- sr-users - kamailio.org

SL absorbed ACK is counted as dropped by core.drop_requests
by Kaufman 03 Jul '23

03 Jul '23

Just checking to see if this is the designed behavior. The SL module will attempt to match ACKs for stateless replies and handle them (absorb them?). The question I have is that it looks as though this ACK gets counted as dropped by the core.drop_requests from the KEX module. Tested using this config: #!KAMAILIO loadmodule "pv" loadmodule "sl" loadmodule "xlog" loadmodule "kex" loadmodule "corex" loadmodule "ctl" modparam("sl", "bind_tm", 0) route { xinfo("[$ci] $rm Request. Src:[$si:$sp] RURI:[$ru] To:[$tu] From:[$fu]\n"); sl_send_reply("404", "Not Found"); } event_route[sl:filtered-ack] { xnotice("sl:filtered-ack ACK [$ci] to local reply absorbed\n"); } Then validate by kamcmd stats.fetch core:drop_requests Is this the designed and "correct" behavior? Kaufman

3 4

Kamailio v5.7.1 Released
by Daniel-Constantin Mierla 28 Jun '23

28 Jun '23

Hello, Kamailio SIP Server v5.7.1 stable release is out. This is a maintenance release of the latest stable branch, 5.7, that includes fixes since the release of v5.7.0. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.7.x. Deployments running previous v5.7.x versions are strongly recommended to be upgraded to v5.7.1. For more details about version 5.7.1 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2023/06/kamailio-v5-7-1-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 Kamailio World Conference - www.kamailioworld.com

1 0

Forward Registration
by nutxase 28 Jun '23

28 Jun '23

Hi All I have setup kamailio using websockets(wss) i am trying to forward any registration that comes to kamailio to go to asterisk, so in theory kamailio is a registration proxy i have set rewritehostport("xxx.xxx.xxx[:5080](http://node3.sipalto.com:5080)"); the registration forwards fine but the contact on asterisk side shows as sip:xxx:kamailioip.xxxx.invalid so therefore asterisk sees the sip peer as offline what am i missing :( Sent with [Proton Mail](https://proton.me/) secure email.

1 0

Releasing v5.7.1
by Daniel-Constantin Mierla 28 Jun '23

28 Jun '23

Hello, I am considering to release Kamailio v5.7.1 (out of branch 5.7) this week (likely on Wednesday or Thursday, Jun 28/29, 2023). If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - www.kamailioworld.com

1 1

Matching the From Username in LCR rule table
by harneet singh 27 Jun '23

27 Jun '23

Hi Experts, We wished to used the LCR Module to be able to select the Gateway(s) to send the calls to based on the From(Caller) Username. Based on the LCR Documentation, the usage of other fields like request URI or prefix etc is pretty clear. But for the From Usename I am having certain trouble understanding the same. It refers to some mtree. Excerpt below: "A gateway matches a request if user part of Request-URI matches "prefix", caller URI matches "from_uri" pattern,* caller URI userpart matches mtree with "mt_value"*, and callee URI matches "request_uri" pattern in a tuple that is associated with the gateway." Can someone explain/ share a sample lcr_rule table entries for the From Username that can be matched against the incoming sip message for gw selection? Do we need to manually create an mtree by the name "*lcr*". Asking based on the below section in LCR documentation? Any pointers are highly appreciated. 3.35. mtree (string) Name of mtree to which load_gws() matches caller URI userpart. *Default value is "lcr".* Regards, Harneet Singh -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

2 3

uac_replace() and uac_restore debug help
by Duarte Rocha 24 Jun '23

24 Jun '23

Hello, I'm currently experiencing some odd behaviours when using these functions and I need some help debugging to find the root of the issue. I receive a call from a client and do some changes to the To and From header using auto mode with password. Encrypted info goes on the Record-Route header parameters (vsf\vst) normally. 1 - On subsequent requests sent by the originating client, domains on To and From header are different from the ones sent in the first INVITE. In this scenario, somehow the automatic modifying has errors on the end result. Can the different domains affect the end result of the automatic modifying or only the vsf\vst parameters and password are needed to the auto replace\restore? 2 - How can I decrypt the info on vsf and vst parameters in order to verify the info sent and the encryption? Cheers, Duarte Rocha

2 1

uac_restore behaviour question
by Duarte Rocha 22 Jun '23

22 Jun '23

Hello, I have some questions regarding uac_replace methods. When I receive the first INVITE from someone I change To and From using uac_replace_to() and uac_replace_from(). I do this only on first INVITES. In order to restore them I use the "auto" so the changes are restored automatically in replies or requests sent to the call originator.. In the case of subsequent requests from the originator kamailio still changes To and From without me using uac_replace functions. Is this caused by the auto restore? Does it work for both sides and is this is the intended behaviour? Cheers

1 1

How to handle 302 redirects
by Trey Ormsbee 22 Jun '23

22 Jun '23

Hello, Im having a hard time figuring out how to handle a 302 reidrect, the uac_redirect modules shows a simple configuration. However for me, what it is not doing is changing the 302 back to invite. It is sending the 302 as is on. if(!t_check_status("3[0-9][0-9]")) { exit; } get_redirects("3:1"); t_relay(); This code just relays the 302 for me. I should add I am using this with the DISPATCHER module so perhaps that has something to do with my issue. Setup is this, kamailio is sitting between to sip servers, so the flow goes like this, sip_server_a -> kamailio -> sip_server_b. sip_server_b only responds with 302 redirects, sip_server_a does not handle 302’s at all. Kamailio will only get invites from sip_server_a, and the redirects from sip_server_b will only have one contact. How can I get kamailio to reply to sip_server_a’s invite with the updated INVITE? My config is a mesh of the dispatcher example and uac_redirect example with this specific section handling the redirect: # Sample failure route failure_route[RTF_DISPATCH] { xlog("L_NOTICE", "RTF_DISPATCH: "); if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } if (t_check_status("302")) { xlog("L_NOTICE", "RTF_DISPATCH: 302 "); if (!get_redirects("1:1")) { xlog("L_NOTICE", "RTF_DISPATCH: failed to get_redirects "); t_reply("480", "Temporarily Unavailable"); exit; } xlog("L_NOTICE", "RTF_DISPATCH: got redirects "); t_relay(“sip_server_a”,”5060”); } } Any ideas what I am doing wrong is this supported? Im expecting once I relay this message that it should be an INVITE not a 302 redirect. I cant find any examples of anyone else really doing this.

3 3

Missing posops module from 5.7 debian repo
by Patrick Karton 20 Jun '23

20 Jun '23

hello , i have installed kamailio 5.7 from debian repo http://deb.kamailio.org/kamailio57 bullseye main but i dont find the posops module in the directory modules. what is the exactly package where i can find it. Thanks.

2 5

KEMI
by iliusha.md＠gmail.com 19 Jun '23

19 Jun '23

Hello, I have the following statement: if KSR.is_method("INVITE") then KSR.log("info", "method not allowed " .. KSR.kx.get_method() .."\n"); KSR.x.exit(); end For some reason, it falls into this condition no matter which method I use, I tried OPTIONS, NOTIFY etc. log file: <core> [core/kemi.c:152]: sr_kemi_core_log(): method not allowed OPTIONS the same with this check: if KSR.is_method_in("I") then KSR.log("info", "method not allowed " .. KSR.kx.get_method() .."\n"); KSR.x.exit(); end this is the lua modparam settings: # ---- lua params ---- modparam("app_lua", "load", "/usr/local/etc/kamailio/main.lua") modparam("app_lua", "reload", 1) modparam("app_lua", "log_mode", 1) cfgengine "lua" version: kamailio 5.7.0 (x86_64/linux) 1c23c2 Any idea what I'm doing wrong?

2 1

Kamailio failover in aws/azure environment
by M Arqum CH 17 Jun '23

17 Jun '23

Hi members, I am setting up kamailio as loadbalcer for asterisk server farm with active-backup conf in Azure cloud using azure loadbalcer, for incoming it works fine external incoming traffic with nated external IP, can any one guide how to do active-backup configuration for internal traffic, as asterisk servers are running on local network. Thanks and regards Arqum

2 1

TLS errors
by nutxase 16 Jun '23

16 Jun '23

Hey All I have been setting up TLS and websocket for kamailio but i keep getting these errors in the log please can someone help me Jun 14 20:51:57 so-sbc-02 /usr/sbin/kamailio[25490]: ERROR: tls [tls_util.h:51]: tls_err_ret(): TLS accept:error:1408A0C1:SSL routines:ssl3_get_client_hello:no shared cipher (sni: [sip.domain.com](http://sip.sipalto.com)) Jun 14 20:51:57 so-sbc-02 /usr/sbin/kamailio[25490]: ERROR: tls [tls_server.c:1333]: tls_h_read_f(): src addr: 1.2.3.4.5:57265 Jun 14 20:51:57 so-sbc-02 /usr/sbin/kamailio[25490]: ERROR: tls [tls_server.c:1336]: tls_h_read_f(): dst addr: 5.4.3.2.1:8089Jun 14 20:51:57 so-sbc-02 /usr/sbin/kamailio[25490]: ERROR: <core> [core/tcp_read.c:1478]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f3c8fc16720 r: 0x7f3c8fc16848 (-1) my config is pretty standard so i am not sure what i am missing using letsencrypt certs #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "tls_method", "TLSv1.2+") modparam("tls", "certificate", "/etc/kamailio/tls/cert.pem") modparam("tls", "private_key", "/etc/kamailio/tls/ckey.pem") modparam("tls", "ca_list", "/etc/kamailio/tls/fullchain.pem")#!endif Sent with [Proton Mail](https://proton.me/) secure email.

3 5

impossible to modify $tU twice
by David VILLAUME 15 Jun '23

15 Jun '23

Hello, I’m in a scenario where i would apply different modification to $tU to User with KEMI, but I can’t achieve it. I managed to reproduce it with legacy config : Scenario 1 (both values concatenates): route[dbg] { $tU ="aaa"; $tU ="bbb"; } To: sip:aaabbb@5.5.5.5 Scenario 1 (both values concatenates and @ disapears): route[dbg] { $tU ="aaa"; $tU=$null; $tU ="bbb"; } To: sip:aaabbb5.5.5.5 --->without @ I tried different combination with [*] as for avps without success, does anyone have an idea about how to perform it ? Regards, David

3 2

Cannot achieve rtpengine HA
by Arsenijs Kabihno 13 Jun '23

13 Jun '23

Hello, We are trying to achieve rtpengine HA solely with kamailio and rtpengine module. In the following example call goes to rtpengine node 10.5.0.183:22222 and then we stop rtpengine service on this node to simulate failure. Client is triggering re-INVITE, but, apparently, kamailio sticks to initially selected rtpengine, producing the following errors: Jun 12 06:59:48 ap07stvp.xxxxxxx.xxx docker[9373]: 27(33) INFO: {2 2 INVITE jdl9kq6fv51jcv1agk2e} <script>: RTPENGINE branch_id:0 ruri: <null>, method:INVITE, status:200, extra_id: z9hG4bK55018600, rtpengine_answer: replace-origin replace-session-connection SIP-source-address direction=priv1 direction=pub1 SIP-source-address via-branch=extra rtcp-mux-offer generate-mid DTLS=passive SDES-off ICE=force RTP/SAVPF Jun 12 06:59:49 ap07stvp.xxxxxxx.xxx docker[9373]: 27(33) NOTICE: {2 2 INVITE jdl9kq6fv51jcv1agk2e} acc [acc.c:287]: acc_log_request(): ACC: transaction answered: timestamp=1686553189;method=INVITE;from_tag=fcn109nfe9;to_tag=01284cc8-bc58-4104-9629-8e6bb827d97f;call_id=jdl9kq6fv51jcv1agk2e;code=200;reason=OK;src_user=90362;src_domain=10.181.0.1;src_ip=10.5.0.145;dst_ouser=123;dst_user=123;dst_domain=10.181.0.1 Jun 12 06:59:49 ap07stvp.xxxxxxx.xxx docker[9373]: 31(37) INFO: {2 2 INVITE jdl9kq6fv51jcv1agk2e} <script>: RTPENGINE branch_id:0 ruri: <null>, method:INVITE, status:200, extra_id: z9hG4bK55018600, rtpengine_answer: replace-origin replace-session-connection SIP-source-address direction=priv1 direction=pub1 SIP-source-address via-branch=extra rtcp-mux-offer generate-mid DTLS=passive SDES-off ICE=force RTP/SAVPF Jun 12 06:59:49 ap07stvp.xxxxxxx.xxx docker[9373]: 51(57) INFO: {1 2 ACK jdl9kq6fv51jcv1agk2e} <script>: RTPENGINE branch_id:0 ruri: sip:10.5.2.192:5060, method:ACK, status:<null>, extra_id: <null>, rtpengine_offer: replace-origin replace-session-connection SIP-source-address direction=pub1 direction=priv1 SIP-source-address rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP Jun 12 06:59:49 ap07stvp.xxxxxxx.xxx docker[9373]: 51(57) INFO: {1 2 ACK jdl9kq6fv51jcv1agk2e} <script>: RTPENGINE branch_id:0 ruri: sip:10.5.2.192:5060, method:ACK, status:<null>, extra_id: <null>, rtpengine_offer: replace-origin replace-session-connection SIP-source-address direction=pub1 direction=priv1 SIP-source-address rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP Jun 12 07:01:26 ap07stvp.xxxxxxx.xxx docker[9373]: 27(33) INFO: {2 3 INVITE jdl9kq6fv51jcv1agk2e} <script>: RTPENGINE branch_id:0 ruri: <null>, method:INVITE, status:200, extra_id: z9hG4bK20888040, rtpengine_answer: replace-origin replace-session-connection SIP-source-address direction=priv1 direction=pub1 SIP-source-address via-branch=extra rtcp-mux-offer generate-mid DTLS=passive SDES-off ICE=force RTP/SAVPF Jun 12 07:01:26 ap07stvp.xxxxxxx.xxx docker[9373]: 27(33) ERROR: {2 3 INVITE jdl9kq6fv51jcv1agk2e} rtpengine [rtpengine.c:3070]: send_rtpp_command(): can't send command "offer" to RTPEngine <udp:10.5.0.183:22222> Jun 12 07:01:26 ap07stvp.xxxxxxx.xxx docker[9373]: 54(60) INFO: {1 3 ACK jdl9kq6fv51jcv1agk2e} <script>: RTPENGINE branch_id:0 ruri: sip:10.5.2.192:5060, method:ACK, status:<null>, extra_id: <null>, rtpengine_offer: replace-origin replace-session-connection SIP-source-address direction=pub1 direction=priv1 SIP-source-address rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP Jun 12 07:01:26 ap07stvp.xxxxxxx.xxx docker[9373]: 54(60) ERROR: {1 3 ACK jdl9kq6fv51jcv1agk2e} rtpengine [rtpengine.c:2706]: rtpp_function_call(): no available proxies We also found Daniel answer on similar issue here: https://github.com/kamailio/kamailio/issues/2713#issuecomment-823138302 Please advise how it can be solved. Kind regards, Arsenijs

4 5

Carrierroute Test API?
by sr.maillists＠gmail.com 13 Jun '23

13 Jun '23

Hi List Does someone know a function to test a number against the carrierroute module, to see which route it picked? My Idea: We have multiple Kamailio-Proxies and I would like to test a number against all hosts to see where it will be routed, without performing testcalls. Regards Stefan

2 1

Kamailio/MySQL Module compatibility with MySQL 8.0+
by Pedro Ulhoa 13 Jun '23

13 Jun '23

Hello friends. I didn't find about it in the documentation, but I will need to update my database version to MySQL 8.0.32 and I'm using Kamailio version 5.4. I did some quick tests, and I only needed to adjust the user due to the authentication mode, and with that, apparently I had no problems. I would like to know if you are using Kamailio with the latest version of MySQL, if you have had any problems in production, and if it really is safe to upgrade. -- Aviso Legal: Este documento pode conter informações confidenciais e/ou privilegiadas. Se você não for o destinatário ou a pessoa autorizada a receber este documento, não deve usar, copiar ou divulgar as informações nele contidas ou tomar qualquer ação baseada nessas informações. Disclaimer: The information contained in this document may be privileged and confidential and protected from disclosure. If the reader of this document is not the intended recipient, or an employee agent responsible for delivering this document to the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited.

3 2

What is the best way to return a lot of data via JSONRPC and jansson?
by Benoit Panizzon 12 Jun '23

12 Jun '23

Hi Team To offer some diagnose functionality I created an API that returns the location list via JSONRPC But this creates huge variables. Jun 12 09:43:32 prod-cpereg02 kamailio[2174548]: ERROR: <core> [core/pvapi.c:1491]: pv_printf_mode(): no more space for spec value - printed:0 token:8599 buffer:8192 Jun 12 09:43:32 prod-cpereg02 kamailio[2174548]: ERROR: <core> [core/pvapi.c:1503]: pv_printf_mode(): buffer overflow -- increase the buffer size... Already hitting the default limit and not sure this is the right way to go as we will end up with thousands of registrations. What is the best way to return larger ammounts of data out of kamailio? Avoid using the JSONRPC and access the database externally? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 1

Dialog DMQ state after restart
by Calvin E. 10 Jun '23

10 Jun '23

Does the dialog module do anything similar to the htable dmq_init_sync behavior? The scenario is restarting the offline member of an HA pair after modifying the script. We're unsure of the state of the replicated dialogs post-restart. Will the standby become aware of the dialogs prior to the restart? In other words, do we need to wait until the pre-restart calls end before we can safely failover? -Calvin E.

5 7

Kamailio - could not allocate private memory from pkg pool
by Muhammad Danish Moosa 09 Jun '23

09 Jun '23

Hi, I have been kamailio services working for few months and suddenly I got this error. ERROR: <core> [core/kemi.c:2923]: sr_kemi_pv_sets(): unable to set pv [$rp] ERROR: <core> [core/pvapi.c:352]: pv_cache_add(): could not allocate private memory from pkg pool ERROR: <core> [core/kemi.c:2914]: sr_kemi_pv_sets(): cannot get pv spec for [$dlg_var(BW0930526150606231188928742(a)10.10.16.100)] ERROR: <core> [core/data_lump.c:544]: dup_lump_list_r(): could not allocate private memory from pkg pool ERROR: <core> [core/data_lump.c:582]: dup_lump_list_r(): out of mem ERROR: tm [t_fwd.c:1759]: t_forward_nonack(): failure to add branches ERROR: <core> [core/msg_translator.c:2501]: build_res_buf_from_sip_req(): could not allocate private memory from pkg pool ERROR: sl [sl_funcs.c:414]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) ERROR: <core> [core/msg_translator.c:2501]: build_res_buf_from_sip_req(): could not allocate private memory from pkg pool ERROR: tm [t_reply.c:475]: _reply_light(): response building failed WARNING: tm [t_lookup.c:1499]: t_unref(): generation of a delayed stateful reply failed Issue was fixed after the restart of kamailio. -- Muhammad Danish Moosa " The core of mans' spirit comes from new experiences. "___ Christopher McCandless

4 10

Kamailio World
by Fred Posner 05 Jun '23

05 Jun '23

1 0

Invalid Links
by Brandon Armstead 04 Jun '23

04 Jun '23

https://www.kamailio.org/w/kamailio-v5-7-0-release-notes/ article references math module and tls_wolfssl, 2 of the 4 modules mentioned link back to 5.6 (these two) instead of 5.7 resulting in a 404. Respective links should be as follows: math - https://www.kamailio.org/docs/modules/5.7.x/modules/math.html tls_wolfssl - https://www.kamailio.org/docs/modules/5.7.x/modules/tls_wolfssl.html Can we get a correction in the article on WP? Thanks - Brandon

1 0

header from failure_route
by David Villasmil 04 Jun '23

04 Jun '23

Hello guys, so i'm doing on a failure_route: $var(xgws_in_reply) = $T_rpl($hdr(X-GWs)); which when it’s present on the reply is cool… but when there is NO REPLY, then kamailio sets it to “0”… anyway over this? i only want to know whether it is present on the reply Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 2

Missing transformations for {val.*}?
by Kaufman 03 Jun '23

03 Jun '23

Beginning in 5.6 new {val.*} transformations are documented as being exported by the pv module, however if I try using it Kamailio fails to start. Wondering if maybe it's actually imported by another module and the documentation is wrong or something like that? This configuration in 5.6.3 on Alpine Linux installed from packages: #!KAMAILIO loadmodule "pv" loadmodule "xlog" request_route { xlog("Null val: $(avp(nothing){val.ne})\n"); exit; } Fails to start with the following: 0(114) INFO: <core> [main.c:3055]: main(): processes (at least): 29 - shm size: 67108864 - pkg size: 8388608 0(114) ERROR: <core> [core/pvapi.c:1774]: tr_lookup(): unknown transformation: [val] in [{val.ne}] 0(114) ERROR: <core> [core/pvapi.c:1799]: tr_lookup(): error parsing [{val.ne}] 0(114) ERROR: <core> [core/pvapi.c:1106]: pv_parse_spec2(): bad tr in pvar name "avp" 0(114) ERROR: <core> [core/pvapi.c:1131]: pv_parse_spec2(): invalid parsing in [$(avp(nothing){val.ne})] at (4) 0(114) ERROR: xlog [xlog.c:551]: xdbg_fixup_helper(): wrong format[Null val: $(avp(nothing){val.ne})] 0(114) ERROR: <core> [core/route.c:1169]: fix_actions(): fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:7 Kaufman Sr. VoIP Engineer - BCM One +1.612.735.9309 bkaufman(a)bcmone.com<mailto:bkaufman@bcmone.com> 24 hour client support: +1.855.639.6300 [https://www.bcmone.com/files/bcmone-logo.png]<https://www.bcmone.com/> [https://www.bcmone.com/files/twitter-icon-circle.png] <https://www.facebook.com/bcmonenyc/> [https://www.bcmone.com/files/twitter-icon-circle.png] <https://twitter.com/bcm_one/> [https://www.bcmone.com/files/linkedin-icon-circle.png] <https://www.linkedin.com/company/business-communications-management/> NOTE: This e-mail and any attachments are from BCM One, Inc. and are intended solely for the use and review of the individual(s) to whom it is addressed. If you believe you received this e-mail in error, please notify the sender immediately, delete the e-mail (and any attachments) from your computer and do not copy or disclose it to anyone else.

2 1

custom mid registrar using kamailio.
by Yuriy Nasida 01 Jun '23

01 Jun '23

Hello, https://lists.kamailio.org/pipermail/sr-users/2020-October/110792.html I read this thread very carefully and I think I managed to implement all the steps described by Daniel. Thank you so much for the detailed description. But it looks like I'm stuck on the last step. I can't delete records from htable when an incoming call cannot reach the registered device (e.g., no more internet access, power off at the device). I try to do this from failure route but looks like i kamailio has not access to htable from it. failure_route[INTERNAL_TO_EXTERNAL_FAILURE] { if (t_check_status("408")) { xlog("L_WARN", "$ci|log| hello \n"); xlog("L_WARN", "$ci|log| mid reg token - $sht(mid_reg_token=>$fU::$si::$sp) \n"); } } I do not see xlog message with mid reg token but i see 'hello'. Yes, htable record exists in case i check using: kamcmd htable.dump mid_reg_token Also I need a way to remove registration at the registrar (Freeswitch in my case). This means I need to send api request from kamailio or just run some script that will do this. What is the best way ? module exec ? Please advice, Thanks

2 3

Kamailio and MS Teams side by side present staus und chat function
by Hans-Jürgen Brand 31 May '23

31 May '23

2 1

Is Min-SE check broken?
by harneet singh 31 May '23

31 May '23

Hi All, I am facing an issue in understanding how the min_se should be working in kamailio. As per the SST documentation, it seems like if the min_se is configured as 500, then any value of Session-Expires OR MIN-SE if lower than 500, can be responded to by a 422. However, I strangely see the reverse happening. To investigate further, I looked in to the ki_sst_check_min() code in the master, and these seems like a potential issue. Ref Code: Inside ki_sst_check_min(), there is an if condition like below: if (sst_min_se < MIN(minse, se.interval)) { However, shouldn't it be the other way around? ie if (sst_min_se > MIN(minse, se.interval)) { because we need to send 422 if the received value(in INVITE etc) is smaller than the sst configure min_se value? I also found a different documentation, at https://git.sgu.ru/oldssu/ex-opensips/blob/cb9df8d59dbb254a9d862569fd5d11f6… where the check is as below? if (sst_min_se > MIN(minse, se.interval)) { Can someone confirm if this is broken, or my understanding is incorrect? Regards, Harneet -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

2 3

http_client using JWT?
by Benoît Panizzon 31 May '23

31 May '23

Hi The http_client module support the somehow old fashioned way to submit credentials as part of the URL. Our 'held' server developer would love to use either JWT or a Bearer Token in the Auth-Header or a 'cookie-jar' to authenticate the requests. Is this somehow possible? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

3 2

Knowing about the uac module with primary and secondary proxy setting
by MUSLIH K 31 May '23

31 May '23

Hi, *sip_provider_proxy1[* uac module-register] ---> *kamailio* [dispatcher Module] ----> *Asterisk1* ----> *Asterisk 2* in my configuration kamailio register to the sip provider using *uac module*, sip information are stored in database (uacreg). when a incoming call(INVITE) come to kamailio it will dispatch call to Asterisk using* "dispatcher module".* Here i have couple of questions 1. If my SIP provider has a secondary proxy as a failover server (sip_provider_proxy2), can I configure Kamailio with the uac module to use the secondary proxy if the primary proxy fails? 2. Since my application is inbound-only, registration failure is only identified when the registration expires (3600 seconds = 1 hour). Can I use the dispatcher module to select the SIP proxy and switch the registration to the secondary proxy if the primary proxy is not reachable (using OPTIONS)?

2 2

DISABLE SDP MEDIA STREAMS
by Patrick Karton 29 May '23

29 May '23

Hello kamailio community, i know its an unusual query but i would like to know if there is a cleaner/proper.right way to disable some (or all) media streams in SDP. i have come with a solution by sdpops module with subst_body function like this. route[SDP_DISABLE_UNAUTHORIZED_MEDIA]{ # disable text,media and video media streams subst_body('#^m=(video|text|audio)[ ]+[0-9/]+(.*)$#m=\1 0 \2#ig'); # update originator ip address subst_body('#^o=([^ ]+)[ ]+([^ ]+)[ ]+([^ ]+)[ ]+([^ ]+)[ ]+([^ ]+)[ ]+([^ ]+)$#o=\1 \2 \3 \4 \5 7.7.7.7#ig'); # 1 2 2 4 5 6 # name sid snumber network net-type addr # update connection ip address subst_body('#^c=([^ ]+)[ ]+([^ ]+)[ ]+([^ ]+)$#c=\1 \2 7.7.7.7#ig'); # 1 2 3 # network net-type addr #remove all media attributes sdp_remove_line_by_prefix("a=fmtp"); sdp_remove_line_by_prefix("a=X-"); sdp_remove_line_by_prefix("a=rtcp"); sdp_remove_line_by_prefix("a=ssrc"); sdp_remove_line_by_prefix("a=sendrecv"); sdp_remove_line_by_prefix("a=sendonly"); sdp_remove_line_by_prefix("a=recvonly"); sdp_remove_line_by_prefix("a=inactive"); sdp_remove_line_by_prefix("b="); } is it the right way to do it. if not can someone provide me an alternative solution. Thanks :)

3 2

Re: [sr-dev] Troubles with outbound kamailio call to other server
by Henning Westerholt 26 May '23

26 May '23

Hello (added sr-users, as its probably more a usage question). The SIP message (the BYE) cannot be parsed apparently. From the log it looks that there is an invalid character in the first line. Do a network trace to analyse the packets that its actually send and try to get some more clues from that. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com > -----Original Message----- > From: aaron.soto(a)dialapplet.com <aaron.soto(a)dialapplet.com> > Sent: Freitag, 26. Mai 2023 10:44 > To: sr-dev(a)lists.kamailio.org > Subject: [sr-dev] Troubles with outbound kamailio call to other server > > Hi everyone, > I'm having the following errors in kamailio when i try to make an outbound call to > other server, I don't know what does it mean, if some one can help me > > 11(25) ERROR: <core> [core/receive.c:310]: receive_msg(): core parsing of SIP > message failed (XX.XX.XX.XX.5060/1) > 13(27) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): > parse_first_line: bad message (offset: 65) > 13(27) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: > parse_msg: message=<BY > sip:asterisk@10.0.5.158:4080;alias=10.0.5.158~44730~1 SIP/2.0 > Via: SIP/2.0/UDP 84.127.226.15:5060;branch=z9hG4bK3bc90b47;rport > Route: <sip:34.252.151.179:4060;r2=on;lr;ftag=5c7f862f-9692-4028-8af0- > ebc224ba0e2a;vsf=AAAAAAAAAAAAAAAAAAAJCQABGgUAAAMJLjE4ODo1MDYw; > nat=yes>,<sip:10.0.5.40:4070;r2=on;lr;ftag=5c7f862f-9692-4028-8af0- > ebc224ba0e2a;vsf=AAAAAAAAAAAAAAAAAAAJCQABGgUAAAMJLjE4ODo1MDYw; > nat=yes> > Max-Forwards: 70 > From: <sip:692598435@kamailio.voip.svc.cluster.local>;tag=as6f32aa26 > To: <sip:963441038@192.168.123.66:4060>;tag=5c7f862f-9692-4028-8af0- > ebc224ba0e2a > Call-ID: b9a49c52-7829-4cdf-a88c-0064cc5bb438 > CSeq: 102 BYE > User-Agent: Asterisk PBX 11.2-cert3 > X-Asterisk-HangupCause: No user responding > X-Asterisk-HangupCauseCode: 18 > Content-Length: 0 > > > > _______________________________________________ > Kamailio (SER) - Development Mailing List To unsubscribe send an email to sr-dev- > leave(a)lists.kamailio.org

1 0

Consistent huntable responses based on failure_route question
by John Hardiman 25 May '23

25 May '23

Hi Kamailio list, I hope you are all well? I am looking for some assistance on a specific error I am trying to work around. Kamailio 5.5.3 - Ubuntu 20.04 Kemi Python script (but I think this is more default failure_route behaviour) Dispatcher module which hunts to a group of FreeSWITCH that I have configured to send 503 responses on. My scenario is fairly simple, I am trying to respond back to a carrier for huntable responses (503 specifically) by calling a failure route below: def ksr_failure_manage(self, msg): if KSR.tm.t_any_timeout()>0 : self.ksr_send_reply_clear_rtpengine(msg, 503, "Timeout to destination") KSR.err("Timeout to Invite for %s from IP %s" % (KSR.pv.get("$ci"),KSR.pv.get("$si"))) return 1 else: self.ksr_send_reply_clear_rtpengine(msg, 503, "Unknown error failover to next node") KSR.err("Failed but not timeout for %s from IP %s" % (KSR.pv.get("$ci"),KSR.pv.get("$si"))) return 1 This works and a 503 with the correct message is sent if the invite doesn't get a response after 2 seconds. (My specific test case here) # Timers for call setup 2 minutes for any 1xx provisional response received # If no invite response received in 2 seconds end the call as per failure route. KSR.tm.t_set_fr(120000, 2000) self.ksr_route_relay(msg) KSR.tm.t_on_failure("ksr_failure_manage") return -255 The issue is after a few more milliseconds after my initial 503 a 408 is sent. Also after the failover if the carrier tries the same Kamailio again it gets a 500 rather than a 408. 408: [cid:image001.png@01D98F15.CD07DCB0] 500: [cid:image002.png@01D98F15.CD07DCB0] My steps to get around this have been to change the default tm code to 503 below, but this has not changed the default response and the above behaviour unfortunately: # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # setting default reason code and response for general errors to allow failover. modparam("tm", "default_code", 503) modparam("tm", "default_reason", "Unknown reason try next node") My question is: How can I make sure my failure route is the final response here. Or if I cannot make sure of this, how can I change the default mapped response to be a consistent SIP response (503 specifically) Many thanks in advance as I am scratching my head on this one a bit. John.

2 2

Override listen advertise address
by Calvin E. 25 May '23

25 May '23

Using 5.6.x, a scenario designed for a public IP now needs to support life in the cloud behind NAT. I'd like to avoid having two listeners to separate on-net from off-net traffic. In a HA scenario, it would be nice to have only one floating IP. Is it possible to use the same local address and port for external/public and internal/private traffic? Should set_advertised_address() override the listen advertise address? Initial testing did not produce the expected result. I want to make sure I'm following a sane pattern before going down a rabbit hole. The idea is to toggle the advertised address by checking if the sender or recipient is rfc1918(). For the dispatcher module, is it possible to use socket names like below? In testing it appeared that only one of the two names was valid. listen=udp:10.0.0.10:5060 name "onnet" listen=udp:10.0.0.10:5060 advertise 11.11.11.11:5060 name "offnet"

3 4

lost module: best practice handling delay caused by held / lis procedure?
by Benoit Panizzon 25 May '23

25 May '23

Hi gang In 2024, Switzerland will start using NG112 (NG911) procedures to transmit the caller location via Geolocation URL via LIS Server to a PSAP. So time to start testing this on my devel plattform. Kamailio ships with the lost module which looks promising. In short, the Procedure is as follows: Customer calls emergency number. Kamailio performs a http request to our inhouse HELD server with the identification of the calling customer. HELD server looks up the location of the caller and pushes a HELD XML object to the Swiss LIS Server, which then is queried by the emergency PSAP receiving the call. LIS server return and URL. HELD server returns the URL to Kamailio. Kamailio adds Geolocation Header with that URL and relays the call. This takes some time and as the call is not yet being relayed while waiting for the held request to complete, there is no 100 trying being sent and the emergency call is bound to time out. So, shall I call t_reply("100","Performing HELD lkup") before calling the lost module? Or is there a better way? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

3 7

cancel at first call after startup kamailio not relay
by Giuseppe Pandolfi 23 May '23

23 May '23

Hi, I have a problem with kamailio v 5.6.1 on mips architecture: version: kamailio 5.6.1 (mips/linux) flags: USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX-NOSMP, USE_PTHREAD_MUTEX, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 14:22:18 Apr 27 2023 Using default script for make a call from sip ua to PSTN system (#!define WITH_PSTN), the cancel sent for the first call after start kamailio doesn't relay to PSTN system and I have the trace error beelow: 2(3203) BUG: {1 27018 CANCEL 0tBgb95Z.0eE5VVrAkoP06nvunIknKNz} tm [t_cancel.c:318]: cancel_branch(): local_cancel buffer=(nil) != BUSY_BUFFER (trying to continue) 12(3203) ERROR: {1 27018 CANCEL 0tBgb95Z.0eE5VVrAkoP06nvunIknKNz} tm [t_fwd.c:1434]: e2e_cancel(): cancel error the next calls works fine. Does anyone have any suggestions? Beelow trace at debug level: 37(3695) INFO: <core> [core/cfg/cfg_ctx.c:602]: cfg_set_now(): core.debug has been changed to 3 37(3695) DEBUG: ctl [io_listener.c:518]: handle_stream_read(): bytes processed: 23 12(3670) DEBUG: <core> [core/parser/parse_fline.c:250]: parse_first_line(): first line type 1 (request) flags 1 12(3670) DEBUG: <core> [core/parser/msg_parser.c:677]: parse_msg(): SIP Request: 12(3670) DEBUG: <core> [core/parser/msg_parser.c:679]: parse_msg(): method: <CANCEL> 12(3670) DEBUG: <core> [core/parser/msg_parser.c:681]: parse_msg(): uri: <sip:100@192.168.2.238:5060> 12(3670) DEBUG: <core> [core/parser/msg_parser.c:683]: parse_msg(): version: <SIP/2.0> 12(3670) DEBUG: <core> [core/parser/parse_hname2.c:302]: parse_sip_header_name(): parsed header name [Via] type 1 12(3670) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 235, <rport> = <n/a>; state=6 12(3670) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKPjLBcUXshEKW2s9T0en9czTYAsciW1vV-W>; state=16 12(3670) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 12(3670) DEBUG: <core> [core/parser/msg_parser.c:555]: parse_headers(): Via found, flags=2 12(3670) DEBUG: <core> [core/parser/msg_parser.c:557]: parse_headers(): this is the first via 12(3670) DEBUG: <core> [core/parser/parse_hname2.c:302]: parse_sip_header_name(): parsed header name [Max-Forwards] type 8 12(3670) DEBUG: <core> [core/parser/parse_hname2.c:302]: parse_sip_header_name(): parsed header name [From] type 4 12(3670) DEBUG: <core> [core/parser/parse_hname2.c:302]: parse_sip_header_name(): parsed header name [To] type 3 12(3670) DEBUG: <core> [core/parser/parse_addr_spec.c:884]: parse_addr_spec(): end of header reached, state=9 12(3670) DEBUG: <core> [core/parser/msg_parser.c:173]: get_hdr_field(): <To> [23]; uri=[sip:100@192.168.2.238] 12(3670) DEBUG: <core> [core/parser/msg_parser.c:176]: get_hdr_field(): to body (23)[sip:100@192.168.2.238 ], to tag (0)[] 12(3670) DEBUG: <core> [core/parser/parse_hname2.c:302]: parse_sip_header_name(): parsed header name [Call-ID] type 6 12(3670) DEBUG: <core> [core/parser/parse_hname2.c:302]: parse_sip_header_name(): parsed header name [CSeq] type 5 12(3670) DEBUG: <core> [core/parser/msg_parser.c:154]: get_hdr_field(): cseq <CSeq>: <17038> <CANCEL> 12(3670) DEBUG: <core> [core/receive.c:394]: receive_msg(): --- received sip message - request - call-id: [CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU] - cseq: [17038 CANCEL] 12(3670) DEBUG: <core> [core/parser/parse_hname2.c:302]: parse_sip_header_name(): parsed header name [User-Agent] type 28 12(3670) DEBUG: <core> [core/parser/parse_hname2.c:302]: parse_sip_header_name(): parsed header name [Content-Length] type 12 12(3670) DEBUG: <core> [core/parser/msg_parser.c:187]: get_hdr_field(): content_length=0 12(3670) DEBUG: <core> [core/parser/msg_parser.c:91]: get_hdr_field(): found end of header 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} <core> [core/receive.c:261]: ksr_evrt_pre_routing(): event route core:pre-routing not defined 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} <core> [core/receive.c:471]: receive_msg(): preparing to run routing scripts... 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} maxfwd [mf_funcs.c:74]: is_maxfwd_present(): value = 70 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} sanity [sanity.c:247]: check_ruri_sip_version(): check_ruri_sip_version entered 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} sanity [sanity.c:270]: check_ruri_sip_version(): check_ruri_sip_version passed 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} sanity [sanity.c:277]: check_ruri_scheme(): check_ruri_scheme entered 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} sanity [sanity.c:297]: check_ruri_scheme(): check_ruri_scheme passed 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} sanity [sanity.c:318]: check_required_headers(): check_required_headers entered 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} sanity [sanity.c:340]: check_required_headers(): check_required_headers passed 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} sanity [sanity.c:348]: check_via1_header(): check via1 header 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} sanity [sanity.c:506]: check_cseq_method(): check_cseq_method entered 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} sanity [sanity.c:539]: check_cseq_method(): check_cseq_method passed 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} sanity [sanity.c:548]: check_cseq_value(): check_cseq_value entered 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} sanity [sanity.c:576]: check_cseq_value(): check_cseq_value passed 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} sanity [sanity.c:585]: check_cl(): check_cl entered 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} sanity [sanity.c:610]: check_cl(): check_cl passed 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} sanity [sanity.c:622]: check_expires_value(): check_expires_value entered 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} sanity [sanity.c:659]: check_expires_value(): no expires header found 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} sanity [sanity.c:758]: check_parse_uris(): check_parse_uris entered 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} sanity [sanity.c:762]: check_parse_uris(): parsing ruri 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} sanity [sanity.c:776]: check_parse_uris(): looking up From header 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=Jjj1QBeRxydZ5rq.-xKCfcJZhlPANj9N 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} <core> [core/parser/parse_addr_spec.c:884]: parse_addr_spec(): end of header reached, state=29 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} sanity [sanity.c:791]: check_parse_uris(): looking up To header 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} sanity [sanity.c:837]: check_parse_uris(): check_parse_uris passed 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} sanity [sanity_mod.c:254]: w_sanity_check(): sanity checks result: 1 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} tm [t_lookup.c:727]: t_lookupOriginalT(): searching on hash entry 10182 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} tm [t_lookup.c:440]: matching_3261(): RFC3261 transaction matched, tid=PjLBcUXshEKW2s9T0en9czTYAsciW1vV-W 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} tm [t_lookup.c:823]: t_lookupOriginalT(): canceled transaction found (0x729bff78)! 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} tm [t_lookup.c:826]: t_lookupOriginalT(): found - lookup completed 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} tm [tm.c:1232]: ki_t_lookup_cancel_flags(): lookup_original: t_lookupOriginalT returned: 0x729bff78 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} tm [t_lookup.c:1348]: t_newtran(): msg (0x771181ec) id=1/3670 global id=0/0 T start=0xffffffff 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} tm [t_lookup.c:498]: t_lookup_request(): start searching: hash=10182, isACK=0 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} tm [t_lookup.c:456]: matching_3261(): RFC3261 transaction matching failed - via branch [z9hG4bKPjLBcUXshEKW2s9T0en9czTYAsciW1vV-W] 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} tm [t_lookup.c:675]: t_lookup_request(): no transaction found 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} <core> [core/crypto/md5utils.c:67]: MD5StringArray(): MD5 calculated: dd71ad9d7511b02a5fc83c3c70ce307f 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} tm [t_lookup.c:727]: t_lookupOriginalT(): searching on hash entry 10182 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} tm [t_lookup.c:432]: matching_3261(): matched skip method - s:0x2 t:0x2 m:0x2 - continue searching 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} tm [t_lookup.c:440]: matching_3261(): RFC3261 transaction matched, tid=PjLBcUXshEKW2s9T0en9czTYAsciW1vV-W 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} tm [t_lookup.c:823]: t_lookupOriginalT(): canceled transaction found (0x729bff78)! 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} tm [t_lookup.c:826]: t_lookupOriginalT(): found - lookup completed 12(3670) BUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} tm [t_cancel.c:318]: cancel_branch(): local_cancel buffer=(nil) != BUSY_BUFFER (trying to continue) 12(3670) ERROR: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} tm [t_fwd.c:1434]: e2e_cancel(): cancel error 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} tm [t_reply.c:1763]: cleanup_uac_timers(): RETR/FR timers reset 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} tm [t_reply.c:638]: _reply_light(): reply sent out - buf=0x77101e68: SIP/2.0 500 cancel e... shmem=0x729c5b60: SIP/2.0 500 cancel e 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} tm [t_funcs.c:120]: put_on_wait(): put T [0x729c36d8] on wait 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} tm [t_reply.c:648]: _reply_light(): finished 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} tm [t_funcs.c:373]: t_relay_to(): new transaction forwarded 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} <core> [core/receive.c:529]: receive_msg(): request-route executed in: 9868 usec 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} <core> [core/usr_avp.c:637]: destroy_avp_list(): destroying list (nil) 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} <core> [core/usr_avp.c:637]: destroy_avp_list(): destroying list (nil) 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} <core> [core/usr_avp.c:637]: destroy_avp_list(): destroying list (nil) 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} <core> [core/usr_avp.c:637]: destroy_avp_list(): destroying list (nil) 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} <core> [core/usr_avp.c:637]: destroy_avp_list(): destroying list (nil) 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} <core> [core/usr_avp.c:637]: destroy_avp_list(): destroying list (nil) 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} <core> [core/xavp.c:541]: xavp_destroy_list(): destroying xavp list (nil) 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} <core> [core/xavp.c:541]: xavp_destroy_list(): destroying xavp list (nil) 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} <core> [core/xavp.c:541]: xavp_destroy_list(): destroying xavp list (nil) 12(3670) DEBUG: {1 17038 CANCEL CYxrNJtcCFZKS.PJN29LlEBd0aejPQUU} <core> [core/receive.c:628]: receive_msg(): cleaning up 34(3692) DEBUG: tm [timer.c:642]: wait_handler(): finished transaction: 0x729c36d8 (p:0x72866768/n:0x729bff78) 34(3692) DEBUG: tm [h_table.c:132]: free_cell_helper(): freeing transaction 0x729c36d8 from timer.c:650

1 0

mk_proxy(): could not resolve hostname
by ajosebar＠gmail.com 23 May '23

23 May '23

I have Kamailio 5.6 on Debian 11. I need to use the function forward within a routing block in my configuration file: forward("udp:192.168.28.51:5060"); But when I start the kamailio service I got: CRITICAL: [core/proxy.c:264]: mk_proxy(): could not resolve hostname: "udp:192.168.28.51:5060" The same occurs when I use the dns nane : forward("udp:raspberry3.emtelco.co:5060"); Any ideas ? Regards

2 1

DNS resolution with ipops
by harneet singh 22 May '23

22 May '23

Hi Experts, I am planning to use *dns_query*() API from the ipops module, which I understand from some documentation that it internally uses *getaddrinfo*(). I wanted to check if someone can comment on the DNS Cache mechanism in this case. * Does Kamailio retain any *dns cache* while using this API, or would it be actually doing the resolution for each call. * In light of the above, does Kamailio honour the *TTL value i*n response to the DNS Query? Regards, Harneet Singh -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

2 1

MySQL duplicate entry error on location after 5.5 => 5.6 upgrade
by Benoit Panizzon 22 May '23

22 May '23

Hi list Since updating 5.5 to 5.6 i often see this message in the logs: ERROR: db_mysql [km_dbase.c:122]: db_mysql_submit_query(): driver error on query: Duplicate entry 'uloc-645ca178-f29e-1' for key 'location.ruid_idx' (1062) ERROR: <core> [db_query.c:244]: db_do_insert_cmd(): error while submitting query ERROR: usrloc [ucontact.c:687]: db_insert_ucontact(): inserting contact in db failed ****** (uloc-645ca178-f29e-1) ERROR: usrloc [urecord.c:420]: wb_timer(): inserting contact into database failed (aor: *******) usrloc is dmq synced between two nodes. Were there some changes which could have caused this issue? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

1 0

Kamailio v5.7.0 Released - new major version is out
by Daniel-Constantin Mierla 21 May '23

21 May '23

Kamailio v5.7.0 is out – it comes with 4 new modules and a large set of improvements touching again more than 60 existing modules. You can read a bit more detailed release notes at: * https://www.kamailio.org/w/kamailio-v5-7-0-release-notes/ Many thanks to all developers and community members that made possible this release. v5.7.0 brings more flexibility and optimizations across many components, new variables and transformations, expression-based preprocessor values and conditions, alternative TLS module using WolfSSL project, improvements to STIR/SHAKEN, IMS/VoLTE, more functions exported to KEMI interface, including a new KEMI Python3 interpreter. ... just to list only a few here. Enjoy Kamailio v5.7.0! Thank you for flying Kamailio! Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com

2 1

db_redis and ursloc - problem after restarting kamailio
by Koldo Aingeru 18 May '23

18 May '23

Hi, I've encountered a problem using db_redis for location, we recently migrated to redis from mysql and just found out this yesterday. We are using the usrloc module with db_mode = 2, so the location info is written to redis every 60 seconds. When restarting, Kamailio reads all the contacts and loads them into memory, but for some contacts it loads the “methods” key with an incorrect value. If in the REGISTER there are no methods kamailio saves it with methods= -1: { "jsonrpc": "2.0", "result": { "AoR": “zzzzzzz", "Contacts": [{ "Contact": { "Address": "sip:1540305477794@W.X.Y.Z:5060", "Expires": 2752, "Q": -1, "Call-ID": "7d3965a85bfd0e88-32357(a)W.X.Y.Z", "CSeq": 11, "User-Agent": “xxxxxxx", "Received": "sip:W.X.Y.Z:5060", "Path": "<sip:A.B.C.D;lr;received=sip:W.X.Y.Z:5060>", "State": "CS_SYNC", "Flags": 0, "CFlags": 64, "Socket": “udp:A.B.C.D:5060", "Methods": -1, "Ruid": "uloc-644a176b-6472-4641", "Instance": "[not set]", "Reg-Id": 0, "Server-Id": 0, "Tcpconn-Id": -1, "Keepalive": 1, "Last-Keepalive": 1684246829, "Last-Modified": 1684246829 } }] }, "id": 19286 } But when restarting kamailio, the same contacts has a methods= 0 { "jsonrpc": "2.0", "result": { "AoR": “zzzzzzz", "Contacts": [{ "Contact": { "Address": "sip:1540305477794@W.X.Y.Z:5060", "Expires": 2697, "Q": -1, "Call-ID": "7d3965a85bfd0e88-32357(a)W.X.Y.Z", "CSeq": 11, "User-Agent": “xxxxxxx", "Received": "sip:W.X.Y.Z:5060", "Path": "<sip:A.B.C.D;lr;received=sip:W.X.Y.Z:5060>", "State": "CS_SYNC", "Flags": 0, "CFlags": 64, "Socket": “udp:A.B.C.D:5060", "Methods”: 0, "Ruid": "uloc-644a176b-6472-4641", "Instance": "[not set]", "Reg-Id": 0, "Server-Id": 0, "Tcpconn-Id": -1, "Keepalive": 1, "Last-Keepalive": 1684246829, "Last-Modified": 1684246829 } }] }, "id": 19286 } In the schema file for redis the key is correctly set as int and in redis I can see it saved as -1. So when a call comes in, the lookup() function returns a -2 as it doesn’t support the INVITE method and the call fails. This is happening in kamailio 5.2.8 and kamailio 5.2.1 (old versions, I know, we are working on updating it); with mysql as a backend it works perfectly after a restart. We do have the param method_filtering=1 , changing to 0 makes it works. We seem to have this since forever and for what I can see, is not unusual as is in the default kamailio.cfg provided (even the default value for it is 0). Has anyone else encountered this? Thanks! ======================================== Koldo Aingeru Marcos Fdez. Ingeniería de Sistemas Sarenet S.A.U. 944209470 Parque Tecnológico, Edificio 103 48170 Zamudio, Bizkaia www.sarenet.es ========================================

2 2

Contact changes, caller dialog attributes and NAT traversal
by Alex Balashov 17 May '23

17 May '23

Hi, When I modify INVITEs with set_contact_alias() / add_contact_alias(), this is not preserved in dialog data, and therefore not honoured in BYEs locally generated due to dialog timeout. For example, if the initial INVITE contained: Contact: sip:line1@192.168.1.100 and, after transit, became: Contact: <sip:line1@192.168.1.100>;alias=1.1.1.1~25844~1 There is no ;alias value stored in the caller dialog data, just the original URI + parameters. A related problem: I tried to work around this by storing an alias in dialog user variables, e.g. $dlg_var(alternate_ct) = $_s($si~$sp~$prid) and in fact, I am able to recover this value later: event_route[tm:local-request] { if($dlg_var(alternate_ct) ne $null) { $ru = $ru + ";alias=" + $dlg_var(alternate_ct); handle_ruri_alias(); } } And in fact, if I print the value of $du after calling handle_ruri_alias(), I can see that it is modified. However, this seems to have no effect on where the request actually goes at the network and transport level. I assume this is because the event_route does not actually allow one to overwrite elements of the spoofed request, and that its essential attributes have been pre-populated elsewhere. Nevertheless, the effect is that these BYEs do not reach NAT'd endpoints. Any suggestions are appreciated! Thanks in advance, -- Alex -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800

2 4

Re: Releasing Kamailio v5.7.0
by Daniel-Constantin Mierla 17 May '23

17 May '23

The header style was wrong for ctl, fixed, thanks. Daniel On 15.05.23 18:55, Дилян Палаузов wrote: > Off-list > > Hello, > > at > https://github.com/kamailio/kamailio-wiki/blob/main/docs/features/new-in-5.… there > is a horizonal line after ctl, which shall not be there. > > Greetings > Дилян > > > На 15 май 2023 г. 7:20:55 UTC, Daniel-Constantin Mierla > <miconda(a)gmail.com> написа: > > Hello, with a few public holidays coming soon around here, I am > considering to release 5.7.0 this week on Wednesday, May 17, 2023. > Rather short notice, but there were no major bug reports that need > to be tracked. The list of new features is being built in the > wiki: - > https://github.com/kamailio/kamailio-wiki/blob/main/docs/features/new-in-5.… > The other tutorials related to this major release should be ready > by then as well. Should anyone think more time is needed, then it > can be postponed, not a problem at all. Cheers, Daniel > -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com

1 0

Releasing Kamailio v5.7.0
by Daniel-Constantin Mierla 17 May '23

17 May '23

Hello, with a few public holidays coming soon around here, I am considering to release 5.7.0 this week on Wednesday, May 17, 2023. Rather short notice, but there were no major bug reports that need to be tracked. The list of new features is being built in the wiki: - https://github.com/kamailio/kamailio-wiki/blob/main/docs/features/new-in-5.… The other tutorials related to this major release should be ready by then as well. Should anyone think more time is needed, then it can be postponed, not a problem at all. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com

2 3

Diameter_authorization_rejected (FHOSS)
by Wilmes, Matthias 16 May '23

16 May '23

Hello sr-users, i use the guide for "how to integrate an IMS using Kamailio with open5gs EPC" from Supreeth Herle. When my CotsUE(ONEPLUS8) try to Register. I get an 403 Forbidden HSS Authorization Rejected (see udp-stream.txt). The output of the fhoss looks like this. de.fhg.fokus.hss.cx.CxFinalResultException: Diameter_Authorization_Rejected at de.fhg.fokus.hss.cx.op.UAR.processRequest(UAR.java:134) at de.fhg.fokus.hss.main.Task.execute(Task.java:169) at de.fhg.fokus.hss.main.Worker.run(Worker.java:66) Insufficient access rights for the mysql hss_db? The user hss@localhost and hss@% have grant access. Maybe someone has had the error and has a hint and can help. Can't find the error. Many thanks in advance for an answer. -M.Wilmes

3 2

How to prevent run-away dialogues?
by Benoît Panizzon 15 May '23

15 May '23

Hi team Is there any way to get correct CDRS when using the dialog module and restarting kamailio? I have configured the dialog timeout to be 12 hours. But today while doing IC billing, we stumbled over a call that lasted for a week. Probably due to a stale entry in the database dialogue tables left over after a kamailio restart. Looking at the logs there definitely was something wrong: dialog [dlg_req_within.c:211]: bye_reply_cb(): inconsistent dlg timer data on dlg 0x7f153f503778 [956:2612] with clid '35BEFD13@7f33ff47' and tags '7f33ff47+1+75300082+dde85140' '3891401256-2097511021' acc [acc_cdr.c:229]: db_write_cdr(): fallback to dlg_only search because of message doesn't exist. -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 1

[auth_radius.c:123]: mod_init(): failed to open configuration file
by Denys Pozniak 12 May '23

12 May '23

Hello! After a major upgrade of Kamailio (5.1 -> 5.6), an error appears that it is not possible to open configuration file (I think this is about /etc/radiusclient/radiusclient.conf), although it exists with the necessary access rights: *0(27075) DEBUG: <core> [core/sr_module.c:969]: init_mod(): auth_radius0(27075) ERROR: auth_radius [auth_radius.c:123]: mod_init(): failed to open configuration file0(27075) ERROR: <core> [core/sr_module.c:975]: init_mod(): Error while initializing module auth_radius (/usr/lib64/kamailio/modules/auth_radius.so)ERROR: error while initializing modules* For the test, I made Kamailio downgrade back to 5.1 and the problem went away. Shortened the Kamailio config to the minimum for reproducing the problem: *#!KAMAILIOmemdbg=5memlog=5children=1log_stderror=nolisten=udp:127.0.0.1:5060 <http://127.0.0.1:5060>loadmodule "kex.so"loadmodule "corex.so"loadmodule "tm.so"loadmodule "sl.so"loadmodule "rr.so"loadmodule "pv.so"loadmodule "maxfwd.so"loadmodule "auth.so"loadmodule "auth_radius.so"modparam("auth_radius", "radius_config", "/etc/radiusclient/radiusclient.conf")request_route { return;}* Could the commit below somehow affect the acceptance of the radius_config module parameter? https://github.com/kamailio/kamailio/commit/f33ba318b20cbd2b34d278a7e2c4e64… -- BR, Denys Pozniak

4 7

kamcmd ul.db_contacts how to specify table name?
by Benoît Panizzon 12 May '23

12 May '23

Good morning. I'm trying to get the number of registered contacts with kamcmd ul.db_contacts. The table has the default name 'location'. But no matter what I try: kamcmd ul.db_contacts location kamcmd ul.db_contacts s:location kamcmd ul.db_contacts database.location kamcmd ul.db_contacts s:database.location Kamcmd complains the table is not found. What would be the correct syntax to provide the table name? Attempted with Kamailio 5.5 and 5.6 -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

1 1

ICMP Port Unreachable using loopback interface with ethernet IP
by Calvin E. 11 May '23

11 May '23

The scenario is Kamailio on Debian 11 talking to an external application server that returns a 302 with a Contact that contains Kamailio's own IP and a different port. Kamailio is listening on the correct port via at the given IP address. However, when Kamailio creates the packet to talk to itself, it it send over the loopback adapter "lo" and gets an ICMP port reply: listen=udp:KAMAILIO_INTERNAL_FLOATING_IP:8888 10.z.a.b.5000 > 10.z.x.y.5060: SIP, length: 849 SIP/2.0 302 Moved Temporarily Contact: <sip:44123456...@10.z.x.y:8888> 15:39:00.548722 lo In IP (tos 0x10, ttl 64, id 5431, offset 0, flags [none], proto UDP (17), length 949) 10.z.x.y.5060 > 10.z.x.y.8888: SIP, length: 921 15:39:00.548732 lo In IP (tos 0xd0, ttl 64, id 19038, offset 0, flags [none], proto ICMP (1), length 576) 10.z.x.y > 10.z.x.y: ICMP 10.44.7.136 udp port 8888 unreachable, length 556 Do I need to configure something in the OS or Kamailio to allow the ethernet IP to work via the loopback adapter? I haven't set up any special networking to cause this, it appears to be a Linux kernel behavior.

3 3

where add contact header to ack for fail messages
by mohsen khashei 11 May '23

11 May '23

Hi I want to know where kamailio sends ack messages to faiurel messages (status >=400) so that I I add Contact header to the ack. I tried WITHINDLG with no luck. thanks

5 6

GeoIP2 and KEMI
by Michal Hajek 11 May '23

11 May '23

Hello, why this commit is not in branch 5.6? https://github.com/kamailio/kamailio/commit/21817906853140322c701f8cbb82010… Without this, is KEMI and GeoIP2 incompatible. Thanks Michal

2 1

kamailio/kamailio-ci docker image
by Sergey Safarov 11 May '23

11 May '23

Hi Now "kamailio/kamailio-ci" docker image is pushed to GitHub packages <https://github.com/kamailio/kamailio-ci/pkgs/container/kamailio-ci> only. I have a question should we completely switch to using the GitHub docker repo and remove the "kamailio/kamailio-ci" docker image from Docker Hub <https://hub.docker.com/repository/docker/kamailio/kamailio-ci>? Or required to push the "kamailio/kamailio-ci" docker image to GitHub packages and Docker Hub repos? Sergey

2 1

Re: where add contact header to ack for fail messages
by Henning Westerholt 11 May '23

11 May '23

Understood. Thanks for sharing your final solution, added the list as well that others can benefit from it. Cheers, Henning From: mohsen khashei <mkhashei(a)gmail.com> Sent: Donnerstag, 11. Mai 2023 12:37 To: Henning Westerholt <hw(a)gilawa.com> Subject: Re: [SR-Users] Re: where add contact header to ack for fail messages I need it for an application server after kamailio. Finally I add contact to the message by ac_extra_hdrs of tm module with a p header. On Thu, May 11, 2023, 15:01 Henning Westerholt <hw(a)gilawa.com<mailto:hw@gilawa.com>> wrote: Hello, why you actually need to add the Contact header to the ACK? According to the RFC 3261 is should be an optional header to that message. Cheers, Henning -----Original Message----- From: James Browne <james(a)frideo.com<mailto:james@frideo.com>> Sent: Mittwoch, 10. Mai 2023 14:08 To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> Subject: [SR-Users] Re: where add contact header to ack for fail messages I made simple tests and it seems that this is a kamailio-TM-internal message being sent, not being processed by the cfg file. If this is simple enough and deterministic, you may consider using the corex module to edit the message as it's being sent "on the wire". I can't think of any other solution. https://kamailio.org/docs/modules/5.6.x/modules/corex.html#corex.evr.networ… You may be able to write some code that manipulates the message to manually add the Contact header. Maybe even the append_hf function may work here, but it seems unlikely. James On Mon, 8 May 2023 at 12:28, mohsen khashei <mkhashei(a)gmail.com<mailto:mkhashei@gmail.com>> wrote: > > thanks for the reply I tried it and it didn't work. also I tried onsend_route with no luck. > > On Sun, May 7, 2023 at 7:00 PM Yuriy G <ovoshlook(a)gmail.com<mailto:ovoshlook@gmail.com>> wrote: >> >> Try here >> https://www.kamailio.org/docs/modules/5.6.x/modules/tm.html#tm.e.loca >> l-request >> >> However I would doubt first do I do something right by adding contact to ACK message wichbasicaly confirmation of the final transaction and contact here is meaningless. >> >> On Sun, 7 May 2023, 16:48 mohsen khashei, <mkhashei(a)gmail.com<mailto:mkhashei@gmail.com>> wrote: >>> >>> Hi I want to know where kamailio sends ack messages to faiurel messages (status >=400) so that I I add Contact header to the ack. >>> I tried WITHINDLG with no luck. >>> thanks >>> __________________________________________________________ >>> Kamailio - Users Mailing List - Non Commercial Discussions To >>> unsubscribe send an email to sr-users-leave(a)lists.kamailio.org<mailto:sr-users-leave@lists.kamailio.org> >>> Important: keep the mailing list in the recipients, do not reply only to the sender! >>> Edit mailing list options or unsubscribe: >> >> __________________________________________________________ >> Kamailio - Users Mailing List - Non Commercial Discussions To >> unsubscribe send an email to sr-users-leave(a)lists.kamailio.org<mailto:sr-users-leave@lists.kamailio.org> >> Important: keep the mailing list in the recipients, do not reply only to the sender! >> Edit mailing list options or unsubscribe: > > __________________________________________________________ > Kamailio - Users Mailing List - Non Commercial Discussions To > unsubscribe send an email to sr-users-leave(a)lists.kamailio.org<mailto:sr-users-leave@lists.kamailio.org> > Important: keep the mailing list in the recipients, do not reply only to the sender! > Edit mailing list options or unsubscribe: __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org<mailto:sr-users-leave@lists.kamailio.org> Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:

1 0

using kamailio to send invite to webrtc clients (with freeswitch)
by shwetank singh 09 May '23

09 May '23

Hello experts I have a setup where webrtc client registrations (from behind a NAT / VPN) are on kamailio and the INVITE to begin a call is sent by freeswitch to kamailio which is then expected to relay it to the subscribed webrtc client. i understand, from documentation and general thinking, that kamailio cannot open a new connection to such a client and should reuse an open connection (a websocket in my case). i have setup kamailio as generic as it may sound and expect t_relay() to do the magic of forwarding the invite, however, from logs i don't see anything indicating kamailio is using any existing connection and hence my outgoing invite requests are stuck showing pointed towards a vpn gateway which was the registered "location" of client (which is the public ip of VPN gateway) please advise what relevant information can i produce here to make the query more readable and clear. headed here after trying just about all the resources available on the topic. e.g. if my location record looks like AoR: 1004(a)mydomain.com Contacts: - Contact: Address: sip:1004@11.22.33.44:49179 CFlags: 64 CSeq: 3087 Call-ID: 189jss21e24ntve1ggsuli Expires: 593 Flags: 0 Instance: <urn:uuid:8a936bbc-514f-4755-bed0-edc644dd4453> KA-Roundtrip: 0 Keepalive: 1 Last-Keepalive: 1683652755 Last-Modified: 1683652755 Methods: 7071 Path: <sip:10.10.6.196:3040;received=sip:mydomain.com ;lr;nat=yes> Q: -1 Received: sip:11.22.33.44:49179;transport=ws Reg-Id: 1 Ruid: uloc-645601c4-131220-3764 Server-Id: 0 Socket: tls:1.2.3.4:4443 State: CS_SYNC Tcpconn-Id: 3916 User-Agent: JsSIP 3.9.0 i try to lookup the location and find the address, and then route to that address by setting $ru in sngrep it does show that kamailio is trying to send INVITE to 11.22.33.44:port (which is the VPN gateway) but it isn't exactly clear from logs if this is being done over an existing connection. how does kamailio identify if a connection exists for a user (for each contact in AOR, if multiple contacts exist for a user -- i.e. different browser sessions?) if(reg_fetch_contacts("location", "$ru", "caller")) { $var(i) = 0; while($var(i) < $(ulc(caller=>count))) { $ru = $(ulc(caller=>addr)[$var(i)]); $var(i) = $var(i) + 1; setflag(FLT_DLGINFO); dlg_manage(); t_relay(); } } -- Shwetank

3 2

Performing async_http calls on stateless Loadbalancers
by Julius Flohr 09 May '23

09 May '23

Hi List, we are running several stateless SIP Loadbalancers using Kamaillio 5.1 and 5.6. Now we would like to perform an async_http request if an INVITE matches certain criteria. The request itself is performed very similar to how it's described here [1] This uses the transaction module in order to assure that the response can be mapped to the original request. However, this also means that the processing of the entire request becomes stateful which we would like to avoid. We tried to use t_release() in the callback route (in [1] this route is called "RELAY_API_RESPONSE"). However, documentation says that this is only allowed in the REQUEST_ROUTE. Is there an easy way to circumnavigate this issue? Cheers [1]: https://wazo-platform.org/blog/kamailio-routing-with-rtjson-and-http-async-… -- Julius Flohr - Voice Engineer - flohr(a)sipgate.de Telefon: +49 (0)211-63 55 55-0 Mobil: +49 (0)1579-000-0205 Telefax: +49 (0)211-63 55 55-22 sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.co.uk

3 3

Upgrade to 5.6.4 => [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer
by Benoît Panizzon 08 May '23

08 May '23

Hi team I upgraded our lab to 5.6.4 from the official repository for ubuntu Jammy. Now I get crashes related to TLS: May 5 14:58:40 dev-cpereg01 kamailio[36147]: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f4fedbed4e8), called from tls: tls_init.c: ser_free(323), first free tls: tls_init.c: ser_f ree(323) - ignoring May 5 14:58:40 dev-cpereg01 kamailio[36147]: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f4fedd72b88), called from tls: tls_init.c: ser_free(323), first free tls: tls_init.c: ser_f ree(323) - ignoring May 5 14:58:40 dev-cpereg01 kamailio[36177]: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f4fedd9e4a0), called from tls: tls_init.c: ser_free(323), first free tls: tls_init.c: ser_f ree(323) - ignoring May 5 14:58:40 dev-cpereg01 kamailio[36157]: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f4fedd9e428), called from tls: tls_init.c: ser_free(323), first free tls: tls_init.c: ser_f ree(323) - ignoring And a bit later on: May 5 14:58:46 dev-cpereg01 kamailio[36179]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 34 May 5 14:58:46 dev-cpereg01 kamailio[36143]: ALERT: <core> [main.c:774]: handle_sigs(): child process 36161 exited by a signal 11 May 5 14:58:46 dev-cpereg01 kamailio[36143]: ALERT: <core> [main.c:777]: handle_sigs(): core was generated May 5 14:58:46 dev-cpereg01 kamailio[36143]: INFO: <core> [main.c:799]: handle_sigs(): terminating due to SIGCHLD Has there been a fundamental change in TLS? Or did I miss something important to do on a 5.5 => 5.6 upgrade? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

4 5

TLS1.2 ciphers
by Sergiu Pojoga 05 May '23

05 May '23

Hi gang, Trying to figure out why Kamailio won't negotiate specific TLS1.2 ciphers, such as for example TLS_ECDHE_RSA_WITH_AES_256_GCM_SHA384 i.e. *ECDHE-RSA-AES256-GCM-SHA384* OpenSSL lists the cipher as supported, however Kamailio won't negotiate it: # nmap --script ssl-enum-ciphers -p 5061 mysbc.domain.com PORT STATE SERVICE 5061/tcp open sip-tls | ssl-enum-ciphers: | TLSv1.2: | ciphers: | TLS_RSA_WITH_AES_256_GCM_SHA384 (rsa 2048) - A | TLS_RSA_WITH_AES_256_CBC_SHA256 (rsa 2048) - A | TLS_RSA_WITH_AES_256_CBC_SHA (rsa 2048) - A | TLS_RSA_WITH_CAMELLIA_256_CBC_SHA (rsa 2048) - A | TLS_RSA_WITH_AES_128_GCM_SHA256 (rsa 2048) - A | TLS_RSA_WITH_AES_128_CBC_SHA256 (rsa 2048) - A | TLS_RSA_WITH_AES_128_CBC_SHA (rsa 2048) - A | TLS_RSA_WITH_SEED_CBC_SHA (rsa 2048) - A | TLS_RSA_WITH_CAMELLIA_128_CBC_SHA (rsa 2048) - A | TLS_RSA_WITH_RC4_128_SHA (rsa 2048) - C | TLS_RSA_WITH_RC4_128_MD5 (rsa 2048) - C | TLS_RSA_WITH_3DES_EDE_CBC_SHA (rsa 2048) - C | compressors: | NULL | cipher preference: server | warnings: | 64-bit block cipher 3DES vulnerable to SWEET32 attack | Broken cipher RC4 is deprecated by RFC 7465 | Ciphersuite uses MD5 for message integrity |_ least strength: C OS: - outdated Debian 8 Jessie - OpenSSL 1.0.1t 3 May 2016 - openssl ciphers -v | grep '*ECDHE-RSA-AES256-GCM-SHA384*' - ECDHE-RSA-AES256-GCM-SHA384 TLSv1.2 Kx=ECDH Au=RSA Enc=AESGCM(256) Mac=AEAD # kamailio -V version: kamailio 5.5.6 (x86_64/linux) ad1244 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. compiled on 21:08:20 Apr 13 2023 with gcc 4.9.2 modparam("tls", "cipher_list", "ALL") method = TLSv1.2+ Elliptic Curve Diffie-Hellman (EDCH) <https://kamailio.org/docs/modules/devel/modules/tls.html#tls.compile>-Ciphers are only supported in OpenSSL 1.0.0e and later. Any suggestions? Thanks.

5 5

Using a second Perl library
by David Cunningham 04 May '23

04 May '23

Hello, We are using a product which already uses the "app_perl" module and sets the "filename" parameter to its own Perl library. However, we want to add more Perl code without having to modify that product's library. Does anyone know of a way to do this? To have Kamailio use Perl code from two different places? I guess in the worst case scenario we could use "app_python3", and have some Python code call subprocess.run() on the Perl code. Thanks very much, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

1 0

Development is open for future release of v5.8.x series
by Daniel-Constantin Mierla 04 May '23

04 May '23

Hello, the branch 5.7 was created, therefore the master branch is open for adding new features, to be part of future release series v5.8.x (or whatever version is decided for next series). Any bug fix committed to master that applies to 5.7.x or older stable branches should be backported as usual with "git cherry-pick -x ..." to appropriate branches like 5.7 or 5.6. Expect that v5.7.0 will be released in a few weeks from now. Based on the workflow used during the past years, the next future release v5.8.0 should be out after another 8-10 months of development, plus 1-2 months of testing, so sometime during the first part of 2024. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com

1 0

Branch 5.7 has been created
by Daniel-Constantin Mierla 04 May '23

04 May '23

Hello, the branch 5.7 has been created, to be used for releasing v5.7.x series. To check out this branch, the following commands can be used: git clone https://github.com/kamailio/kamailio kamailio-5.7 cd kamailio-5.7 git checkout -b 5.7 origin/5.7 Pushing commits in this branch: git push origin 5.7:5.7 Note that 5.7 is an official stable branch, so only bug fixes, missing kemi exports (discuss on sr-dev if not sure) or improvements to documentation or helper tools can be pushed to this branch. As usual, if there is a bug fixed, commit and push first to master branch and then cherry pick to 5.7 branch: git cherry-pick -x COMMITID In few weeks, the first release from branch 5.7 will be out, respectively Kamailio v5.7.0. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com

1 0

send reply from branch route?
by Benoit Panizzon 04 May '23

04 May '23

Hello Is there a possibility to send a reply from within a branch route? Experimental case (yes, I know it's a non working set-up, but I just wonder if that would be possible). Two registrar nodes, location information shared between the two. CPE registers via TLS, therefore the existing connection shall be used towards the CPE and that only exists on one of the registrars. So in the Branch Route I would like to do something like: $var(socket) = $(ulc(aor=>socket)[$T_branch_idx]); if ($var(socket) == 0) { send_reply("503","no local socket"); } This would cause the core routing instance to select the next registrar from the dispatcher list which hopefully holds the active socket. But none of the 'send_reply' variants seem to be allowed within a branch route. Or is there a better solution? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

3 6

Branching 5.7 on git repo for 5.7.x releases
by Daniel-Constantin Mierla 04 May '23

04 May '23

Hello, I am planning to split 5.7.x series on its own git branch 5.7 on Thursday, May 4, 2023. After that, master branch will be open for new features (to become part of 5.8 or what ever version will be decided to be next). Usually testing should continue for branch 5.7 for another 1-2 weeks, then, if no relevant road blockers pop up, we should release 5.7.0 as first stable version in the 5.7.x series. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com

1 1

uac.reg_add to database
by Jon Bonilla (Manwe) 04 May '23

04 May '23

Hi 5.6.4 I'm trying to add uacreg logic to my setup. When I add a new entry via rpc I see that the reg is stored in mem but not inserted in the database for persistency. My uac params are like this: modparam("uac", "reg_db_url",DBURL) modparam("uac", "reg_timer_interval", 900) modparam("uac", "reg_retry_interval", 10) modparam("uac", "reg_random_delay", 0) modparam("uac", "reg_hash_size", 12) ## 5.6 modparam("uac", "reg_db_table", "kam_uacreg") modparam("uac", "reg_contact_addr", "192.168.255.254:5060") #Won't be used #modparam("uac", "default_socket", "udp:192.168.0.125:5060") #Won't be used modparam("uac", "reg_keep_callid", 1) modparam("uac", "reg_active", 1) modparam("uac", "reg_gc_interval", 30) modparam("uac", "reg_use_domain", 0) I'm used to add via rpc and have db persistency. Like for exmaple using the domain module. Is this a bug in my config or does this module behave different? Should I insert in the db and call to enable via RPC? If so, what would be the best way to enable only the new inserted register instead of reloading all the regs? It's expected to have many changes and do not want to have full reloads all the time. cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

1 0

Kamailio - asterisk - ACK
by Péter Dr. Barabás 03 May '23

03 May '23

Hi all, we are developing a softphone mobile application which registers to asterisk through kamailio. Kamailo proxies the request to asterisk and replies from asterisk to the clients. Registration works fine with asterisk authentication. The problem appears in the following flow: 1. Mobile client A calls client B. 2. Asterisk gets INVITE through kamailo, sends back 401, and get INVITE with credentials. 3. Asterisk sends invite to client B towards kamailio. 4. Kamailio forwards INVITE if client is registered or sends a push notification to client B. When client B registers, kamailio continue the suspended transaction and forwards INVITE. 5. Client B gets INVITE, sends a SIP OK. 6. Asterisk gets SIP OK through kamailio and sends an ACK to client B, which is OK. 7. Asterisk sends the SIP OK to client A through kamailio. 8. Kamailio forwards SIP OK to client A but the contact address contains the asterisk IP instead kamailio. 9. Clilent A try to send ACK but it does not arrive to asterisk, and the call will be hungup after 30 secs. So the question is what is missing, what is the correct solution? I try to use the provided configuration given in following link: https://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb The difference is that kamailio has no access to asterisk database. Thanks for your help. With kind regards, Peter

3 3

RFC: selective backports to latest stable as a dedicated git branch
by Daniel-Constantin Mierla 03 May '23

03 May '23

Hello, during past on-site/online devel meetings as well as on mailing lists or tracker, there were various discussions around the idea or requests to make new features available quicker to stable branches. Of course, one can do local git cherry-picking, but I thought to bring this in discussion and see if makes sense to get something more coordinated between developers and community. Therefore I am cross-posting to sr-dev and sr-users to see if there is interest for it from both communities. Somehow inspired from Debian, my first idea would be to create a "x.y-backports" branch, where "x.y" is the branch for the stable release series. For example, with 5.6 release series built from branch "5.6", there could be "5.6-backports" branch which is kept in sync in "5.6" but also can get "selected" new features from devel (master branch) backported. The "selected" new features should be mostly a matter of the people willing to put effort in backporting, but I would consider a list recommendations not to end up with devel and backports branches being more or less the same. For example, in the "no-backporting" list: - no backporting of completely new modules - no backporting of significant changes to the config file language and native scripting interpreter In the "ok-to-backport": - updates to enable use with newer versions of external libraries - changes to make some functions/modules to cope better with the core infrastructure or end points Commits to the backports branch can be proposed via pull requests. The backports branch should be done only for latest stable version, picking from the development version. Right now it would be 5.6, but we can wait till 5.7 is released and then have it for it. No packaging and no official releases should be made from backports branch, only a git branch to be maintained as a community effort. Of course, if someone wants to put more resources into it, we can discuss about. Anyhow, the first questions would be if such branch sounds good to have and if there are people that think they can also contribute to maintain it. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com

3 2

Clarification of 'return false' behaviour
by Patrick Wakano 03 May '23

03 May '23

Hi list, Hope you are all well! I'm using Kamailio version 5.6.4 (installed from the repo rpm.kamailio.org/centos/7) and noticed that every route that uses "return false" is exiting the script, instead of returning.... This was not the case on version 5.4.6 as the same script is running fine..... From this page https://www.kamailio.org/wikidocs/tutorials/faq/main/#how-is-the-function-r…, I would think that when a route returns false, it would return -1 and not stop execution, since negative is equal to false, but it is actually stopping (same as return 0)... So, as an example, this test code doesn't work as expected. In case the source is a public IP, the script doesn't print the "SRC public" it just exits and then of course every other logic meant to be done is not executed.... route[is_src_private] { if (is_ip_rfc1918("$si")) { return true; } return false; #return is_ip_rfc1918("$si"); *# this doesn't work too in case the $si is a public IP* } request_route { ... if (route(is_src_private)) { xlog("L_NOTICE", "SRC private\n"); } else { xlog("L_NOTICE", "SRC public\n"); } ... } If is_src_private is changed to return -1 instead of false, then it all works fine. Also, I noticed that the following code will print "TEST: 0" in case the $si is public and then stop execution. So looks like false is really being converted to 0, but I guess that's unexpected... anyway apologies if I'm missing something obvious.... route[is_src_private] { $var(t) = false; if (is_ip_rfc1918("$si")) { $var(t) = true; } xlog("L_ERR", "TEST: $var(t)\n"); return $var(t); } I could not find a recent ticket or email related to this situation and I've already spent hours trying to understand what is the logic/problem here, so would anyone have been across a similar case that could provide some insight and clarify what is the expected behaviour of the *false* usage (and boolean in general if possible)? Thank you, Kind regards, Patrick Wakano

3 6

rtpengine module: sdp offer in message reply, ip protocol version not passed to rtpengine?
by Benoit Panizzon 02 May '23

02 May '23

Hi Team I think, I might have found something that was overseen in the rtpengine module. A (ipv6) <=> rtpengine <=> B (ipv4) Invite+SDP from B to A It looks like this works as expected. A SDP c= line is created containing an ipv6 address. Invite, no SDP from A to B Reply with SDP offer from B to A The information about A being reachable via ipv6 and therefore an ipv6 c= line being required, could be taken from the via header of the reply. But am I right to fear, that the rtpengine module does not do this at the moment? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 2

Number if typical database connections
by Steve Wilkins 02 May '23

02 May '23

Hello, I moved the Kamailio database to RDS from a local instance and noticed that there were a large number of connections from the Kamailio user. What is the normal number of connection that can be expected? Thank you,

2 1

Rebooting kamailio.org server
by Daniel-Constantin Mierla 02 May '23

02 May '23

Hello, in about 1 hour the kamailio.org server will be rebooted for some system upgrades, it should be quickly back running, but if anyone accesses the web/wiki or mailing lists archives exactly during that time, they might not respond -- just wait for a minute and try again. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com

1 0

Re: [SR-Users] TOPOS (5.1.[23]) breaks PRACK
by Daniel Tryba 02 May '23

02 May '23

On Mon, May 07, 2018 at 04:44:14PM +0200, Daniel Tryba wrote: > Sure. Attached. Problem appears to be that the topos query can't find > callid-totag (from the response). > > I'll try the same scenario with the mysql backend to see if it behaves > different. Config works fine with mysql as topos backend. So the bug is restricted to topos-redis.

5 19

Loading the LCR Module Tables at startup/reload
by harneet singh 30 Apr '23

30 Apr '23

Hi All, Can someone point to a sample as to how we can *load the LCR Module tables* namely the following at startup from a csv file or any other mechanism. * lcr_gw_table * lcr_rule_table * lcr_rule_target_table The idea is to not need to manually feed in the entries into the table but instead add these in the beginning through either a *wrapper script *OR *through the kamailio cfg file* if possible. Any pointers are much appreciated. Thanks & Regards, Harneet Singh -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

2 3

Route Header based routing for initial INVITE
by harneet singh 28 Apr '23

28 Apr '23

Hi Experts, I believe the loose_route() in kamailio can be used for initial Out of the dialog INVITE processing as well. Is that the correct understanding? If so, the request will be forwarded ahead to the destination corresponding to the Route Header remaining (if any) after Kamailio has stripped off it's own Address(es) from the Route Header(s). Is that correct? Regards, Harneet Singh -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

3 3

Strange RTCP issue
by Kiss Zoltán 27 Apr '23

27 Apr '23

Hello, We have a transparent proxy in front of Asterisk boxes. The media is handled by rtpengine, and the "manage" work is done by Kamailio in the route[NATMANAGE] section like this: if(nat_uac_test("8")) { # Packet from backend server if(ds_is_from_list()){ if (rtpengine_manage("replace-origin replace-session-connection direction=internal direction=pub")) { xdbg("NATMANAGE --- Success rtpengine_manage() RTP from INTERNAL to EXTERNAL $ru"); } } else { if (rtpengine_manage("replace-origin replace-session-connection direction=pub direction=internal")) { xdbg("NATMANAGE --- Success rtpengine_manage() RTP from EXTERNAL to INTERNAL $ru"); } } } else { # Packet from backend server if(ds_is_from_list()){ if (rtpengine_manage("replace-origin replace-session-connection trust-address direction=internal direction=pub")) { xdbg("NATMANAGE --- Success rtpengine_manage() RTP from INTERNAL to EXTERNAL $ru"); } } else { if (rtpengine_manage("replace-origin replace-session-connection trust-address direction=pub direction=internal")) { xdbg("NATMANAGE --- Success rtpengine_manage() RTP from EXTERNAL to INTERNAL $ru"); } } } Everything is working fine, but with some clients (like Grandstream phone) the RTCP session wants to go tot he private address of the phone. Here is the log of one of these strange calls: Apr 27 16:54:38 rtp1 rtpengine[2273]: INFO: [312ed76c31f21b71452e91e5184ad25b@172.16.2.210:5060]: [core] --------- Port 178.238.213.14:11088 <> 81.183.216.3:5068 , SSRC 77c19488, 625 p, 107500 b, 0 e, 29 ts Apr 27 16:54:38 rtp1 rtpengine[2273]: INFO: [312ed76c31f21b71452e91e5184ad25b@172.16.2.210:5060]: [core] --------- Port 178.238.213.14:11089 <> 10.0.5.192:5069 (RTCP), SSRC 0, 0 p, 0 b, 0 e, 43 ts As you can see the RTP itself is okay, but the RTCP will somehow go to the private address. If we tracing the sdp messages in sngrep, then we can see that Kamailio transforming IP addresses for the backend servers to the public (and the private address of the rtpengine) addresses. We are missing something? Thank you! With kind regards, Zoltan

2 1

Sensible values for pike mudule?
by Benoit Panizzon 27 Apr '23

27 Apr '23

Hi Before I go too deep into try and error, I guess others have been there too. What are sensible value you use for the pike module to detect / mitigate abusive behavior, especially dictionary attacks? Are there better solutions that the pike module? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 1

Fwd: Sensible values for pike mudule?
by Stefan 27 Apr '23

27 Apr '23

Hi Benoît We use pike in combination with fail2ban. Log a message -> create a fail2ban rule to block the ip for xxx time. example: if ($ua =~ "(friendly-scanner|sipvicious)") { sl_send_reply("488", "Go away."); xlog("L_INFO","$ci Blocking traffic from $si\n"); exit; } Cool thing is, you can write a log entry from every part in your kamailio-config, and then fail2ban does the rest. Regards Stefan Am 27.04.23 um 16:07 schrieb Benoit Panizzon: > Hi > > Before I go too deep into try and error, I guess others have been there > too. > > What are sensible value you use for the pike module to detect / > mitigate abusive behavior, especially dictionary attacks? > > Are there better solutions that the pike module? > > Mit freundlichen Grüssen > > -Benoît Panizzon-

1 0

hooking mechanism for routing blocks
by Valentin Christoph 27 Apr '23

27 Apr '23

Dear all, I am looking for some hooking mechanism for user defined routing blocks. Is something like this already available? 1) a common part of the routing configuration shall not (or only slightly) be changed, after some hooks have been implemented Something like a function "engage_hooks()" of some "hook_module"? : Routing config that should not be changed by application developers engage_hooks(HOOK_TYPE); Routing config that should not be changed by application developers : 2) one application developer may "hook" new parts of the routing configuration into this, by modparam("hook_module", "hook", "name=MY_RB;type=HOOK_TYPE;prio=50); : : Route [MY_RB] { Do something at hook HOOK_TYPE; } : : 3) another application developer might want to "hook" into the same HOOK_TYPE with a higher priority modparam("hook_module", "hook", "name=ANOTHER_RB;type=HOOK_TYPE;prio=20); : : Route [ANOTHER_RB] { Do another thing with higher priority at hook HOOK_TYPE } : : Kr, Christoph

2 2

Kamailio use case
by jdprsaga＠gmail.com 26 Apr '23

26 Apr '23

Hello All, new subscriber here. I've been reading kamailio documentation to check if kamailio has a use case on the architecture for the solution for my company's software, here an small architecture diagram of our solution. sipclient <----> [ Middleware ] <-----> SBC Our middle ware is both a sip proxy and media proxy. we have a large number of sip clients but a small calling number, our current bottleneck is that the middleware can't handle a high number of registration, i would like to deploy Kamailio as a load balancer for registrations, making sure it is statefull so registrations and calls are hashed into the same middleware server. sipclients < ----> [ kamailio ] <------> [ middleware1, middleware2,...,middleware N] <-----> SBC Any thoughts or pointers?

4 3

rtpengine module: Actions caused by rtpengine_manage() on reply-codes?
by Benoît Panizzon 26 Apr '23

26 Apr '23

Hi all I wonder, if there is a documentation on what action is performed by rtpengine_manage() on which kind reply. Am I right, that rtpengine_manage(), on any 4XX reply causes a delete? So if there is a re-invite (for example to switch to T.38) and the other party replies with 488 which in turn is passed through rtpengine_manage() the existing RTP stream is deleted from rtpengine instead of being left running with the previously agreed codec? Has anyone already figured out a list of what reply codes should be exempt from calling rtpengine_manage()? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

8 12

Configuring dynamic Destination identification for Kamailio Routing
by harneet singh 25 Apr '23

25 Apr '23

Hi Experts, We have been using Kamailio as a sip proxy for a fair bit of time now(version 5.3.2 though) and do understand the native routing mechanism that can be instrumented through the kamailio cfg scripts. We initially had only two sides, where the traffic could come from into Kamailio and it just passed through to the other side(ie Traffic from A was routed to B and that from B was routed to A). Off late, a few more sip elements have been introduced in the deployment and now there is a need for a more formal/robust mechanism to select the intended destination to forward an initial SIP INVITE to. I believe that we could still keep all the logic in the native kamailio script and do the routing(for example, an incoming call where the *From URI* contains *sip.elementA.com <http://sip.elementA.com>* should be routed to destination *sip.elementB.com <http://sip.elementB.com>*, where as if the* To URI *contains *sip.elementC.com <http://sip.elementC.com>*, the request should be routed to *sip.elementC.com <http://sip.elementC.com>)**. *Since we have many sip elements like A,B.C,D,E etc where the requests might have to be routed to, based on certain different criteria, this logic would become cumbersome. *What would be the best way to accommodate this?* *Would it be possible to rewire this routing logic at runtime(*like in some csv file that can be reloaded*) without a Kamailio restart? * Regards, Harneet Singh -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

3 5

sips to sip with TLS proxy
by Kiss Zoltán 25 Apr '23

25 Apr '23

Hi all, We have a working Kamailio setup, lets call it a transparent proxy for Asterisk boxes. Its based on domain and dispatcher modules and everything is working as expected with the test clients (more or less microsip, softphone for ios, etc). We are tried to register with a Grandstream deskphone today, and we see that the phone sending sips:xxx in the Reg Contact field for example. Because the sips schema, the register is working, but we cannot initiate calls from this phone. If we are turning SIP scheme to sip from sips in the phone, then everything is working as expected. I think we can transform those requests from sips to sip with Kamailio, but currently we dont know where can we start. Has anybody a suggestion about this issue? I know that we can transform ruri, contact, etc with textops, nathelper and a lot of other modules, but what is the best for this sips->sip translation? Thanks for your help. With kind regards, Zoltan

4 10

TCP/TLS connection (id: 0) for WebSocket could not be found
by Joey Golan 22 Apr '23

22 Apr '23

Hello, I have 2 clients. 1 using websocket and 1 mobile using TLS. Both can call each other perfectly. After 24-48 hours calls from mobile to websocket failed with an error message: TCP/TLS connection (id: 0) for WebSocket could not be found. At the same time I can make calls from the websocket client to the mobile. I know that the search of the connection is done by target address so I printed the addresses + ports and they are matched to the ones that appear on the ul.dump and ws.dump. *kamcmd ul.dump output:* Info: { AoR: usrmt63ly_1040 HashID: 1830036546 Contacts: { Contact: { Address: sip:usrmt63ly_1040@example.com;transport=ws Expires: 283 Q: -1 Call-ID: tvne1pskgkm2a01pnqm677 CSeq: 1060 User-Agent: JsSIP 3.9.0 Received: sip:*10.218.51.86:24061 <http://10.218.51.86:24061>*;transport=ws Path: <sip:10.218.154.228:50443;transport=ws;received=sip:*10.218.51.86:24061 <http://10.218.51.86:24061>*;transport=ws;lr> State: CS_NEW Flags: 0 CFlags: 64 Socket: tls:10.218.154.228:50443 Methods: 7071 Ruid: uloc-61ebba84-64242101-89ba-1 Instance: <urn:uuid:7573726d-7436-116c-915f-313034304077> Reg-Id: 1 Server-Id: 1642838660 Tcpconn-Id: 100420 Keepalive: 1 Last-Keepalive: 1680417282 KA-Roundtrip: 0 Last-Modified: 1680417282 } } } *kamcmd ws.dump* { connections: { 100420: wss:*10.218.51.86:24061 <http://10.218.51.86:24061>* -> wss: 10.218.154.228:50443 (state: OPEN, last used 14s ago, sub-protocol: sip) } info: { wscounter: 1 truncated: no } } *Call from mobile (TLS) to websocket: Failed with error message: TCP/TLS connection (id: 0) for WebSocket could not be found* 30(35263) INFO: {1 2563 INVITE 4248c155-cd92-49b8-b68a-eef5f2cb3e88} <script>: request_route: method [INVITE] from [ sip:usrrcv4o4_1040@example.com] socket [sip:10.218.51.86:50909;transport=tls] to [sip:usrmt63ly_1040@example.com] 30(35263) INFO: {1 2563 INVITE 4248c155-cd92-49b8-b68a-eef5f2cb3e88} <script>: [RELAY]: method [INVITE] from [sip:usrrcv4o4_1040@example.com] socket [sip:10.218.51.86:50909;transport=tls] to [ sip:usrmt63ly_1040@example.com] socket [sip:*10.218.51.86:24061 <http://10.218.51.86:24061>*;transport=ws] 30(35263) WARNING: {1 2563 INVITE 4248c155-cd92-49b8-b68a-eef5f2cb3e88} <core> [core/msg_translator.c:2863]: *via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found* 30(35263) ERROR: {1 2563 INVITE 4248c155-cd92-49b8-b68a-eef5f2cb3e88} <core> [core/msg_translator.c:2040]: build_req_buf_from_sip_req(): could not create Via header 30(35263) ERROR: {1 2563 INVITE 4248c155-cd92-49b8-b68a-eef5f2cb3e88} tm [t_fwd.c:479]: prepare_new_uac(): could not build request 30(35263) ERROR: {1 2563 INVITE 4248c155-cd92-49b8-b68a-eef5f2cb3e88} tm [t_fwd.c:1759]: t_forward_nonack(): failure to add branches 30(35263) ERROR: {1 2563 INVITE 4248c155-cd92-49b8-b68a-eef5f2cb3e88} sl [sl_funcs.c:418]: sl_reply_error(): stateless error reply used: No error (2/SL) 30(35263) NOTICE: {1 2563 INVITE 4248c155-cd92-49b8-b68a-eef5f2cb3e88} acc [acc.c:287]: acc_log_request(): ACC: call missed: timestamp=1680417920;method=INVITE;from_tag=e168ffa6-e577-48ec-b9c9-a73f5be318e8;to_tag=0c3e625e9f223a7ee2b093e8e31b2526-b7165bcb;call_id=4248c155-cd92-49b8-b68a-eef5f2cb3e88;code=500;reason=Server Internal Error;src_user=usrrcv4o4_1040;src_domain=example.com ;src_ip=10.218.51.86;dst_ouser=usrmt63ly_1040;dst_user=usrmt63ly_1040;dst_domain= example.com *Call from websocket to mobile (TLS):* 38(35271) INFO: {1 7488 INVITE urcnete8r71dpkhmngd8} <script>: request_route: method [INVITE] from [sip:usrmt63ly_1040@example.com] socket [sip:*10.218.51.86:24061 <http://10.218.51.86:24061>*;transport=ws] to [ sip:usrrcv4o4_1040@example.com] 38(35271) INFO: {1 7488 INVITE urcnete8r71dpkhmngd8} <script>: [RELAY]: method [INVITE] from [sip:usrmt63ly_1040@example.com] socket [sip:*10.218.51.86:24061 <http://10.218.51.86:24061>*;transport=ws] to [ sip:usrrcv4o4_1040@example.com] socket [sip:10.218.51.86:20674 ;transport=tls] 39(35272) INFO: {2 7488 INVITE urcnete8r71dpkhmngd8} <script>: onreply_route[MANAGE_REPLY] method [INVITE] code [100] from [ sip:usrmt63ly_1040@example.com] to [sip:usrrcv4o4_1040@example.com] 39(35272) INFO: {2 7488 INVITE urcnete8r71dpkhmngd8} <script>: onreply_route[MANAGE_REPLY] method [INVITE] code [200] from [ sip:usrmt63ly_1040@example.com] to [sip:usrrcv4o4_1040@example.com] 39(35272) NOTICE: {2 7488 INVITE urcnete8r71dpkhmngd8} acc [acc.c:287]: acc_log_request(): ACC: transaction answered: timestamp=1680417926;method=INVITE;from_tag=18q0p37nfo;to_tag=a88272a0-25ca-43c2-8ce9-8615831b7e9a;call_id=urcnete8r71dpkhmngd8;code=200;reason=OK;src_user=usrmt63ly_1040;src_domain= example.com ;src_ip=10.218.51.86;dst_ouser=usrrcv4o4_1040;dst_user=usrrcv4o4_1040;dst_domain=10.218.51.86 38(35271) INFO: {1 7488 ACK urcnete8r71dpkhmngd8} <script>: request_route: method [ACK] from [sip:usrmt63ly_1040@example.com] socket [sip:*10.218.51.86:24061 <http://10.218.51.86:24061>*;transport=ws] to [ sip:usrrcv4o4_1040@example.com] 38(35271) INFO: {1 7488 ACK urcnete8r71dpkhmngd8} <script>: [RELAY]: method [ACK] from [sip:usrmt63ly_1040@example.com] socket [sip:*10.218.51.86:24061 <http://10.218.51.86:24061>*;transport=ws] to [ sip:usrrcv4o4_1040@example.com] socket [sip:10.218.51.86:20674 ;transport=tls] *Version: kamailio 5.6.2 (x86_64/linux)* What could be the reason for that? Your help would be much appreciated. Thanks, Joey.

3 4

How to log expiring dialogue?
by Benoît Panizzon 21 Apr '23

21 Apr '23

Hi alle I think I have an issue with the dialogue and session timer module with Kamailio 5.4 I sync the Session Expire Timer with the dialogue timeout via a common AVP. When I check a dialogue via kamcmd dlg.list start_ts: 1682066352 init_ts: 1682066346 end_ts: 0 timeout: 1682066952 lifetime: 600 Timeout is increased on the first session refreshing re-invite, as expected (in this case after 300 seconds). But it looks like the dialogue just disappears (from kamcmd dlg.list, not from the backing database) when the lifetime ist reached, leaving an orphan database entry. I would like to digg deeper into this. Is there a way to log more details from the dialogue module like when it triggers dialogue expiry? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 2

Roadmap to next major release series v5.7.x
by Daniel-Constantin Mierla 20 Apr '23

20 Apr '23

Hello, it is time to make the roadmap to the next major release series v5.7.x. In my opinion, I would try to freeze the development during the first part of April 2023, tentatively proposing April 14 as freezing date, being like 4 weeks from now. Then testing phase and first release (v5.7.0) sometime during May 2023. Alternative timelines can be proposed, of course. If anyone wants to add new features/modules, they have to be published till freezing date, either pushed in the git repository or proposed as pull request. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

3 5

Rtpengine: no audio after kernelization.
by Michel Pelletier 17 Apr '23

17 Apr '23

Hello, I am proxying all RTP through RTPEngine. Everything works fine until about 5 seconds into the call, when rtpengine enters kernelization, after which all RTP forwarding ceases. I've checked the required iptables entries, and all looks good. Here is a description of my environment: # cat os-release PRETTY_NAME="Debian GNU/Linux 11 (bullseye)" NAME="Debian GNU/Linux" VERSION_ID="11" VERSION="11 (bullseye)" VERSION_CODENAME=bullseye ID=debian HOME_URL="https://www.debian.org/" SUPPORT_URL="https://www.debian.org/support" BUG_REPORT_URL="https://bugs.debian.org/" # kamailio -v version: kamailio 5.6.2 (x86_64/linux) 54a9c1 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 54a9c1 compiled on 16:02:49 Dec 1 2022 with gcc 10.2.1 # rtpengine -v Version: 11.1.1.3-1~bpo11+1 Michel Pelletier

4 13

Docker repository removed
by Manel Villar 17 Apr '23

17 Apr '23

Hi, I was using the kamailio/kamailio docker repository in one of my projects, and now the image is not available as the repo has been removed. Is the "kamailio-ci" repo a replacement for it? Thanks! -- Manel Villar Steegerstr. 59 13359 Berlin Germany Tel: (+49) (0)162 6574604 skype: manel.villar manelvf(a)gmail.com *"Ars longa, vita brevis"*

4 5

Development for v5.7.x is frozen
by Daniel-Constantin Mierla 17 Apr '23

17 Apr '23

Hello, the formal notification that the development for the next major version 5.7.0 is now frozen. The focus has to be on testing the master branch. Also, the master branch should not get commits with new features till the branch 5.7 is created, expected to happen in 2-4 weeks, a matter of how testing goes on. Meanwhile, the commits with new features in the C code can be pushed to personal branches, new pull requests can still be done, but they will be merged after branching 5.7. Can still be done commits with documentation improvements, enhancements to related tools (e.g., kamctl, kamcmd), merging exiting pull requests at this moment, exporting missing KEMI functions and completing the functionality of the new modules added for 5.7. Once the branch 5.7 is created, new features can be pushed again to master branch as usual. From that moment, the v5.7.0 should be out very soon, time used for further testing but also preparing the release of packages. If someone is not sure if a commit brings a new feature, just make a pull request and it can be discussed there on github portal or via sr-dev mailing list. A summary of what is new in upcoming 5.7 is going to be built at: * https://www.kamailio.org/wikidocs/features/new-in-5.7.x/ Upgrade guidelines will be collected at: * https://www.kamailio.org/wikidocs/install/upgrade/5.6.x-to-5.7.0/ Everyone is more than welcome to contribute to the above wiki pages, especially to the upgrade guidelines, to help everyone else during the migration process from v5.6.x to 5.7.x. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 2

Assigning a new RTPEngine for a call for the sake of failover
by mehdithreem+kamailio＠gmail.com 15 Apr '23

15 Apr '23

Hello, I am trying to implement RTPEngine failover, where if a selected RTPEngine node fails in the middle of the call (and not responding NG requests), I want to select an alternative node. However, when I call rtpengine_manage(), it does not seem to select an alternative node in case of a failure. To address this, I have attempted to check if $avp(RTP_INSTANCE) is null after calling rtpengine_manage(), and then use rtpengine_delete() to forget the selected RTPEngine node for the specific call, with the expectation that Kamailio would assign a new RTPEngine node for subsequent SIP message retries. However, it appears that rtpengine_delete() does not have the desired effect in this regard. Here is an excerpt of my code for reference: ``` route[RTP] { #!ifdef WITH_RTPENGINE if (has_body("application/sdp")) { if ($avp(x_source) == "INTERNAL") { rtpengine_manage("replace-session-connection replace-origin direction=internal direction=external"); } else if ($avp(x_source) == "EXTERNAL") { rtpengine_manage("replace-session-connection replace-origin direction=external direction=internal"); } if ($avp(RTP_INSTANCE) == $null) { rtpengine_delete(); drop(); } } #!endif } ``` Can you please advise on how I can achieve the desired failover behavior? Is there another function similar to `rtpengine_reset_call` that can make Kamailio forget the RTPEngine node for a specific call? Thank you!

2 1

CLI & destination based LCR
by John Cahill 14 Apr '23

14 Apr '23

Hi, I'd like to build an LCR solution that routes on both CLI & destination e.g. CLI matches ^\+49*, destinations matches ^\+44* => route to gateway 5 for a large rule set (~ tens of thousands). Any recommendations for which modules to use? Can kamailio perform this particular task well or should I look at another solution? I'd be very grateful to hear about any experience of doing this. Many Thanks. Regards, John

5 5

Kamailio + PostgreSQL
by georgy.grigoryev＠snapcom.fr 14 Apr '23

14 Apr '23

Hello everyone, I'm trying to connect Kamailio 5.5.5 to my PostgreSQL 14. They are on two different servers. I have created the Kamailio DB on the Postgre with the help of the command "kamdbctl create" I'm able to connect to the database with no problem. The problem is in the kamailio.cfg file, i have added this so far: #!define WITH_PGSQL #!ifdef WITH_PGSQL #!define DBURL "postgres://USER:PASSWORD@IP:POPT/kamailio" #!endif #!ifdef WITH_PGSQL loadmodule "db_postgres.so" #!endif modparam("db_postgres", "retries", 3) modparam("db_postgres", "timeout", 10) modparam("db_postgres", "tcp_keepalive", 600) modparam("db_postgres", "lockset", 6) modparam("db_postgres", "bytea_output_escape", 0) But my kamailio doenst like this configuration, can someone help me find the problem? Thank you in advance! Best Regards!

3 2

manipulate Authentication header
by David Villasmil 13 Apr '23

13 Apr '23

Hello guys, I can't find a simple way of changing the username and recalculate the has in a REGISTER... do i have to use textops? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 1

How to correctly use the SST module?
by Benoît Panizzon 11 Apr '23

11 Apr '23

Hi I still have odd behaviour with the SST module: modparam("sst", "min_se", 300) if (sstCheckMin("1")) { xlog("L_ERR", "$cfg(route) Session Timer too small\n"); exit; } setflag(FLT_DLG); # set the dialog flag setflag(FLT_SST); # Set the sst flag if(is_present_hf("Session-Expires")) { $avp(dlgtimeout) = $(hdr(Session-Expires){s.int}); } On an INVITE containing those header: Session-Expires: 3600;refresher=uac Min-SE: 600 Supported: 100rel,timer Expires: 330 Now Kamailio replies: SIP/2.0 422 Session Timer Too Small Min-SE: 300 What am I doing wrong? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

1 0

dlg_onroute(): unable to find dialog for INVITE with route param
by Benoît Panizzon 10 Apr '23

10 Apr '23

Happy Eastern Kamailio 5.4 used. On dialogues calls, I noticed this happening on the second re-invite to refresh the session from the B side. dlg_onroute(): unable to find dialog for INVITE with route param 'df9.da75' [2557:22445] and call-id '14148a68-f1b2-485d-b5a2-f9dd7773d2c0' The did df9.da75 is identical to the one in the route-header on the first re-invite from the B side, before this one. No CDR was (yet) generated, and the dialogues is still present and on-going according to the database: CallID [14148a68-f1b2-485d-b5a2-f9dd7773d2c0] H-Entry: 2557 H-ID: 22445 But most probably not in memory as kamcmd dlg.stat_active shows zero (this is on a development node, so no other calls running). There are no sip messages that end the call in between those two re-invites. I see no session timeouts or similar in the logs. Any hint what could cause Kamailio not to find the corresponding dialogue on this re-invite? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

1 2

Fwd: Kamailio + PostgreSQL
by Stefan 07 Apr '23

07 Apr '23

Hi Give it a try without modparams to see if it works, I have the same config but without modparams and it runs without problems. Else have a look into the kamailio logfile for errors. Or check config with "kamailio -c"? Regards Stefan Am 06.04.23 um 15:18 schrieb georgy.grigoryev(a)snapcom.fr: > Hello everyone, > > I'm trying to connect Kamailio 5.5.5 to my PostgreSQL 14. > They are on two different servers. > I have created the Kamailio DB on the Postgre with the help of the > command "kamdbctl create" > I'm able to connect to the database with no problem. > The problem is in the kamailio.cfg file, i have added this so far: > > #!define WITH_PGSQL > > #!ifdef WITH_PGSQL > #!define DBURL "postgres://USER:PASSWORD@IP:POPT/kamailio" > #!endif > > #!ifdef WITH_PGSQL > loadmodule "db_postgres.so" > #!endif > > modparam("db_postgres", "retries", 3) > modparam("db_postgres", "timeout", 10) > modparam("db_postgres", "tcp_keepalive", 600) > modparam("db_postgres", "lockset", 6) > modparam("db_postgres", "bytea_output_escape", 0) > > But my kamailio doenst like this configuration, can someone help me > find the problem? > > Thank you in advance! > > Best Regards! > __________________________________________________________ > Kamailio - Users Mailing List - Non Commercial Discussions > To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only > to the sender! > Edit mailing list options or unsubscribe:

2 2

Malformed To Header
by Serdar GUCLUER 06 Apr '23

06 Apr '23

Hello all, In my situation, it is possible to call forwarding to registered users multiple times on same leg. For each forward i have to update To uri and i am using uac_replace_to to do that. But on second relaying, To uri was malformed. On below, Im attaching configuration to illustrate my problem. (version-5.4.9) First relay - /To: //sip:xyz@192.168.0.51/ Second relay - /To: sip:xyz@192.168.0.51sip:abc@192.168.0.51/ Thanks in advance. Serdar /loadmodule "uac.so"// //modparam("uac","restore_mode","auto")// // // //request_route {// // ...// // $avp(x) = "abc";// // $avp(x) = "xyz";// // $avp(myip) = "192.168.0.51";// // $avp(i) = 0;// // route(SERDAR);// // ...// //}// // //route[SERDAR] {// // if ($(avp(x)[$avp(i)])!=$null) {// // $avp(destination) = $(avp(x)[$avp(i)]);// // $avp(i) = $avp(i) + 1;// // $rU=$_s($avp(destination));// //uac_replace_to("$_s(sip:$avp(destination)@$avp(myip))");// // route(LOCATION);// // } else {// // send_reply("403", "Opps!");// // exit;// // }// //}// // //route[LOCATION] {// // $avp(oexten) = $rU;// // if (!lookup("location") ) {// // t_newtran();// // switch ($var(rc)) {// // case -1:// // case -3:// // send_reply("480", "Temporarily Unavailable");// // exit;// // case -2:// // send_reply("405", "Method Not Allowed");// // exit;// // }// // }// // t_on_failure("SERDAR_FAILURE");// // if(!t_relay()) sl_reply_error();// // exit;// //}// // //failure_route[SERDAR_FAILURE] {// // route(NATMANAGE);// // if (t_is_canceled()) //exit;// // route(SERDAR);// //}/

6 11

Presence module and privacy
by Ihor Olkhovskyi 06 Apr '23

06 Apr '23

Hello! Is there any possibility to control more-or-less privacy related parameters of pua_dialoginfo module (like include_localremote) per subscription? Main idea is I want to show some users full info on the call other user is making, for a different user - just show fact of the call without any details. For now I'm only seeing some manual manipulation over XML based on user who is receiving NOTIFY, but maybe I'm trying to invent a weel here. -- Best regards, Ihor (Igor)

2 2

Re: Repository for modules
by Henning Westerholt 06 Apr '23

06 Apr '23

(please keep the list in the communication) Hello, you can find the modules for example in the Debian package service: https://packages.debian.org/bullseye/amd64/kamailio-tls-modules/filelist Crypto is in TLS module package. Cheers, Henning From: Duarte Rocha <duarterocha91(a)gmail.com> Sent: Donnerstag, 6. April 2023 15:42 To: Henning Westerholt <hw(a)gilawa.com> Subject: Re: [SR-Users] Repository for modules Thanks for the quick answer. How can I see info regarding which modules are in the combined packages? One of the modules i'm looking into is the crypto module for example. Otherwise, where can I get a list of modules not included in Debian packages? Cheers, Duarte Henning Westerholt <hw(a)gilawa.com<mailto:hw@gilawa.com>> escreveu no dia quinta, 6/04/2023 à(s) 13:17: Hello, the majority of modules should be packaged for Debian. What modules are missing for you? Keep in mind that some modules are combined in some packages. Some modules can not properly packaged due to missing dependencies, this needs to be installed from source, right. Cheers, Henning From: Duarte Rocha <duarterocha91(a)gmail.com<mailto:duarterocha91@gmail.com>> Sent: Donnerstag, 6. April 2023 13:55 To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> Subject: [SR-Users] Repository for modules Hello, I'm currently testing Kamailio on a Debian 11 machine. In order to install it I used apt with repositories from Kamailio (https://deb.kamailio.org/). Now I'm trying to add modules and I'm wondering the best way to do it. Using apt install i don't have access to all the modules. Is there another repository for this? Is my only solution downloading and compiling the modules? Cheers

1 0

Repository for modules
by Duarte Rocha 06 Apr '23

06 Apr '23

Hello, I'm currently testing Kamailio on a Debian 11 machine. In order to install it I used apt with repositories from Kamailio (https://deb.kamailio.org/). Now I'm trying to add modules and I'm wondering the best way to do it. Using apt install i don't have access to all the modules. Is there another repository for this? Is my only solution downloading and compiling the modules? Cheers

2 2

issue in kamailio of udp
by satyaprakash ch 06 Apr '23

06 Apr '23

Hi, I had upgraded Kamailio from 5.4.7 to 5.5.5, Then I restarted the Kamailio service. After upgradation im getting the below error, /usr/local/sbin/kamailio[21754]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock, buf: 0x7f7502289200, len: 61012, 0, dst: ( 10.122.45.189:5080), tolen: 16) - err: No buffer space available (105) /usr/local/sbin/kamailio[21754]: ERROR: tm [../../core/forward.h:231]: msg_send_buffer(): udp_send failed /usr/local/sbin/kamailio[21754]: ERROR: tm [uac.c:678]: send_prepared_request_impl(): Attempt to send to precreated request failed ERROR: tm [../../core/forward.h:231]: msg_send_buffer(): udp_send failed ERROR: tm [uac.c:678]: send_prepared_request_impl(): Attempt to send to precreated request failed Anyone have any idea on this errors.

2 3

SCSCF hangs randomly
by solomon.thoobe＠smilecoms.com 05 Apr '23

05 Apr '23

Hi All, We currently experience issues where scscf hangs randomly throughout the day. It started hanging once in a month, the issue has gotten worse over time and now it can hang more than 5 times in a day. I have no prior kamailio experience or experience debugging c applications. Kamailio version: 4.4.0 Running ps shows scscf processes are running. Service status also shows scscf is running but it does not respond to sip requests. Testing with sipsack hangs and print timeout errors. root@NGVI-PCSCF2:/var/log # service scscf status * Status of Kamailio SIP Server: * scscf is running sipsak -vv -s sip:NGVI-PCSCF2:6060 received ICMP message received ICMP message** timeout after 500 ms** Below is the extract from log files Apr 5 12:04:46 NGVI-PCSCF2 /usr/sbin/kamailio[5102]: {1 1 ACK isbc505ramjjjr3g23cs02r53cms0uxbg2es(a)10.18.5.64} WARNING: tm [t_lookup.c:1476]: t_unref(): WARNING: script writer didn't release transaction Apr 5 12:04:46 NGVI-PCSCF2 /usr/sbin/kamailio[5104]: {1 3 REGISTER _A9LGREyW2miE7LhPJzv7g..} ERROR: ims_registrar_scscf [save.c:1034]: update_contacts(): Unable to remove contact <sip:+2347020246874@154.66.2.182:5060;rinstance=28bf37d2cce6e4a1;transport=udp> Apr 5 12:04:47 NGVI-PCSCF2 /usr/sbin/kamailio[5131]: ERROR: rr [rr_mod.c:300]: w_record_route(): Double attempt to record-route Apr 5 12:04:47 NGVI-PCSCF2 /usr/sbin/kamailio[5144]: ERROR: rr [rr_mod.c:300]: w_record_route(): Double attempt to record-route Apr 5 12:04:47 NGVI-PCSCF2 /usr/sbin/kamailio[5108]: {1 1 ACK r8ug398d118i8iaaunrouggisntkigkn(a)10.18.5.64} WARNING: tm [t_lookup.c:1476]: t_unref(): WARNING: script writer didn't release transaction Apr 5 12:04:48 NGVI-PCSCF2 /usr/sbin/kamailio[5190]: CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 23 Apr 5 12:04:48 NGVI-PCSCF2 /usr/sbin/kamailio[5062]: ALERT: <core> [main.c:741]: handle_sigs(): child process 5103 exited by a signal 11 Apr 5 12:04:48 NGVI-PCSCF2 /usr/sbin/kamailio[5062]: ALERT: <core> [main.c:744]: handle_sigs(): core was generated Apr 5 12:06:12 NGVI-PCSCF2 kamailio: WARNING: <core> [daemonize.c:348]: daemonize(): pid file contains old pid, replacing pid Apr 5 12:06:13 NGVI-PCSCF2 /usr/sbin/kamailio[15060]: WARNING: interconnectroute [ported_cache.c:113]: initPortedNumberCount(): Requested records[726049]: Total allocated[871258]: Fat 20% :Total MemUsed [26 MB divided between primary[0x7f22b8667d60] and secondary[0x7f22b93b3390]] pr 5 04:38:24 NGVI-PCSCF2 /usr/sbin/kamailio[14018]: message repeated 107 times: [ ERROR: ims_usrloc_scscf [subscribe.c:430]: free_subscriber(): record not found in hash table] Apr 5 12:50:38 NGVI-PCSCF2 /usr/sbin/kamailio[2604]: WARNING: interconnectroute [icncache_eventprocessor.c:76]: icnRefreshProcess(): total size 2147483648 qm_real_used 700707576 qm_used 387064680 1st_frag 0x7fd5f6d6a690 last_frag 0x7fd676d30fd0 Apr 5 13:00:25 NGVI-PCSCF2 /usr/sbin/kamailio[2437]: ERROR: ims_usrloc_scscf [subscribe.c:430]: free_subscriber(): record not found in hash table Apr 5 13:00:26 NGVI-PCSCF2 /usr/sbin/kamailio[2437]: message repeated 125 times: [ ERROR: ims_usrloc_scscf [subscribe.c:430]: free_subscriber(): record not found in hash table] Apr 5 13:01:25 NGVI-PCSCF2 /usr/sbin/kamailio[2437]: CRITICAL: <core> [main.c:650]: sig_alarm_abort(): shutdown timeout triggered, dying... Apr 5 13:02:30 NGVI-PCSCF2 kamailio: WARNING: <core> [daemonize.c:348]: daemonize(): pid file contains old pid, replacing pid Apr 5 13:02:30 NGVI-PCSCF2 /usr/sbin/kamailio[27629]: WARNING: interconnectroute [ported_cache.c:113]: initPortedNumberCount(): Requested records[726049]: Total allocated[871258]: Fat 20% :Total MemUsed [26 MB divided between primary[0x7f3093a3fd60] and secondary[0x7f309478b390]] Thank you in advance. Regards, Solly Thoobe

3 2

urn:emergency:service:sos as RURI
by a.izquierdo＠zaleos.net 05 Apr '23

05 Apr '23

Hello, I'm running a Kamailio instance with some routing logic in it. When a UA sends an INVITE with urn:emergency:service:sos as RURI, Kamailio responds with a 400 Bad Request URI. I have set debug log level and it looks like the problem is in the parse_uri.c when sanity_check("1511", "7") is called: parse_uri(): bad char ':' in state 3 parsed: <urn:emergency:service> (21) / <urn:emergency:service:sos> (25) According to RFC8141 (https://www.rfc-editor.org/rfc/rfc8141), this is a correct URN, but maybe I'm missing something. Is there a way to make Kamailio parse this RURI? Kamailio version used: 5.6.4 Thank you!

3 2

segfault during Identity verification (stirshaken)
by Yuriy Nasida 04 Apr '23

04 Apr '23

Hello, It's kamailio 5.6.4. I can say that there is no segfault in case Identity is OK but bad attempts sometimes are the reason for segfault. Not sure I can duplicate but it happens several times per day. Please look at logs. in messages: Apr 3 09:40:02 kam1 kernel: [6381565.762987] kamailio[9141]: segfault at 746c75616665 ip 00007f20c4cab6df sp 00007ffe7ec51470 error 4 in libcrypto.so.1.1[7f20c4c3b000+19e000] kamailio.log: Apr 3 09:40:02 kam1 kamailio[9141]: NOTICE: <script>: it is INVITE with Identity. Lets check Identity Apr 3 09:40:02 kam1 kamailio[9141]: CRITICAL: <core> [core/mem/q_malloc.c:501]: qm_free(): BUG: bad pointer 0x645f6d6574737973 (out of memory block!) called from tls: tls_init.c: ser_free(323) - ignoring Apr 3 09:40:02 kam1 kamailio[9182]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 53 Apr 3 09:40:02 kam1 kamailio[9100]: ALERT: <core> [main.c:774]: handle_sigs(): child process 9141 exited by a signal 11 Apr 3 09:40:02 kam1 kamailio[9100]: ALERT: <core> [main.c:777]: handle_sigs(): core was not generated Apr 3 09:40:02 kam1 kamailio[9100]: CRITICAL: <core> [core/mem/q_malloc.c:501]: qm_free(): BUG: bad pointer 0x645f6d6574737973 (out of memory block!) called from tls: tls_init.c: ser_free(323) - ignoring loadmodule "stirshaken.so" ####### STIRSHAKEN ###### modparam("stirshaken", "vs_verify_x509_cert_path", 1) modparam("stirshaken", "vs_ca_dir", "/etc/kamailio/stir_CA") #modparam("stirshaken", "vs_crl_dir", "/etc/kamailio/CRL") modparam("stirshaken", "as_default_key", "/etc/kamailio/key/private_key.pem") modparam("stirshaken", "vs_cache_certificates", 1) modparam("stirshaken", "vs_cache_dir", "/tmp/cert_cache") modparam("stirshaken", "vs_cache_expire_s", 100) if (is_method("INVITE") && is_present_hf("Identity")) { xlog("L_NOTICE", "it is INVITE with Identity. Lets check Identity\n"); if (1 == stirshaken_check_identity()) { xlog("L_NOTICE", "Shaken Identity is OK\n"); } else { xlog("L_NOTICE", "Shaken Identity is invalid\n"); } } Please give me some advice.

2 2

Handling stateless replies with kemi
by Ovidiu Sas 03 Apr '23

03 Apr '23

Hello all, How can we handle stateless replies with kemi? tm has the following param: modparam("tm", "on_sl_reply", "stateless_replies") What's the proper way to map the "stateless_reply" onreply_route to a kemi function. I'm playing with the app_jsdt module. Thanks, Ovidiu

5 7

How to check, if call is already handled by rtpengine module?
by Benoît Panizzon 02 Apr '23

02 Apr '23

Hi Running Kamailio Registrar Node + rtpengine without dialog module, only TRX module. Situation: Some customer are in ranges to which RTP can directly be routed, some are not. So rtpengine shall be engaged only when backhauling of the rtp stream is required. On a message from a CPE, this is done by checking $si and on a message to the CPE by checking $(ulc(aor=>addr)[$T_branch_idx]{uri.host}); On a BYE, rtpengine_delete() is called no matter if RTP was backhauled or not. If rtpengine does not find the call to delete, no harm happens. But now I struggle with Session Refresh Re-Invites TO the CPE as they are routed very early on and do not pass the whole location lookup stuff, therefore I have no access to the aor=>addr to determine if the destination needs backhauling or not. So the obvious solution which came to my mind is to call rtpengine_query() and see if it is finding the call and if so, call rtpengine_manage() to keep backhauling traffic on the Re-Invites refrreshing the session. Unfortunately, I found no way to check if rtpengine_query finds the call. Did I miss something? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 1

Germany unsure about whether Kamailio will survive next winter
by Alex Balashov 01 Apr '23

01 Apr '23

For immediate release: ATLANTA, GA (1 April 2023)--The Federal Ministry for Economic Affairs and Energy of Germany was forced to disclose today that it has been tasked with assessing whether Germany will be able to operate its Kamailio through the next winter. This initiative comes amidst considerable uncertainty, shared in some other EU member states, about whether the enormous fossil fuel energy footprint of Kamailio is sustainable after the Russian invasion of Ukraine in early 2022. Pioneering Atlanta energy market analysts Evariste Systems were tapped to assist with forecasting whether German strategic natural gas reserves and liquefied natural gas (LNG) imports could keep Kamailio's enormous turbines spinning under a variety of stress-test scenarios outlined by regulators, including SIPP and SIPVicious. Alex Balashov, Evariste's principal, said in a news conference earlier this week: "Simply put, the political leadership wanted a dispassionate, technocratic kind of analysis, data-driven and the rest, free of the inflection of ideological bias and tendentious policy angles in charged German domestic politics. Whether it's the centre-right or the Greens, there are strong ideas about what to do with energy-greedy megaprojects like Kamailio." Florida-based energy turnaround vet Fred Posner, parachuted in to assist with the fast-tracked study, agreed: "It's well-understood that the German GDP is almost a perfectly linear function of gross annual SIP packets routed. The Nord Stream 1 shut-off situation in September only exposed Germany to further Kamailio-related energy vulnerabilities." At issue are the basic physics of SIP message routing in Kamailio. Due to a fixed-size worker process pool, Kamailio routing consumes about 1500 BTUs per packet-kilometre travelled. While 1500 BTUs/packet-kilometre is efficient in distance terms as compared to a typical automobile, which consumes roughly 3800 BTUs of fossil fuel energy per passenger-kilometre travelled (at typical load factors), the dizzying number of SIP packets routed through German territory in a typical business day greatly outstrips passenger-kilometres travelled. Balashov noted that sometimes, the activation energy requirement can be higher than 1500 BTUs when INVITEs with large SDPs are launched toward the next hop, though this varies with the altitude of the destination above mean sea level, wind direction, weather and other factors involved in ballistics. "It's a bit of a fool's errand to play these guessing games with averages. The energy budget can vary enormously depending on whether there is DTLS, video codecs, PASSporTs and other stuff. I find it helpful to think in kilowatt-hour terms: sometimes it's less than half a kWh, so like US $0.20 if you're counting the beans, and sometimes it's more like $0.38, we just don't know. The Bundestag always wants these big, round numbers, but if you've ever looked at the TM module, you know that's not how this works." A milder-than-expected winter, a 9% year-over-year increase in US LNG exports (to about 300 million cubic metres per day) in 2022, dependable imports from Norway and the Netherlands, and other favourable factors gave German SIP regulators a reprieve. Despite energy market volatility, premium LNG spot prices, and occasional 408 Request Timed Out scenarios, the country dodged a widely-feared Kamailio energy crisis. However, it is difficult to say whether energy market conditions will be as propitious next year. "Then there's the whole climate change goals thing. Nobody even wants to talk about that, especially for the WebRTC gateway side," says Posner. "There's a persistent fantasy that we can just power Kamailio with solar or wind if we just had enough generating capacity. I think that's really missing the forest for the trees. The worker processes stay running whether you need them or not, it all depends on how many listeners you have set and what the children config value is. Either way, I've got two words for you: base load. Even if we completely ignore that the energy density is just not there per hectare of solar or wind installation, how do you provide the constant power to the child processes?" Past feasibility studies published jointly by Balashov and Posner support this assessment. One such study, initiated in 2018 and concluded during the COVID-19 global pandemic, found that an area equivalent to the size of the entire state of North Rhine-Westphalia would need to be devoted to batteries, even using the latest lithium ferrous phosphate (LFP) technology. "But wait, there's more!" says Posner: "Show me the amperage." "The packet forwarding mechanism is similar to a railgun, at least if you're doing stateful. Where are you going to get the millions of amps? It's either setting up huge banks of capacitors god-knows-where, or the pulsed power system that's undergirded by the original OpenSER turbines. I think we already know the answer to that." Balashov noted that, even beyond the political and economic challenges that Kamailio downtime would introduce, there are other tightropes to walk. As the most economically dynamic, export-oriented EU member state, Germany is prone to flirt with SIP proxy privatisation schemes in its national discourse, all while messaging a stronger Euro to the ECB. "There is a vocal minority," says Balashov, "who push for research into more modular, regionally sited SIP gateways. They're always talking about sclerotic, unresponsive federal regulations and listless, bumbling Brussels bureaucrats and making it all smaller and leaner. Listen, I'm all for the Invisible Hand, but setting the Request URI and adding custom headers requires a truly integrated, national-scale infrastructure. It's the stuff of public-private partnerships and megaprojects. Even if you're a market zealot who is not sold on basic INVITEs as a public good, think about Presence or IMS. You really need the full capacity of a nation-state on deck for that." If devolution to smaller, more numerous SIP routing sites is not feasible--at least, at the unit cost and reliability level demanded by advanced economies--then excessive complexity offers a cautionary tale from the opposite extreme. In a particularly notorious example of technical and policy failure, the Electric Reliability Council of Texas (ERCOT), following the guidance of northern Californian advisors Unicorn & Moonshot, attempted to scale down traditional Kamailio turbine blades into newer, so-called Kubernetes "Pod" chasses to increase exhaust recapture. This made for a much more intricate installation with an exponential growth of moving parts, telemetry and site support skill requirements. This proved unmanageable, and the now-famous explosion that followed the deterioration of the site led to unprecedented casualties in US history. Furthermore, it also resulted in the long-term humidification of large expanses of East Texas, Louisiana, Mississippi and southern Alabama, rendering most of these areas uninhabitable. These events discredited the Unicorn & Moonshot approach in the eyes of most industry analysts. While it will take some months for Evariste to reach a conclusion regarding the resilience of the German Kamailio through the winter and beyond, and it will take still longer to issue policy recommendations, one factor is universally agreed upon already: this prolonged test of Germany's resilience is part of a Russian strategic calculation. A conservative MP from the CDU/CSU faction of the Bundestag, speaking on condition of anonomity, offered this summary: "The Russians think they can wait this out. They are waiting for us to fold and switch to an OpenSIPS reactor, knowing full well this will keep us busy with troubleshooting and diminish our ability to support Ukraine militarily." -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800

2 1

How to get top-most outbound via branch attribute from branched call?
by Benoît Panizzon 31 Mar '23

31 Mar '23

Hi all For rtpengine to correctly distinguish branched calls, the top-most via branch= parameter should be passed to it. I have attempted to use $via0 but that does not yet exist in Kamailio 5.4 So I did try to read $hdr(Via) which I guess returns an array and then access: but this returns the inbound via, not the one generated on the branched call. How can this be accomplished with kamailio 5.4? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

1 1

CANCEL timeout in branch route does not trigger any failure routes
by Richard Chan 31 Mar '23

31 Mar '23

Hi Kamailio users How can I get a failure route to trigger in timeout-to-CANCEL in a parallel forking scenario? Parallel forking test scenario 1. one(the main - 0) branch picks up the call 2. 2nd branch sends CANCEL (expecting 487 etc) - but the callee does not send any replies Result: No failure route handlers are called on timeout of this CANCEL - that is kamailio attempts the CANCEL 4 times but neither tm.t_on_failure_route, tm.t_on_branch_failure handlers are called. Any suggestions on how to resolve this? Regards Shih-Ping

3 2

Rancher, AWS, and Elastic IP
by Calvin E. 30 Mar '23

30 Mar '23

I'm looking for documentation//howto/advice/best practices for running Kamailio under Kubernetes on AWS via Rancher. Specifically, how to manage the public/Elastic IP. This is a load balancer (dispatcher) scenario with rtpengine for receiving inbound calls from a DID provider, but a similar configuration might be used for clients. I'm open to using an AWS load balancer, and would like to scale with multiple Kamailio pods behind a single public IP. I'd rather use Rancher-managed EC2 than pay extra for EKS, unless EKS can cancel out its own cost. This sounds like it should be a solved problem, just complex. I see blogs and videos out there, but it's not always obvious if their content is current. Anyone want to help me peel this onion? -Calvin

4 6

MongoDB query from Kamailio help needed
by Edo 28 Mar '23

28 Mar '23

Help is needed. I know I have seen a similar problem either via email or google search but for the life of me I cannot find it now. I am using: Ubuntu: 5.4.0-144-generic #161-Ubuntu SMP Fri Feb 3 14:49:04 UTC 2023 x86_64 x86_64 x86_64 GNU/Linux Kamailio: version: kamailio 5.3.9 (x86_64/linux) -- This is a dev platform and I can update kamailio if needed. I am trying to query a mongodb collection for the phone number stored under “did” as an integer and make a decision based on the returned result (boolean is sufficient for this decision). I don’t know what I am doing wrong here. Any help will be well appreciated ##### This is the insert statement into the telephone db from a linux script db.c$coll.insertOne({"did":parseInt($(cut -d_ -f 1 <<< $1)), "callDesc":$(cut -d_ -f 1 <<< $1)}))" >> $(cut -d_ -f 1 <<< $1).js did is stored as an integer and the callDesc is stored as presented by $rU { "_id" : ObjectId("641580b4863678d0ba38d69a"), "did" : 1XXXXXXXXXX, "callDesc" : 1XXXXXXXXXXXX } > db.c234.find() { "_id" : ObjectId("641580b4863678d0ba38d69a"), "did" : 2XXXXXXXXXXXX, "callDesc" : 2XXXXXXXXXXXX } > db.c234.find() { "_id" : ObjectId("641580b4863678d0ba38d69a"), "did" : 3XXXXXXXXXXXX, "callDesc" : 3XXXXXXXXXXXX } > Kamailio config: ..... ..... # --- mongodb module loaded 202201082322 loadmodule "ndb_mongodb.so" if (uri==myself) { sl_send_reply("100", "Trying to check if uri is myself $rU $ru"); #return; #Adding this to test connection to mongodb…. This worked and the reply was sent to the endpoint if(*mongodb_cmd_simple*("mongodbsrv1", "telephone", "cXXX”, "{ \"collStats\": \"cXXX\” }", "mgr1")) { sl_send_reply("183", "I got inside this message loop $rU found in mongodb"); xlog("response from mongodb is [[$mongodb(mgr1=>value)]]\n"); } # End of the test #This is to run the actual db query to test if the phone number is in the mongodb…. I am not successful at this query even when the number is known to be in the database if(*mongodb_find_one*("mongodbsrv1", "telephone", "cXXX”, "{\"did\" : \"$rU\" }", "mongoResponse1")) { # Route to confirmed route here -- 202302212032 sl_send_reply("183", "I got inside this mongodb message loop"); route(CONFIRMED_ROUTE); xlog("response from mongodb is [[$mongodb(mongoResponse1=>value)]]\n"); } else { route(UNCONFIRMED_ROUTE); } } #append_hf("P-hint: outbound\r\n"); route(RELAY); #exit; } Please help. -- ----- Ekunwe Tel: 601.497.3932

2 2

Three year old issue with a new CVE vulnerability report being reported
by Olle E. Johansson 28 Mar '23

28 Mar '23

Hi Kamailians! A new CVE with a critical severity level was published recently for an almost three year old bug, which was also fixed and released three years ago (CVE-2020-27507). The issue was fixed in Kamailio 5.4.2 and is not present in newer releases. The Kamailio project has unfortunately not been involved in the CVE process or been informed about this old issue being published at this time. We take vulnerability handling seriously and our process is documented at: https://www.kamailio.org/wikidocs/security/policy/ The latest stable branch is 5.6, with v5.6.4 released out of it. Reference: https://cve.mitre.org/cgi-bin/cvename.cgi?name=2020-27507 Best regards and thanks for flying Kamailio! The Kamailio dev team through /Olle

4 3

json RPC syntax
by Volker Denneberg 28 Mar '23

28 Mar '23

Hello, I am trying to get json RPCs running. simpleequests seem to work, but I can't figure out how to pass parameters. For example curl --header 'Content-Type: application/json' --data-binary '{"id": 1,"jsonrpc": "2.0", "method": "ul.db_users" }' http://localhost:8081/RPC/ { "jsonrpc": "2.0", "error": { "code": 500, "message": "Not enough parameters (table to lookup)" }, "id": 1 } According to http://www.kamailio.org/docs/docbooks/3.2.x/rpc_api/rpc_api.html#rpc.gettin… parameters dont have names and according to http://www.kamailio.org/docs/modules/devel/modules/usrloc.html#usrloc.r.db_… 1 parameter is required. I guessed adding "params":"location" => no error anymore, but still no result: curl --header 'Content-Type: application/json' --data-binary '{"id": 1,"jsonrpc": "2.0", "method": "ul.db_users", "params":"location"}' http://localhost:8081/RPC/ { "jsonrpc": "2.0", "result": { }, "id": 1 } Same result (no error) if I pass "params":"nonsense", so "params" seems to be kind of ignored ... What's the correct json syntax here or in general? Best regards, Volker

5 6

KEMI: app_jsdt_run_ex(): js loading state not initialized (call: ksr_htable_event)
by Ovidiu Sas 27 Mar '23

27 Mar '23

Hello all, While using KEMI with app_jsdt I noticed the following possible regression between 5.5 and 5.6. The following config is working ok on 5.5 but throws an error on 5.6: ERROR: app_jsdt [app_jsdt_api.c:555]: app_jsdt_run_ex(): js loading state not initialized (call: ksr_htable_event) kamailio.cfg: debug=2 children=1 log_facility=LOG_LOCAL0 log_name="k" mpath="/home/osas/src/kamailio/src/modules" loadmodule "htable/htable.so" modparam("htable", "event_callback", "ksr_htable_event") modparam("htable", "event_callback_mode", 1) loadmodule "kemix/kemix.so" loadmodule "app_jsdt/app_jsdt.so" modparam("app_jsdt", "load", "/usr/local/etc/kamailio/kamailio.js") cfgengine "jsdt" kamailio.js: function ksr_htable_event(evname) { KSR.info("ksr_htable_event(" + evname + ")\n"); } function ksr_request_route() { KSR.x.exit(); } If we load the htable module after app_jsdt, then the error is gone. Is there something new during init in 5.6 that requires specific module loading order? -ovidiu

2 2

Handling branch-level append_hf()
by Alex Balashov 27 Mar '23

27 Mar '23

Hi, I’ve got a situation where I catch a 302 redirect in a failure_route (branch #1), and use the Contact to set a custom header (append_hf()). This custom header is utilised on the subsequent branch (branch #2), but if that branch fails (e.g. due to 4xx-5xx response), the custom header is not preserved on subsequent branches (branch #3+). What’s the best way to preserve it across subsequent branches? The documentation for msg_apply_changes() says it cannot be used once the transaction is already created, and in any case, I have other branch-level changes in branch #2 that I _would_ like to revert on subsequent branch attempts (i.e. branch #3). I can certainly devise a transaction-persistent vehicle for this value which can be checked on every branch, but I wondered if there’s an easier and more best-practical way. Thanks! — Alex -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800

2 1

No voice for webrtc call
by Arun K R 25 Mar '23

25 Mar '23

Hello, I am using kamailio to register the webrtc client and routes the calls to asterisk using pjsip. I am having no voice on both ends. Logs are attached witht this email. Mar 24 13:06:58 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [control] Received command 'offer' from 10.13.1.127:59841 Mar 24 13:06:58 debian rtpengine[7345]: NOTICE: [8asmubtpv46ac7nrfk57]: [core] Creating new call Mar 24 13:06:58 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [control] Replying to 'offer' from 10.13.1.127:59841 (elapsed time 0.016067 sec) Mar 24 13:06:58 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [control] Received command 'answer' from 10.13.1.127:52045 Mar 24 13:06:58 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [control] Replying to 'answer' from 10.13.1.127:52045 (elapsed time 0.000997 sec) Mar 24 13:06:58 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [ice] ICE negotiated: peer for component 1 is 10.13.1.170:15634 Mar 24 13:06:58 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [ice] ICE negotiated: local interface 10.13.1.127 Mar 24 13:06:58 debian rtpengine[7345]: ERR: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [core] SRTP output wanted, but no crypto suite was negotiated Mar 24 13:06:58 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [control] Received command 'answer' from 10.13.1.127:52045 Mar 24 13:06:58 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [control] Replying to 'answer' from 10.13.1.127:52045 (elapsed time 0.000786 sec) Mar 24 13:06:58 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [crypto] DTLS: Peer certificate accepted Mar 24 13:06:58 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [crypto] DTLS-SRTP successfully negotiated using AES_CM_128_HMAC_SHA1_80 Mar 24 13:06:58 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [crypto] DTLS-SRTP successfully negotiated using AES_CM_128_HMAC_SHA1_80 Mar 24 13:06:58 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [ice] ICE negotiated: peer for component 1 is 10.13.1.170:15634 Mar 24 13:06:58 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [ice] ICE negotiated: local interface 10.13.1.127 Mar 24 13:07:02 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [core] Confirmed peer address as 10.13.1.170:15634 Mar 24 13:07:02 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [core] Kernelizing media stream: 10.13.1.170:15634 -> 10.13.1.127:10940 | 10.13.1.127:10960 -> 192.168.30.200:53489 Mar 24 13:07:02 debian rtpengine[7345]: WARNING: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [core] No support for kernel packet forwarding available (encryption cipher or HMAC not supported by kernel module) Mar 24 13:07:02 debian rtpengine[7345]: NOTICE: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [core] Setting 'non-forwarding' flag for kernel stream due to lack of sinks Mar 24 13:07:03 debian rtpengine[7345]: ERR: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [rtcp] SRTCP output wanted, but no crypto suite was negotiated Mar 24 13:07:03 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [core] Removing media stream from kernel: local 10.13.1.127:10940 Mar 24 13:07:03 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [core] Kernelizing media stream: 10.13.1.170:15634 -> 10.13.1.127:10940 | 10.13.1.127:10960 -> 192.168.30.200:53489 Mar 24 13:07:03 debian rtpengine[7345]: WARNING: [8asmubtpv46ac7nrfk57/223df8fa-8106-48af-844f-33f97530a0d8/1 port 10940]: [core] No support for kernel packet forwarding available (encryption cipher or HMAC not supported by kernel module) Mar 24 13:07:04 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [control] Received command 'delete' from 10.13.1.127:52045 Mar 24 13:07:04 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [core] Scheduling deletion of call branch '9693uq66f0' (via-branch '') in 30 seconds Mar 24 13:07:04 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [core] Scheduling deletion of call branch '223df8fa-8106-48af-844f-33f97530a0d8' (via-branch '') in 30 seconds Mar 24 13:07:04 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [core] Scheduling deletion of entire call in 30 seconds Mar 24 13:07:04 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [control] Replying to 'delete' from 10.13.1.127:52045 (elapsed time 0.001463 sec) Mar 24 13:07:17 debian /usr/local/sbin/kamailio[13362]: INFO: <script>: 10.13.1.170 authentication bypassed for sip:developer@10.13.1.127 Mar 24 13:07:34 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [core] Final packet stats: Mar 24 13:07:34 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [core] --- Tag '9693uq66f0', created 0:36 ago for branch '' Mar 24 13:07:34 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [core] --- subscribed to '223df8fa-8106-48af-844f-33f97530a0d8' Mar 24 13:07:34 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [core] --- subscription for '223df8fa-8106-48af-844f-33f97530a0d8' Mar 24 13:07:34 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [core] ------ Media #1 (audio over UDP/TLS/RTP/SAVPF) using unknown codec Mar 24 13:07:34 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [core] --------- Port 10.13.1.127:10960 <> 192.168.30.200:53489, SSRC 0, in 0 p, 0 b, 292 e, 36 ts, out 0 p, 0 b, 0 e Mar 24 13:07:34 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [core] --- Tag '223df8fa-8106-48af-844f-33f97530a0d8', created 0:36 ago for branch '' Mar 24 13:07:34 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [core] --- subscribed to '9693uq66f0' Mar 24 13:07:34 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [core] --- subscription for '9693uq66f0' Mar 24 13:07:34 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [core] ------ Media #1 (audio over UDP/TLS/RTP/SAVPF) using opus/48000/2 Mar 24 13:07:34 debian rtpengine[7345]: INFO: [8asmubtpv46ac7nrfk57]: [core] --------- Port 10.13.1.127:10940 <> 10.13.1.170:15634, SSRC 1ada0b2b, in 292 p, 40137 b, 1 e, 30 ts, out 2 p, 1449 b, 0 e

2 3

different sip response on database (mysql) timeout
by Leon de Rooij 24 Mar '23

24 Mar '23

Hi, I need to use a database that may sometimes timeout and I'd like to give a conditional sip response based on whether there was a timeout or not. I'm using sqlops with mysql modules. Does anyone know if that's possible ? Thanks, Leon

3 3

Route: comma separated HOP list
by Benoît Panizzon 24 Mar '23

24 Mar '23

Hi List Cisco-SPA112 does reply to an invite containing Record-Route header with one Route: Header containing a comma separated list of those hops. It looks like kamailio fails to parse them and correctly route for example an ACK to a 200 OK. Is this a known issue? Is there a solution? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 2

WebRTC "client did not present a certificate" error
by David Cunningham 23 Mar '23

23 Mar '23

Hello, We have a Kamailio 5.2.7 server with WebRTC enabled. However, a WebRTC client at https://tryit.jssip.net/ is unable to connect on either Chrome or Firefox. In the Kamailio log we see the lines below. In tls.cfg we have "verify_certificate = no" and "require_certificate = no" for both [server:default] and [client:default]. Would anyone be able to help us with this? Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: tls [tls_domain.c:1159]: tls_lookup_private_key(): Private key lookup for SSL_CTX-0x14baf1cbb090: (nil) Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: tls [tls_domain.c:749]: sr_ssl_ctx_info_callback(): SSL handshake done Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: tls [tls_domain.c:737]: sr_ssl_ctx_info_callback(): SSL handshake started Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: tls [tls_domain.c:741]: sr_ssl_ctx_info_callback(): SSL renegotiation initiated by client Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: tls [tls_domain.c:749]: sr_ssl_ctx_info_callback(): SSL handshake done Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: tls [tls_domain.c:737]: sr_ssl_ctx_info_callback(): SSL handshake started Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: tls [tls_domain.c:741]: sr_ssl_ctx_info_callback(): SSL renegotiation initiated by client Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: tls [tls_domain.c:749]: sr_ssl_ctx_info_callback(): SSL handshake done Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: tls [tls_server.c:424]: tls_accept(): TLS accept successful Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: tls [tls_server.c:431]: tls_accept(): tls_accept: new connection from xx.xx.xx.xx:39816 using TLSv1.3 TLS_AES_256_GCM_SHA384 256 Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: tls [tls_server.c:434]: tls_accept(): tls_accept: local socket: yy.yy.yy.yy:8443 Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: tls [tls_server.c:445]: tls_accept(): tls_accept: client did not present a certificate Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: tls [tls_server.c:1189]: tls_read_f(): Reading on a renegotiation of connection (n:532) (0) Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: <core> [core/tcp_read.c:1527]: tcp_read_req(): EOF Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: <core> [core/io_wait.h:602]: io_watch_del(): DBG: io_watch_del (0x562ffde66d00, 17, -1, 0x10) fd_no=4 called Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: <core> [core/tcp_read.c:1680]: release_tcpconn(): releasing con 0x14baf4cc1ec8, state -1, fd=17, id=665 ([xx.xx.xx.xx]:39816 -> [xx.xx.xx.xx]:8443) Mar 22 01:25:28 br /sbin/kamailio[25259]: DEBUG: <core> [core/tcp_read.c:1684]: release_tcpconn(): extra_data 0x14baf289ea30 Mar 22 01:25:28 br /sbin/kamailio[25291]: DEBUG: <core> [core/tcp_main.c:3320]: handle_tcp_child(): reader response= 14baf4cc1ec8, -1 from 1 Mar 22 01:25:28 br /sbin/kamailio[25291]: DEBUG: tls [tls_server.c:683]: tls_h_close(): Closing SSL connection 0x14baf289ea30 Thanks very much, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

4 7

How to require session timer and set a max session timer value?
by Benoit Panizzon 23 Mar '23

23 Mar '23

Hi All I would like to avoid having run away calls as good as possible. For this, I would like to require timers and set the maximum expires timer to say 1800 I had a look at the sst module, but I see no way to require timers or to set a max timers value. Do my intentions make sense? How did others accomplish this task? Thanks for any hints! Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

1 0

With TOPOH module enabled, transferred call hangup from callee side does not hang up the call
by Arsenijs Kabihno 22 Mar '23

22 Mar '23

Hello, we have faced an issue which appears when TOPOH module is enabled, call is made from WebRTC client, then transferred by callee. We have WebRTC client behind load balancer. We make call to softphone from WebRTC, call from softphone is transferred to external number using Asterisk. When call is hanged up on external number side, it doesn't hangs up on WebRTC client side. If TOPOH is not enabled, this issue doesn't occur. We have the following TOPOH-related sections of kamailio.cfg: <...> #!ifdef WITH_TOPOH loadmodule "topoh.so" #!endif <...> <...> #!ifdef WITH_TOPOH modparam("topoh", "mask_key", "xxxxxxxxxxxxxxxx") modparam("topoh", "mask_ip", "10.0.0.1") modparam("topoh", "mask_callid", 0) modparam("topoh", "sanity_checks", 1) modparam("topoh", "uparam_prefix", "") modparam("topoh", "vparam_prefix", "") modparam("topoh", "callid_prefix", "1337") #!endif <...> <...> #!ifdef WITH_DIALOG modparam("dialog", "db_url", DBURL) modparam("dialog", "db_mode", 1) modparam("dialog", "db_fetch_rows", 800) modparam("dialog", "track_cseq_updates", 1) modparam("dialog", "ka_interval", 30) modparam("dialog", "ka_timer", 30) modparam("dialog", "send_bye", 0) modparam("dialog", "dlg_filter_mode", 1) #!ifdef WITH_TOPOH modparam("dialog", "lreq_callee_headers", "TH: dlh\r\n") #!endif #!endif <...> <...> #!ifdef WITH_DISPATCHER event_route[dispatcher:dst-up] { sql_query("ds_dburl", "select description from dispatcher where destination='$ru'", "ds_dbresult"); xlog("L_WARN", "Destination up: $ru, found name: $dbr(ds_dbresult=>[0,0])\n"); prom_gauge_set("trunk_reachability", "1", "$ru", "$dbr(ds_dbresult=>[0,0])"); } event_route[dispatcher:dst-down] { sql_query("ds_dburl", "select description from dispatcher where destination='$ru'", "ds_dbresult"); xlog("L_WARN", "Destination down: $ru, found name: $dbr(ds_dbresult=>[0,0])\n"); prom_gauge_set("trunk_reachability", "0", "$ru", "$dbr(ds_dbresult=>[0,0])"); } #!endif #!ifdef WITH_TOPOH event_route[topoh:msg-sending] { if(is_reply() and $rs=="302") { drop; } } #!endif (can provide the full config, if needed) We are receiving the following errors: On transfer: https://pastebin.com/H3FQ4Kpj On hangup from external number side: https://pastebin.com/wgfRzGbd On hangup from WebRTC client side: https://pastebin.com/sE0yhBae There are also differences in SIP traffic when TOPOH is disabled and when it is enabled. With TOPOH enabled, apparently, ACK from Asterisk is not reaching kamailio: asterisk INVITE -> kamailio -> WebRTC WebRTC 200 -> kamailio -> asterisk asterisk ACK -> kamailio -> -> -> (SIP flows attached - topoh_on.csv, topoh_off.csv) kamailio -v version: kamailio 5.6.1 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 09:11:15 Nov 15 2022 with gcc 9.3.0 We have dockerized kamailio running on Ubuntu 16.04.3 LTS Docker version 18.03.1-ce, build 9ee9f40 Please advise where to start looking.

1 0

OT: path support fix for asterisk
by Henning Westerholt 21 Mar '23

21 Mar '23

Hello, It's off-topic, but it was several times discussed here on the list. It seems that the upcoming 18.17 and 20.2 asterisk version seems to properly support path headers. From the announcement: * [ASTERISK-30100<https://issues.asterisk.org/jira/browse/ASTERISK-30100>] - res_pjsip: Path is ignored on INVITE to endpoint (Reported by Yury Kirsanov) This bug has been many years open. have not tried it yet, but it looks promising. Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

7 8

Follow-up: Printing Global Parameter Values & Logging JSONRPC Requests over UDP
by sadik.oualla.mohamed＠gmail.com 18 Mar '23

18 Mar '23

Hello everyone, I am following up on my previous email regarding printing Global Parameter values and logging JSONRPC requests over UDP in Kamailio. I understand that everyone might be busy, but I would greatly appreciate any guidance or suggestions you can provide on these topics. To recap my questions: 1)Is there a way to print the Global Parameter values, such as debug, children, flags, etc., within the Kamailio configuration file, to ensure they have been changed? 2°How can I log in Kamailio that it has received a JSONRPC request over UDP? I have been successful in logging this information when using HTTP and JSONRPC but am unsure how to do it with UDP. Any input or advice would be truly valuable. Thank you in advance for your time and assistance. Best.

1 0

Issues/Vulnerabilities in OpenSipS that may affect Kamailio
by Dovid Bender 17 Mar '23

17 Mar '23

Hi All, OpenSipS just released an update to the audit that was done to OpenSips [1]. From my basic coding skills it seems like the changes that were done by the OpenSipS project were not implemented in Kamailio which means that Kamailio is potentially vulnerable? For example you can compare the changes made by OpenSips project here [2] and the Kamailio code here [3] I am not active much on the list so please don't roast me if I am completely wrong here. Regards, Dovid [1] http://lists.opensips.org/pipermail/users/2023-March/046849.html [2] https://github.com/OpenSIPS/opensips/commit/dd9141b6f67d7df4072f3430f628d4b… [3] https://github.com/kamailio/kamailio/blob/master/src/core/parser/digest/par…

4 4

RPC error: 500 - Failed to use table
by Volker Denneberg 17 Mar '23

17 Mar '23

Hi, what can be the reason why usrloc module sometimes can read from database, sometimes not? Normal operation looks ok, the location table is kept up to date. The problem seems to be related with RPC (kamcmd or jsonrpc). For example, kamcmd ul.dump # ok kamcmd ul.lookup location 6351*1 # ok kamcmd ul.db_users location # error: 500 - Failed to use table Error 500 correlates with syslog entry: ERROR: <core> [db.c:481]: db_use_table(): invalid connection parameter Who connects where? In my understanding usrloc via core to mysql. I suppose RPC calls do not fail if usrloc can operate on kamailio memory, does that make sense? If so, what usrloc parameter can I tweak to fix error 500 ? I don't find any usrloc parameter regarding db connection. (Opening another mysql session as user kamailio is no problem, so we re not running out of connections.) Thanks in advance! Volker [cid:fb71bb9b-ad4f-4110-824b-d6805406bf6a] Volker Denneberg Development Telefon: +49 241 980 986 41 Web: www.m3connect.de<https://mx.scientific.de/owa/redir.aspx?C=RW1TSCipXQC6K0Yw1m69CGqyTwJHXjiiN…> E-Mail: v.denneberg(a)m3connect.de<https://mx.scientific.de/owa/redir.aspx?C=BjBXIUrD2WCryLRZ2IrAAwEu0b0W0ciUY…> m3connect GmbH | Pascalstr.18 | 52076 Aachen | DE/Germany | Amtsgericht Aachen | HRB 8773 | USt.- ID: DE219 664 658 | Geschäftsführer/CEO: Emilio Dragas

2 1

Sync NONCE between registrars?
by Benoit Panizzon 16 Mar '23

16 Mar '23

Hi We have two registrars. Today I observed a CPE doing this: CPE REGISTER (no auth) => Registrar 1 => 407 Challenge NONCE A CPE REGISTER (auth to NONCE A => Registrar 2 => 407 Challenge NONCE B CPE REGISTER (auth to NONCE B => Registrar 1 => 407 Challenge NONCE A and so on. Time on both registrars is in sync (I tought NONCE was based on time) Is there a way to sync the NONCE so that both would accept authentication based on the same NONCE? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 1

How to Print Global Parameter Values and Log JSONRPC Requests Received over UDP in Kamailio
by sadik.oualla.mohamed＠gmail.com 16 Mar '23

16 Mar '23

Dear everyone, I am wondering if there is a way to print the Global Parameter values such as debug, children, flags, etc. just to make sure that they have been changed. I am aware that I can use the `kamcmd cfg.get core debug` command to check the value of the debug parameter for example, but I am wondering if there is a way to do this in the configuration file of Kamailio. Additionally, I would like to know if there is a way to print in the log of Kamailio that it received a JSONRPC request over UDP. I have been able to print some logs when using HTTP and JSONRPC to know that the server has received a JSONRPC request, but I am unsure how to do this when using UDP. I would greatly appreciate it if someone could provide some guidance. Thank you in advance for your time and assistance. Best regards.

1 0

Increment in number of "200" responses while calling procedures over jsonrpc with http
by sadik.oualla.mohamed＠gmail.com 16 Mar '23

16 Mar '23

Hello, Thank you very much! I've used your suggestion to accomplish some amazing things. However, I'm having an issue when calling the procedures over jsonrpc with http. I've noticed that the number of "200" responses gets incremented by 2, which is not the case when using the `*kamcmd*` tool. I suspect that this may be due to jsonrpc using *http/https*, which iscausing Kamailio to treat it as a "200" http responses code as "200" SIP codes. Could you advise on how to avoid this issue? Is there any further configuration that needs to be done? Best regards.

3 8

Kamailio store call statistics
by georgy.grigoryev＠snapcom.fr 16 Mar '23

16 Mar '23

Hello everyone, I'm trying to store the call statistics in my Kamailio server. I'm using SQLite just to understand what type of information i can store. I'm using the dialog module with the classic parameters: enable_stats => 1 dlg_flag => 4 hash_size => 4096 db_url => sqlite:///usr/local/etc/kamailio/kamailio.sqlite db_mode => 1 I'm able to open the DB but everything is empty, am i using the wrong modules? or i didn't configure the server well? The goal is to store CALL CDR in the DB, so that i can see who called & how many times etc

2 1

webrtc audio issue
by Arun K R 15 Mar '23

15 Mar '23

Hello, I have installed Kamailio 5.6 in debian 11 and RTPengine 11.3 also in the same server. I have configured kamailio to work as webrtc server and it forwards the registration to asterisk. Now when I am trying call from jssip webrtc client it reaches kamailio and route it through private interface to asterisk server. Asterisk then route it to the provider server. When i make a call asterisk server recives rtp from the provider and convert the rtp to srtp and sending back to kamailio. but there is no sound for webrtc from public internet . Also i am getting warning (SRTCP /RTP output wanted, but no crypto suite was negotiated) webrtc from Local network works fine with VPN For normal udp call without webrtc works fine. Kamailio having two interfaces, interface1 is private 10.13.1.140 and interface 2 is publi ip 100.x.x.x webrtc client sends calls to the public ip interface 100.x.x.x and kamaiio routes the call to the asterisk via private interface 10.13.1.140. asterisk server sends the call to remote server and gets the rtp back from that Asterisk server ---converts RTP to SRTP and forward to ----> kamailio 10.13.1.140 but there is nothing happens after that, please help me on this i am new to kamailio and rtpengine. The issue is only for webrtc from public internet. But when using webrtc from LAN works fine below are the logs Mar 9 20:21:02 debian /usr/local/sbin/kamailio[11778]: INFO: <script>: from domain recived is 10.13.1.152 Mar 9 20:21:02 debian /usr/local/sbin/kamailio[11778]: INFO: <script>: 09496381412 has fromDomain recived is 10.13.1.152 Mar 9 20:21:02 debian /usr/local/sbin/kamailio[11778]: INFO: <script>: Request going to srtp customied arun Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [control] Adding a proxy for control ng stats:127.0.0.1 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Received command 'offer' from 127.0.0.1:53374 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Dump for 'offer' from 127.0.0.1:53374: { "supports": [ "load limit" ], "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 7377619262916888262 2 IN IP4 127.0.0.1 Mar 9 20:21:02 debian rtpengine[11823]: s=- Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=group:BUNDLE 0 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap-allow-mixed Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic: WMS mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB Mar 9 20:21:02 debian rtpengine[11823]: m=audio 9 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126 Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 0.0.0.0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp:9 IN IP4 0.0.0.0 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:NM2h Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:BF2WmB5sOZZWVIFppZ3YTiSH Mar 9 20:21:02 debian rtpengine[11823]: a=ice-options:trickle Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:sha-256 9F:FA:18:B0:8D:17:A5:14:FE:22:BA:1E:73:1E:3E:02:6F:17:EC:B5:FC:77:BD:E1:0B:51:DF:7F:61:B0:2F:55 Mar 9 20:21:02 debian rtpengine[11823]: a=setup:actpass Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=msid:mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB 6f36294e-5d92-4f0c-8ea2-ba415d0ec125 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:111 transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:111 minptime=10;useinbandfec=1 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:63 red/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:63 111/111 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:9 G722/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:0 PCMU/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:8 PCMA/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:13 CN/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:110 telephone-event/48000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=ssrc:2480159127 cname:FJAL1TGxCcnDMPXY Mar 9 20:21:02 debian rtpengine[11823]: a=ssrc:2480159127 msid:mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB 6f36294e-5d92-4f0c-8ea2-ba415d0ec125 Mar 9 20:21:02 debian rtpengine[11823]: ", "call-id": "ui7jc53b9lfbt9lf4nu7", "received-from": [ "IP4", "223.190.206.123" ], "from-tag": "7n38ig1acl", "command": "offer" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Using default bitrate of 32000 bps for 2-channel Opus Mar 9 20:21:02 debian rtpengine[11823]: NOTICE: [ui7jc53b9lfbt9lf4nu7]: [core] Creating new call Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Subscribing '7n38ig1acl' to '' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Subscribing '' to '7n38ig1acl' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating codecs for offerer 7n38ig1acl #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec red/48000/2 (63) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec G722/8000 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec PCMU/8000 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec PCMA/8000 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec CN/8000 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/48000 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating codecs for answerer #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec red/48000/2 (63) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec G722/8000 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec PCMU/8000 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec PCMA/8000 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec CN/8000 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/48000 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for #1 <https://github.com/sipwise/rtpengine/issues/1> -> 7n38ig1acl #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF 110, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec red/48000/2 (63) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for red/48000/2 (63) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] No codec support for red/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for red/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec G722/8000/1 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for G722/8000 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for G722/8000 is G722/8000/1 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec G722/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for G722/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec PCMU/8000/1 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for PCMU/8000 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for PCMU/8000 is PCMU/8000/1 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec PCMU/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for PCMU/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec PCMA/8000/1 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for PCMA/8000 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for PCMA/8000 is PCMA/8000/1 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec PCMA/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for PCMA/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec CN/8000/1 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for CN/8000 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for CN/8000 is CN/8000/1 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec CN/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for CN/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/48000/1 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for telephone-event/48000 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/48000 is telephone-event/48000/1 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/48000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/48000 with DTMF 110, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for #1 <https://github.com/sipwise/rtpengine/issues/1> -> 7n38ig1acl #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF 110, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec red/48000/2 (63) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] No codec support for red/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for red/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec G722/8000/1 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for G722/8000 is G722/8000/1 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec G722/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for G722/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec PCMU/8000/1 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for PCMU/8000 is PCMU/8000/1 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec PCMU/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for PCMU/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec PCMA/8000/1 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for PCMA/8000 is PCMA/8000/1 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec PCMA/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for PCMA/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec CN/8000/1 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for CN/8000 is CN/8000/1 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec CN/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for CN/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/48000/1 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/48000 is telephone-event/48000/1 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/48000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/48000 with DTMF 110, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] set FILLED flag for stream, local 10.13.1.127:30020 remote 0.0.0.0:9 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] Creating active DTLS connection context Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] Creating active DTLS connection context Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] set FILLED flag for stream, local 10.13.1.127:30021 remote 0.0.0.0:9 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [10.13.1.127:30000] Initialized incoming SRTP with SDES crypto params: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [1] Initialized outgoing SRTP with SDES crypto params: suite AEAD_AES_256_GCM, tag 1, key czR43Q05Tlky5pj/m6BomEKXh+b5okz/iSUDQqgvHDI= salt Vov9BA7SKter2Ajj Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] Creating passive DTLS connection context Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] Creating passive DTLS connection context Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [10.13.1.127:30001] Initialized incoming SRTP with SDES crypto params: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [2] Initialized outgoing SRTP with SDES crypto params: suite AEAD_AES_256_GCM, tag 1, key czR43Q05Tlky5pj/m6BomEKXh+b5okz/iSUDQqgvHDI= salt Vov9BA7SKter2Ajj Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Replying to 'offer' from 127.0.0.1:53374 (elapsed time 0.022450 sec) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Response dump for 'offer' to 127.0.0.1:53374: { "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 7377619262916888262 2 IN IP4 127.0.0.1 Mar 9 20:21:02 debian rtpengine[11823]: s=- Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap-allow-mixed Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic: WMS mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB Mar 9 20:21:02 debian rtpengine[11823]: m=audio 30000 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126 Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid Mar 9 20:21:02 debian rtpengine[11823]: a=msid:mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB 6f36294e-5d92-4f0c-8ea2-ba415d0ec125 Mar 9 20:21:02 debian rtpengine[11823]: a=ssrc:2480159127 cname:FJAL1TGxCcnDMPXY Mar 9 20:21:02 debian rtpengine[11823]: a=ssrc:2480159127 msid:mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB 6f36294e-5d92-4f0c-8ea2-ba415d0ec125 Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:111 minptime=10;useinbandfec=1 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:111 transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:63 red/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:63 111/111 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:9 G722/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:0 PCMU/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:8 PCMA/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:13 CN/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:110 telephone-event/48000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp:30001 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:1 AEAD_AES_256_GCM inline:czR43Q05Tlky5pj/m6BomEKXh+b5okz/iSUDQqgvHDJWi/0EDtIq16vYCOM Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:2 AEAD_AES_128_GCM inline:5abyoU5PKLg1lNfmLg1aBLIqyNAgmKNEEo8J1w Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:3 AES_256_CM_HMAC_SHA1_80 inline:i4O28fJjCyvBVQmwjCb7dySy5K20yeg5n1ky+uhPNSpbqkv8vxkcIaerkO9Gbg Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:4 AES_256_CM_HMAC_SHA1_32 inline:OOydiBrJ9nMu23yaBP7JuYNddpbXX8sN/V9Ab/psnTZIBRZKXenHWVPvJSM9Fw Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:5 AES_192_CM_HMAC_SHA1_80 inline:UtkHFdpsaZSO0nwvmEotIoGN+paLt3LHt7yMG4MfDv5rueBsb80 Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:6 AES_192_CM_HMAC_SHA1_32 inline:hgFJ9vS9Roz1vAMhnrCyviKC/TwCPgrk5jeKbxHlIEKtqSvLdK4 Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:7 AES_CM_128_HMAC_SHA1_80 inline:07Q9Z/8DguM0HJgDHv5HF3YDWHdp23hbz+TepHa8 Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:djpscuSL/05sVtFNocQpHmr1x+NTcLXKu8ePb0q+ Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:9 F8_128_HMAC_SHA1_80 inline:5FqC2VY+q3vqaju3JwroIJO/OlCM1i6TNRE/3Atj Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:10 F8_128_HMAC_SHA1_32 inline:kUfTs7OFD6ltRNpDi1dSG5ZjuQAFDeEQD5IvOwme Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:11 NULL_HMAC_SHA1_80 inline:JZ2ruS0914NL4mmS7jxg+ngz0IQaE6FBRlet8abd Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:12 NULL_HMAC_SHA1_32 inline:h21D+oGRkr1dDEoMm1+vEKmKsn+4GxNBwgIew2qA Mar 9 20:21:02 debian rtpengine[11823]: a=setup:actpass Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:sha-256 0C:67:09:6B:AE:6B:30:09:81:C5:9C:65:C6:EC:9E:CE:C2:6E:E2:EA:67:B4:B5:B3:40:C4:F1:0C:3E:05:96:79 Mar 9 20:21:02 debian rtpengine[11823]: a=tls-id:8100c85eb1c58be3580e8acd7f097000 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:hcqW22Sc Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:QjHJKT4qNsfZm80zrcyh8SfI76 Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:GtpEjl9r4IzNpiXr 1 UDP 2130706431 10.13.1.127 30000 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:GtpEjl9r4IzNpiXr 2 UDP 2130706430 10.13.1.127 30001 typ host Mar 9 20:21:02 debian rtpengine[11823]: ", "result": "ok" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7//1 port 30000]: [ice] Created candidate pair GtpEjl9r4IzNpiXr:6b1875839eb037b0:1 between 10.13.1.127 and 10.13.1.152:14450, type prflx Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7//1 port 30000]: [ice] Triggering check for GtpEjl9r4IzNpiXr:6b1875839eb037b0:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7//1 port 30000]: [core] Successful STUN binding request from 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Received command 'answer' from 127.0.0.1:60024 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Dump for 'answer' from 127.0.0.1:60024: { "supports": [ "load limit" ], "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 2654832326 4 IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: s=Asterisk Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic:WMS * Mar 9 20:21:02 debian rtpengine[11823]: m=audio 14450 UDP/TLS/RTP/SAVPF 111 126 Mar 9 20:21:02 debian rtpengine[11823]: a=connection:new Mar 9 20:21:02 debian rtpengine[11823]: a=setup:active Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:SHA-256 62:AD:7E:86:4D:FF:87:6B:44:5A:F0:62:A2:37:D6:56:5C:D4:F4:A8:03:88:A6:E1:DD:2A:15:69:75:1B:04:3C Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:7706bb5e0317151b5749cc8550354244 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:4af879c44cb99216424536fa5defd9d5 Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:Ha0d0198 1 UDP 2130706431 10.13.1.152 14450 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:H6719a439 1 UDP 2130706431 103.25.164.57 14450 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:126 0-16 Mar 9 20:21:02 debian rtpengine[11823]: a=ptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=maxptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=msid:d5be09fd-dce5-4d96-879e-f34f22f54ef1 a3f22c04-6988-4464-a458-5abc099e7bc0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:* transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: ", "call-id": "ui7jc53b9lfbt9lf4nu7", "received-from": [ "IP4", "10.13.1.152" ], "from-tag": "7n38ig1acl", "to-tag": "a5935482-3f00-4dd0-b98c-0b4726da777d", "command": "answer" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Using default bitrate of 32000 bps for 2-channel Opus Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unconfirming peer address for local 10.13.1.127:30000 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unconfirming peer address for local 10.13.1.127:30020 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unconfirming peer address for local 10.13.1.127:30001 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unconfirming peer address for local 10.13.1.127:30021 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unsubscribing '7n38ig1acl' from 'a5935482-3f00-4dd0-b98c-0b4726da777d' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unsubscribing 'a5935482-3f00-4dd0-b98c-0b4726da777d' from '7n38ig1acl' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Subscribing 'a5935482-3f00-4dd0-b98c-0b4726da777d' to '7n38ig1acl' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Subscribing '7n38ig1acl' to 'a5935482-3f00-4dd0-b98c-0b4726da777d' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating codecs for answerer a5935482-3f00-4dd0-b98c-0b4726da777d #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for a5935482-3f00-4dd0-b98c-0b4726da777d #1 <https://github.com/sipwise/rtpengine/issues/1> -> 7n38ig1acl #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Codec answer for a5935482-3f00-4dd0-b98c-0b4726da777d #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Reverse codec for opus/48000/2 (111) is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Reverse codec for telephone-event/8000 (126) is telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] telephone-event/8000/1 payload type 126 already present, skip Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for 7n38ig1acl #1 <https://github.com/sipwise/rtpengine/issues/1> -> a5935482-3f00-4dd0-b98c-0b4726da777d #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for 7n38ig1acl #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for a5935482-3f00-4dd0-b98c-0b4726da777d #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for 7n38ig1acl #1 <https://github.com/sipwise/rtpengine/issues/1> -> a5935482-3f00-4dd0-b98c-0b4726da777d #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for a5935482-3f00-4dd0-b98c-0b4726da777d #1 <https://github.com/sipwise/rtpengine/issues/1> -> 7n38ig1acl #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] set FILLED flag for stream, local 10.13.1.127:30000 remote 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] set FILLED flag for stream, local 10.13.1.127:30001 remote 10.13.1.152:14451 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [10.13.1.127:30001] Initialized incoming SRTP with SDES crypto params: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [2] Initialized outgoing SRTP with SDES crypto params: suite AEAD_AES_256_GCM, tag 1, key czR43Q05Tlky5pj/m6BomEKXh+b5okz/iSUDQqgvHDI= salt Vov9BA7SKter2Ajj Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Replacing previously learned prflx ICE candidate with Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Learning new ICE candidate H6719a439:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Created candidate pair GtpEjl9r4IzNpiXr:H6719a439:1 between 10.13.1.127 and 103.25.164.57:14450, type host Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Recalculating all ICE pair priorities Mar 9 20:21:02 debian rtpengine[11823]: WARNING: [ui7jc53b9lfbt9lf4nu7]: [ice] Priority collision between candidate pairs GtpEjl9r4IzNpiXr:H6719a439:1 and GtpEjl9r4IzNpiXr:Ha0d0198:1 - ICE will likely fail Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Sending ICE/STUN request for candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 from 10.13.1.127 to 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Shutting down ICE agent (nothing to do) Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Replying to 'answer' from 127.0.0.1:60024 (elapsed time 0.003226 sec) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Response dump for 'answer' to 127.0.0.1:60024: { "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 2654832326 4 IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: s=Asterisk Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic:WMS * Mar 9 20:21:02 debian rtpengine[11823]: m=audio 30020 UDP/TLS/RTP/SAVPF 111 126 Mar 9 20:21:02 debian rtpengine[11823]: a=connection:new Mar 9 20:21:02 debian rtpengine[11823]: a=maxptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=msid:d5be09fd-dce5-4d96-879e-f34f22f54ef1 a3f22c04-6988-4464-a458-5abc099e7bc0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:* transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:126 0-16 Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp:30020 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=setup:active Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:sha-256 0C:67:09:6B:AE:6B:30:09:81:C5:9C:65:C6:EC:9E:CE:C2:6E:E2:EA:67:B4:B5:B3:40:C4:F1:0C:3E:05:96:79 Mar 9 20:21:02 debian rtpengine[11823]: a=tls-id:ac2907348568344c9743f21fa1383e08 Mar 9 20:21:02 debian rtpengine[11823]: a=ptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:4xVkmu6h Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:kc54UEYL25S4UjgSjVKJdcEiAR Mar 9 20:21:02 debian rtpengine[11823]: a=ice-options:trickle Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:GtpEjl9r4IzNpiXr 1 UDP 2130706431 10.13.1.127 30020 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=end-of-candidates Mar 9 20:21:02 debian rtpengine[11823]: ", "result": "ok" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Processing incoming DTLS packet Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Sending DTLS packet Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] Received ICE/STUN response code 0 for candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 from 10.13.1.152:14450 to 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] Setting ICE candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 as succeeded Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] Best succeeded ICE pair with all components is GtpEjl9r4IzNpiXr:Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE not completed yet, but can use pair GtpEjl9r4IzNpiXr:Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE negotiated: peer for component 1 is 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE negotiated: local interface 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Processing incoming DTLS packet Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [crypto] DTLS: Peer certificate accepted Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [crypto] DTLS handshake successful Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [crypto] DTLS-SRTP successfully negotiated using AES_CM_128_HMAC_SHA1_80 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] SRTP keys, incoming: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] --- AES_CM_128_HMAC_SHA1_80 key AtAWESyBg5e6Ub3EPRQ5ag== salt rgwpYXp6KpHCmfjMxVA= Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] SRTP keys, outgoing: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] --- AES_CM_128_HMAC_SHA1_80 key 8+zHcQ5a1Bv5L0dv2AfKyg== salt 7YaPNc68d5ALtKKIOCQ= Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [crypto] DTLS-SRTP successfully negotiated using AES_CM_128_HMAC_SHA1_80 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] SRTP keys, incoming: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] --- AES_CM_128_HMAC_SHA1_80 key AtAWESyBg5e6Ub3EPRQ5ag== salt rgwpYXp6KpHCmfjMxVA= Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] SRTP keys, outgoing: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] --- AES_CM_128_HMAC_SHA1_80 key 8+zHcQ5a1Bv5L0dv2AfKyg== salt 7YaPNc68d5ALtKKIOCQ= Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Sending DTLS packet Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Successful STUN binding request from 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1]: [ice] Start nominating ICE pairs Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1]: [ice] Nominating ICE pair GtpEjl9r4IzNpiXr:Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1]: [ice] Sending nominating ICE/STUN request for candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 from 10.13.1.127 to 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] Received ICE/STUN response code 0 for candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 from 10.13.1.152:14450 to 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] Setting nominated ICE candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 as valid Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE completed, using pair GtpEjl9r4IzNpiXr:Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE negotiated: peer for component 1 is 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE negotiated: local interface 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Received command 'answer' from 127.0.0.1:60024 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Dump for 'answer' from 127.0.0.1:60024: { "supports": [ "load limit" ], "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 2654832326 4 IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: s=Asterisk Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic:WMS * Mar 9 20:21:02 debian rtpengine[11823]: m=audio 14450 UDP/TLS/RTP/SAVPF 111 126 Mar 9 20:21:02 debian rtpengine[11823]: a=connection:new Mar 9 20:21:02 debian rtpengine[11823]: a=setup:active Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:SHA-256 62:AD:7E:86:4D:FF:87:6B:44:5A:F0:62:A2:37:D6:56:5C:D4:F4:A8:03:88:A6:E1:DD:2A:15:69:75:1B:04:3C Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:7706bb5e0317151b5749cc8550354244 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:4af879c44cb99216424536fa5defd9d5 Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:Ha0d0198 1 UDP 2130706431 10.13.1.152 14450 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:H6719a439 1 UDP 2130706431 103.25.164.57 14450 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:126 0-16 Mar 9 20:21:02 debian rtpengine[11823]: a=ptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=maxptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=msid:d5be09fd-dce5-4d96-879e-f34f22f54ef1 a3f22c04-6988-4464-a458-5abc099e7bc0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:* transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: ", "call-id": "ui7jc53b9lfbt9lf4nu7", "received-from": [ "IP4", "10.13.1.152" ], "from-tag": "7n38ig1acl", "to-tag": "a5935482-3f00-4dd0-b98c-0b4726da777d", "command": "answer" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Using default bitrate of 32000 bps for 2-channel Opus Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating codecs for answerer a5935482-3f00-4dd0-b98c-0b4726da777d #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for a5935482-3f00-4dd0-b98c-0b4726da777d #1 <https://github.com/sipwise/rtpengine/issues/1> -> 7n38ig1acl #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Codec answer for a5935482-3f00-4dd0-b98c-0b4726da777d #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Reverse codec for opus/48000/2 (111) is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Reverse codec for telephone-event/8000 (126) is telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] telephone-event/8000/1 payload type 126 already present, skip Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for 7n38ig1acl #1 <https://github.com/sipwise/rtpengine/issues/1> -> a5935482-3f00-4dd0-b98c-0b4726da777d #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for 7n38ig1acl #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for a5935482-3f00-4dd0-b98c-0b4726da777d #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for 7n38ig1acl #1 <https://github.com/sipwise/rtpengine/issues/1> -> a5935482-3f00-4dd0-b98c-0b4726da777d #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for a5935482-3f00-4dd0-b98c-0b4726da777d #1 <https://github.com/sipwise/rtpengine/issues/1> -> 7n38ig1acl #1 <https://github.com/sipwise/rtpengine/issues/1> Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Dropping new ICE candidate Ha0d0198 in favour of Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Dropping new ICE candidate H6719a439 in favour of H6719a439:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Shutting down ICE agent (nothing to do) Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Replying to 'answer' from 127.0.0.1:60024 (elapsed time 0.002244 sec) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Response dump for 'answer' to 127.0.0.1:60024: { "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 2654832326 4 IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: s=Asterisk Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic:WMS * Mar 9 20:21:02 debian rtpengine[11823]: m=audio 30020 UDP/TLS/RTP/SAVPF 111 126 Mar 9 20:21:02 debian rtpengine[11823]: a=connection:new Mar 9 20:21:02 debian rtpengine[11823]: a=maxptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=msid:d5be09fd-dce5-4d96-879e-f34f22f54ef1 a3f22c04-6988-4464-a458-5abc099e7bc0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:* transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:126 0-16 Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp:30020 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=setup:active Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:sha-256 0C:67:09:6B:AE:6B:30:09:81:C5:9C:65:C6:EC:9E:CE:C2:6E:E2:EA:67:B4:B5:B3:40:C4:F1:0C:3E:05:96:79 Mar 9 20:21:02 debian rtpengine[11823]: a=tls-id:ac2907348568344c9743f21fa1383e08 Mar 9 20:21:02 debian rtpengine[11823]: a=ptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:4xVkmu6h Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:kc54UEYL25S4UjgSjVKJdcEiAR Mar 9 20:21:02 debian rtpengine[11823]: a=ice-options:trickle Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:GtpEjl9r4IzNpiXr 1 UDP 2130706431 10.13.1.127 30020 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=end-of-candidates Mar 9 20:21:02 debian rtpengine[11823]: ", "result": "ok" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] New ingress SSRC for: 10.13.1.152:14450 SSRC: 46ca8abf Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Handling packet: remote 10.13.1.152:14450 (expected: 10.13.1.152:14450) -> local 10.13.1.127:30000 (RTP seq 31783 TS 528 SSRC 46ca8abf) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Generated session key: master key f3ecc771..., master salt ed868f35..., label 00, length 16, result 6806f5cc... Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Generated session key: master key f3ecc771..., master salt ed868f35..., label 01, length 20, result 044b0cce... Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Generated session key: master key f3ecc771..., master salt ed868f35..., label 02, length 14, result 4426c95d... Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] New egress (direct) SSRC for: 0.0.0.0:9 SSRC: 46ca8abf Mar 9 20:21:02 debian rtpengine[11823]: ERR: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] SRTP output wanted, but no crypto suite was negotiated Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Error when sending message. Error: Inappropriate ioctl for device Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Handling packet: remote 10.13.1.152:14450 (expected: 10.13.1.152:14450) -> local 10.13.1.127:30000 (RTP seq 31784 TS 1488 SSRC 46ca8abf) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Handling packet: remote 10.13.1.152:14450 (expected: 10.13.1.152:14450) -> local 10.13.1.127:30000 (RTP seq 31785 TS 2448 SSRC 46ca8abf) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Handling packet: remote 10.13.1.152:14450 (expected: 10.13.1.152:14450) -> local 10.13.1.127:30000 (RTP seq 31786 TS 3408 SSRC 46ca8abf) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Handling packet: remote 10.13.1.152:14450 (expected:

2 1

Dispatcher behavior
by Jawaid Bazyar 15 Mar '23

15 Mar '23

Hi, is this expected behavior? I have two target hosts in a dispatcher set, 10.20.20.118 (prio 5) and 10.20.20.252 (prio 10). When I have 10.20.20.252 turned off, dispatcher still tries to send packets to that host, even when the status flags are IP. root@vs-rs01:/etc/kamailio# kamcmd dispatcher.list { NRSETS: 1 RECORDS: { SET: { ID: 1 TARGETS: { DEST: { URI: sip:10.20.20.252:5060 FLAGS: IP PRIORITY: 10 LATENCY: { AVG: 0.000000 STD: 0.000000 EST: 0.000000 MAX: 0 TIMEOUT: 62 } } DEST: { URI: sip:10.20.20.118:5060 FLAGS: AP PRIORITY: 5 LATENCY: { AVG: 3.876000 STD: 2.965000 EST: 5.058000 MAX: 14 TIMEOUT: 0 } } } } } }

5 14

Error on transfer when calling from WebRTC with TOPOH
by Arsenijs Kabihno 15 Mar '23

15 Mar '23

Hello, We have an issue which appears when TOPOH module is enabled, call is made from WebRTC, then transferred by calee: Mar 15 11:55:27 ap07stvp.xxxxxxx.xxx docker[2062]: 29(35) INFO: {2 2 INVITE g2rgvnioadg8m7l6i7k6} <script>: RTPENGINE branch_id:0 ruri: <null>, method:INVITE, status:200, extra_id: z9hG4bK66807520, rtpengine_answer: replace-origin replace-session-connection SIP-source-address direction=priv1 direction=pub1 SIP-source-address via-branch=extra rtcp-mux-offer generate-mid DTLS=passive SDES-off ICE=force RTP/SAVPF Mar 15 11:55:27 ap07stvp.xxxxxxx.xxx docker[2062]: 29(35) NOTICE: {2 2 INVITE g2rgvnioadg8m7l6i7k6} acc [acc.c:287]: acc_log_request(): ACC: transaction answered: timestamp=1678881327;method=INVITE;from_tag=p9qqi7hrua;to_tag=ea7d2850-3e76-4c11-821c-a0d1e3cbb011;call_id=g2rgvnioadg8m7l6i7k6;code=200;reason=OK;src_user=92010;src_domain=10.181.0.1;src_ip=10.5.0.145;dst_ouser=27221;dst_user=27221;dst_domain=10.181.0.1 Mar 15 11:55:38 ap07stvp.xxxxxxx.xxx docker[2062]: 31(37) INFO: {1 27689 BYE e034a187-0e4d-4c4e-ab4b-f2e5f1281330} <script>: RTPENGINE branch_id:0 ruri: sip:97093@10.128.150.67:46058;rinstance=d403efb922420649;transport=UDP, method:BYE, status:<null>, extra_id: <null>, rtpengine_delete Mar 15 11:55:38 ap07stvp.xxxxxxx.xxx docker[2062]: 28(34) NOTICE: {2 27689 BYE e034a187-0e4d-4c4e-ab4b-f2e5f1281330} acc [acc.c:287]: acc_log_request(): ACC: transaction answered: timestamp=1678881338;method=BYE;from_tag=0c29be45-6aff-4d4e-8503-650893a5a5ac;to_tag=b1a8a401;call_id=e034a187-0e4d-4c4e-ab4b-f2e5f1281330;code=200;reason=OK;src_user=22628;src_domain=10.5.2.192;src_ip=10.5.2.192;dst_ouser=97093;dst_user=97093;dst_domain=10.128.150.67 Mar 15 11:55:39 ap07stvp.xxxxxxx.net docker[2062]: 32(38) INFO: {1 27798 INVITE g2rgvnioadg8m7l6i7k6} <script>: RTPENGINE branch_id:0 ruri: sip:jgqtuev6@2idi5cucan2v.invalid;transport=ws;ob, method:INVITE, status:<null>, extra_id: <null>, rtpengine_offer: replace-origin replace-session-connection SIP-source-address direction=priv1 direction=pub1 SIP-source-address direction=priv1 direction=pub1 rtcp-mux-offer generate-mid DTLS=passive SDES-off ICE=force RTP/SAVPF Mar 15 11:55:39 ap07stvp.xxxxxxx.xxx docker[2062]: 27(33) INFO: {1 4728 INVITE 133ccd4c-031b-4d10-b073-745f35433132} <script>: RTPENGINE branch_id:0 ruri: sip:0037126048570@193.138.189.220, method:INVITE, status:<null>, extra_id: z9hG4bKPjedcf2d96-01cd-43fd-bed0-4e601e3e542f0, rtpengine_offer: replace-origin replace-session-connection SIP-source-address direction=priv1 direction=pub1 via-branch=extra Mar 15 11:55:39 ap07stvp.xxxxxxx.xxx docker[2062]: 50(56) INFO: {2 27798 INVITE g2rgvnioadg8m7l6i7k6} <script>: RTPENGINE branch_id:0 ruri: <null>, method:INVITE, status:200, extra_id: z9hG4bKPj1cd87bda-7862-4ca9-8475-ac6b342c45ef0, rtpengine_answer: replace-origin replace-session-connection SIP-source-address direction=pub1 direction=priv1 SIP-source-address via-branch=extra rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP Mar 15 11:55:39 ap07stvp.xxxxxxx.xxx docker[2062]: 25(31) ERROR: {1 27798 ACK g2rgvnioadg8m7l6i7k6} <core> [core/resolve.c:1730]: sip_hostport2su(): could not resolve hostname: "2idi5cucan2v.invalid" Mar 15 11:55:39 ap07stvp.xxxxxxx.xxx docker[2062]: 25(31) ERROR: {1 27798 ACK g2rgvnioadg8m7l6i7k6} <core> [core/forward.c:516]: forward_request(): bad host name 2idi5cucan2v.invalid, dropping packet Mar 15 11:55:39 ap07stvp.xxxxxxx.xxx docker[2062]: 25(31) ERROR: {1 27798 ACK g2rgvnioadg8m7l6i7k6} sl [sl_funcs.c:418]: sl_reply_error(): stateless error reply used: Unresolvable destination (478/SL) We have the following setup: WebRTC client behind load-balanser -> kamailio -> asterisk. When call is transferred, asterisk makes re-invite. If TOPOH is disabled, it happens this way and issue is not present: asterisk INVITE -> kamailio -> WebRTC WebRTC 200 -> kamailio -> asterisk asterisk ACK -> kamailio -> WebRTC ACK example: 03/09/2023 09:35:48.126 +02:00: 10.5.2.192:5060 -> 10.181.0.1:5060 ACK sip:98dn3met@0j62ln34srjn.invalid;transport=ws;ob;alias=10.5.0.145~27562~6 SIP/2.0 Via: SIP/2.0/UDP 10.5.2.192:5060;rport;branch=z9hG4bKPj1d11dae2-10bf-466e-9221-5710f801d725 From: <sip:27221@10.181.0.1>;tag=b6b73135-efe1-4d86-bbc8-bf4313130eb5 To: <sip:92010@10.181.0.1>;tag=namgu127j9 Call-ID: t0hmmk50h5i4ok88asjj CSeq: 28632 ACK Route: <sip:10.181.0.1:5060;lr;r2=on;ftag=namgu127j9;did=aaa.7b92;rtp=bridge;rtp=ws> Route: <sip:10.181.0.1:4444;transport=ws;lr;r2=on;ftag=namgu127j9;did=aaa.7b92;rtp=bridge;rtp=ws> Max-Forwards: 70 User-Agent: Asterisk PBX 18.16.0 Content-Length: 0 If TOPOH is enabled, kamailio doesn't pass this ACK to WebRTC: asterisk INVITE -> kamailio -> WebRTC WebRTC 200 -> kamailio -> asterisk asterisk ACK -> kamailio -> -> -> ACK example: 03/09/2023 09:33:03.544 +02:00: 10.5.2.192:5060 -> 10.181.0.1:5060 ACK sip:10.0.0.1;line=Bn7Ipzh8eb8DxMSFyk.rWzitxzVFBoirxRG5xYeOxb7hpoXTlMGDBbuTCk3oBDAqlH** SIP/2.0 Via: SIP/2.0/UDP 10.5.2.192:5060;rport;branch=z9hG4bKPjf00fc541-f011-4e52-b63a-2c569478ca39 From: <sip:27221@10.181.0.1>;tag=7d402872-7355-4269-b912-e52ee28ed415 To: <sip:92010@10.181.0.1>;tag=t0q4g9am9g Call-ID: t0hmmetickvlvn5k7nco CSeq: 25565 ACK Route: <sip:10.181.0.1:5060;lr;r2=on;ftag=t0q4g9am9g;did=4c8.a182;rtp=bridge;rtp=ws> Route: <sip:10.0.0.1;line=Bn7IpzdIwzd8Vf8Iwzd9WkyFWkAFBELJBo.qBYyuCoVgBz1uxn8gx61uxn8geYXOeD3FV6dFeD7Oxj7YpnX5ekFFlDHJljd8VzATC60ulYi5ebC7poiFBk3oBI**> Max-Forwards: 70 User-Agent: Asterisk PBX 18.16.0 Content-Length: 0 We have an assumption that issue might be related to the way how TOPOH handles alias in contact. Did someone have encountered something like this?

1 0

Advertise NAT IP address causing Multiple SIP INVITEs is that expected ?
by Adil Mafzool Zaka 15 Mar '23

15 Mar '23

Hi I am using advertise in the listen interface for our NATed setup within AWS. Using following listen statement in kamailio: - version: kamailio 5.5.1 listen=udp:10.0.1.4:5060 advertise 3X.1XX.1XX.2XX:5060 dispatcher.list 1 sip:10.0.4.23:5060 8 10 weight=11 #FreeSwitch 1 We are seeing that kamailio (10.0.1.4) is sending two SIP INVITEs via dispatcher to FreeSwitch at 10.0.4.23 which seems strange or is it expected behaviour? (As seen below) SIP INVITE Request: - WAN Address >> Kamailio AWS NATed address >> FreeSwitch 1 1X.2X.2X.2X >> 3X.1XX.1XX.2XX 10.0.1.4 >> 10.0.4.23 10.0.1.4 >> 10.0.4.23 Many thanks Regards Muhammad Zaka

2 1

Use Kamailio and RTPEngine to play wav file as early media to announce a server maintenance
by jhayhoe＠liquidweb.com 15 Mar '23

15 Mar '23

I am looking to configure Kamailio and RTPEngine to play a wav file for any new calls during a scheduled asterisk server maintenance. I have set up the following within route{} to send the call to a early_media route based on a variable that I can set on the server using kamcmd. route { ... if ($sht(config=>maint)=="1") { if (is_method("INVITE") && !has_totag()) { xlog ("L_INFO","Ongoing maintenance. Call-ID:$ci | Method: $rm"); route(early_media); } } ... } route[early_media] { xlog("Maintenance: handling early_media\n"); rtpengine_manage("direction=external direction=external"); $var(body) = $(rb{re.subst,/(IP4.).*/\1##server IP redacted##/g}); set_reply_body($var(body), "application/sdp"); t_send_reply("183", "Session Progress"); sleep(2); unblock_dtmf("all"); play_media("file=/etc/kamailio/media/maintenance.wav"); sleep(21); rtpengine_delete(); t_reply("480","Temporarily Unavailable"); } This mostly works as I hear the recorded message in place of the ringing tone when a call is processed by kamailio and then a busy signal is heard as part of the 480 reply at the end. The sleep's are in place to match the duration of the recorded message as otherwise the 480 is sent immediately following the start of the play_media. Feel free to let me know if there is a better solution to wait the duration of the media file and then proceed with processing instead of using sleep(). Is there a way to instead send a CANCEL or a BYE instead of a 480 after the wav file is played? I tried using t_cancel_callid("$ci", "$cs", "0") as well as a dlg_bye("all") at the end of the early_media route and neither seemed to send any headers out.

3 3

stuck with webrtc on kamailio
by Arun K R 14 Mar '23

14 Mar '23

hello, I have issue of no voice on kamailio configured for webrtc. I have configured rtpengine with public and private interface. My kamilio has two interface of pubic and private ip. Somebody has suggested me that there is no ice trickling happens. I am using jssip webrtc client. Kindly suggest me or help me with this. I have send another email for same issue to this group and didnt get any reply. Kindly help me

1 0

Re: Your message to sr-users@lists.kamailio.org awaits moderator approval
by Arun K R 14 Mar '23

14 Mar '23

I need a help on issue with rtpengine configuration and I didn't get any repy for it. Kindly check about it and help me. If i won't get any help kindly let me know, so I can stop waiting for your reply On Mon, Mar 13, 2023 at 12:50 PM <sr-users-bounces(a)lists.kamailio.org> wrote: > Your mail to 'sr-users(a)lists.kamailio.org' with the subject > > webrtc audio issue > > Is being held until the list moderator can review it for approval. > > The message is being held because: > > The message is larger than the 150 KB maximum size > > Either the message will get posted to the list, or you will receive > notification of the moderator's decision. >

1 0

send messages from source ip in Linux anyIP list
by mohsen khashei 14 Mar '23

14 Mar '23

1 0

Dispatcher Incoming
by nutxase 14 Mar '23

14 Mar '23

Hi All I have setup kamailio using dispatcher to proxy registrations from the UAC to asterisk but when asterisk sends an incoming call it does not seem to keep the path header and therefore kamailio sends 404 is there anyway around this? Thanks Sent with [Proton Mail](https://proton.me/) secure email.

5 18

Use MySQL connector instead of mariadb connector
by suchendra adiga 14 Mar '23

14 Mar '23

Hi, I have been working on setting up the database with the Kamailio server. But facing a problem when my db user has configured with SSL required. But I learnt that there is no support to enable the SSL in mariadb connector. So I am thinking of using the MySQL connector instead. Is there anyway I could configure the kamailio to use MySQL instead of Mariadb? Thanks and regards, Suchendra

2 7

Advertise NAT IP address causing Multiple SIP INVITEs is that expected ?
by Muhammad Zaka 13 Mar '23

13 Mar '23

Hi We are using advertise in the listen interface for our NATed setup within AWS. Using following listen statement in kamailio: - version: kamailio 5.5.1 listen=udp:10.0.1.4:5060 advertise 3X.1XX.1XX.2XX:5060 dispatcher.list 1 sip:10.0.4.23:5060 8 10 weight=11 #FreeSwitch 1 We are seeing that kamailio (10.0.1.4) is sending two SIP INVITEs via dispatcher to FreeSwitch at 10.0.4.23 which seems strange or is it expected behaviour? (As seen below) SIP INVITE Request: - WAN Address --> Kamailio AWS NATed address --> FreeSwitch 1 1X.2X.2X.2X --> 3X.1XX.1XX.2XX 10.0.1.4 --> 10.0.4.23 10.0.1.4 --> 10.0.4.23 [cid:image001.png@01D955CE.9F0C2EE0] Many Thanks Regards Muhammad Zaka Muhammad Zaka UCaaS Solutions Engineer Tel: 03330147908 Email: Muhammad.Zaka(a)gamma.co.uk This is an email from Gamma Telecom Ltd, trading as “Gamma”. The contents of this email are confidential to the ordinary user of the email address to which it was addressed. This email is not intended to create any legal relationship. No one else may place any reliance upon it, or copy or forward all or any of it in any form (unless otherwise notified). If you receive this email in error, please accept our apologies, we would be obliged if you would telephone our postmaster on +44 (0) 808 178 9652 or email postmaster(a)gamma.co.uk Gamma Telecom Limited, a company incorporated in England and Wales, with limited liability, with registered number 04340834, and whose registered office is at The Scalpel, 18th Floor, 52 Lime Street, London EC3M 7AF and whose principal place of business is at Kings House, Kings Road West, Newbury, Berkshire, RG14 5BY.

1 0

(no subject)
by gui 10 Mar '23

10 Mar '23

Hello I have an issue with Rx in my VolTE call Flow. Please refer to picture and pcap in annex. I am using same rtp engine at origination and termination. I am sending rtpmanage for invite and 183 session progress at both origination and termination. My issue is that Rx is not sending for the proper media flow component to PCRF. At origination it is considering a media flow between A 50034 and RTPEngine 49186 (instead of considering RTP Engine Port 49152) At Termination it is considering a media flow between B 50054 and RTPEngine 49132 (instead of considering RTP EnginePort 49170) Any idea why?

3 4

use_dns_failover and dispatcher
by Lars Jansson 10 Mar '23

10 Mar '23

Hi, I had the intention to route calls to some SRV destinations via dispatcher and to other SRV destinations via DNS lookup and 'use_dns_failover=on'. Currently in a route block using dispatcher two out of three targets within the SRV destination are unavailable (by intention). During my tests to the dispatcher destination I have found that: A) If I have 'use_dns_failover=on' this setting "overrules" the dispatcher routing and the unavailable targets are also tried before DNS failover finds the available target. B) If I have 'use_dns_failover=off' the dispatcher routing logic works as expected and the call is always routed to the only available target directly, making use of the result of the dispatcher OPTIONS pinging. -- So my conclusion is now that you have to choose: Either use dispatcher routing, or use DNS failover based routing. - I'm fine with this, but I just would like to know if my conclusion is correct, or if there is actually a way to use dispacther for some SRV destinations and DNS failover for other SRV destinations. Regards, Lars

1 0

New 'reverse' transaction not using correct transport protocol
by Benoit Panizzon 10 Mar '23

10 Mar '23

Hi CPE registering with TCP (or TLS). 2nd leg is UDP. Within one transaction (INVITE to OK including all ACK and PRACK) transport is kept as desired for both legs. But when the B side disconnects (sending BYE, new reverse transaction), this is sent via UDP to the A CPE which initially talked TCP and thus ignored. record_route() is called on each leg. What could I be missing? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

4 6

Silo extra_hdrs duplication
by Dr. Barabás Péter 10 Mar '23

10 Mar '23

Hi, I’m using kamailio’s silo module to store offline messages in it. I set extra_hdrs modparam to e.g. : ‘Resent-from-silo’ so the clients get information that the message has arrived directly (immediatelly) from the sender or it was stored offline and comes from silo. So the problem is the following: * if receiver’s socket is broken (e.g. airplane mode on mobile) and kamailio ‘thinks’ it is registered, server tries to send the message * message will be timed out and based on config it will be stored in silo. * the next step is to check if receiver is registered or not (since the client can re-register during 30sec until the message has timed out) * If client is registered, kamailio tries to send the message immediately with m_dump(). * If the receiver’s connection is still broken, the message will be timed out and store in silo. * In every store, a new ‘resent’ extra_hdrs value is appended. When extra_hdrs length reaches 1024 bytes, the m_dump will fail and it blocks the dumping of messages to the given receiver. Question: can extra_hdrs value be removed before store? Or what can be the solution not to duplicate the extra_hdrs value in some bad network situation? Peter

3 2

Re: [sr-dev] KEMI python 'KSR' has no attribute 'sanity'
by Henning Westerholt 10 Mar '23

10 Mar '23

(adding sr-users) Hello, maybe because you're spelling it wrong? Compare e.g., to here: https://kamailio.org/docs/tutorials/5.6.x/kamailio-kemi-framework/modules/#… Cheers, Henning -----Original Message----- From: agural5(a)gmail.com <agural5(a)gmail.com> Sent: Freitag, 10. März 2023 08:32 To: sr-dev(a)lists.kamailio.org Subject: [sr-dev] KEMI python 'KSR' has no attribute 'sanity' Hello, I am trying to use KEMI KSR.sanity in python but got error: AttributeError: module 'KSR' has no attribute 'sanity' appreciate your help for solve this. Thanks! _______________________________________________ Kamailio (SER) - Development Mailing List To unsubscribe send an email to sr-dev-leave(a)lists.kamailio.org

2 1

webrtc from public internet have no voice
by mail4arunkr＠gmail.com 10 Mar '23

10 Mar '23

Hello, I have installed Kamailio 5.6 in debian 11 and RTPengine 11.3 also in the same server. I have configured kamailio to work as webrtc server and it forwards the registration to asterisk. Now when I am trying call from jssip webrtc client it reaches kamailio and route it through private interface to asterisk server. Asterisk then route it to the provider server. When i make a call asterisk server recives rtp from the provider and convert the rtp to srtp and sending back to kamailio. but there is no sound for webrtc from public internet . Also i am getting warning (SRTCP /RTP output wanted, but no crypto suite was negotiated) webrtc from Local network works fine with VPN For normal udp call without webrtc works fine. Kamailio having two interfaces, interface1 is private 10.13.1.140 and interface 2 is publi ip 100.x.x.x webrtc client sends calls to the public ip interface 100.x.x.x and kamaiio routes the call to the asterisk via private interface 10.13.1.140. asterisk server sends the call to remote server and gets the rtp back from that Asterisk server ---converts RTP to SRTP and forward to ----> kamailio 10.13.1.140 but there is nothing happens after that, please help me on this i am new to kamailio and rtpengine. The issue is only for webrtc from public internet. But when using webrtc from LAN works fine below are the logs Mar 9 20:21:02 debian /usr/local/sbin/kamailio[11778]: INFO: <script>: from domain recived is 10.13.1.152 Mar 9 20:21:02 debian /usr/local/sbin/kamailio[11778]: INFO: <script>: 09496381412 has fromDomain recived is 10.13.1.152 Mar 9 20:21:02 debian /usr/local/sbin/kamailio[11778]: INFO: <script>: Request going to srtp customied arun Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [control] Adding a proxy for control ng stats:127.0.0.1 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Received command 'offer' from 127.0.0.1:53374 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Dump for 'offer' from 127.0.0.1:53374: { "supports": [ "load limit" ], "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 7377619262916888262 2 IN IP4 127.0.0.1 Mar 9 20:21:02 debian rtpengine[11823]: s=- Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=group:BUNDLE 0 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap-allow-mixed Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic: WMS mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB Mar 9 20:21:02 debian rtpengine[11823]: m=audio 9 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126 Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 0.0.0.0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp:9 IN IP4 0.0.0.0 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:NM2h Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:BF2WmB5sOZZWVIFppZ3YTiSH Mar 9 20:21:02 debian rtpengine[11823]: a=ice-options:trickle Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:sha-256 9F:FA:18:B0:8D:17:A5:14:FE:22:BA:1E:73:1E:3E:02:6F:17:EC:B5:FC:77:BD:E1:0B:51:DF:7F:61:B0:2F:55 Mar 9 20:21:02 debian rtpengine[11823]: a=setup:actpass Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=msid:mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB 6f36294e-5d92-4f0c-8ea2-ba415d0ec125 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:111 transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:111 minptime=10;useinbandfec=1 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:63 red/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:63 111/111 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:9 G722/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:0 PCMU/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:8 PCMA/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:13 CN/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:110 telephone-event/48000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=ssrc:2480159127 cname:FJAL1TGxCcnDMPXY Mar 9 20:21:02 debian rtpengine[11823]: a=ssrc:2480159127 msid:mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB 6f36294e-5d92-4f0c-8ea2-ba415d0ec125 Mar 9 20:21:02 debian rtpengine[11823]: ", "call-id": "ui7jc53b9lfbt9lf4nu7", "received-from": [ "IP4", "223.190.206.123" ], "from-tag": "7n38ig1acl", "command": "offer" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Using default bitrate of 32000 bps for 2-channel Opus Mar 9 20:21:02 debian rtpengine[11823]: NOTICE: [ui7jc53b9lfbt9lf4nu7]: [core] Creating new call Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Subscribing '7n38ig1acl' to '' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Subscribing '' to '7n38ig1acl' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating codecs for offerer 7n38ig1acl #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec red/48000/2 (63) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec G722/8000 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec PCMU/8000 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec PCMA/8000 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec CN/8000 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/48000 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating codecs for answerer #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec red/48000/2 (63) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec G722/8000 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec PCMU/8000 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec PCMA/8000 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec CN/8000 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/48000 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for #1 -> 7n38ig1acl #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF 110, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec red/48000/2 (63) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for red/48000/2 (63) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] No codec support for red/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for red/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec G722/8000/1 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for G722/8000 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for G722/8000 is G722/8000/1 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec G722/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for G722/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec PCMU/8000/1 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for PCMU/8000 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for PCMU/8000 is PCMU/8000/1 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec PCMU/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for PCMU/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec PCMA/8000/1 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for PCMA/8000 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for PCMA/8000 is PCMA/8000/1 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec PCMA/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for PCMA/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec CN/8000/1 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for CN/8000 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for CN/8000 is CN/8000/1 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec CN/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for CN/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/48000/1 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for telephone-event/48000 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/48000 is telephone-event/48000/1 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/48000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/48000 with DTMF 110, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for #1 -> 7n38ig1acl #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF 110, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec red/48000/2 (63) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] No codec support for red/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for red/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec G722/8000/1 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for G722/8000 is G722/8000/1 (9) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec G722/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for G722/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec PCMU/8000/1 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for PCMU/8000 is PCMU/8000/1 (0) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec PCMU/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for PCMU/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec PCMA/8000/1 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for PCMA/8000 is PCMA/8000/1 (8) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec PCMA/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for PCMA/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec CN/8000/1 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for CN/8000 is CN/8000/1 (13) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec CN/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for CN/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/48000/1 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/48000 is telephone-event/48000/1 (110) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/48000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/48000 with DTMF 110, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN 13 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] set FILLED flag for stream, local 10.13.1.127:30020 remote 0.0.0.0:9 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] Creating active DTLS connection context Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] Creating active DTLS connection context Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] set FILLED flag for stream, local 10.13.1.127:30021 remote 0.0.0.0:9 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [10.13.1.127:30000] Initialized incoming SRTP with SDES crypto params: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [1] Initialized outgoing SRTP with SDES crypto params: suite AEAD_AES_256_GCM, tag 1, key czR43Q05Tlky5pj/m6BomEKXh+b5okz/iSUDQqgvHDI= salt Vov9BA7SKter2Ajj Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] Creating passive DTLS connection context Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] Creating passive DTLS connection context Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [10.13.1.127:30001] Initialized incoming SRTP with SDES crypto params: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [2] Initialized outgoing SRTP with SDES crypto params: suite AEAD_AES_256_GCM, tag 1, key czR43Q05Tlky5pj/m6BomEKXh+b5okz/iSUDQqgvHDI= salt Vov9BA7SKter2Ajj Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Replying to 'offer' from 127.0.0.1:53374 (elapsed time 0.022450 sec) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Response dump for 'offer' to 127.0.0.1:53374: { "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 7377619262916888262 2 IN IP4 127.0.0.1 Mar 9 20:21:02 debian rtpengine[11823]: s=- Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap-allow-mixed Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic: WMS mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB Mar 9 20:21:02 debian rtpengine[11823]: m=audio 30000 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126 Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 Mar 9 20:21:02 debian rtpengine[11823]: a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid Mar 9 20:21:02 debian rtpengine[11823]: a=msid:mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB 6f36294e-5d92-4f0c-8ea2-ba415d0ec125 Mar 9 20:21:02 debian rtpengine[11823]: a=ssrc:2480159127 cname:FJAL1TGxCcnDMPXY Mar 9 20:21:02 debian rtpengine[11823]: a=ssrc:2480159127 msid:mP5OKOsd0Yb5JQystk9oCQJVA8LOp4pWhuJB 6f36294e-5d92-4f0c-8ea2-ba415d0ec125 Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:111 minptime=10;useinbandfec=1 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:111 transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:63 red/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:63 111/111 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:9 G722/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:0 PCMU/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:8 PCMA/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:13 CN/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:110 telephone-event/48000 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp:30001 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:1 AEAD_AES_256_GCM inline:czR43Q05Tlky5pj/m6BomEKXh+b5okz/iSUDQqgvHDJWi/0EDtIq16vYCOM Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:2 AEAD_AES_128_GCM inline:5abyoU5PKLg1lNfmLg1aBLIqyNAgmKNEEo8J1w Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:3 AES_256_CM_HMAC_SHA1_80 inline:i4O28fJjCyvBVQmwjCb7dySy5K20yeg5n1ky+uhPNSpbqkv8vxkcIaerkO9Gbg Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:4 AES_256_CM_HMAC_SHA1_32 inline:OOydiBrJ9nMu23yaBP7JuYNddpbXX8sN/V9Ab/psnTZIBRZKXenHWVPvJSM9Fw Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:5 AES_192_CM_HMAC_SHA1_80 inline:UtkHFdpsaZSO0nwvmEotIoGN+paLt3LHt7yMG4MfDv5rueBsb80 Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:6 AES_192_CM_HMAC_SHA1_32 inline:hgFJ9vS9Roz1vAMhnrCyviKC/TwCPgrk5jeKbxHlIEKtqSvLdK4 Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:7 AES_CM_128_HMAC_SHA1_80 inline:07Q9Z/8DguM0HJgDHv5HF3YDWHdp23hbz+TepHa8 Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:djpscuSL/05sVtFNocQpHmr1x+NTcLXKu8ePb0q+ Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:9 F8_128_HMAC_SHA1_80 inline:5FqC2VY+q3vqaju3JwroIJO/OlCM1i6TNRE/3Atj Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:10 F8_128_HMAC_SHA1_32 inline:kUfTs7OFD6ltRNpDi1dSG5ZjuQAFDeEQD5IvOwme Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:11 NULL_HMAC_SHA1_80 inline:JZ2ruS0914NL4mmS7jxg+ngz0IQaE6FBRlet8abd Mar 9 20:21:02 debian rtpengine[11823]: a=crypto:12 NULL_HMAC_SHA1_32 inline:h21D+oGRkr1dDEoMm1+vEKmKsn+4GxNBwgIew2qA Mar 9 20:21:02 debian rtpengine[11823]: a=setup:actpass Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:sha-256 0C:67:09:6B:AE:6B:30:09:81:C5:9C:65:C6:EC:9E:CE:C2:6E:E2:EA:67:B4:B5:B3:40:C4:F1:0C:3E:05:96:79 Mar 9 20:21:02 debian rtpengine[11823]: a=tls-id:8100c85eb1c58be3580e8acd7f097000 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:hcqW22Sc Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:QjHJKT4qNsfZm80zrcyh8SfI76 Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:GtpEjl9r4IzNpiXr 1 UDP 2130706431 10.13.1.127 30000 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:GtpEjl9r4IzNpiXr 2 UDP 2130706430 10.13.1.127 30001 typ host Mar 9 20:21:02 debian rtpengine[11823]: ", "result": "ok" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7//1 port 30000]: [ice] Created candidate pair GtpEjl9r4IzNpiXr:6b1875839eb037b0:1 between 10.13.1.127 and 10.13.1.152:14450, type prflx Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7//1 port 30000]: [ice] Triggering check for GtpEjl9r4IzNpiXr:6b1875839eb037b0:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7//1 port 30000]: [core] Successful STUN binding request from 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Received command 'answer' from 127.0.0.1:60024 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Dump for 'answer' from 127.0.0.1:60024: { "supports": [ "load limit" ], "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 2654832326 4 IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: s=Asterisk Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic:WMS * Mar 9 20:21:02 debian rtpengine[11823]: m=audio 14450 UDP/TLS/RTP/SAVPF 111 126 Mar 9 20:21:02 debian rtpengine[11823]: a=connection:new Mar 9 20:21:02 debian rtpengine[11823]: a=setup:active Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:SHA-256 62:AD:7E:86:4D:FF:87:6B:44:5A:F0:62:A2:37:D6:56:5C:D4:F4:A8:03:88:A6:E1:DD:2A:15:69:75:1B:04:3C Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:7706bb5e0317151b5749cc8550354244 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:4af879c44cb99216424536fa5defd9d5 Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:Ha0d0198 1 UDP 2130706431 10.13.1.152 14450 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:H6719a439 1 UDP 2130706431 103.25.164.57 14450 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:126 0-16 Mar 9 20:21:02 debian rtpengine[11823]: a=ptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=maxptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=msid:d5be09fd-dce5-4d96-879e-f34f22f54ef1 a3f22c04-6988-4464-a458-5abc099e7bc0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:* transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: ", "call-id": "ui7jc53b9lfbt9lf4nu7", "received-from": [ "IP4", "10.13.1.152" ], "from-tag": "7n38ig1acl", "to-tag": "a5935482-3f00-4dd0-b98c-0b4726da777d", "command": "answer" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Using default bitrate of 32000 bps for 2-channel Opus Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unconfirming peer address for local 10.13.1.127:30000 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unconfirming peer address for local 10.13.1.127:30020 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unconfirming peer address for local 10.13.1.127:30001 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unconfirming peer address for local 10.13.1.127:30021 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unsubscribing '7n38ig1acl' from 'a5935482-3f00-4dd0-b98c-0b4726da777d' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Unsubscribing 'a5935482-3f00-4dd0-b98c-0b4726da777d' from '7n38ig1acl' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Subscribing 'a5935482-3f00-4dd0-b98c-0b4726da777d' to '7n38ig1acl' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Subscribing '7n38ig1acl' to 'a5935482-3f00-4dd0-b98c-0b4726da777d' Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating codecs for answerer a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for a5935482-3f00-4dd0-b98c-0b4726da777d #1 -> 7n38ig1acl #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Codec answer for a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Reverse codec for opus/48000/2 (111) is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Reverse codec for telephone-event/8000 (126) is telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] telephone-event/8000/1 payload type 126 already present, skip Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for 7n38ig1acl #1 -> a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Creating codec handler for telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for 7n38ig1acl #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for 7n38ig1acl #1 -> a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for a5935482-3f00-4dd0-b98c-0b4726da777d #1 -> 7n38ig1acl #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] set FILLED flag for stream, local 10.13.1.127:30000 remote 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] set FILLED flag for stream, local 10.13.1.127:30001 remote 10.13.1.152:14451 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [10.13.1.127:30001] Initialized incoming SRTP with SDES crypto params: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [crypto] [2] Initialized outgoing SRTP with SDES crypto params: suite AEAD_AES_256_GCM, tag 1, key czR43Q05Tlky5pj/m6BomEKXh+b5okz/iSUDQqgvHDI= salt Vov9BA7SKter2Ajj Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Replacing previously learned prflx ICE candidate with Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Learning new ICE candidate H6719a439:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Created candidate pair GtpEjl9r4IzNpiXr:H6719a439:1 between 10.13.1.127 and 103.25.164.57:14450, type host Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Recalculating all ICE pair priorities Mar 9 20:21:02 debian rtpengine[11823]: WARNING: [ui7jc53b9lfbt9lf4nu7]: [ice] Priority collision between candidate pairs GtpEjl9r4IzNpiXr:H6719a439:1 and GtpEjl9r4IzNpiXr:Ha0d0198:1 - ICE will likely fail Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Sending ICE/STUN request for candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 from 10.13.1.127 to 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Shutting down ICE agent (nothing to do) Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Replying to 'answer' from 127.0.0.1:60024 (elapsed time 0.003226 sec) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Response dump for 'answer' to 127.0.0.1:60024: { "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 2654832326 4 IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: s=Asterisk Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic:WMS * Mar 9 20:21:02 debian rtpengine[11823]: m=audio 30020 UDP/TLS/RTP/SAVPF 111 126 Mar 9 20:21:02 debian rtpengine[11823]: a=connection:new Mar 9 20:21:02 debian rtpengine[11823]: a=maxptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=msid:d5be09fd-dce5-4d96-879e-f34f22f54ef1 a3f22c04-6988-4464-a458-5abc099e7bc0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:* transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:126 0-16 Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp:30020 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=setup:active Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:sha-256 0C:67:09:6B:AE:6B:30:09:81:C5:9C:65:C6:EC:9E:CE:C2:6E:E2:EA:67:B4:B5:B3:40:C4:F1:0C:3E:05:96:79 Mar 9 20:21:02 debian rtpengine[11823]: a=tls-id:ac2907348568344c9743f21fa1383e08 Mar 9 20:21:02 debian rtpengine[11823]: a=ptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:4xVkmu6h Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:kc54UEYL25S4UjgSjVKJdcEiAR Mar 9 20:21:02 debian rtpengine[11823]: a=ice-options:trickle Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:GtpEjl9r4IzNpiXr 1 UDP 2130706431 10.13.1.127 30020 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=end-of-candidates Mar 9 20:21:02 debian rtpengine[11823]: ", "result": "ok" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Processing incoming DTLS packet Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Sending DTLS packet Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] Received ICE/STUN response code 0 for candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 from 10.13.1.152:14450 to 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] Setting ICE candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 as succeeded Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] Best succeeded ICE pair with all components is GtpEjl9r4IzNpiXr:Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE not completed yet, but can use pair GtpEjl9r4IzNpiXr:Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE negotiated: peer for component 1 is 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE negotiated: local interface 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Processing incoming DTLS packet Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [crypto] DTLS: Peer certificate accepted Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [crypto] DTLS handshake successful Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [crypto] DTLS-SRTP successfully negotiated using AES_CM_128_HMAC_SHA1_80 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] SRTP keys, incoming: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] --- AES_CM_128_HMAC_SHA1_80 key AtAWESyBg5e6Ub3EPRQ5ag== salt rgwpYXp6KpHCmfjMxVA= Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] SRTP keys, outgoing: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] --- AES_CM_128_HMAC_SHA1_80 key 8+zHcQ5a1Bv5L0dv2AfKyg== salt 7YaPNc68d5ALtKKIOCQ= Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [crypto] DTLS-SRTP successfully negotiated using AES_CM_128_HMAC_SHA1_80 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] SRTP keys, incoming: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] --- AES_CM_128_HMAC_SHA1_80 key AtAWESyBg5e6Ub3EPRQ5ag== salt rgwpYXp6KpHCmfjMxVA= Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] SRTP keys, outgoing: Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] --- AES_CM_128_HMAC_SHA1_80 key 8+zHcQ5a1Bv5L0dv2AfKyg== salt 7YaPNc68d5ALtKKIOCQ= Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Sending DTLS packet Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Successful STUN binding request from 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1]: [ice] Start nominating ICE pairs Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1]: [ice] Nominating ICE pair GtpEjl9r4IzNpiXr:Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1]: [ice] Sending nominating ICE/STUN request for candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 from 10.13.1.127 to 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] Received ICE/STUN response code 0 for candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 from 10.13.1.152:14450 to 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] Setting nominated ICE candidate pair GtpEjl9r4IzNpiXr:Ha0d0198:1 as valid Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE completed, using pair GtpEjl9r4IzNpiXr:Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE negotiated: peer for component 1 is 10.13.1.152:14450 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [ice] ICE negotiated: local interface 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Received command 'answer' from 127.0.0.1:60024 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Dump for 'answer' from 127.0.0.1:60024: { "supports": [ "load limit" ], "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 2654832326 4 IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: s=Asterisk Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic:WMS * Mar 9 20:21:02 debian rtpengine[11823]: m=audio 14450 UDP/TLS/RTP/SAVPF 111 126 Mar 9 20:21:02 debian rtpengine[11823]: a=connection:new Mar 9 20:21:02 debian rtpengine[11823]: a=setup:active Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:SHA-256 62:AD:7E:86:4D:FF:87:6B:44:5A:F0:62:A2:37:D6:56:5C:D4:F4:A8:03:88:A6:E1:DD:2A:15:69:75:1B:04:3C Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:7706bb5e0317151b5749cc8550354244 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:4af879c44cb99216424536fa5defd9d5 Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:Ha0d0198 1 UDP 2130706431 10.13.1.152 14450 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:H6719a439 1 UDP 2130706431 103.25.164.57 14450 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:126 0-16 Mar 9 20:21:02 debian rtpengine[11823]: a=ptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=maxptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=msid:d5be09fd-dce5-4d96-879e-f34f22f54ef1 a3f22c04-6988-4464-a458-5abc099e7bc0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:* transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: ", "call-id": "ui7jc53b9lfbt9lf4nu7", "received-from": [ "IP4", "10.13.1.152" ], "from-tag": "7n38ig1acl", "to-tag": "a5935482-3f00-4dd0-b98c-0b4726da777d", "command": "answer" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [core] Using default bitrate of 32000 bps for 2-channel Opus Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating codecs for answerer a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Adding codec telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for a5935482-3f00-4dd0-b98c-0b4726da777d #1 -> 7n38ig1acl #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Codec answer for a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Reverse codec for opus/48000/2 (111) is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Reverse codec for telephone-event/8000 (126) is telephone-event/8000 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] telephone-event/8000/1 payload type 126 already present, skip Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for 7n38ig1acl #1 -> a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for 7n38ig1acl #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Updating supplemental codecs for a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for 7n38ig1acl #1 -> a5935482-3f00-4dd0-b98c-0b4726da777d #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Setting up codec handlers for a5935482-3f00-4dd0-b98c-0b4726da777d #1 -> 7n38ig1acl #1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Default sink codec is opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for opus/48000/2 is opus/48000/2 (111) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec opus/48000/2 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for opus/48000/2 with DTMF -1, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Checking receiver codec telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink codec for telephone-event/8000 is telephone-event/8000/1 (126) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Sink supports codec telephone-event/8000 for passthrough Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [codec] Using passthrough handler for telephone-event/8000 with DTMF 126, CN -1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Dropping new ICE candidate Ha0d0198 in favour of Ha0d0198:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Dropping new ICE candidate H6719a439 in favour of H6719a439:1 Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [ice] Shutting down ICE agent (nothing to do) Mar 9 20:21:02 debian rtpengine[11823]: INFO: [ui7jc53b9lfbt9lf4nu7]: [control] Replying to 'answer' from 127.0.0.1:60024 (elapsed time 0.002244 sec) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7]: [control] Response dump for 'answer' to 127.0.0.1:60024: { "sdp": "v=0 Mar 9 20:21:02 debian rtpengine[11823]: o=- 2654832326 4 IN IP4 10.13.1.152 Mar 9 20:21:02 debian rtpengine[11823]: s=Asterisk Mar 9 20:21:02 debian rtpengine[11823]: c=IN IP4 10.13.1.127 Mar 9 20:21:02 debian rtpengine[11823]: t=0 0 Mar 9 20:21:02 debian rtpengine[11823]: a=msid-semantic:WMS * Mar 9 20:21:02 debian rtpengine[11823]: m=audio 30020 UDP/TLS/RTP/SAVPF 111 126 Mar 9 20:21:02 debian rtpengine[11823]: a=connection:new Mar 9 20:21:02 debian rtpengine[11823]: a=maxptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=msid:d5be09fd-dce5-4d96-879e-f34f22f54ef1 a3f22c04-6988-4464-a458-5abc099e7bc0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-fb:* transport-cc Mar 9 20:21:02 debian rtpengine[11823]: a=mid:0 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:111 opus/48000/2 Mar 9 20:21:02 debian rtpengine[11823]: a=rtpmap:126 telephone-event/8000 Mar 9 20:21:02 debian rtpengine[11823]: a=fmtp:126 0-16 Mar 9 20:21:02 debian rtpengine[11823]: a=sendrecv Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp:30020 Mar 9 20:21:02 debian rtpengine[11823]: a=rtcp-mux Mar 9 20:21:02 debian rtpengine[11823]: a=setup:active Mar 9 20:21:02 debian rtpengine[11823]: a=fingerprint:sha-256 0C:67:09:6B:AE:6B:30:09:81:C5:9C:65:C6:EC:9E:CE:C2:6E:E2:EA:67:B4:B5:B3:40:C4:F1:0C:3E:05:96:79 Mar 9 20:21:02 debian rtpengine[11823]: a=tls-id:ac2907348568344c9743f21fa1383e08 Mar 9 20:21:02 debian rtpengine[11823]: a=ptime:20 Mar 9 20:21:02 debian rtpengine[11823]: a=ice-ufrag:4xVkmu6h Mar 9 20:21:02 debian rtpengine[11823]: a=ice-pwd:kc54UEYL25S4UjgSjVKJdcEiAR Mar 9 20:21:02 debian rtpengine[11823]: a=ice-options:trickle Mar 9 20:21:02 debian rtpengine[11823]: a=candidate:GtpEjl9r4IzNpiXr 1 UDP 2130706431 10.13.1.127 30020 typ host Mar 9 20:21:02 debian rtpengine[11823]: a=end-of-candidates Mar 9 20:21:02 debian rtpengine[11823]: ", "result": "ok" } Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] New ingress SSRC for: 10.13.1.152:14450 SSRC: 46ca8abf Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Handling packet: remote 10.13.1.152:14450 (expected: 10.13.1.152:14450) -> local 10.13.1.127:30000 (RTP seq 31783 TS 528 SSRC 46ca8abf) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Generated session key: master key f3ecc771..., master salt ed868f35..., label 00, length 16, result 6806f5cc... Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Generated session key: master key f3ecc771..., master salt ed868f35..., label 01, length 20, result 044b0cce... Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [srtp] Generated session key: master key f3ecc771..., master salt ed868f35..., label 02, length 14, result 4426c95d... Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] New egress (direct) SSRC for: 0.0.0.0:9 SSRC: 46ca8abf Mar 9 20:21:02 debian rtpengine[11823]: ERR: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] SRTP output wanted, but no crypto suite was negotiated Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Error when sending message. Error: Inappropriate ioctl for device Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Handling packet: remote 10.13.1.152:14450 (expected: 10.13.1.152:14450) -> local 10.13.1.127:30000 (RTP seq 31784 TS 1488 SSRC 46ca8abf) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Handling packet: remote 10.13.1.152:14450 (expected: 10.13.1.152:14450) -> local 10.13.1.127:30000 (RTP seq 31785 TS 2448 SSRC 46ca8abf) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Handling packet: remote 10.13.1.152:14450 (expected: 10.13.1.152:14450) -> local 10.13.1.127:30000 (RTP seq 31786 TS 3408 SSRC 46ca8abf) Mar 9 20:21:02 debian rtpengine[11823]: DEBUG: [ui7jc53b9lfbt9lf4nu7/a5935482-3f00-4dd0-b98c-0b4726da777d/1 port 30000]: [core] Handling packet: remote 10.13.1.152:14450 (expected:

2 1

Re: Kamailio stops processing the calls - restart fixes it.
by Henning Westerholt 10 Mar '23

10 Mar '23

Hello, please keep the list in the mail flow. Have you tried to set the acc_function modparam to an empty string, as commented in the linked issue? modparam("uac_redirect","acc_function","") If you don't need the acc function from the uac_redirect. Regarding the other warnings, they should be fixed - but different compilers can report different warnings, so it's sometimes happening. If they are just warnings, they should not create an issue. If you are using Debian/Ubuntu you can just use the prebuild packages from deb.kamailio.org - you don't need to build by yourself. Cheers, Henning -----Original Message----- From: Muhammad Danish Moosa <danishmoosa(a)gmail.com> Sent: Donnerstag, 2. März 2023 14:25 To: Henning Westerholt <hw(a)gilawa.com> Subject: Re: [SR-Users] Kamailio stops processing the calls - restart fixes it. Hi, I tried to install 5.6.4 (https://www.kamailio.org/pub/kamailio/5.6.4/src/) and get this error. Mar 3 00:05:39 lab-danish /usr/local/sbin/kamailio[24642]: INFO: rr [rr_mod.c:188]: mod_init(): outbound module not available Mar 3 00:05:39 lab-danish /usr/local/sbin/kamailio[24642]: ERROR: uac_redirect [../../modules/acc/acc_api.h:191]: acc_load_api(): cannot find bind_acc Mar 3 00:05:39 lab-danish /usr/local/sbin/kamailio[24642]: ERROR: uac_redirect [uac_redirect.c:259]: redirect_init(): cannot bind to ACC API Mar 3 00:05:39 lab-danish /usr/local/sbin/kamailio[24642]: ERROR: <core> [core/sr_module.c:975]: init_mod(): Error while initializing module uac_redirect (/usr/local/lib64/kamailio/modules/uac_redirect.so) It seems this issue was reported earlier and supposed to be fixed but apparently it's not. https://github.com/kamailio/kamailio/issues/3188 Besides that , I had seen warnings during compilation. What should be the most tested and supported version ? Example Warnings: dmq_funcs.c: In function ‘ki_dmq_send_message’: dmq_funcs.c:303:3: warning: missing braces around initializer [-Wmissing-braces] dmq_peer_t new_peer = {0}; ^ dmq_funcs.c:303:3: warning: (near initialization for ‘new_peer.peer_id’) [-Wmissing-braces] dmq_funcs.c: In function ‘ki_dmq_bcast_message’: dmq_funcs.c:373:3: warning: missing braces around initializer [-Wmissing-braces] dmq_peer_t new_peer = {0}; ^ dmq_funcs.c:373:3: warning: (near initialization for ‘new_peer.peer_id’) [-Wmissing-braces] CC (gcc) [M dmq.so] notification_peer.o CC (gcc) [M dmq.so] dmq.o dmq.c:61:1: warning: missing braces around initializer [-Wmissing-braces] sip_uri_t dmq_server_uri = {0}; On Wed, Mar 1, 2023 at 7:26 PM Henning Westerholt <hw(a)gilawa.com> wrote: > > Hello, > > better take the latest one, e.g. 5.6.4 released yesterday. Minor releases only contains bugfixes, documentation enhancements and similar. Only rarely regressions happen. But you should of course test it. > > Cheers, > > Henning > > -----Original Message----- > From: Muhammad Danish Moosa <danishmoosa(a)gmail.com> > Sent: Mittwoch, 1. März 2023 09:22 > To: Henning Westerholt <hw(a)gilawa.com> > Cc: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> > Subject: Re: [SR-Users] Kamailio stops processing the calls - restart fixes it. > > Thank you for your email. > > Is v5.6.1 (July 6, 2022) stable? > > obtained from > > https://www.kamailio.org/pub/kamailio/latest-stable-version-number > > > On Wed, Mar 1, 2023 at 5:50 PM Henning Westerholt <hw(a)gilawa.com> wrote: > > > > Hello, > > > > hard to say without more information, a backtrace etc... As I first step, I would suggest you to update the system to one of the supported releases, e.g. the latest 5.6.x or 5.5.x. > > > > Cheers, > > > > Henning > > > > -----Original Message----- > > From: Muhammad Danish Moosa <danishmoosa(a)gmail.com> > > Sent: Mittwoch, 1. März 2023 01:39 > > To: sr-users(a)lists.kamailio.org > > Subject: [SR-Users] Kamailio stops processing the calls - restart fixes it. > > > > Hi, > > > > I have a very simple proxy (stateful) where kamailio acts as a proxy between 2 endpoints. Everything works fine for weeks and suddenly I see kamailio stops responding. From pcap I can see kamailio is not proxying the session progress and bombarding invites to one endpoint without any reason. Even that invite was stripped on the Body part. > > > > Restarting kamailio fixes it immediately. Unfortunately I could not take bt full yet. > > > > Version is > > > > kamailio 5.5.3 (x86_64/linux) 473cef > > > > configuration is very simple , routing is based on tm.t_relay ( based on KEMI). > > > > Any help will be welcome. > > > > Danish > > > > > > > > -- > > Muhammad Danish Moosa > > __________________________________________________________ > > Kamailio - Users Mailing List - Non Commercial Discussions To > > unsubscribe send an email to sr-users-leave(a)lists.kamailio.org > > Important: keep the mailing list in the recipients, do not reply only to the sender! > > Edit mailing list options or unsubscribe: > > > > -- > Muhammad Danish Moosa > > " The core of mans' spirit comes from new experiences. "___ > Christopher McCandless -- Muhammad Danish Moosa " The core of mans' spirit comes from new experiences. "___ Christopher McCandless

4 21

WebRTC Error
by Social Boh 09 Mar '23

09 Mar '23

Hello, I have this error when trying to register this softphone: https://github.com/InnovateAsterisk/Browser-Phone 61096#015#012Host: sip1.kamailio.xyz:8443#015#012Connection: Upgrade#015#012Pragma: no-cache#015#012Cache-Control: no-cache#015#012Upgrade: websocket#015#012Origin: https://campus.voztovoice.org#015#012Sec-WebSocket-Version: 13#015#012User-Agent: Mozilla/5.0 (Windows NT 10.0; Win64; x64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/110.0.0.0 Safari/537.36#015#012Accept-Encoding: gzip, deflate, br#015#012Accept-Language: es-419,es;q=0.6#015#012sec-gpc: 1#015#012Sec-WebSocket-Key: uj5cXle8zzBFUzv4kWB2pg==#015#012Sec-WebSocket-Extensions: permessage-deflate; client_max_window_bits#015#012Sec-WebSocket-Protocol: sip#015#012#015#012> Mar 9 08:10:55 sip1 /sbin/kamailio[194807]: WARNING: <script>: HTTP request received on 8443 Mar 9 08:10:55 sip1 /sbin/kamailio[194807]: DEBUG: xhttp [xhttp_mod.c:426]: xhttp_send_reply(): sending out response: 403 Forbidden Any help is really apreciated -- --- I'm SoCIaL, MayBe

1 1

Location / Registrar, TCP: Use existing TCP connection
by Benoit Panizzon 09 Mar '23

09 Mar '23

Hi List CPE behind Firewall, registering to Kamailio via TCP (or TLS). When a call is sent to the CPE, kamailio attemts to open a new TCP connection and is blocked by Firewall. Is there an option to tell kamailio to use the existing registered TCP connection? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 2

Call recovery for HA with sems
by Patrick Karton 09 Mar '23

09 Mar '23

1 0

TLS -> UDP ReInvite
by Social Boh 08 Mar '23

08 Mar '23

hello, I'm configuring Kamailio to permite calls like: UDP -> UDP TLS -> TLS UDP -> TLS TLS -> UDP Until now all is ok; the only problem is with Microsip SoftPhone. When make a call from Microsip TLS to another phone UDP, when the callee answer, Microsip send a new INVITE and Kamailio don't delete all references to SRTP so the call happens but without audio. I know I have to process the reINVITE where I do has_totag stuff but I don't know how. Regards -- --- I'm SoCIaL, MayBe

1 0

Re: sr-users Digest, Vol 213, Issue 52
by Mohamed SADIK 08 Mar '23

08 Mar '23

Hello, Thank you for your reply! I've used your suggestion to accomplish some amazing things. However, I'm having an issue when calling the *"sl.stats"* or any other procedures. I've noticed that the number of "200" responses gets incremented by 2, which is not the case when using the `*kamcmd*` tool. I suspect that this may be due to jsonrpc using *http/https*, which is causing Kamailio to treat it as a "200" response code. Could you advise on how to avoid this issue? Is there any further configuration that needs to be done? Best regards. On Sat, Feb 25, 2023 at 12:00 PM <sr-users-request(a)lists.kamailio.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via email, send a message with subject or > body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..."Today's Topics: > > 1. Kamailio RPC configuration (sadik.oualla.mohamed(a)gmail.com) > 2. Re: Kamailio RPC configuration (Alex Balashov) > > > > ---------- Forwarded message ---------- > From: sadik.oualla.mohamed(a)gmail.com > To: sr-users(a)lists.kamailio.org > Cc: > Bcc: > Date: Fri, 24 Feb 2023 17:07:33 -0000 > Subject: [SR-Users] Kamailio RPC configuration > Dear Kamailio community, > > I am a newbie in Kamailio and I am currently working on a project that > involves communicating with Kamailio using Python. I have been exploring > the Kamailio documentation and I have come across RPC, JSONRPCS..., which I > believe can help me achieve my goal. > > However, I am not sure about the steps to follow in the Kamailio > configuration file to create my own script to communicate with Kamailio > using Python (the script is on an other machine on the same LAN). I would > be grateful if someone could guide me through the process. > > I have already set up Kamailio (builed it from the source code 5.5.5) and > Python on my machines and I have basic knowledge of Kamailio configuration. > I just need some guidance on how to set up the RPC module in Kamailio and > how to create a Python script to communicate with it. I found that I can > use RPyC library of python. > > I would appreciate any help or suggestions from the Kamailio community. > Thank you in advance for your time and assistance. > > Best regards, > Wild Coder. > > > > ---------- Forwarded message ---------- > From: Alex Balashov <abalashov(a)evaristesys.com> > To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> > Cc: > Bcc: > Date: Fri, 24 Feb 2023 13:22:16 -0500 > Subject: [SR-Users] Re: Kamailio RPC configuration > Hi, > > The easiest way to communicate with the JSONRPCS module is probably to use > HTTP, in the form of the `xhttp` module: > > https://kamailio.org/docs/modules/5.6.x/modules/xhttp.html > > You will need... > > 1) To enable the HTTP transport in the `jsonrpcs` module; > > 2) A TCP or TLS listener (listen= config directive) to accept the traffic; > > 3) An xhttp `event_route[xhttp:request]` route, as detailed in that > module's docs: > > > https://kamailio.org/docs/modules/5.6.x/modules/xhttp.html#xhttp.f.xhttp_re… > > 4) Probably some measure of security / authentication for #3. > > The example from jsonrpc_dispatch() is perfectly sufficient to get you > started: > > > https://kamailio.org/docs/modules/5.6.x/modules/jsonrpcs.html#jsonrpcs.f.js… > > You should then be able to do things like: > > # curl \ > -X POST \ > -d '{"jsonrpc": "2.0", "id": "abc123", "method": "ul.dump"}' \ > http://${KAMAILIO_IP}:5060/RPC <http://${kamailio_ip}:5060/RPC> > > -- Alex > > > On Feb 24, 2023, at 12:07 PM, sadik.oualla.mohamed(a)gmail.com wrote: > > > > Dear Kamailio community, > > > > I am a newbie in Kamailio and I am currently working on a project that > involves communicating with Kamailio using Python. I have been exploring > the Kamailio documentation and I have come across RPC, JSONRPCS..., which I > believe can help me achieve my goal. > > > > However, I am not sure about the steps to follow in the Kamailio > configuration file to create my own script to communicate with Kamailio > using Python (the script is on an other machine on the same LAN). I would > be grateful if someone could guide me through the process. > > > > I have already set up Kamailio (builed it from the source code 5.5.5) > and Python on my machines and I have basic knowledge of Kamailio > configuration. I just need some guidance on how to set up the RPC module in > Kamailio and how to create a Python script to communicate with it. I found > that I can use RPyC library of python. > > > > I would appreciate any help or suggestions from the Kamailio community. > Thank you in advance for your time and assistance. > > > > Best regards, > > Wild Coder. > > __________________________________________________________ > > Kamailio - Users Mailing List - Non Commercial Discussions > > To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org > > Important: keep the mailing list in the recipients, do not reply only to > the sender! > > Edit mailing list options or unsubscribe: > > -- > Alex Balashov > Principal Consultant > Evariste Systems LLC > Web: https://evaristesys.com > Tel: +1-706-510-6800 e > > _______________________________________________ > sr-users mailing list -- sr-users(a)lists.kamailio.org > To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org >

1 0

Using jsonrpc_dispatch in xhttp:request - is it possible to restrict the RPC that can be invoked ?
by Chaigneau, Nicolas 07 Mar '23

07 Mar '23

Hello, I'm using jsonrpc_dispatch in a "xhttp:request" section (with xhttp and jsonrpcs modules), e.g.: event_route[xhttp:request] { if ($hu =~ "^/RPC") { jsonrpc_dispatch(); exit; } xhttp_reply("404", "Not Found", "", ""); exit; } This works, but I would like to restrict the RPC functions that can be invoked this way. Any advice on how I could do that ? The xhttp module provides "url_skip" and "url_match" but this is not useful because in the URL I only have something like "RPC". And the method is in the JSON body. I could parse the JSON body before calling jsonrpc_dispatch, but it's a bit complicated (I don't know if it's feasible in Kamailio script ?), and seems redundant because jsonrpc_dispatch already does that... Maybe somehow this could be checked from the "void *" second argument passed to the RPC function ? Thanks. Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

2 4

Info: Kamailio World 2023 - Announcing First Group Of Speakers
by Daniel-Constantin Mierla 06 Mar '23

06 Mar '23

Hello, it's now like 3 months till the start of the next Kamailio World Conference (taking place in Berlin, Germany, during June 5-8, 2023) and the first group of speakers has just been published, the details are available at: - https://www.kamailioworld.com/k2023/speakers/ Not all proposals were reviewed and Call for Speakers is still open, therefore if you want to share your interesting experiences or knowledge in the RTC space, submit it via: - https://www.kamailioworld.com/k2023/call-for-speakers/ All together, expect another edition of Kamailio World conference with great content covering topics such as RTC scalability, security, cloud-based deployments and elasticity, NG112/911, IoT, WebRTC, VoLTE and 4G/5G. Looking forward to meeting many of you in Berlin! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

1 0

Remove silo extra_hdrs before store
by Dr. Barabás Péter 06 Mar '23

06 Mar '23

Hi, I’m using kamailio’s silo module to store offline messages in it. I set extra_hdrs modparam to e.g. : ‘Resent-from-silo’ so the clients get information that the message has arrived directly (immediatelly) from the sender or it was stored offline and comes from silo. So the problem is the following: * if receiver’s socket is broken (e.g. airplane mode on mobile) and kamailio ‘thinks’ it is registered, server tries to send the message * message will be timed out and based on config it will be stored in silo. * the next step is to check if receiver is registered or not (since the client can re-register during 30sec until the message has timed out) * If client is registered, kamailio tries to send the message immediately with m_dump(). * If the receiver’s connection is still broken, the message will be timed out and store in silo. * In every store, a new ‘resent’ extra_hdrs value is appended. When extra_hdrs length reaches 1024 bytes, the m_dump will fail and it blocks the dumping of messages to the given receiver. Question: can extra_hdrs value be removed before store? Or what can be the solution not to duplicate the extra_hdrs value in some bad network situation? Peter

1 0

How to access rtpengine rtcp-metrics?
by Benoit Panizzon 06 Mar '23

06 Mar '23

Hi all I'm trying to incorporate rtcp data from rtpengine into our CDRS. So trying with the first interesting value, the average mos. modparam("rtpengine", "mos_average_pv", "$avp(mos_average)") If I understood right, to get this variable set, I need to call rtpengine_manage() on the message that terminates the call. So on an established call, that is either BYE or the 200 OK to the BYE. Right? if ($rm == "BYE") { rtpengine_manage(); xlog("L_INFO", "$cfg(route): $rm: MOSS: $avp(mos_average)\n"); } onreply_route[MANAGE_REPLY] { [...] rtpengine_manage(); xlog("L_INFO", "$cfg(route): $rm reply MOS: $avp(mos_average)\n"); } Messages pass those blocks, $avp(mos_average) is 'null' no mater what. What am I missing? In the syslog output of rtpengine I see there is rtcp data. Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

3 2

tcp_set_connection_lifetime => in respect to 'idle' connection?
by Benoit Panizzon 06 Mar '23

06 Mar '23

Hi List Digging further into my tcp/tls issues... Upon successful authentication I call: tcp_keepalive_enable("60", "5", "5"); tcp_set_connection_lifetime("120"); I tought, this would do to keep the connection alive. I noticed, despite keepalive packets being exchanged, kamailio is closing the connection after a bit more than 2 minutes. I am right at assuming, that tcp_set_connection_lifetime is not related to how long the connection is idle, but is counted from the SYN, no matter if keepalives were successfully exchanged? Does calling tcp_set_connection_lifetime() again restart the lifetime like for 2 more minutes like in this example? Or would I have to keep increase the value of the lifetime on each successful re-register to keep it open? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

1 0

Kamailio v5.6.3 Released
by Daniel-Constantin Mierla 05 Mar '23

05 Mar '23

Hello, Kamailio SIP Server v5.6.3 stable release is out. This is a maintenance release of the latest stable branch, 5.6, that includes fixes since the release of v5.6.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.6.x. Deployments running previous v5.6.x versions are strongly recommended to be upgraded to v5.6.3. For more details about version 5.6.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2023/01/kamailio-v5-6-3-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 Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com

5 10

Kamailio v5.5.6 Released
by Daniel-Constantin Mierla 02 Mar '23

02 Mar '23

Hello, Kamailio SIP Server v5.5.6 stable release is out. This is a maintenance release of the stable branch 5.5 that includes fixes since the release of v5.5.5. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.5.x. Deployments running previous v5.5.x versions are strongly recommended to be upgraded to v5.5.6. Note that 5.5 is the second last stable branch, still officially maintained by Kamailio development team. The latest stable branch is 5.6, with v5.6.4 being release out of it. For more details about version 5.5.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2023/03/kamailio-v5-5-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

1 0

Releasing Kamailio v5.5.6
by Daniel-Constantin Mierla 02 Mar '23

02 Mar '23

Hello, I am considering to release Kamailio v5.5.6 soon, branch on branch 5.5, likely on Thursday or Wednesday next week (Mar 2/3, 2023). This is the usual heads up notification to see if anyone is aware of issues not yet reported to bug tracker and if yes, do it as soon as possible to give them a chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

1 1

Websocket closing
by David Villasmil 01 Mar '23

01 Mar '23

Hello guys, We're seeing corner cases where the following happens: On proxy-ws - IP1:PORT1 connects via websocket from Client1 - Registration happens on an upstream kamailio - for any reason, the TCP socket closes or times out. - IP1:PORT1 (same IP:PORT combination) connects via websocket from Client2 - Registration happens on an upstream kamailio Now a call comes in to Client1. Because the first connection was never cleaned up, it is sent to the proxy-ws and the proxy will send it to the IP1:PORT1 where Client2 is connected. Short story, proxy-ws doesn't check the IP1:PORT1 where it is sending the INVITE is the actual client it is supposed to be sending... It seems that when a socket is closed, the mapping IP:PORT to Address (i.e.: sip:d4f27e34@994f31243be9.invalid;transport=ws) doesn't seem to be cleared... is this by design? Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 6

Kamailio stops processing the calls - restart fixes it.
by Muhammad Danish Moosa 01 Mar '23

01 Mar '23

Hi, I have a very simple proxy (stateful) where kamailio acts as a proxy between 2 endpoints. Everything works fine for weeks and suddenly I see kamailio stops responding. From pcap I can see kamailio is not proxying the session progress and bombarding invites to one endpoint without any reason. Even that invite was stripped on the Body part. Restarting kamailio fixes it immediately. Unfortunately I could not take bt full yet. Version is kamailio 5.5.3 (x86_64/linux) 473cef configuration is very simple , routing is based on tm.t_relay ( based on KEMI). Any help will be welcome. Danish -- Muhammad Danish Moosa

3 5

Kamailio v5.6.4 Released
by Daniel-Constantin Mierla 27 Feb '23

27 Feb '23

Hello, Kamailio SIP Server v5.6.4 stable release is out. This is a maintenance release of the latest stable branch, 5.6, that includes fixes since the release of v5.6.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.6.x. Deployments running previous v5.6.x versions are strongly recommended to be upgraded to v5.6.4. For more details about version 5.6.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2023/02/kamailio-v5-6-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. An additional note to say that call for speakers and registration for Kamailio World Conference 2023 are now open, more details at: * https://www.kamailioworld.com 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 Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

1 0

Kamailio MS Teams TLS Issue
by iliusha.md＠gmail.com 27 Feb '23

27 Feb '23

Hello, We have one Kamailio Instance connected with MS Teams (based on this instruction: https://skalatan.de/en/blog/kamailio-sbc-teams), which worked fine for a while until recently we noticed that calls from teams are not working anymore. When I looked through the logs I found that Microsoft cannot establish a TLS connection to our server because of the cipher: TLS accept:error:1408A0C1:SSL routines:ssl3_get_client_hello:no shared cipher (sni: sbc.example.com - domain is obfuscated). Certificate is valid, the configuration is below: [server:default] method = TLSv1.2+ verify_certificate = no require_certificate = no private_key = /usr/local/etc/kamailio/certs/example.net/sbc1-teams_example_net.key certificate = /usr/local/etc/kamailio/certs/example.net/sbc1-teams_example_net.crt server_name = sbc1-teams.example.net ca_list = /usr/local/etc/kamailio/certs/sectigo_ca.pem #ca_list=/etc/ssl/certs/ca-bundle.crt [client:default] method = TLSv1.2+ verify_certificate = no require_certificate = no private_key = /usr/local/etc/kamailio/certs/example.net/sbc1-teams_example_net.key certificate = /usr/local/etc/kamailio/certs/example.net/sbc1-teams_example_net.crt ca_list = /usr/local/etc/kamailio/certs/sectigo_ca.pem #ca_list=/etc/ssl/certs/ca-bundle.crt We use a certificate from Sectigo, but I've tried with Let's Encrypt - and it's the same. Any idea what could be the reason?

5 7

Releasing v5.6.4
by Daniel-Constantin Mierla 27 Feb '23

27 Feb '23

Hello, I am considering to release Kamailio v5.6.4 (out of branch 5.6) next week (likely on Monday or Tuesday, Feb 27/28, 2023). If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

1 1

Kamailio RPC configuration
by sadik.oualla.mohamed＠gmail.com 24 Feb '23

24 Feb '23

Dear Kamailio community, I am a newbie in Kamailio and I am currently working on a project that involves communicating with Kamailio using Python. I have been exploring the Kamailio documentation and I have come across RPC, JSONRPCS..., which I believe can help me achieve my goal. However, I am not sure about the steps to follow in the Kamailio configuration file to create my own script to communicate with Kamailio using Python (the script is on an other machine on the same LAN). I would be grateful if someone could guide me through the process. I have already set up Kamailio (builed it from the source code 5.5.5) and Python on my machines and I have basic knowledge of Kamailio configuration. I just need some guidance on how to set up the RPC module in Kamailio and how to create a Python script to communicate with it. I found that I can use RPyC library of python. I would appreciate any help or suggestions from the Kamailio community. Thank you in advance for your time and assistance. Best regards, Wild Coder.

2 1

OT: MS Teams source IP address ranges
by Jon Bonilla (Manwe) 24 Feb '23

24 Feb '23

Hi Sorry for the OT but I think here's the place where I an find a lot of Ms teams integrations I've been working on MS teams direct routing integration for PekePBX. It works. I guess I've done it as everybody else, using Henning's guide as base and extending it for multitenant setup (thanks Henning!) What I've realized is that the source IP address of calls coming from MS are not always matching dispatcher hosts. Sometimes they come from another source IP and failover to the dispatcher hosts when they receive no response. That makes some of the calls to have an additional latency Searching in the MS doc I see that they document these nets as source of their signaling: 52.112.0.0/14 52.120.0.0/14 But I've seen IP addresses outside of this range as source. In this blog https://erwinbierens.com/microsoft-teams-direct-routing-ip-addresses/ The ranges are listed as 52.112.0.0/16 52.113.0.0/16 52.114.0.0/16 52.115.0.0/16 52.120.0.0/16 52.121.0.0/16 52.122.0.0/16 52.123.0.0/16 which looks better but scares me out. Having no auth is it secure to bind so many ranges to MS? Do you use anything else than certificate verification for these calls? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

4 6

Dispatcher failover when SRV failover is on?
by Jon Bonilla (Manwe) 23 Feb '23

23 Feb '23

Hi Again, fighting with MS Teams but this time it's a kamailio question. I have MSteams hosts in the dispatcher for OPTIONS and for destination. Acording to MS you need to try sip.pstnhub.foo then sip2.pstnhub.foo and sip3.pstnhub.foo 2 | sip:sip.pstnhub.microsoft.com;transport=tls | 0 | 1 2 | sip:sip2.pstnhub.microsoft.com;transport=tls | 0 | 2 2 | sip:sip3.pstnhub.microsoft.com;transport=tls | 0 | 3 I was going to implement dispatcher failover as usual ds_select_domain (alg 8) with failure routes but checking the SRV of the first domain I see it already includes 4 hosts: host -t srv _sips._tcp.sip.pstnhub.microsoft.com _sips._tcp.sip.pstnhub.microsoft.com has SRV record 10 0 5061 sip.pstnhub.microsoft.com. _sips._tcp.sip.pstnhub.microsoft.com has SRV record 20 0 5061 sip2.pstnhub.microsoft.com. _sips._tcp.sip.pstnhub.microsoft.com has SRV record 30 0 5061 sip3.pstnhub.microsoft.com. _sips._tcp.sip.pstnhub.microsoft.com has SRV record 40 0 5061 sip4.pstnhub.microsoft.com. So... If I have dns srv failover enabled in kamailio... what would be the point of the dispatcher failover? In case I only want failover in case 408 or 5XX wouldn't the tm module do it for me? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

3 2

What does l_username in UAC do?
by Дилян Палаузов 23 Feb '23

23 Feb '23

Hello, the uac module-registrations have the fields l_uuid - user for Contact: in REGISTER (before @). Described at https://kamailio.org/docs/db-tables/kamailio-db-5.6.x.html#idm9037 as “Local unique id used to build and match contact addresses.”. Described at https://www.kamailio.org/docs/modules/devel/modules/uac#idm700 as “local unique user id, e.g.,: 12345678” l_username - Described at https://kamailio.org/docs/db-tables/kamailio-db-5.6.x.html#idm9037 as “Local username”. Described at https://www.kamailio.org/docs/modules/devel/modules/uac#idm700 as “local user name, e.g.,: test ”. For me there is no difference between “local unique id” (as for l_uuid) and “local username“ (here). l_domain - used for Contact: in REGISTER (after @), described at https://kamailio.org/docs/db-tables/kamailio-db-5.6.x.html#idm9037 as “local domain” When is l_username used, what is it purpose? Does it solely serve as possible filter criterion for the RPC commands: uac.reg_info l_username abc uac.reg_enable l_username abc uac.reg_disable l_username abc, and uac.reg_unregister l_username abc Do I see correctly, that via UAC-RPC an unREGISTER can be triggered, but there is no way to trigger a REGISTER over UAC-RPC? Thanks for the clarifications Дилян

2 2

change default 477 response code per case
by Patrick Karton 22 Feb '23

22 Feb '23

Hello , i would like to know if there is a way to change the default 477 response code generated by kamailio in some cases. for example if kamalio generates 477 because its not able to reach a peer i would like to change it to another code before to forward it. Thanks.

2 1

Call History
by Anthony Wittig 21 Feb '23

21 Feb '23

Hello, I'd like to start tracking a call history/log as soon as I receive a call. I'd like the customer to be able to see that there's an incoming call to one of their numbers in real time. I'm thinking of using the RABBITMQ Module to manually emit an event from my main route when I receive an INVITE. Is that a bad idea? What if the broker isn't located on the same host (latency/availability)? I don't want to negatively affect the call processing. I was looking at the Dialog Module and it has three event routes: * event_route[dialog:start] * event_route[dialog:end] * event_route[dialog:failed] It's really close to what I want, but I'd like to capture the dialog before it starts (Unconfirmed or Early). Maybe another option is to scan the Dialog table with an external process and emit events from there. Has anyone had success in creating a realtime call history? Is there a better way to go about doing so? Thanks! -- Anthony Wittig Mango Voice Developer

7 7

Info: Registration to Kamailio World 2023 is open!
by Daniel-Constantin Mierla 21 Feb '23

21 Feb '23

Hello, a quick note to the community forums to announce that the registration for the next Kamailio World Conference (June 5-7, 2022, in Berlin, Germany) is now open! The event returns to an in-person conference format, at the same wonderful location in the city center of Berlin! More details are available at: * https://www.kamailioworld.com/k2023/registration/ Call for presentations is also open, there is already a group of very interesting submissions, we aim to publish details about them in the near future. Looking forward to meeting many of you in Berlin! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

2 2

topos breaking route set on hairpin calls?
by Benoît Panizzon 20 Feb '23

20 Feb '23

Hi Having this situation while elaborating topos in lab. * Kamailio Registration Instance with topos enabled where customers are registered. * Kamailio Core Instance with Dialogue enabled and handling interconnections. Calls in transit from customer to IC work flawlessly, topos does what it should, hides and restores headers needed on the core. But I have run into an issue with customer calling another customer on same registrar instance. To generate a CDR with diaglogue and for decision where to route the calls to (registrar is kept simple and only handles registrations and locations), those calls are also routed via core. This works as long as PRACK is disabled. When PRACK is enabled and CPE B is requesting 100rel, the PRACK send from CPE A has no route set (it was removed by topos). Therefore the PRACK only makes it to the core who then does not know where to route that PRACK. It looks like topos on the REG only restores ONE Route pointing to the core (maybe because the 2nd one pointed to itself?) header when restoring them. I have attempted to spam more record_route() here and there in the config of the core and the registrar to try to force creation of route headers, but failed. Is this a known limitation of using topos? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 4

Re: OT: MS Teams source IP address ranges
by Henning Westerholt 20 Feb '23

20 Feb '23

Hi Jon, you could of course try to add some heuristic, in the end the used IPs are a smaller sub-set. But to just quote from the official docs, it seems to be pretty clear: "You need to open ports for all these IP address ranges in your firewall to allow incoming and outgoing traffic to and from the addresses for signaling." Cheers, Henning ________________________________ From: Jon Bonilla (Manwe) Sent: Monday, February 20, 2023 15:49 To: Henning Westerholt; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] OT: MS Teams source IP address ranges El Mon, 20 Feb 2023 13:06:30 +0000 Henning Westerholt <hw(a)gilawa.com> escribió: > Hello, > > correct me if I am wrong, but the second group of addresses ("/16") is > included in the first group ("/14"), right? > > Regarding the large IP scope, this is the way Microsoft designed it, > unfortunately. There is not that much what you can do about. > > Right. Didn't check the masks. they all were 16 in my mind :) So, if I add those ranges in the permissions table my only way of avoiding spoofed calls is certificate validation? I'm already doing that and also checking the company domain and ms account email address. Just wanted to be sure that nothing "essential" was missing. -- PekePBX, the multitenant PBX solution https://pekepbx.com

1 0

sendig 4 byte udp nat pings
by Bernd Krueger-Knauber 18 Feb '23

18 Feb '23

Hi, maybe I'm stupid, but I can not find out how to send 4 byte UDP NAT packets to the sip server. loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 15) modparam("nathelper|registrar", "received_avp", "$avp(s:rcv)") I'm a absolute beginner with kamailio and the page about the nathelper Module is not really helpful. At the moment no pings are sendet. But I receive 4 byte packets from the sip server. After 15 minutes my calls are cancelled because of the missing pings. Without proxy, the phone send the pings directly to the sip server and the call stays alive. When I experiment with some other nathelper settings, kamailo start results in errors. Any help would be appreciated. Best regards, Bernd

3 11

enable topology hiding on one side or dynamic ip in topoh module Contact and Via Header
by Patrick Karton 17 Feb '23

17 Feb '23

Hello , im using kamailio with two interfaces external and internal. i need a way either to : * enable topoh hiding only when outgoing interface is external ( mask contact and Via ip only when ougoing interface is external * or if it is not possible to enable it only in one direction.i want to know how to configure dynamic(for example in xavp or avp) ip to put in Contact and Via when topoh is enabled. i see that the 'mask_ip' parameter of topoh module is a string. so we can not set a dynamic value here unfortunately. Thanks

4 8

Calling uac_replace multiple times?
by Benoît Panizzon 17 Feb '23

17 Feb '23

Hey all I have started using the uac module to be able to mirror username changes on both legs of a call. But now I found an issue: Assume this situation: Call IC => Customer IC is e164, Customer wants 'local' numbers. IC From: +41991234567 => Customer From: 0991234567 uac_replace_from("sip:" + $avp(from_local) + "@" + $(fu{uri.host})); Customer is absent, after timeout in failure_route call is dispatched to Voicemail Server, which is configured for e164 numbers: uac_replace_from("sip:" + $avp(from_e164) + "@" + $(fu{uri.host})); But then uac complains: replace_uri(): Already called uac_replace for this dialog Why isn't just the most from version kept and used? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 1

TLS version and cipher from WSS
by Michal Hajek 17 Feb '23

17 Feb '23

Hi, I can log incoming SIPS TLS version and cipher with PV $tls_version and $tls_cipher_info. How can I log these variables on incoming HTTP/WebSocket. Thanks Michal H.

2 2

Identify error in log
by Евгений Мастратов 16 Feb '23

16 Feb '23

Hello colleagues. I'm starting to use Kamailio for educational purposes, as an IMS test zone. Registration of subscribers, calls go successfully, but the following errors are constantly pouring into the log: Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/rvalue.c:1378]: int_intop2(): rv div by 0 Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/rvalue.c:1378]: int_intop2(): rv div by 0 Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/rvalue.c:2381]: rval_expr_eval(): rval expression evaluation failed (735,15-735,55) Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/rvalue.c:2381]: rval_expr_eval(): rval expression evaluation failed (735,15-735,55) Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/lvalue.c:386]: lval_assign(): rval expression evaluation failed (735,15-735,55) Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/lvalue.c:386]: lval_assign(): rval expression evaluation failed (735,15-735,55) Apparently they don’t carry anything critical, but they clog the log, natively I can’t understand the meaning of the errors. Please tell me what they mean and how to eliminate their cause? If necessary, I will provide the pcscf config.

2 1

uac reg reload frequency
by Dmytro M 16 Feb '23

16 Feb '23

When you do "kamcmd uac.reg_reload" frequently enough, you'll get an error: # kamcmd uac.reg_reload error: 500 - Failed to shift records - check log messages Logs: ERROR: uac [uac_reg.c:416]: uac_reg_ht_shift(): shifting in-memory table is not possible in less than 150 secs What is the reason behind 150 sec limitation for how frequent reloads can be done? Is it configurable in any way? Use case: Remote app needs to disable uac registration. It deletes record in DB and calls kamcmd uac.reg_reload to activate the change. With 150 sec limit it can't apply DB changes immediately. Please advise. Thanks, Dmytro

2 2

ACK on RE-Invite is send to UAC NATed Internal port instead of public port
by Angelo Sipper 15 Feb '23

15 Feb '23

Hi, My setup has: (TLS)UAC 77.1.1.1 -> (TLS)Kamailio 17.17.17.211 (UDP) -> (UDP)Kamailio 17.17.17.181 (UDP) -> Carrier(UDP) Both Kamailio are 4.3 with minor modifications on default cfg, and Topoh and TLS modules are used. I am having an issue on some outbound calls from nated UACs. On these calls our SBC sometimes receives reverse invites from the carrier side within the call. When this happens, my SBC(17.17.17.211) sends the invite to UAC and the UAC sends back 200 OK (SDP). Unfortunately my SBC(17.17.17.211) sends back the ACK not to the correct UAC public port but instead to the UAC internal port making the call to crash with several retries from my SBC to send 200 OK as ACK was never received from UAC! If you look the below traces of reverse invite past within the call, the reinvite has port private 20125 and alias the public port 2314 of UAC but the ACK bellow is send to 20125 and not to 2314 2023/01/26 09:48:30.668634 17.17.17.181:5060 -> 17.17.17.211:5060 INVITE sip:301234567890@77.1.1.1:20125;alias=77.1.1.1~2314~1 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Supported: timer Session-Expires: 600;refresher=uas Min-SE: 90 Content-Length: 297 Content-Disposition: session; handling=required Content-Type: application/sdp Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> v=0 o=Sonus_UAC 14218 13519 IN IP4 14.1.1.30 s=SIP Media Capabilities c=IN IP4 14.1.1.30 t=0 0 m=audio 18600 RTP/AVP 8 18 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=sendrecv a=rtcp:18601 2023/01/26 09:48:30.679743 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0;rport=5060 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Server: Supervoice Content-Length: 0 2023/01/26 09:48:30.740803 17.17.17.211:5060 -> 77.1.1.1:2314 INVITE sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Max-Forwards: 68 Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Supported: timer Session-Expires: 600;refresher=uas Min-SE: 90 Content-Length: 297 Content-Disposition: session; handling=required Content-Type: application/sdp Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> v=0 o=Sonus_UAC 14218 13519 IN IP4 14.1.1.28 s=SIP Media Capabilities c=IN IP4 14.1.1.28 t=0 0 m=audio 13288 RTP/AVP 8 18 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=sendrecv a=rtcp:13289 2023/01/26 09:48:30.893512 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Content-Length: 0 2023/01/26 09:48:31.026742 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:31.093656 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 276 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 14.1.1.28 s=3cxPS Audio call c=IN IP4 14.1.1.28 t=0 0 m=audio 13278 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=sendrecv a=rtcp:13279 2023/01/26 09:48:31.254591 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.48f49235d45a0f6f9efdad9d6d8224bd.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:31.260901 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.e8fb34e79623843d516ae77257014e2d.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.48f49235d45a0f6f9efdad9d6d8224bd.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Any help will be much appreciated. Thank you! Regards, Angelo

2 5

SCA Presence : Record-Route Filed Not Update in sca_subscriptions Table
by Vinayak Makwana 14 Feb '23

14 Feb '23

Hello all, I am using the sca module in kamailio script.so when i subscribe with my ip phone for SCA Presence i get all the information in a sca_subscriptions table for the first time. When I reboot the device I get a new record-route header in trace but It does not update *record-route *Filed in sca_subscriptions table. Can anyone help me out with this issue ? *Here's modparam configuration * modparam("sca", "db_update_interval", 10 ) modparam("sca", "db_url", DBURL_LOCAL) modparam( "sca", "line_seize_max_expires", 30) modparam( "sca", "purge_expired_interval", 1800) -- -- * <https://www.ecosmob.com/itexpo/> * *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.

3 8

SCA Feature:: Record-route query.
by satyaprakash ch 13 Feb '23

13 Feb '23

Hello I have a query regarding the record-route field update in DB for SCA SUBSCRIBE. Here are my test cases *Initial SCA subscription,* We got the below SUBSCRIBE packet in kamailio SUBSCRIBE sip:1234.xyz.com:5060 SIP/2.0 Via: SIP/2.0/UDP kam-ip:5060;branch=z9hG4bK2sansay1081381rdb66 *Record-Route* : <sip:sansay1081381rdb66spid17urcvjtr139807186744464nat0@kam-ip:5060;lr;transport=udp> To: <sip:372427@1234.xyz.com:6060> From: "708" <sip:372427@1234.xyz.com:6060>;tag=CFFBAF19-19A68CBC Call-ID: bd4b50d9384523c14ea05c532d895bf2 CSeq: 2 SUBSCRIBE Contact: < sip:372427@192.168.1.3> Expires: 3600 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER Event: call-info User-Agent: PolycomVVX Accept-Language: en Max-Forwards: 69 Content-Length: 0 also in the sca_subscriptions table record-route header is updated as per Initial SUBSCRIBE record-route. *record_route: <sip:sansay1081381rdb66spid17urcvjtr139807186744464nat0@kam-ip:5060;lr;transport=udp> * But After rebooting the device, we got a new record-route in SUBSCRIBE Packet but in DB Record-route Field not updated after getting an updated record-route in Trace. *Kamailio Trace *SUBSCRIBE sip:1234.xyz.com:5060 SIP/2.0 Via: SIP/2.0/UDP kam-ip:5060;branch=z9hG4bK2sansay1081415rdb48 *Record-Route*: <sip:sansay1081415rdb48spid17urcvjtr139807186744464nat0@kam-ip:5060;lr;transport=udp> To: <sip:372427@1234.xyz.com:6060> From: "708" <sip:372427@1234.xyz.com:6060>;tag=35852031-CF749E20 Call-ID: 41fa031d79e35fae4a61c61984895bf2 CSeq: 2 SUBSCRIBE Contact: < sip:372427@192.168.1.3> Expires: 3600 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER Event: call-info User-Agent: PolycomVVX Accept-Language: en Max-Forwards: 69 Content-Length: 0 as per logic it should update the record-route field in sca_subscriptions table . but it didn't update it : record_route: <sip:sansay1081381rdb66spid17urcvjtr139807186744464nat0@kam-ip:5060;lr;transport=udp> So can you suggest to us how we can update the record-route field as per SUBSCRIBE's packet new record-route ?

2 1

Issues matching regex
by Nathan Stratton 11 Feb '23

11 Feb '23

I am having an issue matching toll-free. I tested my regex on regex101.com, and it looks good, but it does not match my if statement. Am I doing something wrong? if (method=="INVITE") { xlog("L_INFO", "Checking Routes $rU"); if ($rU=~"^(\+?1)?(8(00|44|55|66|77|88)[2-9]\d{6})$") { xlog("L_INFO", "Tollfree call"); }; }; Logs show "<script>: Checking Routes +18002211212" so I don't get why it does not match. ><> Nathan Stratton Founder, CTO Vocinity, Inc. <https://www.vocinity.com/gabbi>

2 1

kemi equivalent config
by mohsen khashei 11 Feb '23

11 Feb '23

2 1

KEMI Equivalent of script
by mohsen khashei 10 Feb '23

10 Feb '23

Hi I have an existing kamailio config and I want to migrate to kemi. I want to know what is the equivalent of this code in kemi javascript ? $avp(s:contact) = $hdr(Contact); $avp(s:contact_fix) = $hdr(Contact); avp_subst("$avp(s:contact_fix)", "/@([0-9]+).([0-9]+).([0-9]+).([0-9]+)/@$sel(cfg_get.kamailio.ss_ip)/gi"); remove_hf("Contact"); append_hf("Contact: $avp(s:contact_fix)\r\n"); t_on_reply("REG_FIX"); thanks for your attention.

1 0

Return value from lua/python/js/bash/etc... function
by Dmytro M 10 Feb '23

10 Feb '23

What is the simplest way to return values from external application into Kamailio config and assign it to a variable? I'm looking for the most light weight mechanism to do it and to offload some logic to external app. Something like this would be ideal: $var(myVar), $var(err) = lua_run("lua_func1", "$rU"); OR $var(encString) = exec_cmd("echo 'abc $rU' | base64"); There is this method: http_client_query("http://...", "", "", "$var(result)"); but there is an overhead of networking and HTTP protocol. Please advise. Thanks, Dmytro

3 4

ACK on Reverse Invite is send to NATed Internal port instead of public port
by Angelo Sipper 09 Feb '23

09 Feb '23

Hi, I am having an issue on some outbound calls from nated UAC. On these calls our SBC receive reverse invites from the carrier side within the call. When this happens, my SBC(Kamailio1) sends the invite to UAC and the UAC sends 200 OK (SDP). Unfortunately my SBC(Kamailio1) sends back the ACK not to the correct UAC public port but instead to UAC internal port making the call to crash! My setup has: (TLS)UAC -> (UDP)Kamailio1(UDP) -> (UDP)Kamailio2(TLS) -> Carrier(UDP) Bellow you can find the trace and you can see the ACK send to wrong port on 2023/01/26 09:48:31.254591 Your help will be much appreciated! 2023/01/26 09:48:23.225326 77.1.1.1:2314 -> 17.17.17.211:5060 INVITE sip:300987654321@17.17.17.211:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport Max-Forwards: 70 Contact: <sip:301234567890@77.1.1.1:20125> To: <sip:300987654321@17.17.17.211:5060> From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Remote-Party-ID: <sip:301234567890>;party=called Content-Length: 283 v=0 o=3cxPS 792019937001472 2393157908037633 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 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=sendrecv 2023/01/26 09:48:23.240310 17.17.17.211:5060 -> 77.1.1.1:2314 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.80.228:5060 ;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314;received=77.1.1.1 To: <sip:300987654321@17.17.17.211:5060> From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Server: Supervoice Content-Length: 0 2023/01/26 09:48:23.301454 17.17.17.211:5060 -> 17.17.17.181:5060 INVITE sip:300987654321@sip.mydomain.com:5060 SIP/2.0 Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bKc2d8.34fe87ef37b154d931766bddfd3da41d.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Max-Forwards: 69 Contact: <sip:301234567890@77.1.1.1:20125;alias=77.1.1.1~2314~1> To: <sip:300987654321@sip.mydomain.com> From: 301234567890<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 300 P-Asserted-Identity: <sip:301234567890@sip.mydomain.com> v=0 o=3cxPS 792019937001472 2393157908037633 IN IP4 14.1.1.28 s=3cxPS Audio call c=IN IP4 14.1.1.28 t=0 0 m=audio 13278 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=fmtp:18 annexb=no a=sendrecv a=rtcp:13279 2023/01/26 09:48:23.302899 17.17.17.181:5060 -> 17.17.17.211:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bKc2d8.34fe87ef37b154d931766bddfd3da41d.0;rport=5060 Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 To: <sip:300987654321@sip.mydomain.com> From: 301234567890<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Server: Supervoice Content-Length: 0 2023/01/26 09:48:24.077941 17.17.17.181:5060 -> 17.17.17.211:5060 SIP/2.0 183 Session Progress From: 301234567890<sip:301234567890@sip.mydomain.com>;tag=07ece97c To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Content-Length: 250 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 17.17.17.211;rport=5060;branch=z9hG4bKc2d8.34fe87ef37b154d931766bddfd3da41d.0,SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 14.1.1.30 s=SIP Media Capabilities c=IN IP4 14.1.1.30 t=0 0 m=audio 18600 RTP/AVP 8 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:18601 2023/01/26 09:48:24.138348 17.17.17.181:5060 -> 17.17.17.211:5060 SIP/2.0 183 Session Progress From: 301234567890<sip:301234567890@sip.mydomain.com>;tag=07ece97c To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Content-Length: 250 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 17.17.17.211;rport=5060;branch=z9hG4bKc2d8.34fe87ef37b154d931766bddfd3da41d.0,SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 14.1.1.30 s=SIP Media Capabilities c=IN IP4 14.1.1.30 t=0 0 m=audio 18600 RTP/AVP 8 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:18601 2023/01/26 09:48:24.139050 17.17.17.211:5060 -> 77.1.1.1:2314 SIP/2.0 183 Session Progress From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Content-Length: 250 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 14.1.1.28 s=SIP Media Capabilities c=IN IP4 14.1.1.28 t=0 0 m=audio 13288 RTP/AVP 8 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:13289 2023/01/26 09:48:24.200575 17.17.17.211:5060 -> 77.1.1.1:2314 SIP/2.0 183 Session Progress From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Content-Length: 250 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 14.1.1.28 s=SIP Media Capabilities c=IN IP4 14.1.1.28 t=0 0 m=audio 13288 RTP/AVP 8 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:13289 2023/01/26 09:48:30.115324 17.17.17.181:5060 -> 17.17.17.211:5060 SIP/2.0 200 OK From: 301234567890<sip:301234567890@sip.mydomain.com>;tag=07ece97c To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Require: timer Supported: timer Session-Expires: 600;refresher=uac Content-Length: 250 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 17.17.17.211;rport=5060;branch=z9hG4bKc2d8.34fe87ef37b154d931766bddfd3da41d.0,SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 14.1.1.30 s=SIP Media Capabilities c=IN IP4 14.1.1.30 t=0 0 m=audio 18600 RTP/AVP 8 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:18601 2023/01/26 09:48:30.336507 17.17.17.211:5060 -> 77.1.1.1:2314 SIP/2.0 200 OK From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Require: timer Supported: timer Session-Expires: 600;refresher=uac Content-Length: 250 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 14.1.1.28 s=SIP Media Capabilities c=IN IP4 14.1.1.28 t=0 0 m=audio 13288 RTP/AVP 8 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:13289 2023/01/26 09:48:30.432457 17.17.17.181:5060 -> 17.17.17.211:5060 SIP/2.0 200 OK From: 301234567890<sip:301234567890@sip.mydomain.com>;tag=07ece97c To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Require: timer Supported: timer Session-Expires: 600;refresher=uac Content-Length: 242 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 17.17.17.211;rport=5060;branch=z9hG4bKc2d8.34fe87ef37b154d931766bddfd3da41d.0,SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 194.154.142.193 s=SIP Media Capabilities c=IN IP4 194.154.142.196 t=0 0 m=audio 10764 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 2023/01/26 09:48:30.433017 17.17.17.211:5060 -> 77.1.1.1:2314 SIP/2.0 200 OK From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 INVITE Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Server: Voiceland SBC Require: timer Supported: timer Session-Expires: 600;refresher=uac Content-Length: 242 Content-Disposition: session; handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---9bbfa2727a1f3803;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Record-Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> v=0 o=Sonus_UAC 14218 13518 IN IP4 194.154.142.193 s=SIP Media Capabilities c=IN IP4 194.154.142.196 t=0 0 m=audio 10764 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 2023/01/26 09:48:30.457268 77.1.1.1:2314 -> 17.17.17.211:5060 ACK sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181 SIP/2.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;branch=z9hG4bK-524287-1---5ff61b7097f10965;rport Max-Forwards: 70 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:301234567890@77.1.1.1:20125> To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 ACK User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 0 2023/01/26 09:48:30.481470 17.17.17.211:5060 -> 17.17.17.181:5060 ACK sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bKc2d8.44ee18452b274c8de283aa3c27549f4e.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---5ff61b7097f10965;rport=2314 Max-Forwards: 69 Contact: <sip:301234567890@77.1.1.1:20125;alias=77.1.1.1~2314~1> To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 From: "My, User"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 ACK User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 0 2023/01/26 09:48:30.555470 77.1.1.1:2314 -> 17.17.17.211:5060 ACK sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181 SIP/2.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;branch=z9hG4bK-524287-1---5ff61b7097f10965;rport Max-Forwards: 70 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:301234567890@77.1.1.1:20125> To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 ACK User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 0 2023/01/26 09:48:30.557824 17.17.17.211:5060 -> 17.17.17.181:5060 ACK sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bKc2d8.44ee18452b274c8de283aa3c27549f4e.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---5ff61b7097f10965;rport=2314 Max-Forwards: 69 Contact: <sip:301234567890@77.1.1.1:20125;alias=77.1.1.1~2314~1> To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 From: "My, User"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 1 ACK User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 0 2023/01/26 09:48:30.668634 17.17.17.181:5060 -> 17.17.17.211:5060 INVITE sip:301234567890@77.1.1.1:20125;alias=77.1.1.1~2314~1 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Supported: timer Session-Expires: 600;refresher=uas Min-SE: 90 Content-Length: 297 Content-Disposition: session; handling=required Content-Type: application/sdp Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> v=0 o=Sonus_UAC 14218 13519 IN IP4 14.1.1.30 s=SIP Media Capabilities c=IN IP4 14.1.1.30 t=0 0 m=audio 18600 RTP/AVP 8 18 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=sendrecv a=rtcp:18601 2023/01/26 09:48:30.679743 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0;rport=5060 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Server: Supervoice Content-Length: 0 2023/01/26 09:48:30.740803 17.17.17.211:5060 -> 77.1.1.1:2314 INVITE sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Max-Forwards: 68 Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Supported: timer Session-Expires: 600;refresher=uas Min-SE: 90 Content-Length: 297 Content-Disposition: session; handling=required Content-Type: application/sdp Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> v=0 o=Sonus_UAC 14218 13519 IN IP4 14.1.1.28 s=SIP Media Capabilities c=IN IP4 14.1.1.28 t=0 0 m=audio 13288 RTP/AVP 8 18 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=sendrecv a=rtcp:13289 2023/01/26 09:48:30.893512 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Content-Length: 0 2023/01/26 09:48:31.026742 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:31.093656 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 276 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 14.1.1.28 s=3cxPS Audio call c=IN IP4 14.1.1.28 t=0 0 m=audio 13278 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=sendrecv a=rtcp:13279 2023/01/26 09:48:31.254591 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.48f49235d45a0f6f9efdad9d6d8224bd.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:31.260901 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.e8fb34e79623843d516ae77257014e2d.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.48f49235d45a0f6f9efdad9d6d8224bd.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:31.530836 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:31.531333 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:31.609667 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.6cff4a8159c123f434b422c73e17a08c.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:31.612312 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.63edf375f6b899dfa9d2d9a2dcee4dfd.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.6cff4a8159c123f434b422c73e17a08c.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:32.536512 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:32.536839 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:32.623207 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.48a5467fc248682a755e70a672268f79.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:32.641928 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.745a55762e847bc3cf56d27926f7e62d.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.48a5467fc248682a755e70a672268f79.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:34.540735 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:34.541310 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:34.590013 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.6c3274e7b34c3b780cfbe9180a4d5eb1.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:34.595763 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.6c6654fa1c2f4b1991d5b6d792f2f081.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.6c3274e7b34c3b780cfbe9180a4d5eb1.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:38.561756 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:38.562039 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:38.599703 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.ef7afaa26d39ee76ce9000b1203574e7.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:38.603466 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.e022b4e31f8266525e62f213dc4dd8ae.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.ef7afaa26d39ee76ce9000b1203574e7.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:42.584025 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:42.584257 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:42.622809 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.03eec221779917d79f0daa500f7a0cc6.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:42.626545 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.0e211693731e67705c85dd4844e42ce8.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.03eec221779917d79f0daa500f7a0cc6.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:46.629501 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:46.629961 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:46.676968 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.701d02748c09dee63b88ca1e96a4e371.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:46.682458 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.cc80042ad9e1838346f4cf6d66a4c13f.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.701d02748c09dee63b88ca1e96a4e371.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:50.636998 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:50.637287 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:50.675272 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.041bbf55bf3941bfbd16df0f3ca92cf1.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:50.681861 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.be45a1f76f0fcd5ce85292397ff68acc.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.041bbf55bf3941bfbd16df0f3ca92cf1.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:54.677049 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:54.677633 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:54.721399 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.6ce35eaf9a5805bf47c868cd80c09478.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:54.729950 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.365607f0beb6e7e5dcb047c1ad6fcf5f.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.6ce35eaf9a5805bf47c868cd80c09478.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:58.681010 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:58.681307 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:58.723730 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.abb70cc570d323c867c112765ec50432.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:58.731143 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.12bbae879e42215b50512c04c4fc0214.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.abb70cc570d323c867c112765ec50432.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:49:02.684483 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:49:02.684985 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:49:02.727999 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.c2b81601667ffb2a688deba47abd85ef.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:49:02.732652 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.fb4ff819be250670479341711417a554.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.c2b81601667ffb2a688deba47abd85ef.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:49:03.028200 77.1.1.1:2314 -> 17.17.17.211:5060 BYE sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181 SIP/2.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;branch=z9hG4bK-524287-1---7c113301b578bb42;rport Max-Forwards: 70 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:301234567890@77.1.1.1:20125> To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 2 BYE User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 0 2023/01/26 09:49:03.090194 17.17.17.211:5060 -> 17.17.17.181:5060 BYE sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK92d8.6e9f4e9b390a591c1c54d79389ba087d.0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---7c113301b578bb42;rport=2314 Max-Forwards: 69 Contact: <sip:301234567890@77.1.1.1:20125;alias=77.1.1.1~2314~1> To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 From: "My, User"<sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 2 BYE User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 0 2023/01/26 09:49:03.261453 17.17.17.181:5060 -> 17.17.17.211:5060 SIP/2.0 200 OK From: "My, User"<sip:301234567890@sip.mydomain.com>;tag=07ece97c To: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 2 BYE Content-Length: 0 Via: SIP/2.0/UDP 17.17.17.211;rport=5060;branch=z9hG4bK92d8.6e9f4e9b390a591c1c54d79389ba087d.0,SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---7c113301b578bb42;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:49:03.261941 17.17.17.211:5060 -> 77.1.1.1:2314 SIP/2.0 200 OK From: "My, User"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c To: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 2 BYE Content-Length: 0 Via: SIP/2.0/UDP 192.168.80.228:5060 ;received=77.1.1.1;branch=z9hG4bK-524287-1---7c113301b578bb42;rport=2314 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Thank you! Regards, Angelo

2 4

DB_CLUSTER failure with carrierroute (and others?)
by Kaufman 08 Feb '23

08 Feb '23

All, I've been trying to chase down an issue with db_cluster and permissions in combination with something else, where with my full configuration using the same cluster in multiple modules has failures. I've been trying to create a simplified configuration to consistently reproduce the problem without much success, however I did stumble upon THIS particular problem. The configuration here works fine as is - avpops uses the cluster URL, and carrierroute uses the direct SQLite URL. If I define WITH_DBCLUSTER, then carrierroute fails fetching rows using the cluster connection. Note that AVPOPS is only used as a general test to see if the cluster config works. If I remove references to avpops , carrierroute still fails loading from the cluster URL. Thoughts? #!KAMAILIO loadmodule "db_sqlite" loadmodule "pv" loadmodule "db_cluster" loadmodule "carrierroute" loadmodule "avpops" modparam("db_cluster", "connection", "db1=>sqlite:////etc/db.sqlite") modparam("db_cluster", "cluster", "cl1=>db1=9s9s") modparam("db_cluster", "inactive_interval", 30) #!ifdef WITH_DBCLUSTER modparam("carrierroute", "db_url", "cluster://cl1") #!else modparam("carrierroute", "db_url", "sqlite:////etc/db.sqlite") #!endif modparam("carrierroute", "config_source", "db") modparam("avpops", "db_url", "cluster://cl1") modparam("avpops", "avp_table", "usr_preferences") request_route { exit; } Failure log: 0(1) ERROR: carrierroute [cr_db.c:345]: load_route_data_db(): Fetching rows failed 0(1) ERROR: carrierroute [cr_data.c:179]: reload_route_data(): could not load routing data 0(1) ERROR: carrierroute [carrierroute.c:240]: mod_init(): could not prepare route data 0(1) ERROR: <core> [core/sr_module.c:942]: init_mod(): Error while initializing module carrierroute (/usr/lib/kamailio/modules/carrierroute.so) Kaufman

3 5

Drops on async_route with delay
by Alex Balashov 07 Feb '23

07 Feb '23

Hi, Some percentage of requests processed with async_route("REQ_PROCESS", "5") seem to end up with a resumed transaction, though most do. Not sure what the exact percentage is. The ones that don't Is this due to excessive requests? Is there a limit on internal IPC queue depth? Is it conceptually similar to a generic shared blocking queue internally, along the lines of 'mqueue'? Is there any reasonable way to troubleshoot this? Thanks! -- Alex -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800

3 3

Dispatcher + ENUM
by Jawaid Bazyar 07 Feb '23

07 Feb '23

Hi, Can DISPATCHER be used in conjunction with ENUM? E.g., to avoid sending requests to SIP destinations that are returned by ENUM, but, which DISPATCHER knows are down? Thanks, Jawaid

2 1

nating private uas to public ip addr
by Jack Murphy 06 Feb '23

06 Feb '23

I have a multi-homed kam with a uas behind one private interface and a public interface for the uac+sip provider. When the uac initiates a call, kam sends the invite to the private uas which sets up a b-leg call back to kam with a Contact header that has the uas' private ip address. I want to change that private ip address to be Kamailio's advertised/public ip address before it gets forwarded to the sip provider. I am rewriting Contact in a route from request_route like this: > if ($rP != $null) { > $var(ct) = ";transport=" + $rP; > }else{ > $var(ct) = ""; > } > $var(cr) = "<sip:" + $fU + "@" + $RAi + ":" + $RAp + $var(ct) + ">"; > > remove_hf("Contact"); > append_hf("Contact: $var(cr)\r\n"); This seems to work fine for the INVITE going to the provider. There's also the 200 OK that the uas sends for the uac (a-leg) with a private ip address in Contact -- I want to rewrite that before sending it to the UAC, too. Using the same method as above (but inside onreply_route), what happens is the UAC receives the modified OK and sends back an ACK with the modified Contact (kam's public ip addr) in the ACK r-uri, and kam gets in a loop of sending an ACK from its private ip address to its public ip address and replying to itself. call flows: https://imgur.com/a/Yj2DJpn Is there a right way to do this?

2 2

htable: autoexpire vs updatedelete
by Дилян Палаузов 06 Feb '23

06 Feb '23

Hello, https://www.kamailio.org/docs/modules/devel/modules/htable#htable.p.htable says for modparam('htable', 'htable', '…autoexpire … updateexpire…') autoexpire -time in seconds to delete an item from a hash table if no update was done to it. If is missing or set to 0, the items won't expire. updateexpire - if set to 1 (default), the time until expiration of an item is reset when that item is updated. Certain uses of htable may dictate that updates should not reset the expiration timeout, however, in which case this attribute can be set to 0. My reading is that, when a value is updated and autoexpire is positive, autoexpire will reset the timer, but updateexpire may or may not reset the same timer. That is: modparam('htable', 'htable', 'a=>autoexpire=5;updatedelete=1') modparam('htable', 'htable', 'b=>autoexpire=5;updatedelete=0') If b was not updated for 5s, will it be removed, as the documentation for autoxpire suggests, irrespective of updatedelete? Greetings Dilyan

2 2

Depending on topoh.mask_key Kamailio either cannot parse RURIs or reads invalid data
by Дилян Палаузов 06 Feb '23

06 Feb '23

Hello, when I use dns_try_naptr=on corelog=-1 debug=-1 enable_tls=yes use_dns_cache=off dns_cache_init=off modparam("topoh", "mask_key", "TEAI32l)- eauiDEUIA!?()") and run Kamailio under valgrind, Kamailio logs: 20(21) ERROR: kemix [kemix_mod.c:229]: ki_kx_get_ruri_attr(): failed to parse the R-URI 20(21) ERROR: rr [loose.c:1011]: loose_route_mode(): failed to parse Request URI The workflow is: I start Kamailio. A UDP-client registers. Then a websocket client calls the UDP-client. Finally the websocket client hangs up. Valgrind does not report anything suspicious. When I use instead modparam("topoh", "mask_key", "TEAI32l") for the same workflow, with the same configuration, valgrind logs: ==14== Invalid read of size 4 ==14== at 0x4867E5: atomic_cmpxchg_int (atomic_x86.h:224) ==14== by 0x486830: futex_get (futexlock.h:99) ==14== by 0x490CE4: dns_hash_get (dns_cache.c:673) ==14== by 0x4972A6: dns_get_entry (dns_cache.c:2001) ==14== by 0x499288: dns_srv_get_he (dns_cache.c:2455) ==14== by 0x597AAD: no_naptr_srv_sip_resolvehost (resolve.c:1599) ==14== by 0x598332: naptr_sip_resolvehost (resolve.c:1675) ==14== by 0x5983C5: _sip_resolvehost (resolve.c:1707) ==14== by 0x49943B: dns_srv_sip_resolvehost (dns_cache.c:2516) ==14== by 0x49B3DD: dns_sip_resolvehost (dns_cache.c:2738) ==14== by 0x59846A: sip_hostport2su (resolve.c:1727) ==14== by 0x4CB951: forward_request (forward.c:515) ==14== by 0x99993CD: t_relay_to (t_funcs.c:300) ==14== by 0x99EBA32: _w_t_relay_to (tm.c:1764) ==14== by 0x99F4DEB: ki_t_relay (tm.c:2917) ==14== by 0xA80A94B: sr_kemi_lua_exec_func_ex (app_lua_api.c:1022) ==14== by 0xA81237D: sr_kemi_lua_exec_func (app_lua_api.c:1706) ==14== by 0xA81B93F: sr_kemi_lua_exec_func_209 (app_lua_kemi_export.c:1717) ==14== by 0xA8383C0: luaD_precall (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA84C4CA: luaV_execute (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA838C90: luaD_callnoyield (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA837029: luaD_rawrunprotected (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA8391DF: luaD_pcall (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA8333DE: lua_pcallk (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA806F0C: app_lua_run_ex (app_lua_api.c:773) ==14== by 0xA825F06: sr_kemi_config_engine_lua (app_lua_mod.c:119) ==14== by 0x5048BE: sr_kemi_route (kemi.c:3784) ==14== by 0x588909: receive_msg (receive.c:502) ==14== by 0xA609ADF: ws_frame_receive (ws_frame.c:644) ==14== by 0x4BCAAB: sr_event_exec (events.c:299) ==14== Address 0x0 is not stack'd, malloc'd or (recently) free'd ==14== ==14== ==14== Process terminating with default action of signal 11 (SIGSEGV): dumping core ==14== Access not within mapped region at address 0x0 ==14== at 0x4867E5: atomic_cmpxchg_int (atomic_x86.h:224) ==14== by 0x486830: futex_get (futexlock.h:99) ==14== by 0x490CE4: dns_hash_get (dns_cache.c:673) ==14== by 0x4972A6: dns_get_entry (dns_cache.c:2001) ==14== by 0x499288: dns_srv_get_he (dns_cache.c:2455) ==14== by 0x597AAD: no_naptr_srv_sip_resolvehost (resolve.c:1599) ==14== by 0x598332: naptr_sip_resolvehost (resolve.c:1675) ==14== by 0x5983C5: _sip_resolvehost (resolve.c:1707) ==14== by 0x49943B: dns_srv_sip_resolvehost (dns_cache.c:2516) ==14== by 0x49B3DD: dns_sip_resolvehost (dns_cache.c:2738) ==14== by 0x59846A: sip_hostport2su (resolve.c:1727) ==14== by 0x4CB951: forward_request (forward.c:515) ==14== by 0x99993CD: t_relay_to (t_funcs.c:300) ==14== by 0x99EBA32: _w_t_relay_to (tm.c:1764) ==14== by 0x99F4DEB: ki_t_relay (tm.c:2917) ==14== by 0xA80A94B: sr_kemi_lua_exec_func_ex (app_lua_api.c:1022) ==14== by 0xA81237D: sr_kemi_lua_exec_func (app_lua_api.c:1706) ==14== by 0xA81B93F: sr_kemi_lua_exec_func_209 (app_lua_kemi_export.c:1717) ==14== by 0xA8383C0: luaD_precall (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA84C4CA: luaV_execute (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA838C90: luaD_callnoyield (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA837029: luaD_rawrunprotected (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA8391DF: luaD_pcall (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA8333DE: lua_pcallk (in /lib64/kamailio/modules/app_lua.so) ==14== by 0xA806F0C: app_lua_run_ex (app_lua_api.c:773) ==14== by 0xA825F06: sr_kemi_config_engine_lua (app_lua_mod.c:119) ==14== by 0x5048BE: sr_kemi_route (kemi.c:3784) ==14== by 0x588909: receive_msg (receive.c:502) ==14== by 0xA609ADF: ws_frame_receive (ws_frame.c:644) ==14== by 0x4BCAAB: sr_event_exec (events.c:299) The stacktrace with the values of the variables is available at https://github.com/kamailio/kamailio/issues/3350 . Or at least these things seem very similar to me. Any idea? I can share the OCI-image and the full configuration. Greetings Дилян

3 4

topoh: th_mask_decode (th_mask_encode (x)) ≠ x
by Дилян Палаузов 06 Feb '23

06 Feb '23

Hello, after receiving INVITE sip:test-gnome-calls@bapha.be SIP/2.0 Via: SIP/2.0/WSS bapha.be;branch=z9hG4bK974100 Max-Forwards: 70 To: <sip:test-gnome-calls@bapha.be> From: "Online https://sip.bapha.be" <sip:online@bapha.be>;tag=12q2efc423 Call-ID: fkhe8faq0fh80glmq59i CSeq: 5622 INVITE Contact: <sip:4q5ne5no@bapha.be;transport=ws;ob> Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound User-Agent: SIP.js/0.7.8 Content-Type: application/sdp Content-Length: 1964 v=0 o=mozilla...THIS_IS_SDPARTA-99.0 8263261229339246488 0 IN IP4 0.0.0.0 s=- … kamailio calls th_mask_encode( char *in = "sip:test-gnome-calls@bapha.be;gr=urn:uuid:f2f5a3cf-a0fb-0047-be50- 740fb9bdc562;alias=87.118.146.153~60722~2>;+sip.instance=\"<urn:uuid:f2f5a3cf-a0fb-0047-be50- 740fb9bdc562>";+org.linphone.specs=\"conference/1.0,ephemeral/1.1,groupchat/1.1,groupchat/1.2,lime\n"\ "Content-Type: application/sdp and so on", int ilen = 107, str prefix = { .len = 23, .s ="sip:127.3.4.84;line=sr-" }, int *olen). It returns olen=167, and the string "sip:127.3.4.84;line=sr- if7s1mg7i36PNf0AbdwPpfzlbqWEpLzsSGItpLwyN39ZMY4t1mCTSd6DNo4LWdIOpfpPp5FLpUQscX63Kd47W5EPWO6sNL90pLgoW5p-1fzlSdzO25n3KoIk1vHkWXptc5wOeopsWO9-eo9*". Then Kamailio forwards (sends): Contact: <sip:127.3.4.84;line=sr-if7s1mg7i36PNf0AbdwPpfzlbqWEpLzsSGItpLwyN39ZMY4t1mCTSd6DNo4LWdIOpfpPp5FLpUQscX63Kd47W5EPWO6sNL90pLgoW5p- 1fzlSdzO25n3KoIk1vHkWXptc5wOeopsWO9-eo9*>;+sip.instance="<urn:uuid:f2f5a3cf-a0fb-0047-be50- 740fb9bdc562>";+org.linphone.specs="conference/1.0,ephemeral/1.1,groupchat/1.1,groupchat/1.2,lime" which at some point leads to ACK sip:127.3.4.84;line=sr-if7s1mg7i36pnf0abdwppfzlbqweplzssgitplwyn39zmy4t1mctsd6dno4lwdiopfppp5flpuqscx63kd47w5epwo6snl90plgow5p- 1fzlsdzo25n3koik1vhkwxptc5woeopswo9-eo9* SIP/2.0 and BYE sip:127.3.4.84;line=sr-if7s1mg7i36pnf0abdwppfzlbqweplzssgitplwyn39zmy4t1mctsd6dno4lwdiopfppp5flpuqscx63kd47w5epwo6snl90plgow5p- 1fzlsdzo25n3koik1vhkwxptc5woeopswo9-eo9* SIP/2.0 Below you can find th_test.c . The functions th_mask_encode() and th_mask_decode() there are identical to the same functions in src/modules/topoh/th_mask.c . main() provides some tests: Passing «sip:test-gnome-calls@bapha.be;gr=urn:uuid:f2f5a3cf-a0fb-0047-be50- 740fb9bdc562;alias=87.118.146.153~60472~2>;+sip.instance="<urn:uuid:f2f5a3cf-a0fb-0047-be50- 740fb9bdc562>";+org.linphone.specs="conference/1.0,ephemeral/1.1,groupchat/1.1,groupchat/1.2,lime» to th_mask_encode() returns sip:127.3.4.84;line=sr-if7s1mg7i36PNf0AbdwPpfzlbqWEpLzsSGItpLwyN39ZMY4t1mCTSd6DNo4LWdIOpfpPp5FLpUQscX63Kd47W5EPWO6sNL90pLgoW5p- 1fzlSdzO25n3KoIk1vHkWXptc5wOeopsWXi-eo9* (this is what Kamailio does send out). Passing the last string to th_mask_decode() returns sip:test-gnome-calls@bapha.be;gr=urn:uuid:f2f5a3cf-a0fb-0047-be50-740fb9bdc562;alias=87.118.1 the ;alias= is incomplete, compared to the origin. The above ACK leads to passing "sr-if7s1mg7i36pnf0abdwppfzlbqweplzssgitplwyn39zmy4t1mctsd6dno4lwdiopfppp5flpuqscx63kd47w5epwo6snl90plgow5p-1fzlsdzo25n3koik1vhkwxptc5woeopswo9-eo9* SIP/2.0\n"\ "Route: <sip:Q+L06HX94DMPtQWQTI5OE8SQTI5O4MY=@144.76.142.78;transport=ws;r2=on;lr;nat=yes>\n"\ "Route: <sip:127.3.4.84;line=sr-if7s17IB5XEfVzn0WIgWwqggCTzwV5C2g5R5wCg4WwrQ5CuZ6XIQWvH3WUHkWX9tWOnyMq4RbmWsb34Q2YgoiXP-coTAboPlioPtpY6Z8dCO>\n"\ to th_mask_decode() and the result is callub.p.b;rn:s.d2lQg#cfa=bA07beQ7R409bcQ62;alac=87718*1P.15#~60R22~2 with the dots being unprintable characters. (result is garbage). I have seen that callub.p.b in the unparsable R-URI reports from Kamailio. That is an indication that th_mask_decode() does not decode the Contact encoded by th_mask_encode() correctly. Valgrind does not report improper memory usage. Greetings Dilyan gcc -g -o th-test ./th-test.c && ./th-test th-test.c #include <stdio.h> #include <stdlib.h> #include <string.h> #define TH_EB64I \ "ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz0123456789.-" char _th_EB64[65] = "EFvXIzGq94xKcW126gV5wCdYpNSbiM8enRUou7LmhajBlPtAsk-OQTf3H0DyrZ.J"; int _th_DB64[256]; char *_th_PD64 = "*"; struct str_ { char* s; /**< Pointer to the first character of the string */ int len; /**< Length of the string */ }; typedef struct str_ str; str prefix = { .len = 23, .s ="sip:127.3.4.84;line=sr-" }; char* th_mask_encode(char *in, int ilen, const str *prefix, int *olen); char* th_mask_decode(char *in, int ilen, const str *prefix, int extra, int *olen); char* th_mask_encode(char *in, int ilen, const str *prefix, int *olen) { char *out; int left; int idx; int i; int r; char *p; int block; *olen = (((ilen+2)/3)<<2) + ((prefix!=NULL&&prefix->len>0)?prefix->len:0); out = (char*)malloc((*olen+1)*sizeof(char)); if(out==NULL) { fprintf(stderr, "malloc error\n"); *olen = 0; return NULL; } memset(out, 0, (*olen+1)*sizeof(char)); if(prefix!=NULL&&prefix->len>0) memcpy(out, prefix->s, prefix->len); p = out + (int)((prefix!=NULL&&prefix->len>0)?prefix->len:0); for(idx=0; idx<ilen; idx+=3) { left = ilen - idx - 1 ; left = (left>1)?2:left; block = 0; for(i=0, r=16; i<=left; i++, r-=8) block += ((unsigned char)in[idx+i]) << r; *(p++) = _th_EB64[(block >> 18) & 0x3f]; *(p++) = _th_EB64[(block >> 12) & 0x3f]; *(p++) = (left>0)?_th_EB64[(block >> 6) & 0x3f]:_th_PD64[0]; *(p++) = (left>1)?_th_EB64[block & 0x3f]:_th_PD64[0]; } return out; } char* th_mask_decode(char *in, int ilen, const str *prefix, int extra, int *olen) { char *out; int n; int block; int idx; int i; int j; int end; char c; for(n=0,i=ilen-1; in[i]==_th_PD64[0]; i--) n++; *olen = (((ilen-((prefix!=NULL&&prefix->len>0)?prefix->len:0)) * 6) >> 3) - n; if (*olen<=0) { return NULL; } out = (char*)malloc((*olen+1+extra)*sizeof(char)); if(out==NULL) { fprintf(stderr, "malloc error\n"); *olen = 0; return NULL; } memset(out, 0, (*olen+1+extra)*sizeof(char)); end = ilen - n; i = (prefix!=NULL&&prefix->len>0)?prefix->len:0; for(idx=0; i<end; idx+=3) { block = 0; for(j=0; j<4 && i<end ; j++) { c = _th_DB64[(int)in[i++]]; if(c<0) { free(out); *olen = 0; return NULL; } block += c << (18 - 6*j); } for(j=0, n=16; j<3 && idx+j< *olen; j++, n-=8) out[idx+j] = (char)((block >> n) & 0xff); } return out; } int main() { //this is from th_mask_init for(int i=0; i<256; i++) _th_DB64[i] = -1; for(int i=0; i<64; i++) _th_DB64[(int)_th_EB64[i]] = i; int olen, o2; char *end; char *input = "sip:test-gnome-calls@bapha.be;gr=urn:uuid:f2f5a3cf-a0fb-0047-be50- 740fb9bdc562;alias=87.118.146.153~60472~2>;+sip.instance=\"<urn:uuid:f2f5a3cf-a0fb-0047-be50- 740fb9bdc562>\";+org.linphone.specs=\"conference/1.0,ephemeral/1.1,groupchat/1.1,groupchat/1.2,lime\""; printf("input=[%s]\n", input); char* middle = th_mask_encode (input, 107, &prefix, &olen); printf("olen = %i, middle=[%s]\n", olen, middle); end = th_mask_decode(middle, 147, &prefix, 0, &o2); printf("o2 = %i, end=[%s]\n", o2, end); char * b1 = "sr-if7s1mg7i36pnf0abdwppfzlbqweplzssgitplwyn39zmy4t1mctsd6dno4lwdiopfppp5flpuqscx63kd47w5epwo6snl90plgow5p- 1fzlsdzo25n3koik1vhkwxptc5woeopswo9-eo9*"; o2 = 0; end = th_mask_decode(b1, 147, &prefix, 0, &o2); printf("o2 = %i, end=[%s]\n", o2, end); char* b2 = "sr-if7s1mg7i36pnf0abdwppfzlbqweplzssgitplwyn39zmy4t1mctsd6dno4lwdiopfppp5flpuqscx63kd47w5epwo6snl90plgow5p-1fzlsdzo25n3koik1vhkwxptc5woeopswo9-eo9* SIP/2.0\n"\ "Route: <sip:Q+L06HX94DMPtQWQTI5OE8SQTI5O4MY=@144.76.142.78;transport=ws;r2=on;lr;nat=yes>\n"\ "Route: <sip:127.3.4.84;line=sr-if7s17IB5XEfVzn0WIgWwqggCTzwV5C2g5R5wCg4WwrQ5CuZ6XIQWvH3WUHkWX9tWOnyMq4RbmWsb34Q2YgoiXP-coTAboPlioPtpY6Z8dCO>\n"\ "Via: SIP/2.0/WSS bapha.be;branch=z9hG4bK9069155\n"\ "Max-Forwards: 70\n"\ "To: <sip:test-gnome-calls@bapha.be>;tag=Pn4npAj\n"\ "From: \"Online https://sip.bapha.be\" <sip:online@bapha.be>;tag=12q2efc423\n"\ "Call-ID: fkhe8faq0fh80glmq59i\n"\ "CSeq: 5623 ACK\n"\ "Supported: outbound\n"\ "User-Agent: SIP.js/0.7.8\n"\ "Content-Length: 0\n"\ "\n"; o2 = 0; end = th_mask_decode(b2, 147, &prefix, 0, &o2); printf("o2 = %i, end=[%s]\n", o2, end); return 0; }

2 1

no autocomplete for kamcmd for latest debian packages
by Ovidiu Sas 04 Feb '23

04 Feb '23

Hello, The latest debian packages are built without autocomplete for kamcmd. Do we have the libreadline-dev package installed on the build machine? I know that we have kamctl rpc and kamcli, but if we are still shipping kamcmd, it should have the autocomplete feature enabled. Regards, Ovidiu Sas -- VoIP Embedded, Inc. http://www.voipembedded.com

3 2

Sip messages responses counters per interface
by Patrick Karton 02 Feb '23

02 Feb '23

2 1

rtpengine - SRTP <> RTP
by kdrewicz+kamailio＠cludo.pl 01 Feb '23

01 Feb '23

I try to workout if - currently it would work, or - where and how to debug more: I face - 2 interfacec - public internet (so, TLS + sRTP) is desired and private - old infrastructure - i mus only use plain RTP 172.23.9.70 - private ip - from this endpoint of kamailio and rtpengine should send only basic RTP 172.23.210.75:5060 private - target for kamailio 1.2.3.24 obfuscated public IP (TLS + sRTP required) kamailio 5.4.4 (x86_64/linux) rtpengine -v Version: 11.1.1.4-1~bpo11+1 all i do is: if (proto==TLS) { rtpengine_manage("RTP/AVP ICE=remove replace-session-connection replace-origin pad-crypto ptime=20 codec-transcode-PCMA record-call=on allow-transcoding direction=external direction=internal record-call=on"); } else if ($ru =~ "transport=tls") { rtpengine_manage("DTLS=on SRTP AVPF ICE=remove replace-session-connection replace-origin pad-crypto ptime=20 codec-transcode-PCMA record-call=on allow-transcoding direction=internal direction=external record-call=on media-address=1.2.3.24"); } # 1.2.3.24 obfuscated public IP 172.23.210.75:5060 is in dispatch.cfg, as '11' route[SBC_CORE] { append_hf("X-My-SRTP: removed31337\r\n"); ### i see this text in invtes from kamailio 172.23.9.70 towards 172.23.210.75:5060 ### i see only RTP, so as expected if (!ds_select_dst("11", "0")) { xwarn("I:$var(i) DROP(DOWN!) FWD:$rm [$fU->$tU] [SBCVIP] to $du\n"); sl_send_reply("503", "Destination down"); exit; } what i did: certificate is a paid one (the public party needs it) TLS works i deleted - entries in (not kamailo) cryptosuite that caused this: 13:08:05 localhost rtpengine[15140]: ERR: [51ad8758-b64d-4d2f-9fd0-41d03a38f74d]: [core] Failed to parse a=crypto attribute, ignoring: unknown crypto suite Tried to search for any "ready" examples for this - only found old threads and - that this should be possible, but - no example for woking config. what i see: Jan 19 19:00:57 localhost rtpengine[17301]: DEBUG: [core] timer run time = 0.000038 sec Jan 19 19:00:58 localhost rtpengine[17301]: DEBUG: [core] timer run time = 0.000036 sec Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] Closing call due to timeout Jan 19 19:00:59 localhost rtpengine[17301]: DEBUG: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] Redis delete_async=0 Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] Final packet stats: Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --- Tag 'JVR5LTs', created 60:00 ago for branch '' Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --- subscribed to '' Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --- subscription for '' Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] ------ Media #1 (audio over RTP/SAVP) using unknown codec Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --------- Port 1.2.3.24:30136 <> 52.129.106.28:17030, SSRC 0, in 0 p, 0 b, 0 e, 3600 ts, out 0 p, 0 b, 0 e Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --------- Port 1.2.3.24:30137 <> 52.129.106.28:17031 (RTCP), SSRC 0, in 0 p, 0 b, 0 e, 3600 ts, out 0 p, 0 b, 0 e Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --- Tag '', created 60:00 ago for branch '' Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --- subscribed to 'JVR5LTs' Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --- subscription for 'JVR5LTs' Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] ------ Media #1 (audio over RTP/AVP) using unknown codec Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --------- Port 172.23.9.70:30014 <> :0 , SSRC 0, in 0 p, 0 b, 0 e, 3600 ts, out 0 p, 0 b, 0 e Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] --------- Port 172.23.9.70:30015 <> :0 (RTCP), SSRC 0, in 0 p, 0 b, 0 e, 3600 ts, out 0 p, 0 b, 0 e Jan 19 19:00:59 localhost rtpengine[17301]: INFO: [c17bab16-5eea-492e-b1c4-ac9387f3e265]: [core] Moved metadata file "/var/spool/rtpengine/tmp/rtpengine-meta-c17bab16-5eea-492e-b1c4-ac9387f3e265-7003946f152c6c8d.tmp" to "/var/spool/rtpengine/metadata" Jan 19 19:00:59 localhost rtpengine[17301]: DEBUG: [core] timer run time = 0.000828 sec Jan 19 19:01:00 localhost rtpengine[17301]: DEBUG: [core] timer run time = 0.000053 sec route(SBC_CORE); maybe any hint or - someone has working exmaple of kamailio config + rtpengine settings ? i use only userspace daemon rtp forwarding (this is a test, dont need any performance here) Thanks,

5 9

transport=ws causing kamailio to use wrong listen socket
by nathan.bruning＠talksome.com 31 Jan '23

31 Jan '23

Hi all, We're hitting an issue while integrating secure websockets in our existing SIP infrastructure using Kamailio. When the registration comes in, it causes an entry in our AOR table, with ";transport=ws" appended. When we want to send a message to this client (using t_relay), kamailio seems to take 'ws' as being *unsecure* websockets. In turn, this makes Kamailio try to send out the message using a TCP listener - while it should have picked the TLS listener. There are some remarks in the sources about ws vs. wss, so i'm struggling to figure out where things go wrong. I've also created github issue #3340 with more details. Any help would be appreciated. If this turns out to be a Kamailio bug, i'm happy to provide a patch.

6 7

How can I setup Kamailio for failover on LCR ?
by scramatte 31 Jan '23

31 Jan '23

Hello, How can I setup Kamailio for failover on LCR ? Currently my Kamailio 4.x works with lcr but doesn’t care about gateway status. I mean that if a gateway match rule , it sends the call through without check if the gateway is available an never try the next one if the first fail. I’ve tried to implement inactivate_gw() But I received error that lcr_id_avp is not defined. I’ve added the parameters to lcr module for Ping and enable defunc but I’m unable to get it working. By the way I ‘ve trad that defunct is not a good practice as hacker sending malformed sip header can throw down the gateway. So what is the good and working way to setup outbound call failover ? Regards Sebastien

1 0

Evapi Dispatcher 100% cpu.
by Sebastian Sastre 30 Jan '23

30 Jan '23

Hello We are currently using EVAPI to push messages into kamailio from a go app. For the most part ift works without problems, messages get delivered into the event route and we use it to update presence status. Recently we started noticing delays on the evapi processing. After 500 requests x min the pid corresponding to the EVAPI dispatcher gets pegged at 100% CPU and processing of requests starts slowing down until in some cases the connection just drops and we have to reconnect. After that pid drops, the dispatcher goes back to working. Unfortunately the error we get back is very generic and is not telling us much logger.go:39: ERR Handler returned error (write tcp 127.0.0.1:48448->127.0.0.1:8228: write: broken pipe) This only happens on prod servers and with high load which makes it hard to debug. There is no other slowness or increase of traffic or drop SIP traffic, only evapi dispatcher at 100% and the EVAPI workers on idle. Curious if anyone has run into this issue. We've tried different versions (5.4 , 5.5 and 5.6) and it happens on all of them. Any feedback appreciated ! Thanks !

2 1

Is preloading openssl_mutex_shared necessary for Kamailio 5.6 or is it not necessary?
by Дилян Палаузов 30 Jan '23

30 Jan '23

Hello, https://www.kamailio.org/docs/modules/devel/modules/tls says: For OpenSSL (libssl) v1.1.x, it is required to preload 'openssl_mutex_shared' library shipped by Kamailio. For more details see 'src/modules/tls/utils/openssl_mutex_shared/README.md'. 'src/modules/tls/utils/openssl_mutex_shared/README.md says: **IMPORTANT: the workaround of using this preloaded shared library is no longer needed starting with Kamailio v5.3.0-pre1 (git master branch after September 14, 2019). The code of this shared library has been included in the core of Kamailio and the same behaviour is now achieved by default.** This is a shared library required as a short term workaround for using Kamailio with OpenSSL (libssl) v1.1. It has to be pre-loaded before starting Kamailio. In v1.1, libssl does not allow setting custom locking functions, using internally pthread mutexes and rwlocks, but they are not initialized with process shared option (PTHREAD_PROCESS_SHARED), which can result in blocking Kamailio worker processes. Is preloading openssl_mutex_shared necessary for Kamailio 5.6 or is it not necessary? If it is not necessary, why isn’t it removed from the source code? Greetings Дилян

1 0

Websocket-answers sent by K. log unparseable errors
by Дилян Палаузов 29 Jan '23

29 Jan '23

Hello, when I connect to Kamailio over websocket, according to the websocket protocol K. sends “HTTP/1.1 101 Switching Protocols” and K. logs: 17(18) DEBUG: websocket [ws_handshake.c:179]: ws_handle_handshake(): found Upgrade: websocket 17(18) DEBUG: websocket [ws_conn.c:195]: wsconn_add(): connection id [14] 17(18) DEBUG: websocket [ws_conn.c:213]: wsconn_add(): new wsc => [0x7fcad6e10160], ref => [0] 17(18) DEBUG: websocket [ws_conn.c:233]: wsconn_add(): added to conn_table wsc => [0x7fcad6e10160], ref => [1] 17(18) DEBUG: sl [sl.c:310]: send_reply(): reply in stateless mode (sl) 17(18) DEBUG: <core> [core/msg_translator.c:162]: check_via_address(): (22.222.222.222, 22.222.222.222, 0) 17(18) DEBUG: <core> [core/tcp_main.c:1644]: _tcpconn_find(): found connection by id: 14 17(18) DEBUG: <core> [core/tcp_main.c:2528]: tcpconn_send_put(): send from reader (18 (17)), reusing fd 17(18) DEBUG: <core> [core/tcp_main.c:2763]: tcpconn_do_send(): sending... 17(18) DEBUG: <core> [core/tcp_main.c:2796]: tcpconn_do_send(): after real write: c= 0x7fcad6da7b38 n=238 fd=9 17(18) DEBUG: <core> [core/tcp_main.c:2797]: tcpconn_do_send(): buf= 17(18) DEBUG: <core> [core/parser/parse_fline.c:255]: parse_first_line(): bad request first line 17(18) DEBUG: <core> [core/parser/parse_fline.c:258]: parse_first_line(): at line 0 char 22: 17(18) DEBUG: <core> [core/parser/parse_fline.c:264]: parse_first_line(): parsed so far: HTTP/1.1 101 Switching 17(18) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 22) 17(18) DEBUG: <core> [core/parser/msg_parser.c:675]: parse_msg(): invalid message 17(18) ERROR: <core> [core/parser/msg_parser.c:749]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 101 Switching Protocols Sia: SIP/2.0/TLS 22.222.222.222:33776 Sec-WebSocket-Protocol: sip Upgrade: websocket Connection: upgrade Sec-WebSocket-Accept: NeswoJX5ZQp7ER8hTKM5B3a3HDQ= Content-Length: 0 > 17(18) ERROR: <core> [core/msg_translator.c:3256]: build_sip_msg_from_buf(): parsing failed 17(18) DEBUG: app_lua [app_lua_api.c:1924]: sr_kemi_lua_exit(): script exit call 17(18) DEBUG: app_lua [app_lua_api.c:789]: app_lua_run_ex(): ksr error call from Lua: ~~ksr~exit~~ 17(18) DEBUG: app_lua [app_lua_mod.c:164]: sr_kemi_config_engine_lua(): execution of route type 513 with name [ksr_xhttp_event] returned 1 22.222.222.222 is my public address (the address of my router). It is not the IP-address of Kamailio. While the returned address is not parseable in terms of SIP, it is a valid Websocket-answer message. This valid condition triggers logging an error, but does not represent an error condition. How can I avoid that the above non-error is logged as an error? Thanks for your feedback Дилян

1 0

Locally generated BYE sent to private IP
by James Lipski 27 Jan '23

27 Jan '23

Hi all, I'm doing some experimenting with kamailio -- currently looking for a replacement of our existing (aging) SIP infrastructure. One of the experiments that I'm doing is testing the dialog module's dlg_set_timeout function. The issue that I'm having is while I do see that kamailio is generating a BYE, and sends it out to the B-leg of the call properly, the BYE goes to a private address on the A-leg. To Note: - kamailio is behind a NAT (on AWS) - both UA endpoints are remote and are both behind a NAT - Not sure if this matters, but I do have the topos module enabled, though I see the same behavior regardless if it's enabled or not. - Have set nathelpers (fix_nated_register, fix_nated_contact,etc...) which helps calls between endpoints and the endpoints themselves are able to send/receive BYEs when they generate them; however doesn't seem to help if kamailio locally generates the BYE. The only thing I do see within my logs is when the BYE is being generated by kamailio, it uses the private IP of my UA and not public for the caller. Callee, the public address is referenced. If more information is needed, please let me know (just didn't want to spam my script/ logs). Thanks. Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: dialog [dlg_req_within.c:399]: send_bye(): sending BYE to caller Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: tm [uac.c:458]: t_uac_prepare(): next_hop=<sip:1000@10.0.0.47:38606;line=9k4wsgc9> Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: tm [uac.c:158]: dlg2hash(): hashid 40484 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_fline.c:249]: parse_first_line(): first line type 1 (request) flags 1 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:679]: parse_msg(): SIP Request: Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:680]: parse_msg(): method: <BYE> Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:682]: parse_msg(): uri: <sip:1000@10.0.0.47:38606;line=9k4wsgc9> Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:684]: parse_msg(): version: <SIP/2.0> Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [Via] type 1 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_via.c:1300]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK42e9.4a962087000000000000000000000000.0>; state=16 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:555]: parse_headers(): Via found, flags=2 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:557]: parse_headers(): this is the first via Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [To] type 3 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=ueisi99vbf Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_addr_spec.c:884]: parse_addr_spec(): end of header reached, state=29 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): <To> [61]; uri=[sip:1000@fakedomain.com] Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:174]: get_hdr_field(): to body (44)[<sip:1000@fakedomain.com>], to tag (10)[ueisi99vbf] Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [From] type 4 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [CSeq] type 5 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <1> <BYE> Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [Call-ID] type 6 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [Content-Length] type 12 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:187]: get_hdr_field(): content_length=0 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [Max-Forwards] type 8 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/msg_parser.c:91]: get_hdr_field(): found end of header Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=1132060721 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/parser/parse_addr_spec.c:884]: parse_addr_spec(): end of header reached, state=29 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: topos [tps_msg.c:1013]: tps_request_sent(): handling outgoing request (1, 1) Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: topos [tps_msg.c:417]: tps_pack_message(): compacted headers - x_via1: [SIP/2.0/UDP <KAMAILIO PUBLIC IP>:5060;branch=z9hG4bK42e9.4a962087000000000000000000000000.0](85) - x_via2: [](0) - x_vbranch1: [z9hG4bK42e9.4a962087000000000000000000000000.0](46) Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: topos [tps_msg.c:539]: tps_pack_message(): compacted headers - a_rr: [](0) - b_rr: [](0) - s_rr: [](0) Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: topos [tps_msg.c:544]: tps_pack_message(): compacted headers - as_contact: [](0) - bs_contact: [](0) Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: topos [tps_msg.c:1027]: tps_request_sent(): no x-uuid header - nothing to do Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/msg_translator.c:1799]: check_boundaries(): no multi-part body Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: topos [topos_mod.c:567]: tps_msg_sent(): new outbound buffer generated Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: tm [uac.c:686]: send_prepared_request_impl(): uac: 0x7f22f94448e8 branch: 0 to 10.0.0.47:38606 Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: <core> [core/onsend.c:50]: run_onsend(): required parameters are not available - ignoring Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: dialog [dlg_req_within.c:432]: send_bye(): BYE sent to caller Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: dialog [dlg_req_within.c:399]: send_bye(): sending BYE to callee Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: tm [uac.c:458]: t_uac_prepare(): next_hop=<sip:1001@<UA PUBLIC IP>:5066>Jan 26 16:33:29 localhost /opt/kamailio-5.6.2/sbin/kamailio[1238222]: DEBUG: tm [uac.c:158]: dlg2hash(): hashid 40482

3 3

Kamailio not responding after stress test
by Igor Olhovskiy 27 Jan '23

27 Jan '23

Hello, I'm continuing investigations on Kamailio and stress test. Got it again in the state where it's not accepting any new TCP/TLS connections (UDP still works though), but all looks good from lsof/netnstat part, like system is not reporting any zombie connections. Restart of Kamailio process helps This time I got output of kamctl trap Put it here: https://pastebin.com/iYrNZ8U9 kamailio --version version: kamailio 5.6.2 (x86_64/linux) 54a9c1 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 54a9c1 compiled on 14:01:01 Oct 18 2022 with gcc 4.8.5 It's statically linked with tlsa pointing on openssl-1.1.1q Settings related to TLS are: fork=yes children=4 tcp_children=12 enable_tls=yes tcp_accept_no_cl=yes tcp_max_connections=63536 tls_max_connections=63536 tcp_accept_aliases=no tcp_async=yes tcp_connect_timeout=10 tcp_conn_wq_max=63536 tcp_crlf_ping=yes tcp_delayed_ack=yes tcp_fd_cache=yes tcp_keepalive=yes tcp_keepcnt=3 tcp_keepidle=30 tcp_keepintvl=10 tcp_linger2=30 tcp_rd_buf_size=80000 tcp_send_timeout=10 tcp_wq_blk_size=2100 tcp_wq_max=10485760 open_files_limit=63536 Can you please help to read gdb output and understand where I missed in config? Thanks in advance! -- Best regards, Ihor (Igor)

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