- sr-users - kamailio.org

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

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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

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