opensuse.org/repositories/home:/kamailio:/v5.2.x-rpms/CentO… repo it fails with: kamailio-5.2.7-7.el7.centos.x8 FAILED =================== ] 579 kB/s | 12 MB 00:00:04 ETA https://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.2.x-rpms/…: [Errno -1] Package does not match intended download. Suggestion: run yum --enablerepo=home_kamailio_v5.2.x-rpms clean metadata Trying other mirror. Error downloading packages: kamailio-5.2.7-7.el7.centos.x86_64: [Errno 256] No more mirrors to try. Would someone be able to have a look at that? Thank you! Patrick Wakano

2 1

Registrar - registered match_received
by Ilie Soltanici 27 Jul '20

27 Jul '20

Hello, Yes, i have it there but the variable name was mismatching with the module variable name: $xavp(regcfg=>match_received) = $su; modparam("registrar", "xavp_cfg", "reg") It's all good now. Thanks for your help.

1 0

Registrar - registered match_received
by Ilie Soltanici 27 Jul '20

27 Jul '20

Hello, Trying to check if an extension it's already registered with the same received path. if (registered("location","$fu",2)) { sl_send_reply("200","OK"); exit; }; But, it's not working for me, getting true even if the received path it's different: This is what I'm getting from Kamailio: [root@dev ilie.soltanici]# /usr/local/sbin/kamcmd ul.dump { Domains: { Domain: { Domain: location Size: 1024 AoRs: { Info: { AoR: 101 HashID: -1698832128 Contacts: { Contact: { Address: sip:101@192.168.1.10:12767;ob Expires: 2452 Q: 1.000000 Call-ID: 3fa6c9f7db404e8e93a9664c3634ddf7 CSeq: 31114 User-Agent: MicroSIP/3.19.31 Received: sip: 192.168.1.10:12767 Path: [not set] State: CS_SYNC Flags: 0 CFlags: 12288 Socket: udp: 192.168.1.1:5060 Methods: 8159 Ruid: uloc-b-5f1eea5d-57b0-1 Instance: [not set] Reg-Id: 0 Server-Id: 11 Tcpconn-Id: -1 Keepalive: 1 Last-Keepalive: 1595862742 Last-Modified: 1595861637 } } This is the SIP Trace: 2020/07/27 16:17:12.214151 192.168.1.10:12610 -> 192.168.1.1:5060 OPTIONS sip:dev.local SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK-91-1-0 From: "Test Account" <sip:101@dev.local>;tag=91SIPpTag001 To: sip:dev.local Call-ID: 1-91(a)192.168.1.10 User-Agent: SIPP CSeq: 1 OPTIONS Content-Length: 0 As you can see the source port it's different 12767/12610, but the registered function is still getting true. Thank you.

2 1

Install 5.3 on Ubuntu 20.04 With MySQL 8
by Mark Boyce 27 Jul '20

27 Jul '20

Hi all This is installing 5.3 for Bionic (18.04) on Focal (20.04) but as I know you’re in the throws of building 5.4 thought I'd throw this out there in case it changes anything; Installing kamailio from; deb http://deb.kamailio.org/kamailio53 bionic main The following packages have unmet dependencies: kamailio : Depends: libreadline7 (>= 6.0) but it is not installable kamailio-mysql-modules : Depends: libmysqlclient20 (>= 5.7.11) but it is not installable kamailio-redis-modules : Depends: libhiredis0.13 (>= 0.13.1) but it is not installable libreadline7 -> libreadline8 (v 8.0-4) libmysqlclient20 -> libmysqlclient21 (v 8.0.21-1ubuntu20.04) libhiredis0.13 -> libhiredis-dev (v 0.14.0-6). Ubuntu Server 20.04 LTS (focal) MySQL 8 (From MySQL Repo) Redis 6 from latest stable Best regards Mark -- Mark Boyce Dark Origins Ltd

2 6

Using GeoIP2 AS Lookup?
by Mark Boyce 27 Jul '20

27 Jul '20

Hi all Just looking at the latest GeoIP2 MaxMind databases (now requires registration, but still free) and noticed that they also include the AS (ISP) lookup one in the free offering. Wondering if this is another way to facilitate better security for users on dynamic IP. Typically working from home these days. So, rather than just limiting an end device to a country we could limit it to a particular ISP within that country. Has anyone tried this? Have I missed a reason why this wouldn’t help? Admin overhead not worth it? Thoughts? Best regards Mark -- Mark Boyce Dark Origins Ltd

2 5

kamailio 5.3.5 - none local route header
by Jonathan Hunter 27 Jul '20

27 Jul '20

Hi Guys, Sorry if this is too generic or silly a question but I am testing kamailio 5.3.5 within Azure and its behind NAT. I am having an issue with passing an ACK from our internal side back out to the carrier side and it appears having run loose_route a none local route-header is consistently being removed by kamailio at this point. ie Route: <sip:CARRIER_IP;lr> Is this normal or configuration related? As I am unsure within what scenario I would remove a route header which doesnt contain the local Proxy IP. Please can someone advise? Many thanks Jon Sent from Mail<https://go.microsoft.com/fwlink/?LinkId=550986> for Windows 10

2 3

Redis error: Server closed the connection / nbd_redis / kamailio 5.3.5
by Karsten Horsmann 27 Jul '20

27 Jul '20

Hi everyone, I get some strange errors from nbd_redis but my setup is working. kamailio 5.3.5 on CentOS 7.7.1908. Redis-Server is an haproxy frontend that pushed the stuff to an redis-sentinel cluster (with and really old version redis-3.2.1-2.el7.remi.x86_64). Any ideas on that? Jul 22 13:42:52 siptrunk5 kamailio[112377]: ERROR: ndb_redis [redis_client.c:1083]: redisc_exec_argv(): Redis error: Server closed the connection Jul 22 13:43:34 siptrunk5 kamailio[112359]: ERROR: ndb_redis [redis_client.c:1083]: redisc_exec_argv(): Redis error: Server closed the connection Jul 22 13:43:34 siptrunk5 kamailio[112378]: ERROR: ndb_redis [redis_client.c:1083]: redisc_exec_argv(): Redis error: Server closed the connection Jul 22 13:44:33 siptrunk5 kamailio[112363]: ERROR: ndb_redis [redis_client.c:1083]: redisc_exec_argv(): Redis error: Server closed the connection Jul 22 13:45:03 siptrunk5 kamailio[112362]: ERROR: ndb_redis [redis_client.c:1083]: redisc_exec_argv(): Redis error: Server closed the connection Jul 22 14:00:01 siptrunk5 kamailio[112381]: ERROR: ndb_redis [redis_client.c:1083]: redisc_exec_argv(): Redis error: Server closed the connection Jul 22 14:00:01 siptrunk5 kamailio[112365]: ERROR: {24519 CANCEL 59644df6-68ac-4dd6-9fa5-1a9c2eda3b59} ndb_redis [redis_client.c:1083]: redisc_exec_argv(): Redis error: Server closed the connection # grep redis /etc/kamailio/kamailio.cfg loadmodule "ndb_redis.so" loadmodule "topos_redis.so" modparam("ndb_redis", "server", "name=CFG_REDIS_NAME;addr=CFG_REDIS_ADDR;port=6379;db=CFG_REDIS_DB") modparam("ndb_redis", "connect_timeout", 1500) # default 1000ms/1sec modparam("ndb_redis", "cmd_timeout", 1500) # default 1000ms/1sec modparam("topos", "storage", "redis") modparam("topos_redis", "serverid", "CFG_REDIS_NAME") -- Cheers *Karsten Horsmann*

3 2

failed to lock the memory pages (disable swap) WARNING
by Joel Serrano 27 Jul '20

27 Jul '20

Hi all, Can anyone give me some info on what this warning means: Jul 7 17:02:29 csbc02 csbc[16006]: WARNING: <core> [core/daemonize.c:596]: mem_lock_pages(): failed to lock the memory pages (disable swap): Cannot allocate memory [12] Yes, if I set mlock_pages=no I don't see the warning, but I'm trying to understand why can't it allocate with mlock_pages=yes. Joel.

4 4

Kamailio Installation Steps
by Abdul Rasheed 26 Jul '20

26 Jul '20

Hello All , I am trying to setup Kamailio on Debian . May i know which document should be followed for installing the latest version of Kamailio on Debian 10 Buster. ? Thanks and Regards Abdul Rasheed

2 2

Dialplan: using $avp("key") on subst rule with regexp capturing group
by João Vitor Arruda 24 Jul '20

24 Jul '20

Hello, I'm trying to use the dialplan with $avp("key") on match_exp and subst_exp but i'm getting an error message when there is a regex capturing group on subst_exp to be used on repl_exp This is the config used: modparam("dialplan", "match_dynamic", 1) $avp(s:PSTNAC) = "0"; Dialplan entry: dpid = 1 match_op = 1 match_exp = ^$avp(s:PSTNAC) match_len = 0 subst_exp = ^$avp(s:PSTNAC)(.*) repl_exp = 9\1 After executing dp_translate while the dialplan is properly matched the following error message occurs and the repl_exp didn't work as expected: ERROR: dialplan [dp_repl.c:422]: rule_translate(): illegal access to 1-th subexpr of subst expr (max 0) I've tried some other dialplan entries like: dpid = 1 match_op = 1 match_exp = ^$(avp(s:PSTNAC)[*]) match_len = 0 subst_exp = ^$(avp(s:PSTNAC)[*])(.*) repl_exp = 9\1 But that resulted in the same error message and no replacement either. Any suggestion on what I'm doing wrong here? Thanks, Joao Arruda

1 2

how to achive destination number based routing before falling back to dispatcher-module
by Christoph Russow 24 Jul '20

24 Jul '20

Hi all, im fairly new to Kamailio and currently setting up our new sip-server systems so, hello everyone! right now i have a Kamailio SBC which routes all calls using the dispatcher module to our internally setup application servers (freeswitch). i just got the idea that we might want specific routing based on destination phone number in addition to simple loadbalancing. can anyone point me in the right direction where i should look for something i can place before the dispatcher routing to route calls based on some entries in a database like destination number XYZ goes to server A, number ABC goes to server Z and so on. Or is there a way to select the dispatcher group id from the database based on destination number? best regards Christoph

4 9

HEP siptrace HEPv3 truncated and ignored by heplify-server (Homer 7)
by José Lopes 24 Jul '20

24 Jul '20

Hello all, I am using the siptrace module on Kamailio 5.3.5 and I am trying to monitor the SIP over websockets traffic with Homer 7.8.1 (heplify-server). I notice a strange behavior, some packets are not presented on Homer. I checked heplify-server and i notice the next warning: heplify-server: 2020/07/22 18:41:09.425513 decoder.go:103: WARN HEP packet length is 8192 but should be 11663 So I think the packets are truncated by 8192 (packet size), so they are ignored by heplify-server. I have done a tcpdump on the Homer machine and have the next output, where I can see some packets with size greater than 8192. 19:58:29.788927 IP X.X.X.X.6666 > Y.Y.Y.Y.9060: UDP, bad length 11658 > 1472 19:58:29.793539 IP X.X.X.X.6666 > Y.Y.Y.Y.9060: UDP, length 626 19:58:29.810733 IP X.X.X.X.6666 > Y.Y.Y.Y.9060: UDP, length 579 19:58:29.811121 IP X.X.X.X.6666 > Y.Y.Y.Y.9060: UDP, length 579 19:58:29.838824 IP X.X.X.X.6666 > Y.Y.Y.Y.9060: UDP, bad length 11953 > 1472 19:58:29.852286 IP X.X.X.X.6666 > Y.Y.Y.Y.9060: UDP, length 473 19:58:29.865388 IP X.X.X.X.6666 > Y.Y.Y.Y.9060: UDP, bad length 3521 > 1472 19:58:29.875311 IP X.X.X.X.6666 > Y.Y.Y.Y.9060: UDP, length 562 Can you guide me to identify where the packets are truncated? Os melhores cumprimentos / Best regards, *José Lopes* Research and Development Technician Phone: +351 256 370 980 Email: jose.lopes(a)itcenter.com

2 2

new in upcoming Kamailio 5.4.0
by Henning Westerholt 23 Jul '20

23 Jul '20

Hello, I've updated the "New in version 5.4.0" document in our wiki. It is again an impressive list! * In this development period (started October 2019) almost 1500 commits have been added to the repository * 5 new modules have been added (pv_headers, kafka, secsipid, systemdops, dlgs) * Over 70 modules have been extended, many of them substantially (dispatcher, pv, tls, presence etc..) * Countless bugfixes and documentation extensions/fixes have been added as well * Many refactorings, extensions and architectural improvements were done in Kamailio core, tooling and modules You'll find the full details at the usual place: https://www.kamailio.org/wiki/features/new-in-5.4.x Many thanks to everybody that contributed to the Kamailio project. Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

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