- sr-users - kamailio.org

SRTP/TLS BYE Problem with One of the Legs of the Call
by conradcordeiro＠gmail.com 20 Sep '20

20 Sep '20

Hello, Thank you for your reading this and for your help. I'm a Kamailio newbie and managed to set up an edge proxy, which works perfectly on UDP traffic. I'm now attempting to deploy TLS/SRTP and everything almost works perfectly. The single issue I'm having is that when either of the parties to an SRTP/TLS call disconnect, the other party's call remains active. With UDP, when one of the parties disconnects the call, the other leg of the call receives the BYE command and the call automatically disconnects. This is how I have our infrastructure set up: 1. Twilio SIP Trunk with Secure Media enabled. 2. Kamailio 5.4.1 set up with TLS module, set to listen on TLS port 5061, SSL certificates from Let's Encrypt, route set to our phone system. 3. Phone system is Asterisk. As per above, everything works almost perfectly with TLS/SRTP. The only issue is that calls will not disconnect when one of the sides hang up. If I disable TLS/SRTP and use UDP only, everything works. Audio is just fine with TLS/SRTP. Does anyone know why calls with SRTP/TLS will not disconnect automatically when one of the parties ends the call? Thank you, Conrad

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Kamailio support for 5g
by Pavithra Mohanraja 18 Sep '20

18 Sep '20

Hi, Does kamailio IMS support for 5g Network functions since in 5g everything is handled via http requests. Could anyone please respond Thanks, Pavithra

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IMS Diameter server module
by Pavithra Mohanraja 18 Sep '20

18 Sep '20

Hi, Can anybody reply what is the exact use of diameter server module. I am unable to understand from the kamailio doc provided. It is mentioned that it converts diameter messages to json messages. But without diameter.xml file, it is not working. Whats the exact use of it. Can i not be able to send the http request alone via Rx interface instead of using diameter.xml file. Kindly reply as i am blocked in this for few days. Thanks, Pavithra

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Connection-oriented socket closure event hooks
by Alex Balashov 18 Sep '20

18 Sep '20

Hi, The `tcpops` module exposes, * event_route[tcp:closed] * event_route[tcp:timeout] * event_route[tcp:reset] These fire on closure of TCP connections and TLS connections. TLS is considered a separate transport in Kamailio, having distinct identifiers in all cases where transport protocol is relevant, configured by separate and specific parameters, and provided by a separate module (tls). But TLS runs on top of TCP, of course, so treating a TLS disconnection as a superset of TCP disconnection makes sense. WebSocket is considered a separate transport in Kamailio, having distinct identifiers in all cases where transport protocol is relevant, configured by separate and specific parameters, and provided by a separate module (websocket). But WebSocket runs on top of TCP, of course... ... yet it doesn't trigger these TCP connection events. Instead, one must use a separate `event_route[websocket_closed]` event route for this. Is there a reason for this inconsistency? I would think that the common denominator of TCP would lead to the same hierarchy of abstraction when it comes to events. It becomes important because if one wishes for Kamailio to log or notify disconnections for vanilla SIP or browser-based clients, one must essentially implement the same code twice. Thank you, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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webrtc sound problem in version 85.0 google chrome version
by Hamed Rastegar 17 Sep '20

17 Sep '20

Hello I am using kamailio 5.1 for WebRTC (websocket) But in the latest version of Google Chrome (85.0) I have a problem, No sound is transmitted between the source and the destination. This problem does not exist in Firefox browser If you have experience in this field, please comment

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Diameter Messages for Rx
by Pavithra Mohanraja 17 Sep '20

17 Sep '20

Hi, Kindly give me some guidelines on how to convert diameter messages to http messages when establishing connection between PCSCF and PCF Kindly help Thanks, Pavithra

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Few newbie questions
by Adam Rutkowski 17 Sep '20

17 Sep '20

Hi Everyone As I am just dipping my foot in the VOIP waters so please bear with me. Although it may be more appropriate for Freeswitch group I am sure there are plenty of knowledgeable folks here. I'm playing with freeswitch in my lab environment and really like what Kamailio has got to offer. That being said, there are few concepts I need help understanding. Consider this ( for hosted services I suppose ) [image: Screenshot 2020-09-16 at 22.21.22.png] 1 Does this topology seem right ? 2 If Kamailio is used as registar and load balancer, does that mean all domains, extensions must be present on all the FS boxes ? 3 How will auto provisioning work in this case ? Now, in the below scenario, clients have got FS box deployed on prem [image: Screenshot 2020-09-16 at 22.28.09.png] 4 Is there any reason to keep the FS farm ? or Kamailio poxying calls to VOIP providers is enough ? 5 What roles Kamailio could be used for in this case ? 6 Is deploying on prem PBX for a customer a popular solution ? 7 Anyone deployed above on Docker/ Kubernetes ? How's the performance and the bad bits if any ? I do realize this is pretty basic stuff for everyone in here, however, I would appreciate any helpful comments and if there is some good literature/ links you can recommend, I am all ears. Regards Adam

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Kamailio IMS Diameter Messages
by Pavithra Mohanraja 17 Sep '20

17 Sep '20

Hi, Is there any possibility of converting diameter messages to http requests in IMS. Since I am working in 5g based network, In order to connect pcscf to pcf, Rx interface has to be established. But PCF is not accepting diameter messages. It is failing with error "No connected DefaultRoute peer found for app_id 16777236 and vendor id 10415" I used IMS Diameter server since it is mentioned as converting diameter messages to http json messages. But even before calling that diameter server, It is failing with App_id issue. How it works? Need to understand, how the diameter messages are working since in 5g , everything is based on http rest apis. Is there any possibility of connecting to it or there is no support in kamailio IMS for 5g. If you have any idea, please let me know. It will be very useful for me. Thanks, Pavithra

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multi netwotk interface kamailio stops processing packets in one of it's network interfaces although keeps processing in the others
by Nuno Miguel Reis 17 Sep '20

17 Sep '20

Hello everyone. I've started to have an issue with a kamailio 4.4 instance listening on multiple network interfaces where it stops processing SIP on one of it's network interfaces but still continues to work fine on the others. If I restart kamailio everything starts working fine again. I'm using the default 'children=8' and one of my guesses on why this could be happening is that the number of childs processing couldn't be enough. Do you remember anything else on an issue like this where I should be looking? Thanks. -- Nuno Miguel Reis Departamento de Engenharia Informática Faculdade de Ciências e Tecnologia Universidade de Coimbra

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Regular Expression Evaluation
by Mack Hendricks 16 Sep '20

16 Sep '20

I already run into issues with this. If I want to only execute a statement if it’s an INVITE with a request domain of 10.10.*.* how do I format? Below is what I have. I always have to play with it a bit. Looking to get a better understanding. The regular expression tester I’m using states that it’s valid. What am I doing wrong? if ((is_method("INVITE")) && ($rd =~ "10\.10\..+")) { # Do something xlog("fix got a 10.10 address"); } Thanks

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CockroachDB and Kamailio
by Noah Mehl 16 Sep '20

16 Sep '20

Has anyone been down this path before? We are trying to test this out and the results are pretty promising so far. I realize the lack of Stored Procedures and Triggers make this untenable for many Postgres based implementations. Thanks! ~Noah

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Kamailio does not forward BYE to Asterisk
by egemen ulus 16 Sep '20

16 Sep '20

Hi all, I have an infrastructure with Kamailio (reg), Asterisks farm, and Kamailio (4.3) proxy which dispatches calls. Trunk-1 (Vlan_A) Kam(reg) <> Asterisk Farm (5) <> Kam(proxy) <> Trunk-2 (Vlan_B) Trunk-3 (Vlan_C) For each trunk, I have to use the IP of the VLAN interface in "contact header". That means when I need to send a call to Trunk-1 I have to set Vlan_A IP address on the contact header. I have achieved that by rewriting Contact Header. ### remove_hf("Contact:"); append_hf("Contact: <sip:$fU@$var(recieved)>\r\n", "Call-ID"); #by using db ### But this causes a problem, Kamailio does not forward the BYE message that comes from Trunk-X to Asterisk (In before, the BYE message had Asterisk IP in R-URI). However, when I use topoh module it forward properly for one trunk. But I can not set multiple mask IP in topoh module. I'm using the dialog module and I thought it can handle all requests in one dialog to forward to the correct asterisk but it did not. Is there any suggestion for this case? And also how many sub-interfaces can Kamailio handle for a good performance? Or do you suggest multiple Kamailio instances instead of multiple sub-interfaces on one server? [https://ipmcdn.avast.com/images/icons/icon-envelope-tick-round-orange-anima…]<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> Virus-free. www.avast.com<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…>

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Kamailio used as proxy for Conference factories
by Adrian Tabacioiu 16 Sep '20

16 Sep '20

Hello, I am new in this VOIP field, yet I have to make a decision in using a SIP proxy that would allow the configuration of multiple conference factories, make load balancing when creating new conference, and provide presence information related to active conferences. Can you please tell me if this is possible with Kamailio, and give me some reference to a module or some documentation that would relate to that ? Thank you in advance. Best regards, Adrian Tabacioiu

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Proxying BFCP
by René Hansen 15 Sep '20

15 Sep '20

Hi, I'm rather new to Kamailio and RTPEngine, so I'm looking for some help in locating any insights, documentation, guides or examples that resemble my use case. Currently I've got both running in working order and I can do SIP calls with both video and audio. Now I'm trying to figure out how to pass through BFCP as well. From what I understand so far, since BFCP isn't strictly RTP, it's not something that's outright supported. Please correct me if my assumption is wrong. My observation is that "*m=application PORT {UDP,TCP}/BFCP **" SDP lines doesn't seem to open for traffic as other media lines do. (Obviously expected if it's not supported) So I'm really just looking for any suggestions for, or where to start looking, in getting BFCP support. I do have the option of BFCP over both TCP and UDP, so if I could somehow include the originator IP as an added attribute in the re-written SDP answer, I could then just connect directly, around the proxy. I've yet to figure out how to do that as well however. Any pointers are much appreciated. Regards, René Hansen

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Called with SDP without DTLS fingerprint from WebRTC to SIP
by Benjamin Flügel | vio:networks 15 Sep '20

15 Sep '20

Hey guys and girls, I'm trying to configure Kamailio so that I can communicate over it using WebRTC and SIP. Client 1 is a softphone based on SIPJS. Client 2 is a Yealink phone. Both are registered with the Kamailio. Behind the Kamailio there is another Asterisk. The Kamailio acts as a proxy here. So far everything works. Only if I accept the call, which is made from Client 1 to Client 2 or vice versa, then I get this error in Client 1: Called with SDP without DTLS fingerprint. I can see that the fingerprint is present in the INVITE from Client 1 to Client 2. However, it is missing in the OK. The lower part contains my NATMANAGE config. This could probably be optimized a lot, but for now it should work. Unfortunately I have only recently started with Kamailio. I hope you can help me there. I can also send you the SIP-Log if you want to. Many thanks in advance. Greetings Benny route[NATMANAGE] { if (is_request()) { if(has_totag()) { if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (check_route_param("rtp=bridge")) { setbflag(FLB_BRIDGE); } if (check_route_param("rtp=ws")) { setbflag(FLB_RTPWS); } } } if (!isbflagset(FLB_BRIDGE)) { return; } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB) || isbflagset(FLB_RTPWS))) { return; } $xavp(r=>$T_branch_idx) = "replace-origin replace-session-connection"; if (!nat_uac_test("8")) { xlog("IS TRUSTED"); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " trust-address"; } if (is_request()) { if (!has_totag()) { if (!t_is_failure_route()) { $avp(extra_id) = @via[1].branch + $T_branch_idx; $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " via-branch=extra"; xlog("BRANCH: $T_branch_idx"); } } } if (is_reply()) { $avp(extra_id) = @via[2].branch + $T_branch_idx; $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " via-branch=extra"; xlog("BRANCH: $T_branch_idx"); } if(isbflagset(FLB_RTPWS)){ xlog("IS WS"); if(is_request()){ xlog("IS REQUEST"); if ($proto =~ "ws") { xlog("WEB --> SIP"); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP"; rtpengine_manage($xavp(r=>$T_branch_idx)); if (route(FROMASTERISK)) { xlog("FROM INTERNAL"); } else { xlog("FROM EXTERNAL"); rtpengine_manage(" direction=external direction=internal ICE=remove loop-protect"); } } else { xlog("SIP --> WEB"); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-offer generate-mid DTLS=passive SDES-off ICE=force RTP/SAVPF direction=internal direction=external loop-protect"; } } else { xlog("IS RESPONSE"); if ($proto =~ "ws") { xlog("WEB --> SIP"); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP direction=external direction=internal loop-protect"; } else { xlog("SIP --> WEB"); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-offer generate-mid DTLS=passive SDES-off ICE=force RTP/SAVPF direction=internal direction=external loop-protect"; } rtpengine_manage($xavp(r=>$T_branch_idx)); } } else { xlog("SIP --> SIP"); if (route(FROMASTERISK)) { xlog("FROM INTERNAL"); rtpengine_manage(" direction=internal direction=external ICE=remove loop-protect"); } else { xlog("FROM EXTERNAL"); rtpengine_manage(" direction=external direction=internal ICE=remove loop-protect"); } } xlog("NATMANAGE branch_id:$T_branch_idx ruri: $ru, method:$rm, status:$rs, extra_id: $avp(extra_id), rtpengine_manage: $xavp(r=>$T_branch_idx)\n"); if (is_request()) { if (!has_totag()) { if (t_is_branch_route()) { if (isbflagset(FLB_NATB)) { add_rr_param(";nat=yes"); } if (isbflagset(FLB_BRIDGE)) { add_rr_param(";rtp=bridge"); } if (isbflagset(FLB_RTPWS)) { add_rr_param(";rtp=ws"); } } } } if (is_reply()) { if (isbflagset(FLB_NATB)) { if (is_first_hop()) { if (af == INET) { set_contact_alias(); } } } } return; }

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UAC module supports registration with Remote registrar using WSS ?
by sagar malam 15 Sep '20

15 Sep '20

Hello , I want to use UAC module to register with a remote registrar using WSS. Does it support it ? https://kamailio.org/docs/modules/5.3.x/modules/uac.html#idm614 -- Thanks, Sagar

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Installing Diameter server module (IMS_DIAMETER_SERVER)
by Pavithra Mohanraja 15 Sep '20

15 Sep '20

Hi, How to install Diameter server module (IMS_DIAMETER_SERVER) using apt-get package. I need to have DRA in my application. Can anybody please help on this on how to install and how to use it. Kindly help. Thanks, Pavithra

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Topos_redis and loose_route
by Ernest Mavrel 15 Sep '20

15 Sep '20

Dear Community, Call scenario: Calling number 33825462354 Called number 44656646820 UAC (IP=1.1.1.1) => Kamailio (IP=2.2.2.2) => SIP proxy (IP=3.3.3.3) I stocked on strange issue with module TOPOS_REDIS and PRACK message (IP=2.2.2.2 kamailio version 5.2.3). Configuration with module TOPOS works, but because of a lot of calls we would like to use TOPOS_REDIS (avoid mysql). I already check this: https://lists.kamailio.org/pipermail/sr-users/2018-May/101641.html and I already have fixed version of module. In attach you can find traces (pcap file and kamailio log with debug=4) Your help will be greatly appreciated Kind Regards Ernest Mavrel

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Regex in the switch statement
by Ilie Soltanici 14 Sep '20

14 Sep '20

Hello, Is there any way to use Regex or 'OR/AND' operators in the switch/case statements? I've tried the following syntax but it's not working, are there any other ways to do this? switch($dlg_var(ISP)) { case "TRUNK1|TRUNK2": } Thank you.

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RTPENGINE question
by Andrew Chen 14 Sep '20

14 Sep '20

Hey guys, Is this the right place to ask about rtpengine (ngcp) related issues with kernel packet forwarding? Thanks. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

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Kamailio IMS - 5.3.5 with 5g core
by Pavithra Mohanraja 14 Sep '20

14 Sep '20

Hi, I am working with kamailio IMS 5.3.5. I am trying to connect kamailio IMS with 5g core. Did anybody come across this scenario. If so, Please help me how Rx interface works with 5g and how UDM and HSS can be an converged application incase of 5g. Kindly help as it is an urgent requirement. Thanks, Pavithra

1 0

Kamailio question
by alexandros vlachos 14 Sep '20

14 Sep '20

Hello, I am a new user and i am trying to use kamailio as remote sip server for clients (softphones particularly) The error message is "SIP: SIP/2.0 403 Not relaying" The kamailio listens with one public ip address in a datacenter ( linux debian). root@vmdeb1:~# kamctl version /usr/sbin/kamctl 5.2.0 The xlog shows that the register message arrives in kamailio. I am sure that the user 410 exists. I have added the remote public address to address via kamctl. I have added the location via ul add command. I am not sure about kamailio.cfg because most of it except the tls is the default. I would appreciate your help. Alexandros

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Is there any way to remove stale registrations before expires time from usrloc ?
by sagar malam 14 Sep '20

14 Sep '20

Hello , I am using the usrloc module to store registrations. I could not find any way to remove stale registrations before expires time. Most of the registrations are mobile app so they wont run in background to save battery of phone hence we end up having lots of registrations in usrloc which were not removed because app was killed in background by OS without letting it send unregister request. Is there any way to remove stale registrations before expires time from usrloc ? -- Thanks, Sagar

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SRTP-DTLS and rtpengine
by Gholamreza Sabery 12 Sep '20

12 Sep '20

Hi, I checked rtpengine's documentation, and I wonder what happens if I use SRTP-DTLS with rtpengine? My connection will be hop by hop encrypted with rtpengine, or like TURN my connection will remain end-to-end encrypted?? Thx a lot

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Re: [SR-Users] xlog print sdp parameters
by Angel Elena 11 Sep '20

11 Sep '20

Ok, thanks for the info. -------------------------------- Ángel Elena Medina _o) craem(a)craem.net / \\ http://diablo.craem.net _(___V @craem_ www.linkedin.com/in/angel-elena-medina -------------------------------- Kopano Webapp. -----Original message----- From: Henning Westerholt <hw(a)skalatan.de> Sent: Friday 11th September 2020 14:03 To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Cc: Angel Elena <craem(a)craem.net> Subject: RE: [SR-Users] xlog print sdp parameters Hello, you can use e.g. https://www.kamailio.org/docs/modules/stable/modules/sdpops.html#idm239to print in all xlog. Have a look to the other functions to get access to certain lines. Cheers, Henning -- Henning Westerholt –https://skalatan.de/blog/ Kamailio services –https://gilawa.com From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Angel Elena Sent: Friday, September 11, 2020 1:36 PM To: sr-users(a)lists.kamailio.org Subject: [SR-Users] xlog print sdp parameters Hi, good moorning. I have a Kamailio 4.4.7 in a debian box and I need print with xlog (if is possible) the o,c and m values from SDP headers. Is possible make a xlog() print this parameters with sdpops module or another ? Thanks -------------------------------- Ángel Elena Medina _o) craem(a)craem.net <mailto:craem@craem.net> / \\ http://diablo.craem.net <http://diablo.craem.net> _(___V @craem_ www.linkedin.com/in/angel-elena-medina <http://www.linkedin.com/in/angel-elena-medina> -------------------------------- Kopano Webapp.

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REST API to initiate a call in Kamailio
by Ameed Jamous 11 Sep '20

11 Sep '20

Hello Everyone, I was wondering what’s the most recommended module by the community today to initiate calls via Kamailio using REST API. Call Flow: Application —> HTTP (Async) —> Kamailio —> B2BUA —> Callee(s). Does a reliable module for this already exist today? Thanks in advance ! Stay safe. -- [image: photo] *Ameed Jamous* Founder, TelecomsXChange +1 954 246 3356 <+1+954+246+3356> | a.jamous(a)telecomsxchange.com www.telecomsxchange.com | Skype: telecomsxchange <#SignatureSanitizer_SafeHtmlFilter_> 1800 N Bayshore Dr, Suite CU1, Miami, FL 33132

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Getting memory error on kamailio v5.1.2 tm:Free fragment not found!
by M Arqum CH 11 Sep '20

11 Sep '20

Hello Users, I am facing memory-related error on my Kamailio server, using this kamailio version kamailio -v version: kamailio 5.1.2 (x86_64/linux) with -m 512 -M 32 childern=16 System Memory : 24 GB System Cores: 12 On console getting the following error : Sep 08 14:40:29 debian /usr/sbin/kamailio[99894]: ERROR: <core> [core/mem/q_malloc.c:286]: qm_find_free(): qm_find_free(0x7fdf51eb9000, 11320); Free fragment not found! Sep 08 14:40:29 debian /usr/sbin/kamailio[99894]: ERROR: <core> [core/mem/q_malloc.c:419]: qm_malloc(): qm_malloc(0x7fdf51eb9000, 11320) called from tm: h_table.c: build_cell(320), module: tm; Free fragment not found! Sep 08 14:40:29 debian /usr/sbin/kamailio[99894]: ERROR: tm [t_lookup.c:1279]: new_t(): out of mem: Sep 08 14:40:29 debian /usr/sbin/kamailio[99894]: ERROR: tm [t_lookup.c:1422]: t_newtran(): new_t failed Sep 08 14:40:29 debian /usr/sbin/kamailio[99894]: ERROR: sl [sl_funcs.c:362]: sl_reply_error(): stateless error reply used: No error (2/SL) Please guide about the possible reason of failure or its memory leakage, possible solution. Thanks -- Regards M Arqum

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Re: [SR-Users] How to handle BYE in case of serial fork
by Björn Klasen 11 Sep '20

11 Sep '20

Hi Hennning, loose_route() is what we are using. But we use also dialog module for call counting etc. We did not really change the default config. This is the snippet in our config. if (has_totag()) { # sequential requests within a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { t_on_reply("bye_reason"); setflag(FLT_ACC); # ... even if the transaction fails But I think that is not the point. The responses 18x responses are my real problem, because they have the same did in record-route but different to-tags because of the call forwarding scenario. That leads to the BYE of the first fork. With this BYE they try to stop the call with the first to-tag. But the first fork ist already canceld. So Kamailio behaves correctly by not answering or forwarding anythin, but throwing a warning with tight dialog matching not possible. Thats because at that point Kamailio does only know about the call with the second. So when the BYE is received by Kamailio the dialog modules can find the did but not the to-tog as Kamailio only has the second one left in memory. Now the point is, is there a way to hold the first early dialog fork in memory until it is either terminatet with a CANCEL or BYE or the last fork with the corresponding did is terminatet. I have read that ims_dialog or dialog_ng in our can handle things like this, but I also can't get it to work. By the way cdr generation does also not work with dialog_ng or ims_dialog (i also made tests with kamailio 5.3.5). This is the problematic BYE Message (i deleted customer relatet infomations) BYE sip:bob@kamailio:5060 SIP/2.0 Via: SIP/2.0/UDP carrier:5060;branch=z9hG4bKh6e7ie103g9amkc8gdo0.1 Max-Forwards: 65 To: <sip:bob@kamailio>;tag=11A2CF0C-5F578E7E000376B5-AA5BD700 From: <sip:anonymous@anonymous.invalid>;tag=h7g4Esbg_h7g4Esbg_c5478d8b Call-ID: ae36c13c31b5e910(a)a.b.c.d CSeq: 1313626 BYE Privacy: user;id Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, OPTIONS, PRACK, PUBLISH, REGISTER, SUBSCRIBE, UPDATE Route: <sip:kamailio-ip:5070;r2=on;lr;ftag=h7g4Esbg_h7g4Esbg_c5478d8b;did=c4851.7b1c5> Route: <sip:sems-ip;r2=on;lr;ftag=h7g4Esbg_h7g4Esbg_c5478d8b;did=c4851.7b1c5> This one matches to the 180 of the call to callee b SIP/2.0 180 Ringing Record-Route: <sip:sems-ip;r2=on;lr;ftag=h7g4Esbg_h7g4Esbg_c5478d8b;did=c4851.7b1c5> Record-Route: <sip:kamailio-ip:5070;r2=on;lr;ftag=h7g4Esbg_h7g4Esbg_c5478d8b;did=c4851.7b1c5> Via: SIP/2.0/UDP bob@kamailiko:5060;branch=z9hG4bK1s9rar309o1uf6j41ik0.1 Call-ID: ae36c13c31b5e910(a)a.b.c.d CSeq: 1313625 INVITE To: <sip:bob@kamailio>;tag=11A2CF0C-5F578E7E000376B5-AA5BD700 From: <sip:anonymous@anonymous.invalid>;tag=h7g4Esbg_h7g4Esbg_c5478d8b Content-Length: 0 Contact: <bob-pjhq8ddet3tu9@bobdomain;transport=udp> I hope you can understand my problem a little better now. BR Björn Am 11.09.20 um 12:26 schrieb Henning Westerholt: > Hello Björn, > > I was not referring to dialog stateful routing e.g. with dialog module. I was referring to the loose_route() function in your kamailio.cfg (e.g. compare to the default cfg in github). > > Maybe you can quote the problematic BYE SIP message here. > > Cheers, > > Henning > -- Björn Klasen, Specialist TNG Stadtnetz GmbH, Network Management (VoIP) Projensdorfer Straße 324 24106 Kiel Germany T +49 431/ 530530 F +49 431/ 7097-555 mailto: bklasen(a)tng.de http://www.tng.de Register: Amtsgericht Kiel HRB 6002 KI Executive board (Geschäftsführung): Dr.-Ing. Volkmar Hausberg, Sven Schade, Carsten Tolkmit, Dr. Sven Willert Tax-Id (Steuernr.): 2029047020, VAT-Id (USt-Id): DE225201428

2 2

How to check which memory allocator kamailio is using ?
by sagar malam 11 Sep '20

11 Sep '20

Hello , Is there any way to check if the memory allocator is used by kamailio ? qmalloc,fmalloc or tlsf. -- Thanks, Sagar

2 3

xlog print sdp parameters
by Angel Elena 11 Sep '20

11 Sep '20

Hi, good moorning. I have a Kamailio 4.4.7 in a debian box and I need print with xlog (if is possible) the o,c and m values from SDP headers. Is possible make a xlog() print this parameters with sdpops module or another ? Thanks -------------------------------- Ángel Elena Medina _o) craem(a)craem.net / \\ http://diablo.craem.net _(___V @craem_ www.linkedin.com/in/angel-elena-medina -------------------------------- Kopano Webapp.

2 1

Using Kamalio as a proxy for internal servers
by Moshe Katz 11 Sep '20

11 Sep '20

Hello all, (Note: I previously posted a more detailed version of this question on StackOverflow at https://stackoverflow.com/q/63760506/829970 . This version is simplified to fit better in an email.) I have Kamailio 5.4.1 (and RTPEngine) running on an internal server with a private IP address 172.31.7.96 and One-to-one NAT to an external IP address. The external IP is 192.0.2.100. (Note: The internal IP addresses are all unedited, but the public IPs have been replaced with TEST-NET-1 and TEST-NET-2 example addresses.) I will eventually be doing transcoding with RTPEngine, but for now this is a simple SIP Proxy. Kamailio is installed on Ubuntu 18.04 using the DEB packages from dev.kamailio.org/kamailio54 and is using the stock configuration that comes with those packages, except for the following changes: #!define WITH_NAT #!define WITH_RTPENGINE #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_IPAUTH listen=udp:0.0.0.0:5060 advertise 192.0.2.100:5060 #!define DBURL "mysql://kamailio:REAL_PASSWORD_HERE@127.0.0.1/kamailio" I have internal SIP servers with private IP addresses in the 172.31.7.0/24 range that I want to have send all SIP traffic through the Kamailio server. The internal servers are running a Java SIP client with the `OUTBOUND_PROXY` setting set to 172.31.7.96. The problem I have is that the SIP `200 OK` message sent by Kamailio to my SIP server has its `Record-Route` header set to the public IP address `192.0.2.100` instead of the private address `172.31.7.96`. The SIP client therefore tries to send the `ACK` message back to the public address, but it has no route to the public address so the ACK never gets sent. How can I configure Kamailio to use the public IP for external traffic but the private IP for communicating with internal machines on the same subnet? I tried setting `mhomed=1`, but the machine isn't actually multi-homed so that didn't work. I thought of adding a second listen line `listen=udp:172.31.7.96:5061` and having the internal servers talk to port 5061, but that doesn't work because Kamailio uses the 5061 definition for the external side too. I see in the docs that it is possible to name the listener lines, but I don't understand how to use those names in a way that would be relevant to my issue. Thank you very much for your help, Moshe

5 6

How to handle BYE in case of serial fork
by Björn Klasen 11 Sep '20

11 Sep '20

Hi we are using Kamailio 4.3.7 and I have a question concerning BYE handling. We have the following situation: A carrier a call to us. We are forwarding the call to the destination, but it does not pick up the call. We have call-forward active with use of tm module, so we forward the call to the new destination. The first call generates a to-tag in 180, let's call it TT1. The second call, so the c-destination also answer with 180 but with another to-tag (TT2). Both ringing are passed to the originating carrier. When the timer for the first call is timed out, Kamailio cancels the branch with a CANCEL so it does not exist any more Now the c-destination picks up the phone, and reply with 200 OK with TT2 The carrier then sends a BYE with TT1 to us, but Kamailio doesn't know anything about it, so it does not reply and the carrier sends a timeout and terminates the complete call. So the question is, how handle such a situation. Is it possible to store the first branch, or fake a OK reply to the BYE request of the carrier as for my understanding this behaviour is not against RFC. To clarify the situation here comes a call-flow diagram A Kamailio B C |---INV Bob@P1->| | | | |--INV Bob@B--->| | | |<-100----------| | |<-100----------| | | | |<-180 TT1------| | |<-180 TT1------| | | Forward after 10 seconds | |--CANCEL------>| | | |<-487----------| | | |<-200 OK-------| | | |--ACK--------->| | |<-181 TT1------| | | | |--INV Carol@C----------------->| | |<-100--------------------------| | |<-180 TT2----------------------| |<-180 TT2------| | | | |<-200 OK TT2-------------------| |<-200 OK TT2---| | | |--BYE TT1----->| | | |--BYE TT1----->| | | |--BYE TT1----->| | | |--408--------->| | | |--ACK--------->| | | | |--ACK------------------------->| | |<-BYE TT2----------------------| |<-BYE TT2------| | |--481--------->| | | |--481------------------------->| To get around the the situation with the second to-tag, we passing every call through SEMS now, because its B2B function rewrites the to-tag, so there is always only one to-tag towards the carrier A. But it's not the gold solution. I hope somebody can help me. BR, Björn -- Björn Klasen, Specialist TNG Stadtnetz GmbH, Network Management (VoIP) Projensdorfer Straße 324 24106 Kiel Germany T +49 431/ 530530 F +49 431/ 7097-555 mailto: bklasen(a)tng.de http://www.tng.de Register: Amtsgericht Kiel HRB 6002 KI Executive board (Geschäftsführung): Dr.-Ing. Volkmar Hausberg, Sven Schade, Carsten Tolkmit, Dr. Sven Willert Tax-Id (Steuernr.): 2029047020, VAT-Id (USt-Id): DE225201428

1 0

Kamailio v5.3.6 Released
by Daniel-Constantin Mierla 10 Sep '20

10 Sep '20

Hello, Kamailio SIP Server v5.3.6 stable release is out. This is a maintenance release of the latest stable branch, 5.3, that includes fixes since the release of v5.3.5. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.3.x. Deployments running previous v5.3.x versions are strongly recommended to be upgraded to v5.3.6. Note that 5.3 is the second last stable branch, still officially maintained by Kamailio development team. The latest stable branch is 5.4, with v5.4.1 being release out of it. For more details about version 5.3.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/09/kamailio-v5-3-6-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

1 0

Releasing v5.3.6
by Daniel-Constantin Mierla 10 Sep '20

10 Sep '20

Hello, I am considering to release v5.3.6 from the branch 5.3 in the next days, aiming from Thursday (Sep 10, 2020) if everything goes as planned at this moment, or on Monday (Sep 14, 2020) if a little bit of time is needed. Should you be aware of any issue that needs to be addressed in the branch 5.3 (e.g., a bug, or a missing backport), inform us as soon as possible to try to sort it out in time. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 1

Dispatcher / ds_select_routes
by Peter Gradwell 09 Sep '20

09 Sep '20

Hello I'm doing a project where I need to route using dispatcher across multiple groups. i.e. when the customer makes a call, the "core" server makes a request to cgrates to get back a list of carriers that support that, and then want to work through them in order of preference. e.g. the string we get back from cgrates via evapi might be "2,1,3". i.e. try all Group 2 destinations, then group 1, then group 3. Each of these is then a dispatcher group (with multiple dispatcher routes in each group). When the call then gets to the "Group 2" server that in turn is also using dispatcher to handle multiple destinations, and if all of those destinations fail, it returns a 503 to the "core", which is then meant to try "Group 1", and so on. The documentation on this is a bit sparse, but what I think I am meant to do (at the core level) is use ds_select_routes to load up the list of groups (https://kamailio.org/docs/modules/5.3.x/modules/dispatcher.html#dispatcher.…) if(ds_select_routes("2=4;1=4;3=4", "d", "0")) { t_on_failure("REROUTE"); t_relay(); exit; } and then in my failure_route, if the error code is 503 I call ds_next_domain, e.g. failure_route[REROUTE] { if(t_check_status("503")) { if(ds_next_domain()) { t_on_failure("REROUTE"); t_relay(); exit; } } } My problem is, that in applying theory to practice, I have ended up with a bunch of looping and it's not working right! Does anyone have an example code for such a scenario? many thanks peter

2 3

testing SIP address
by Nicholas Saunders 09 Sep '20

09 Sep '20

How do I test connectivity of a SIP address? Is there a "time" or "echo" type SIP address to call or message? I'm just using a free address from onsip.com thanks, Nick

4 3

how to reload kemi scripts? kamailio 5.3.x or 5.4.x
by Karsten Horsmann 09 Sep '20

09 Sep '20

Hello List, i read Daniels Presentation https://www.kamailio.org/events/2018-KamailioWorld/Day0/W07-Daniel-Constant… and there is an hint about python(2/3) - that kemi script reloading is not supported with python kemi scripts. Is that correct for the actual versions (5.3.x and 5.4.x)? And if i use for example lua as kemi script language, how i restart that script only? Drop in a new file and thats it (like FreeSWITCH did that)? Thanks in advance Cheers Karsten -- *Karsten Horsmann*

3 5

$HN(key) for IPv6
by Sergey Safarov 05 Sep '20

05 Sep '20

I want to deploy dual-stack kamailio and config file in need use in one place IPv4 kamailio IP address and in another place I want to use IPv6 address. Could I get something like $HN(i4) and $HN(i6)?

2 1

Change FROM to hide topology
by Andre Pires dos Santos 05 Sep '20

05 Sep '20

Hi guys, We have SIP Server > Kamailio > webrtc client. The only call flow we have is SIP server calling kamailio tha sends the call to webrtc client. We set up TOPOH to hide the sip server info but once the sip server initiates the call we still have the FROM details identifying SIP Server IP address. We couldn't find a way to mask this ip using TOPOH. What would be the best way to hide the sip server IP address in this case? Thanks a lot! Andre

3 2

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

03 Sep '20

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

2 2

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

03 Sep '20

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

2 1

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

03 Sep '20

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

2 1

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

02 Sep '20

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

4 16

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

02 Sep '20

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

1 0

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

02 Sep '20

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

1 0

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

02 Sep '20

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

3 5

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

02 Sep '20

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

3 2

worker processes
by Ali Taher 01 Sep '20

01 Sep '20

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

3 9

kamailio coring ver 5.1.2
by Andrew Chen 01 Sep '20

01 Sep '20

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

2 2

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

01 Sep '20

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

2 5

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

01 Sep '20

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

1 0

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

31 Aug '20

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

1 0

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

30 Aug '20

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

2 2

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

27 Aug '20

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

2 5

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

25 Aug '20

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

3 2

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

25 Aug '20

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

2 1

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

25 Aug '20

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

4 3

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

24 Aug '20

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

1 1

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

22 Aug '20

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

2 1

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

21 Aug '20

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

2 1

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

20 Aug '20

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

2 2

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

20 Aug '20

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

2 2

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

20 Aug '20

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

2 1

t_load_contacts in 5.4.0
by Cindy Leung 20 Aug '20

20 Aug '20

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

2 4

Domain Name changes
by Pavithra Mohanraja 19 Aug '20

19 Aug '20

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

2 4

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

19 Aug '20

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

2 4

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

19 Aug '20

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

2 1

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

18 Aug '20

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

4 3

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

18 Aug '20

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

9 10

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

18 Aug '20

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

2 1

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

18 Aug '20

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

2 2

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

16 Aug '20

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

2 1

Kamailio Query
by Amit Pal 15 Aug '20

15 Aug '20

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

1 0

Kamailio Team Attentions
by Amit Pal 15 Aug '20

15 Aug '20

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

1 0

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

15 Aug '20

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

1 0

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

14 Aug '20

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

2 2

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

14 Aug '20

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

4 5

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

14 Aug '20

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

2 1

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

14 Aug '20

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

3 3

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

13 Aug '20

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

1 0

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

13 Aug '20

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

1 0

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

13 Aug '20

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

4 14

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

13 Aug '20

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

1 0

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

13 Aug '20

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

3 6

Mysql lastinsertid
by Kjeld Flarup 12 Aug '20

12 Aug '20

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

4 6

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

12 Aug '20

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

3 5

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

12 Aug '20

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

1 0

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

12 Aug '20

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

3 5

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

12 Aug '20

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

1 0

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

12 Aug '20

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

3 2

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

12 Aug '20

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

3 7

Kamailio configuration ??
by Amit Pal 12 Aug '20

12 Aug '20

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

1 0

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

12 Aug '20

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

2 3

Change Record Route onreply
by Aidar Kamalov 11 Aug '20

11 Aug '20

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

2 2

dns_naptr_ignore_rfc default value
by Juha Heinanen 10 Aug '20

10 Aug '20

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

2 3

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

10 Aug '20

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

3 3

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

08 Aug '20

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

2 2

In memory ISP Info
by Ilie Soltanici 07 Aug '20

07 Aug '20

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

2 2

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

07 Aug '20

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

4 4

Deletion of Prometheus metrics
by Marat Gareev 07 Aug '20

07 Aug '20

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

2 4

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

07 Aug '20

Hi, I have kamailio connect to Teams, and works form Asterisk -> Teams calls. For Teams -> Asterisk calls I'd worked using extension and register Asterisk with that extension. But I'd like to use direct routing with IP. In kamailio.cfg I activate define WITH_PSTN. I configured the IP and PORT for my PSTN. I'm using the default route[PSTN]: route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined xlog("L_INFO","PSTN ACTIVADO"); if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")){ xlog("L_INFO", "Error en el formato numerico!!"); return; } # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + #if (starts_with("$rU", "00")) { # strip(2); # prefix("+"); #} if (strempty($sel(cfg_get.pstn.gw_port))) { #$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); xlog("L_INFO","SELECCION CON PUERTO"); $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } else { xlog("L_INFO","SELECCION CON PUERTO"); $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } And in my request_route: remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { if($src_ip != "IP ASTERISK"){ xlog("L_INFO", "***********ROUTE PSTN***********"); route(PSTN); } else { xlog("L_INFO","LLamada desde $si con puerto $sp"); record_route_preset("FQND:5061;transport=tls", "IP KAMAILIO:5060"); add_rr_param(";r2=on"); route(DISPATCH); route(RELAY); } } But never see that the call go to PSTN route.. I'd made any wrong?? Thanks

3 2

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

06 Aug '20

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

2 2

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

06 Aug '20

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

2 1

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

06 Aug '20

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

2 1

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

05 Aug '20

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

1 2

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

05 Aug '20

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

2 1

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

04 Aug '20

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

2 3

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

04 Aug '20

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

2 1

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

04 Aug '20

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

2 2

Kamailio Dockerization
by Edward Romanenco 04 Aug '20

04 Aug '20

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

4 7

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

04 Aug '20

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

2 3

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

04 Aug '20

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

4 3

search_append_body?
by Steve Wilkins 04 Aug '20

04 Aug '20

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

2 1

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

04 Aug '20

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

3 3

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

04 Aug '20

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

2 1

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

02 Aug '20

Hello, I'm setting up a Kamailio MSFT Teams SBC that is connected to our SIP trunk provider. This server is directly connected to the Internet and not behind a NAT routing. It is successfully processing invites from both sides (MSFT Teams and SIP trunk). Unfortunately the ACKs and CANCEL messages are not relaying to Teams. In the sipdump i can't find these messages. Does anyone see what I am doing wrong? Below you can find my config #!KAMAILIO ####### Defined Values ######### #!define MULTIDOMAIN 0 # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 #!define FROM_TEAMS 11 #!define FROM_PBX 12 ######## Define Modules ########### #!define WITH_RTPENGINE #!define WITH_TLS #!define WITH_SIPDUMP #!define WITH_DISPATCH ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR debug=2 log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes */ children=2 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes alias=SBC_FQDN /* listen addresses */ listen=udp:PUBLIC_IP:5060 advertise SBC_FQDN:5060 #!ifdef WITH_TLS listen=tls:PUBLIC_IP:5061 advertise SBC_FQDN:5061 #!endif server_header= "Server: ABC SBC" user_agent_header= "User-Agent: ABC SBC" ###### TLS Enable ###### #!ifdef WITH_TLS enable_tls=yes #tcp_connect_timeout=1000 tcp_accept_no_cl=yes tcp_async = yes tcp_connection_lifetime=600 /* upper limit for TLS connections */ tls_max_connections=2048 #!endif ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ ####### Modules Section ######## /* set paths to location of modules */ loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" #!ifdef WITH_RTPENGINE loadmodule "rtpengine.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_SIPDUMP loadmodule "sipdump.so" #!endif #!ifdef WITH_DISPATCH loadmodule "dispatcher.so" #!endif # ----------------- setting module-specific parameters --------------- #!ifdef WITH_RTPENGINE # ----- rtpengine params -----+ modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:12222") #!endif #!ifdef WITH_TLS # ----- tls params -----+ modparam("tls", "xavp_cfg", "tls") modparam("tls", "config", "/etc/kamailio/tls.cfg") modparam("tls", "connection_timeout", 10) modparam("tls", "ssl_release_buffers", 1) modparam("tls", "send_close_notify", 1) modparam("tls", "session_cache", 0) #!endif #!ifdef WITH_SIPDUMP modparam("sipdump", "enable", 1) #!endif #!ifdef WITH_DISPATCH #---------- dispatch modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_interval", 300) #!endif # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/run/kamailio/kamailio_ctl") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks route(REQINIT); #check who is the sender route(INITCHECK); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # update $du to set the destination address for proxying #$du = "sip:" + $rd + ":9"; route(RELAY); exit; } route[INITCHECK] { if(from_uri =~ ".*microsoft.com") { setflag(FROM_TEAMS); $du = "sip:" + "PBX_IP"; route(HANDLE_RTP_FROM_TEAMS); } else if(from_uri =~ ".*" + "PBX_IP") { setflag(FROM_PBX); $du="sip:sip.pstnhub.microsoft.com;transport=tls"; route(HANDLE_RTP_FROM_PBX); } else { exit; } } #Manage RTP & transcoding comming from Teams to PBX route[HANDLE_RTP_FROM_TEAMS] { if (has_body("application/sdp")) { t_on_reply("PBX_REPLY_TO_TEAMS"); rtpengine_manage("RTP codec-mask=all codec-transcode=PCMA replace-origin replace-session-connection ICE=remove"); record_route(); t_relay_to_udp("PBX_IP","5060"); } } #Manage RTP & transcoding comming from PBX to Teams route[HANDLE_RTP_FROM_PBX] { if (has_body("application/sdp")) { t_on_reply("TEAMS_REPLY_TO_PBX"); rtpengine_manage("SRTP codec-mask=all ICE=force codec-transcode=PCMA replace-origin replace-session-connection"); record_route_preset("SBC_FQDN:5061;transport=tls"); add_rr_param(";r2=on"); $rd = "sip.pstnhub.microsoft.com"; $td = "SBC_FQDN"; $fd = "SBC_FQDN"; #Set TLS SNI (server name & server id) $xavp(tls=>server_name) = "SBC_FQDN"; $xavp(tls=>server_id) = "SBC_FQDN"; t_relay(); } } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS")) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; #Teams reINVITEs if(isflagset(FROM_TEAMS)) { t_relay_to_udp("PBX_IP","5060"); exit; } # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails #set coresponding cert on transactions if($fd == "SBC_FQDN") { $xavp(tls=>server_name) = "SBC_FQDN"; $xavp(tls=>server_id) = "SBC_FQDN"; } } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); } #PBX On Reply onreply_route[PBX_REPLY_TO_TEAMS] { if (has_body("application/sdp")) { rtpengine_manage("SRTP codec-mask=all codec-transcode=PCMA replace-origin replace-session-connection media-address=PUBLIC_IP"); } } #From Teams On Reply onreply_route[TEAMS_REPLY_TO_PBX] { if (has_body("application/sdp")) { rtpengine_manage("RTP codec-mask=all codec-transcode=PCMA replace-origin replace-session-connection media-address=PUBLIC_IP"); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { if (t_is_canceled()) exit; } event_route[tm:local-request] { if(is_method("OPTIONS") && $ru =~ "pstnhub.microsoft.com") { append_hf("Contact: <sip:SBC_FQDN:5061;transport=tls>\r\n"); } xlog("L_INFO", "Sent out tm request: $mb\n"); }

6 5

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

02 Aug '20

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

2 1

lua_dostring sr vs KSR
by Julien Chavanton 31 Jul '20

31 Jul '20

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

2 4

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

31 Jul '20

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

3 11

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

30 Jul '20

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

2 2

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

29 Jul '20

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

1 0

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

29 Jul '20

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

1 2

PHONENUM Module on Centos7
by Duarte Rocha 28 Jul '20

28 Jul '20

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

2 1

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

28 Jul '20

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

1 0

PHONENUM Module on Centos7
by Ilie Soltanici 28 Jul '20

28 Jul '20

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

1 0

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

28 Jul '20

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

2 1

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

28 Jul '20

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

2 1

Registrar - registered match_received
by Ilie Soltanici 27 Jul '20

27 Jul '20

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

1 0

Registrar - registered match_received
by Ilie Soltanici 27 Jul '20

27 Jul '20

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

2 1

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

27 Jul '20

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

2 6

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

27 Jul '20

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

2 5

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

27 Jul '20

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

2 3

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

27 Jul '20

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

3 2

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

27 Jul '20

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

4 4

Kamailio Installation Steps
by Abdul Rasheed 26 Jul '20

26 Jul '20

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

2 2

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

24 Jul '20

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

1 2

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

24 Jul '20

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

4 9

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

24 Jul '20

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

2 2

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

23 Jul '20

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

5 15

TAS recommendation
by BALL SUN 23 Jul '20

23 Jul '20

Hi We recently setup Kamailio as the IMS core in our test lab, and we would like to test scenarios with TAS, is there any recommendation? Does Kamailio able to setup as a SIP AS? if yes, any online instruction on how we can make it works? Thanks RBK

1 0

Dispatcher does not go to 2nd server
by Aristeidis Tsitras 22 Jul '20

22 Jul '20

I have a Kamailio and 2 asterisk servers. All users are created in both of the asterisk servers. I am forwarding the registration to asterisk. The problem is that it is always used on only one server from the list. Even if one goes to shutdown, then there is not any registration sent to the available server. Even if *some *of the extensions can be seen registered in both of the asterisk's, if the secondary goes down, then there are no services for the phones. I am attaching the kamailio.cfg. My dispatch list is: 1 sip:192.168.0.100:5080 0 0 maxload=20 2 sip:192.168.0.101:5080 0 0 maxload=20 In both of the asterisk servers i am using sip.conf to create users and s sip trunk for the Kamailio. Nothing special about it. I am looking to find what is wrong with my config and i cannot loadbalance/failover to the asterisk servers.

6 12

Siremis - PassThru Authentication
by Adam - UC-Reseller 21 Jul '20

21 Jul '20

I have recently installed Siremis And have set up a domain pointing to it, how do I configure it so registration requests to this domain are passed thru to the actual PBX Adam

2 1

P-Asserted-Identity in SIP INVITE from Kamailio IMS to PSTN (MGCF)
by BALL SUN 21 Jul '20

21 Jul '20

Hi We have a question. We are preparing a setup to bridge the SIP call from Kamailio IMS to PSTN, but we are puzzling how we can present the P-Asserted-Identity in the SIP INVITE. By default, the PAI is in sip:user@domain, but we believed we should use<tel> format in the PAI when bridge to PSTN. question is 1. should we have two PAI headers?, or 2. should we paste <tel> format in PAI User Part and domain in PAI Host Part?, or 3. Should we only use one PAI header with tel format? In our test setup, we are using FHoSS, and the IMPU sip:user@domain have below entry in "IMPUs from Implicit-Set" - sip:user@domain - tel:number Question is how we can extract the number from the HSS and presented in PAI? - RBK

1 1

Kamailio load balance to PBXes
by John Tuxies 20 Jul '20

20 Jul '20

Trying to setup a lab where i would like tohave: -kamailio (5.3 from deb repo) in front to load balance. Keep registrations. IP: 192.168.1.100 -Asterisk _1 (16 LTS). IP: 192.169.1.101.It <http://192.169.1.232.It> will serve as media server for IVR, VM, etc. Bind port is 5080. A sip trunk to Kamailio in sip.conf -Asterisk-2 (16 LTS). IP: 192.168.1.102. Identical config to Asterisk-1. Tried to follow this https://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb, but it is quite old and it does jot apply to newer versions of Asterisk and Kamailio. Up to jow i have the Kamailio setup and users can register on Kamailio following the basic setup. I would like advice on setup and configuration, to integrate Kamailio with Asterisk boxes Some advice please? Sincerely yours, John

2 2

Kamailio load balance to PBXes
by John Tuxies 20 Jul '20

20 Jul '20

Trying to setup a lab where i would have: -kamailio in front to load balance. Keep registrations. IP: 192.168.1.100 -Asterisk _1 (16 LTS). IP: 192.169.1.101.It <http://192.169.1.232.It> will serve as media server for IVR, VM, etc. Bind port is 5080. A sip trunk to Kamailio in sip.conf -Asterisk-2 (16. IP: 192.168.1.102. Identical config to Asterisk-1. Tried to follow this https://kb.asipto.com/asterisk:realtime:kamailio-3. 0.x-asterisk-1.6.2-astdb, but it is quite old and it does jot apply to newer versions of Asterisk and Kamailio. Up to jow i have the Kamailio setup and users can register on Kamailio following the basic setup. I would like advice on setup and configuration, to integrate Kamailio with Asterisk boxes

1 0

Kamailio load balance to PBXes
by John Tuxies 20 Jul '20

20 Jul '20

Trying to setup a lab where i would have: -kamailio in front to load balance. Keep registrations. IP: 192.168.1.218 -Asterisk _1. IP: 192.169.1.232.It will serve as media server for IVR, VM, etc -Asterisk-2. IP: 192.168.1.233. Identical config to Asterisk-1. Tried to follow this https://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb, but it is quite old and it does jot apply to newer versions of Asterisk and Kamailio. Up to jow i have the I would like advice on setup and configuration,

1 0

Kamailio & Freeswitch as media and SBC
by John Tuxies 20 Jul '20

20 Jul '20

Trying to setup Kamailio with Freeswitch in a Debian machine as in the http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc, but Kamailio fails to start and gives a ton of errors. Before of the alterations Kamailio and Siremis start correctly and i can easily create users through Siremis. After the changes to include Freeswitch's blocks, it fails to start. All run in the same machine with IP 192.168.1.213. I am attaching kamailio.cfg (named as test1.txt) if someone could spot the error, please. Also i am attaching the kamailio.cfg(Original1.txt) when it was working fine with Siremis.

3 8

SIP Security handling in Kamailio IMS
by BALL SUN 20 Jul '20

20 Jul '20

Hi During the testing with Kamailio IMS, we found that SUBSCRIBE and NOTIFY message flow between UE and P-CSCF is shown below. SUBSCRIBE and its response: 192.168.1.102:9101 (port-c) -> 192.168.2.66:6101 (port-s) NOTIFY: 192.168.2.66:5101 (port-c) -> 192.168.1.102:9100 (port-s) based on the sip security, is it the expected behavior that they are using port-c and port-s? The reason why we asked is because we found that the NOTIFY did not properly propagate to the UE due to different port addresses. Does anyone have this experience? and how we can resolve it? - RBK

2 2

the list with sip compact header names
by Daniel-Constantin Mierla 17 Jul '20

17 Jul '20

Hello, anyone knowing by heart or a place listing all compact header names? I found: - https://www.cs.columbia.edu/sip/compact.html but it says: Last updated 01/10/2008 Maybe newer RFC introduced other compact header names. Or, in case you are aware of some other compact names not listed there, just say it here, as I am working a bit on updating the parser on this scope. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

2 2

CommCon 2020 Online - Now presentation about Kamailio
by Daniel-Constantin Mierla 16 Jul '20

16 Jul '20

Hello, being a bit late with this announcement, as it started a couple of minutes ago, but if anyone wants to watch the live presentation about Kamailio by Fred Posner at CommCon 2020 Online, here is the link: - https://www.youtube.com/watch?v=FGZF7dYdRng Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 0

Reset Prometheus counter once
by Marat Gareev 16 Jul '20

16 Jul '20

Hello. Are there any ways to reset the Prometheus counter once? I can do something like this, but counter will be reset on each request loadmodule "xhttp_prom.so" > modparam("xhttp_prom", "prom_counter", "name=mycounter;") > route { > prom_counter_reset("mycounter"); > } Otherwise, I will not receive the counter until a special and rare event occurs. Regards, Marat

3 4

RPC Socket stops answering after a few time.
by David VILLAUME 16 Jul '20

16 Jul '20

Hello, I’m having problem with RPC socket that stops answering after approximatively one or two hours. Version is 5.2.7 My principal host with exact same version and configuration has no problem, only the backup node (Keepalived + DMQ) encounters the problem. I could catch this log about the problem, but didn’t find any reason for this : Jun 18 10:53:33 sbc-kam-partner-b kamailio[83170]: ERROR: ctl [binrpc_run.c:600]: rpc_send(): ERROR: binrprc: rpc_send: send failed My configuration about rpc : loadmodule "jsonrpcs.so" modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") modparam("ctl", "binrpc_max_body_size", 1024) Do you have any idea to troubleshoot this ? Regards, David

2 13

Routing specific number from kamailio to asterisk
by Hendra 834 16 Jul '20

16 Jul '20

Hi, Im new in VoIP, especially Kamailio. I want to connect my kamailio with asterisk by routing the call coming from kamailio to asterisk for specific number, also from asterisk to kamailio. For example, if kamailio client call ext number 0123, the will be routed to asterisk, if kamailio client call 0233 the call will not be routed to external proxy. And also how can I receive call form asterisk to kamailio ? Is there any real example, ? because I dont understand what to change in documentation example. Thanks

2 1

Is it possible to load both dialog and ims_dialog modules in the same configuration?
by Sumesh Soman 16 Jul '20

16 Jul '20

Hi, Is it possible to load and use both dialog and ims_dialog modules in the same server ? Even though ims_dialog docs says ims_dialog extends dialog module, I do see that some of the dialog module features are not present in ims_dialog. Example, wait_ack , track_cseq_updates are not present in ims_dialog. I have requirements to use ims_dialog mainly for ims_charging module and dialog module with wait_ack and track_cseq_updates parameters are enabled. -- Regards, Sumesh S.

1 0

5.4 compiler warning
by Juha Heinanen 14 Jul '20

14 Jul '20

Got this when building 5.4; CC (gcc) [sip-proxy] core/dset.o core/dset.c: In function 'uri_trim_rcv_alias': core/dset.c:1042:6: warning: variable 'proto' set but not used [-Wunused-but-set-variable] int proto; ^~~~~ -- Juha

2 2

Re: [SR-Users] xHTTP_PROM with older Kamailio
by Marat Gareev 13 Jul '20

13 Jul '20

Hello, Karsten, Thanks for the link. I will definitely examine it.

1 0

No of registartions
by Pavithra Mohanraja 13 Jul '20

13 Jul '20

Hi, Is there any way to query number of registrations happened in kamailio IMS. Kindly reply . Thanks, Pavithra

2 1

subsequent ds_select_dst with different groups
by David VILLAUME 13 Jul '20

13 Jul '20

Hello, Is it possible to use the following logic : Route() ds_select_dst (1,2) failure_route(1) ds_next_dst(first_dispatcher_ secondgateway) failure_route(newdispatcher_group) ds_select_dst (2,2) failure_route(secondispatcher_secondgateway) ds_next_dst() Thanks in advance. Regards, David

4 5

Compact SIP
by Abdoul Osséni 13 Jul '20

13 Jul '20

Hello, I use Kamailio and Asterisk. SIP Compact header is activated by ASTERISK ( https://www.cs.columbia.edu/sip/compact.html). I am trying to disable the SIP Compact header on KAMAILIO. Is it possible? Regards Abdoul OSSENI

3 4

Re: [SR-Users] xHTTP_PROM with older Kamailio
by Marat Gareev 12 Jul '20

12 Jul '20

I'm sorry. I mean *xhttp_prom.c* file instead of *xhttp_mod.c*. >

1 0

xHTTP_PROM with older Kamailio
by Marat Gareev 11 Jul '20

11 Jul '20

Hello, I have a customized Kamailio (kamailio 5.1.1 (x86_64/linux)), xHTTP_PROM <https://www.kamailio.org/docs/modules/devel/modules/xhttp_prom.html> was released in the 5.3.0 <https://www.kamailio.org/w/kamailio-v5-3-0-release-notes/> version. I'm trying to import the latest version of xhttp_prom to older Kamailio, but something goes wrong. This is simple config for test: log_stderror=yes > listen=5060 > debug=2 > mpath="/home/devel/build_dir/build/lib64/kamailio/modules" > loadmodule "tm.so" > loadmodule "sl.so" > loadmodule "xlog.so" > loadmodule "pv.so" > disable_tcp=no > tcp_accept_no_cl=yes > loadmodule "xhttp.so" > loadmodule "xhttp_prom.so" > modparam("xhttp_prom", "xhttp_prom_stats", "all") > route { > sl_send_reply("200", "OK"); > exit; > } > event_route[xhttp:request] { > if (prom_check_uri()) { > prom_dispatch(); > } else { > xhttp_reply("404", "Not Found", "text/plain", "Wrong URL $hu\n"); > } > } then I send HTTP request $ curl http://localhost:5060/metrics but get error: > 6(110968) CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 11 > 0(110962) ALERT: <core> [main.c:746]: handle_sigs(): child process 110967 > exited by a signal 11 > 0(110962) ALERT: <core> [main.c:749]: handle_sigs(): core was generated > 0(110962) INFO: <core> [main.c:771]: handle_sigs(): terminating due to > SIGCHLD note: for http://localhost:5060/test request I get the expected message "Wrong URL /test". Backtrace for killed process > (gdb) c > Continuing. > Program received signal SIGSEGV, Segmentation fault. > 0x00007fefff3ff096 in atomic_cmpxchg_int (var=0x0, old=0, new_v=1) at > ../../core/mem/../atomic/atomic_x86.h:224 > 224 ATOMIC_FUNC_CMPXCHG(cmpxchg, "cmpxchgl %2, %1", int , int) > (gdb) bt > #0 0x00007fefff3ff096 in atomic_cmpxchg_int (var=0x0, old=0, new_v=1) at > ../../core/mem/../atomic/atomic_x86.h:224 > #1 0x00007fefff3ff0e2 in futex_get (lock=0x0) at > ../../core/mem/../futexlock.h:99 > #2 0x00007fefff4130bf in prom_metric_list_print (ctx=0x7fefff628da0 > <ctx>) at prom_metric.c:1319 > #3 0x00007fefff416468 in prom_stats_get (ctx=0x7fefff628da0 <ctx>, > stat=0x7fefff6281f0 <xhttp_prom_stats>) at prom.c:178 > #4 0x00007fefff3d4d10 in prom_send (ctx=0x7fefff628da0 <ctx>) at > xhttp_prom.c:361 > #5 0x00007fefff3d5faf in ki_xhttp_prom_dispatch (msg=0x7ffecb730ee0) at > xhttp_prom.c:418 > #6 0x00007fefff3d5feb in w_prom_dispatch (msg=0x7ffecb730ee0) at > xhttp_prom.c:428 > #7 0x0000000000461dbe in do_action (h=0x7ffecb730e00, a=0x7ff0007c7b78, > msg=0x7ffecb730ee0) at core/action.c:1067 > #8 0x000000000046e571 in run_actions (h=0x7ffecb730e00, a=0x7ff0007c7b78, > msg=0x7ffecb730ee0) at core/action.c:1565 > #9 0x0000000000461d2d in do_action (h=0x7ffecb730e00, a=0x7ff0007ca628, > msg=0x7ffecb730ee0) at core/action.c:1058 > #10 0x000000000046e571 in run_actions (h=0x7ffecb730e00, a=0x7ff0007c76f8, > msg=0x7ffecb730ee0) at core/action.c:1565 > #11 0x00007fefff62f649 in xhttp_process_request (orig_msg=0x7ff0007ebbd8, > new_buf=0x7ff0007cc280 "GET /metrics HTTP/1.1\r\nVia: SIP/2.0/TCP > 127.0.0.1:37000\r\nUser-Agent: curl/7.29.0\r\nHost: > localhost:5060\r\nAccept: */*\r\n\r\n", new_len=119) at xhttp_mod.c:298 > #12 0x00007fefff630d78 in xhttp_handler (msg=0x7ff0007ebbd8) at > xhttp_mod.c:385 > #13 0x0000000000520141 in nonsip_msg_run_hooks (msg=0x7ff0007ebbd8) at > core/nonsip_hooks.c:111 > #14 0x000000000058cef1 in receive_msg (buf=0x7feffee65bb0 "GET /metrics > HTTP/1.1\r\nUser-Agent: curl/7.29.0\r\nHost: localhost:5060\r\nAccept: > */*\r\n\r\n", len=85, rcv_info=0x7feffee658d0) > at core/receive.c:216 > #15 0x000000000061233f in receive_tcp_msg (tcpbuf=0x7feffee65bb0 "GET > /metrics HTTP/1.1\r\nUser-Agent: curl/7.29.0\r\nHost: > localhost:5060\r\nAccept: */*\r\n\r\n", len=85, rcv_info=0x7feffee658d0, > con=0x7feffee658b8) at core/tcp_read.c:1379 > #16 0x000000000061479e in tcp_read_req (con=0x7feffee658b8, > bytes_read=0x7ffecb731c7c, read_flags=0x7ffecb731c78) at > core/tcp_read.c:1611 > #17 0x00000000006174e9 in handle_io (fm=0x7ff0007d9720, events=1, idx=-1) > at core/tcp_read.c:1785 > #18 0x00000000006069b4 in io_wait_loop_epoll (h=0xaa76c0 <io_w>, t=2, > repeat=0) at core/io_wait.h:1065 > #19 0x00000000006194ff in tcp_receive_loop (unix_sock=14) at > core/tcp_read.c:1955 > #20 0x00000000004f1a43 in tcp_init_children () at core/tcp_main.c:4853 > #21 0x0000000000423f99 in main_loop () at main.c:1716 > #22 0x000000000042a5ee in main (argc=9, argv=0x7ffecb7323b8) at main.c:2646 > (gdb) > (gdb) p 'prom_metric.c'::prom_lock > $4 = (gen_lock_t *) 0x0 If I understood correctly, the problem is in an uninitialized variable static gen_lock_t *prom_lock = NULL; /* Lock to protect Prometheus metrics. */ So, is it possible to import xHTTP_PROM to older Kamailio? I will be glad for any advice. Thanks.

1 0

loose_route failing with topos ?
by Mark Boyce 10 Jul '20

10 Jul '20

Hi all Just enabled topos with redis backend (topos/topos_redis/ndb_redis) The interesting bits of the calls are going from kamailio (5.3.5) to Asterisk with topos enabled for that leg All seems to be ok until the far end (Asterisk) sends a BYE. At which point kamailo passes message to WITHINDLG, in loose_route comes up false (-1) and the BYE gets missed, falling through to the “404 Not here” at the end of WITHINDLG. I’m guessing that loose_route should be able to “see inside” topos, or should I be throwing loose_route away when using topos? Happy to upgrade to 5.4 and do diags if it appears to be a bug. Best regards Mark -- Mark Boyce Dark Origins Ltd

3 4

rtcp-mux is missing in the OK for WebRTC-Calls
by Benjamin Flügel | vio:networks 10 Jul '20

10 Jul '20

Hey guys, I'm trying to configure a Kamailio to work with a browser softphone based on SIPJS using WebRTC. So far it works great on Firefox but have a specific problem with chrome, when I want to make call from the softphone to another extension. After anwsering the call Chrome/the softphone sends a BYE immediately, because this line "a=rtcp-mux" is missing in the OK. The Kamailio is a proxy. Behind the Kamailio there is an Asterisk, which is responsible for the pbx-features. Those are my rtpengine Flags for the Invite: rtpengine_manage: replace-origin replace-session-connection trust-address via-branch=extra rtcp-mux-demux DTLS=off SDES-on ICE=remove RTP/AVP And those are the flags for the response, in this case the OK: rtpengine_manage: replace-origin replace-session-connection rtcp-mux-offer rtcp-mux-accept generate-mid DTLS=off SDES-on ICE=force RTP/SAVPF direction=internal direction=external loop-protect It seems that the Kamailio ignores ths "rtcp-mux-offer rtcp-mux-accept" in the response. Can you help me get it to work? Here is the SIP-Dialog. The call is from extension 201 to the extension 2. INVITE: INVITE sip:2@mydomain SIP/2.0 Via: SIP/2.0/WSS g4n0lpfirgpv.invalid;branch=z9hG4bK5104892 To: <sip:2@mydomain.com> From: <sip:201@mydomain.com>;tag=aqplr71k05 CSeq: 2 INVITE Call-ID: sde09v49f8gqtt59oddn Max-Forwards: 70 Proxy-Authorization: Digest algorithm=MD5, username="201", realm="mydomain.com", nonce="sfdsdfdsfsdfdsfsdfaseqww", uri="sip:2@mydomain.com", response="81rewtrega23423r" Contact: <sip:kf6iduon@g4n0lpfirgpv.invalid;transport=ws;ob> Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound User-Agent: SIPJS Content-Type: application/sdp Content-Length: 1916 v=0 o=- 5826889093965459811 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE 0 a=msid-semantic: WMS Jk5GQwzPaVSTxyIZER7RBqkkMNWCdmKmMdyO m=audio 54274 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126 c=IN IP4 mycomputerIP a=rtcp:9 IN IP4 0.0.0.0 a=candidate:1577908739 1 udp 2113937151 efd6d297-d186-4c07-87d8-933e5846a82d.local 54274 typ host generation 0 network-cost 999 a=candidate:842163049 1 udp 1677729535 mycomputerIP 54274 typ srflx raddr 0.0.0.0 rport 0 generation 0 network-cost 999 a=ice-ufrag:Ktxl a=ice-pwd:x5xDUb41GeOXAcHNQlra4yUN a=ice-options:trickle a=fingerprint:sha-256 2C:7C:32:AF:34:A3:9D:AE:C7:FD:92:68:DD:D8:AB:82:DB:F0:32:51:14:97:20:60:66:5C:2F:CF:B7:98:B8:A8 a=setup:actpass a=mid:0 a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id a=sendrecv a=msid:Jk5GQwzPaVSTxyIZER7RBqkkMNWCdmKmMdyO dc187be3-4cd0-4129-98e4-f804cd8d2c94 a=rtcp-mux a=rtpmap:111 opus/48000/2 a=rtcp-fb:111 transport-cc a=fmtp:111 minptime=10;useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:110 telephone-event/48000 a=rtpmap:112 telephone-event/32000 a=rtpmap:113 telephone-event/16000 a=rtpmap:126 telephone-event/8000 a=ssrc:2376769177 cname:RgPkYyxDrEJTpmO2 a=ssrc:2376769177 msid:Jk5GQwzPaVSTxyIZER7RBqkkMNWCdmKmMdyO dc187be3-4cd0-4129-98e4-f804cd8d2c94 a=ssrc:2376769177 mslabel:Jk5GQwzPaVSTxyIZER7RBqkkMNWCdmKmMdyO a=ssrc:2376769177 label:dc187be3-4cd0-4129-98e4-f804cd8d2c94 RINGING: SIP/2.0 180 Ringing Via: SIP/2.0/WSS g4n0lpfirgpv.invalid;rport=60196;received=mycomputerIP;branch=z9hG4bK5104892 Record-Route: <sip:internalIP;r2=on;lr=on;ftag=aqplr71k05;nat=yes;rtp=ws> Record-Route: <sip:externalIP;transport=ws;r2=on;lr=on;ftag=aqplr71k05;nat=yes;rtp=ws> From: <sip:201@mydomain.com>;tag=aqplr71k05 To: <sip:2@mydomain.com>;tag=as64fe0fc8 Call-ID: sde09v49f8gqtt59oddn CSeq: 2 INVITE Server: myserver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <mycontact> P-Asserted-Identity: "Phone 2" <sip:2@mydomain.com> Content-Length: 0 OK: SIP/2.0 200 OK Via: SIP/2.0/WSS g4n0lpfirgpv.invalid;rport=60196;received=mycomputerIP;branch=z9hG4bK5104892 Record-Route: <sip:internal-ip;r2=on;lr=on;ftag=aqplr71k05;nat=yes;rtp=ws> Record-Route: <sip:external-ip;transport=ws;r2=on;lr=on;ftag=aqplr71k05;nat=yes;rtp=ws> From: <sip:201@mydomain.com>;tag=aqplr71k05 To: <sip:2@mydomain.com>;tag=as64fe0fc8 Call-ID: sde09v49f8gqtt59oddn CSeq: 2 INVITE Server: myserver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <mycontact> P-Asserted-Identity: "Phone 2" <sip:2@mydomain.com> Content-Type: application/sdp Content-Length: 806 v=0 o=root 882840402 882840402 IN IP4 externalIP s=Asterisk PBX 11.22.0 c=IN IP4 externalIP t=0 0 a=rtpengine:4d633e3022f5 m=audio 16416 RTP/SAVPF 111 9 8 0 126 a=maxptime:60 a=mid:0 a=rtpmap:111 opus/48000/2 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:126 telephone-event/8000 a=fmtp:111 maxplaybackrate=16000; stereo=0; sprop-stereo=0; useinbandfec=0 a=fmtp:126 0-16 a=sendrecv a=rtcp:16417 a=setup:active a=fingerprint:sha-1 B1:67:4B:B8:47:89:E8:49:CD:DD:F8:FF:41:5C:83:72:D9:DE:4D:45 a=ptime:20 a=ice-ufrag:WBZN8m7c a=ice-pwd:mKVR6Jv0G0pvrTv7OfEm2wPW9Q a=ice-options:trickle a=candidate:3kddB3zBUV2n2jt0 1 UDP 2130706431 externalIP 16416 typ host a=candidate:3kddB3zBUV2n2jt0 2 UDP 2130706430 externalIP 16417 typ host a=end-of-candidates I hope, you can help me with this. If you have further question, I will try to answer them as best I can. Greetings, Ben

2 1

Development is open for future release of v5.5.x series
by Daniel-Constantin Mierla 10 Jul '20

10 Jul '20

Hello, the branch 5.4 was created, therefore the master branch is open for adding new features, to be part of future release series v5.5.x. Any bug fix committed to master that applies to 5.4.x or older stable branches should be backported as usual with "git cherry-pick -x ..." to appropriate branches like 5.4 or 5.3. Expect that v5.4.0 will be released in a few weeks from now. Based on the workflow used during the past years, the next future release v5.5.0 should be out after another 8-10 months of development, plus 1-2 months of testing, so sometime in the first part of 2021. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Branch 5.4 has been created
by Daniel-Constantin Mierla 10 Jul '20

10 Jul '20

Hello, the branch 5.4 has been created, to be used for releasing v5.4.x series. To check out this branch, the following commands can be used: git clone https://github.com/kamailio/kamailio kamailio-5.4 cd kamailio-5.34 git checkout -b 5.4 origin/5.4 Pushing commits in this branch: git push origin 5.4:5.4 Note that 5.4 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 5.4 branch: git cherry-pick -x COMMITID In few weeks, the first release from branch 5.4 will be out, respectively Kamailio v5.4.0. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Drop local generated requests
by Patrick Wakano 10 Jul '20

10 Jul '20

Hello list, Hope you all doing fine! I have a case where I want to drop a local generated request but it looks like you can't drop the request from the tm:local-request route. I found this ticket https://github.com/kamailio/kamailio/issues/980, but I could not find somewhere in the docs or emails definitely saying that drop() is not supported from tm:local-request.... And in case we definitely can't drop the request in there, any ideas on how to avoid a local generated request of going out? I am specifically interested in dropping Registers from the uacreg module under certain situations, but I don't want to disable it or delete it from the DB.... Thank you, Kind regards, Patrick Wakano

2 4

Help on dumping running config in Kamailio
by BALL SUN 09 Jul '20

09 Jul '20

Hi I need an urgent help, I accidentally pipe the output to the kamailio config, and now I lost the copy of my latest setting, is there a chance that I can dump the running config back to file? Thanks PLEASE HELP RBK

2 1

Branching 5.4 in git repository
by Daniel-Constantin Mierla 08 Jul '20

08 Jul '20

Hello, so far it looks pretty calm during the testing phase, some static analysis didn't reveal any new major issues (well, this is done also during development phase). Therefore, if nothing else is proposed or shows up, I am considering to create the git branch 5.4 next week on Thursday -- this branch will hold the code for 5.4.x release series. After branching 5.4, the master branch will be open again for new features. The 5.4.0 enters the release candidate phase, likely to have the first release out in 1-2 weeks after branching. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 1

Maintenance work on kamailio.org server - Wed, July 8, 2020
by Daniel-Constantin Mierla 08 Jul '20

08 Jul '20

Hello, on Wednesday, July 8, 2020, there will be some work on Kamailio project infrastructure. Therefore there could be short intervals of downtime for online resources like the web server, wiki, documentation and downloads portals, mailing lists or the GIT repository mirror. Once the maintenance work is finished, we will post an update. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 2

30X redirect in interconnects - better alternative?
by Gerry | Rigatta.com 08 Jul '20

08 Jul '20

Hi, I would like to use Kamailio for load balancing incoming carrier traffic. We do currently IP authentication and call logic in Yate boxes. Ideally I would like to distribute calls with 30X redirects with the Kamailio dispatcher so that IP authentication and all logic can stay in the Yate boxes. However I have doubts that 30X redirects are generally accepted in interconnects. What is your experience with this? What is the possible alternative to redirects if one wants to keep IP authentication and call logic in the boxes behind the Kamailio SIP router? E.g. how can one reliably check the carrier source IPs behind Kamailio? Custom headers injected by Kamailio? Of cause I can check source IPs with a database lookup in Kamailio but I try to avoid that as this makes the setup much more complicated and error prone. Thank you for your ideas. Gerry

3 2

Re: [SR-Users] sanity module checks fail for ACK with parameters in RURI - possible bug?
by George Diamantopoulos 08 Jul '20

08 Jul '20

Update: Disabling the topoh module on the proxy which produces the error seems to stop the failure from manifesting. I'll try using topos_redis instead, but should this be treated as a bug? BR, George On Wed, 8 Jul 2020 at 12:37, George Diamantopoulos <georgediam(a)gmail.com> wrote: > Hello again, > > Indeed $mb seems to contain garbage: > > SCRIPT_MB: ACK <BC><EA><8F> SIP/2.0#015#012Via: SIP/2.0/UDP > 172.30.154.189;TH=dcv;branch=z9hG4bK629b.6af9302cd78dc58dffe817e60124f4ed.0#015#012Route: > <sip:RJ2U2c7mrFzQjgG5SSvyE8RVS9omgAA=@172.30.155.1 > ;lr;received=sip:2.2.2.2:32768 > ;ob;r2=on>,<sip:RJ2U2c7mrFzQjgG5SSvyE8RVS9omgAA=@3.3.3.3 > ;lr;received=sip:2.2.2.2:32768;ob;r2=on>#015#012Max-Forwards: > 68#015#012From: "Anonymous" <sip:unknown@voip.domain.com>;tag=as4bc9e324#015#012To: > <sip:voip-test-user-02@voip.domain.com>;tag=jw7z5s0zvc#015#012Call-ID: > 0138c6370346d1dd7f1b47f604b01f92(a)voip.domain.com#015#012CSeq > <http://0138c6370346d1dd7f1b47f604b01f92@voip.domain.com#015%23012CSeq>: > 102 ACK#015#012Content-Length: 0#015#012TH: dch#015#012#015#012 > > How can this be possible? Capturing traffic on wire shows the RURI I > pasted in my original message and there are no script operations on the > RURI before sanity_check() (message buffer above is printed just before > sanity_check() is run in REQINIT). > > BR, > George > > On Wed, 8 Jul 2020 at 11:18, George Diamantopoulos <georgediam(a)gmail.com> > wrote: > >> I'll post the message buffer ASAP, but in the meantime I don't see how >> config operations could affect the RURI. Here's everything that's happening >> until the sanity check involved: >> >> request_route { >> >> if(is_method("KDMQ")) { >> dmq_handle_message(); >> } >> >> # no connect for sending replies >> set_reply_no_connect(); >> >> if($ua =~ "friendly-scanner|sipcli|sipvicious|VaxSIPUserAgent") { >> # silent drop for scanners - uncomment next line if want to reply >> # sl_send_reply("200", "OK"); >> exit; >> } >> >> if (!mf_process_maxfwd_header("10")) { >> force_rport(); >> sl_send_reply("483","Too Many Hops"); >> exit; >> } >> >> # OPTIONS and NOTIFYs directed to myself >> if(is_method("OPTIONS|NOTIFY") && uri==myself && $rU==$null) { >> force_rport(); >> sl_send_reply("200","Keepalive"); >> exit; >> } >> >> # All keep-alive methods regardless of destination >> if ( $hdr(Event) == "keep-alive") { >> force_rport(); >> sl_send_reply("200","Keepalive"); >> exit; >> } >> >> if(!sanity_check("17895", "7")) { >> xlog("Malformed SIP request from $si:$sp\n"); >> exit; >> } >> >> BR, >> George >> >> On Wed, 8 Jul 2020 at 10:58, Daniel-Constantin Mierla <miconda(a)gmail.com> >> wrote: >> >>> Hello, >>> >>> check your config operations, because the R-URI seems to be the next >>> string (without quotes): "<BC><EA><8F>" >>> >>> You can try to print $mb in such case to see the entire SIP message >>> buffer. >>> >>> Cheers, >>> Daniel >>> On 08.07.20 09:48, George Diamantopoulos wrote: >>> >>> Hello Daniel, >>> >>> Thanks for the reply. Indeed there is, not sure how I managed to miss >>> that. And it wasn't about the schema after all: >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: DEBUG: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>> [sanity.c:277]: check_ruri_scheme(): check_ruri_scheme entered >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: DEBUG: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - <core> >>> [core/parser/parse_uri.c:1254]: parse_uri(): uri too short: <<BC><EA><8F>> >>> (3) >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: DEBUG: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - <core> >>> [core/parser/parse_uri.c:1328]: parse_sip_msg_uri(): bad uri <<BC><EA><8F>> >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: WARNING: >>> {1 <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>> [sanity.c:282]: check_ruri_scheme(): failed to parse request uri >>> [<BC><EA><8F>] >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: DEBUG: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>> [sanity_mod.c:254]: w_sanity_check(): sanity checks result: 0 >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: ERROR: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - <script>: Malformed >>> SIP request from 172.30.154.189:5060 >>> >>> Still, not sure what the problem is though... >>> >>> BR, >>> George >>> >>> On Wed, 8 Jul 2020 at 09:30, Daniel-Constantin Mierla <miconda(a)gmail.com> >>> wrote: >>> >>>> Hello, >>>> >>>> when the ruri scheme check fails, there should be another debug message >>>> saying that. Have you pasted all log messages for the failure case? >>>> >>>> Cheers, >>>> Daniel >>>> On 07.07.20 22:23, George Diamantopoulos wrote: >>>> >>>> Sorry, I realised I copy pasted wrong log messages for Call 1. Here's >>>> the relevant part showing success for call 1 in contrast with Call 2: >>>> >>>> grep 2a859fcc4e1c8f840191a81d7c16e76d kamailio.log | egrep >>>> 'check_ruri_scheme|w_sanity_check' | grep ACK >>>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[907]: DEBUG: >>>> {1 <null> 172.30.154.189 102 ACK >>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>> [sanity.c:277]: check_ruri_scheme(): check_ruri_scheme entered >>>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[907]: DEBUG: >>>> {1 <null> 172.30.154.189 102 ACK >>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>> [sanity.c:297]: check_ruri_scheme(): check_ruri_scheme passed >>>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[907]: DEBUG: >>>> {1 <null> 172.30.154.189 102 ACK >>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>> [sanity_mod.c:254]: w_sanity_check(): sanity checks result: 1 >>>> >>>> On Tue, 7 Jul 2020 at 21:34, George Diamantopoulos < >>>> georgediam(a)gmail.com> wrote: >>>> >>>>> Hello all, >>>>> >>>>> I'm not 100% sure this is the only culprit in an issue I'm >>>>> investigating, but superficially it appears that RURI scheme sanity module >>>>> checks from the default config (flags 17895 in REQINIT) fail if the >>>>> RURI in an ACK following a 487 includes parameters. Example from two calls >>>>> from a kamailio instance acting as registrar/usrloc server, INVITE RURIs >>>>> are after usrloc lookup: >>>>> >>>>> Call 1: INVITE sip:voip-test-gd@172.17.173.14:5063 SIP/2.0 >>>>> Call 2: INVITE sip:voip-test-user-02@10.2.24.142:32768;line=moo62e08 >>>>> SIP/2.0 >>>>> >>>>> These INVITEs produce no complaints. Later, the same registrar >>>>> produces ACKs to acknowledge 487 (thus, same transaction ACKs) responses >>>>> from the next proxy in the path following a CANCEL: >>>>> >>>>> Call 1: ACK sip:voip-test-gd@172.17.173.14:5063 SIP/2.0 >>>>> Call 2: ACK sip:voip-test-user-02@10.2.24.142:32768;line=moo62e08 >>>>> SIP/2.0 >>>>> >>>>> The next proxy (which produced/relayed the 487) processes the ACK for >>>>> Call 1 successfully, but sanity_check at the proxy drops the request for >>>>> Call 2 with: >>>>> >>>>> DEBUG: {1 <null> 172.30.154.189 102 ACK >>>>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>>>> [sanity.c:277]: check_ruri_scheme(): check_ruri_scheme entered >>>>> DEBUG: {1 <null> 172.30.154.189 102 ACK >>>>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>>>> [sanity_mod.c:254]: w_sanity_check(): sanity checks result: 0 >>>>> >>>>> whereas Call 1 seems OK: >>>>> >>>>> DEBUG: {1 <null> 172.30.154.189 102 ACK >>>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>>> [sanity.c:305]: check_required_headers(): check_required_headers entered >>>>> DEBUG: {1 <null> 172.30.154.189 102 ACK >>>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>>> [sanity.c:313]: check_required_headers(): check_required_headers passed >>>>> >>>>> Could this be a bug in sanity module? Is there anything one can do in >>>>> config which could result in illegal ACKs being produced for hop-by-hop >>>>> transactions? schema appears to be sip: in both cases... >>>>> >>>>> Thank you. Best regards, >>>>> George >>>>> >>>> >>>> _______________________________________________ >>>> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>> >>>> -- >>>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>>> Funding: https://www.paypal.me/dcmierla >>>> >>>> -- >>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>> Funding: https://www.paypal.me/dcmierla >>> >>>

2 2

Setting up Kamailio as Teams SBC
by Роман С. 08 Jul '20

08 Jul '20

Hello. I'm trying to set up Kamailio as SBC for Teams using https://skalatan.de/en/blog/kamailio-sbc-teams. Setup is completely default (except things mentioned at article), but I use wildcard certificate for TLS. Well, I can't even pass dispatcher: kamcmd dispatcher.list | egrep "RI|FLAG" URI: sip:sip.pstnhub.microsoft.com ;transport=tls FLAGS: IP PRIORITY: 3 URI: sip:sip2.pstnhub.microsoft.com ;transport=tls FLAGS: IP PRIORITY: 2 URI: sip:sip3.pstnhub.microsoft.com ;transport=tls FLAGS: IP PRIORITY: 1 Where do I start to dig?

3 6

sanity module checks fail for ACK with parameters in RURI - possible bug?
by George Diamantopoulos 08 Jul '20

08 Jul '20

Hello all, I'm not 100% sure this is the only culprit in an issue I'm investigating, but superficially it appears that RURI scheme sanity module checks from the default config (flags 17895 in REQINIT) fail if the RURI in an ACK following a 487 includes parameters. Example from two calls from a kamailio instance acting as registrar/usrloc server, INVITE RURIs are after usrloc lookup: Call 1: INVITE sip:voip-test-gd@172.17.173.14:5063 SIP/2.0 Call 2: INVITE sip:voip-test-user-02@10.2.24.142:32768;line=moo62e08 SIP/2.0 These INVITEs produce no complaints. Later, the same registrar produces ACKs to acknowledge 487 (thus, same transaction ACKs) responses from the next proxy in the path following a CANCEL: Call 1: ACK sip:voip-test-gd@172.17.173.14:5063 SIP/2.0 Call 2: ACK sip:voip-test-user-02@10.2.24.142:32768;line=moo62e08 SIP/2.0 The next proxy (which produced/relayed the 487) processes the ACK for Call 1 successfully, but sanity_check at the proxy drops the request for Call 2 with: DEBUG: {1 <null> 172.30.154.189 102 ACK 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity [sanity.c:277]: check_ruri_scheme(): check_ruri_scheme entered DEBUG: {1 <null> 172.30.154.189 102 ACK 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity [sanity_mod.c:254]: w_sanity_check(): sanity checks result: 0 whereas Call 1 seems OK: DEBUG: {1 <null> 172.30.154.189 102 ACK 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity [sanity.c:305]: check_required_headers(): check_required_headers entered DEBUG: {1 <null> 172.30.154.189 102 ACK 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity [sanity.c:313]: check_required_headers(): check_required_headers passed Could this be a bug in sanity module? Is there anything one can do in config which could result in illegal ACKs being produced for hop-by-hop transactions? schema appears to be sip: in both cases... Thank you. Best regards, George

2 5

Some Functions not defined in Documentation of Module ims_registrar_pcscf
by Hamid Hashmi 08 Jul '20

08 Jul '20

I have been using a sample code of PCSCF<https://github.com/kamailio/kamailio/blob/master/misc/examples/ims/pcscf/ro…>, where a function "pcscf_assert_identity" is used but it's not defined in module ims_registrar_pcscf documentation. Can you please explain its functionality. The same way other functions are not defined in the documentation. pcscf_assert_called_identity reginfo_handle_notify pcscf_unregister Regards Hamid R. Hashmi

1 0

Kamailio 5.3.4 - Segfault on http_async_client
by José Seabra 07 Jul '20

07 Jul '20

Hello there, Kamailio is terminating with signal 11 and is generating a coredump file. I opened the core file generated by it and it seems to be related to the module http_async_client. In the following link you can find the output of core dump backtrace full: https://pastebin.com/rapPZy9N Below is the http_async_client config that I have: modparam("http_async_client", "workers", 2) modparam("http_async_client", "connection_timeout", 500) modparam("http_async_client", "hash_size", 2048) modparam("http_async_client", "curl_verbose", 0) modparam("http_async_client", "memory_manager", "shm") If the information provided is not enough to troubleshoot this issue let me know. Many thanks Best Regards -- Cumprimentos José Seabra

1 1

Kamailio Buildroot
by Benjamin Marty 07 Jul '20

07 Jul '20

Hello I would like to ask if someone already tried to build Kamailio for the Embedded Distribution Buildroot. I tried it myself but it looks like Kamailio is not that easy to cross compile. Thanks Benjamin

3 3

siptrace not sending data to hepic
by Andrew Chen 07 Jul '20

07 Jul '20

Hi all, I've been assigned a task to investigate why hepic doesn't generate MOS scores ever since we changed the kamailio interface MTU from 9001 -> 1500. During my investigation, this is what I have found out so far: - Putting MTU back to 9001 removes this condition where no MOS was generated. - I did a tcpdump of port 9063 which is the port used to send data to Hepic capture server and saw it reports packets are fragmented: 3:48:55.895505 IP (tos 0x68, ttl 64, id 38049, offset 0, flags [+], proto UDP (17), length 1500) 206.81.181.40.5060 > 10.225.88.239.9063: SIP 13:48:55.896451 IP (tos 0x68, ttl 64, id 38050, offset 0, flags [+], proto UDP (17), length 1500) 206.81.181.40.5060 > 10.225.88.239.9063: SIP - Hepic support also caused by using 1500 MTU on the interface. - I tried using the following core parameters to see if they could remedy this issue but didn't seem to work: ## MTU related core settings udp4_raw=-1 udp4_raw_mtu=1500 I can't seem to find any parameters in the siptrace module that controls fragmentation of the packets to Hepic. Any suggestions on how else to troubleshoot this? Thanks. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 2

Keep one registration per device(src_ip+port)
by Zhan Bazarov 07 Jul '20

07 Jul '20

Hello. I'd like to keep one registration per device in usrloc table, for other cases I want to store and maintain one contact per AoR. Does kamailio support it out of the box? For second case save("location", "0x01"); working fine. But how about first case? for example, if one of device located at home, and second located at office. - there are two different network and NAT - how I can make both devices ringing when INVITE is sent? If I do classic multiAor scenario - it can lead to issues caused by multiple contacts in 200OK response for a single device and devices are confusing sometimes. So, what I did now: if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); add_contact_alias(); } [REGISTRAR] { if (nat_uac_test(64)) { if (!save("location", "0x04")) { sl_reply_error(); } } else { if (!save("location")) { sl_reply_error(); } } } modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 3) modparam("usrloc", "use_domain", MULTIDOMAIN) modparam("usrloc", "xavp_contact", "usrlocattrs") modparam("usrloc", "desc_time_order", 1) modparam("usrloc", "db_timer_clean", 1) modparam("usrloc", "close_expired_tcp", 1) modparam("usrloc", "db_insert_update", 1) modparam("usrloc", "expires_type", 1) modparam("usrloc", "matching_mode", 1) modparam("registrar", "method_filtering", 1) modparam("registrar", "max_expires", 3530) modparam("registrar", "min_expires", 1800) modparam("registrar", "received_avp", "$avp(s:rcv)") modparam("registrar", "gruu_enabled", 0) modparam("registrar", "use_path", 1) modparam("registrar", "path_check_local", 1) modparam("registrar", "path_mode", 1) modparam("path", "use_received", 1) How do u think, maybe modparam("registrar", "append_branches", 0) will make some help? -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

1 0

Nathelper ping from dedicated kamailio instance
by Angelo Pellegrinon 07 Jul '20

07 Jul '20

Hi! I am new to kamailio so please forgive me if I write some errors. I am configuring some kamailio instances in different environments to handle big traffic and I have a problem with usrloc + db + nathelper. What have I done: 1. Configured a kamailio instance to handle sip registers with usrloc to database with database mode 2 (memory + db), disabled nathelper pings. This is ok and performs well; 2. Configured a kamailio instance to handle nathelper ping. Same usrloc + db + nathelper config from 1 except that I use db mode 3 (only db) and that I enabled natping_interval . My problem is that it seems that the second kamailio instance does not send pings. The second one connects correctly to db (if I lookup from kamcmd a user registered from the first kamailio I can find it) but it seems that does not read all locations. Can you help me? Many many thanks Angelo

1 0

rtpengine with mutiple IPs on the same network.
by David Villasmil 07 Jul '20

07 Jul '20

Hello guys, I have rtpengine running on a box with ips like this: external: 172.10.0.10 (advertise 10.10.10.1) internal: 172.10.0.11 Both Ips are on the same network. When i do rtpengine_manage i set "direction=external direction=internal", and the SDP offer is correct. The problem I'm having is that when rtp tried to send to the internal network, the OS (i believe) selects the first interface (external) ip 172.10.0.10 to reach the internal network. This I suppose is not a great deal, but I would like to separate them and only use the internal for internal proxying. Help is appreciated! Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 5

Retrieving cert details from tls peer
by Mark Boyce 07 Jul '20

07 Jul '20

Hi all Been trying to grab the TLS cert details from incoming connections, but failing :-( So with lines just before AUTH is called like this; if (proto == TLS) { xlog("L_INFO", "TLSDUMP $ci peer_subject : $tls_peer_subject\n"); Gets met with a log line line this; INFO: tls [tls_server.c:431]: tls_accept(): tls_accept: new connection from 1.2.3.4:11797 using TLSv1.2 ECDHE-RSA-AES256-GCM-SHA384 256 INFO: tls [tls_server.c:434]: tls_accept(): tls_accept: local socket: 5.6.7.8:5061 INFO: tls [tls_server.c:445]: tls_accept(): tls_accept: client did not present a certificate ... INFO: tls [tls_select.c:168]: get_cert(): Unable to retrieve peer TLS certificate from SSL structure This is with verify_certificate and require_certificate set to no in tls.cfg If I try and set the following in tls.cfg [server:default] method = TLSv1.2+ verify_certificate = no require_certificate = yes I see in the logs; INFO: tls [tls_domain.c:303]: ksr_tls_fill_missing(): TLSs<default>: tls_method=22 INFO: tls [tls_domain.c:315]: ksr_tls_fill_missing(): TLSs<default>: certificate='/etc/kamailio/tls-certs/cert.pem' INFO: tls [tls_domain.c:322]: ksr_tls_fill_missing(): TLSs<default>: ca_list='(null)' INFO: tls [tls_domain.c:329]: ksr_tls_fill_missing(): TLSs<default>: crl='(null)' INFO: tls [tls_domain.c:333]: ksr_tls_fill_missing(): TLSs<default>: require_certificate=1 INFO: tls [tls_domain.c:340]: ksr_tls_fill_missing(): TLSs<default>: cipher_list='(null)' INFO: tls [tls_domain.c:347]: ksr_tls_fill_missing(): TLSs<default>: private_key='/etc/kamailio/tls-certs/privkey.pem' INFO: tls [tls_domain.c:351]: ksr_tls_fill_missing(): TLSs<default>: verify_certificate=0 INFO: tls [tls_domain.c:354]: ksr_tls_fill_missing(): TLSs<default>: verify_depth=9 NOTICE: tls [tls_domain.c:1095]: ksr_tls_fix_domain(): registered server_name callback handler for socket [:0], server_name='<default>' ... INFO: tls [tls_domain.c:692]: set_verification(): TLSs<default>: Client MUST present valid certificate INFO: tls [tls_domain.c:303]: ksr_tls_fill_missing(): TLSc<default>: tls_method=20 INFO: tls [tls_domain.c:315]: ksr_tls_fill_missing(): TLSc<default>: certificate='(null)' INFO: tls [tls_domain.c:322]: ksr_tls_fill_missing(): TLSc<default>: ca_list='(null)' INFO: tls [tls_domain.c:329]: ksr_tls_fill_missing(): TLSc<default>: crl='(null)' INFO: tls [tls_domain.c:333]: ksr_tls_fill_missing(): TLSc<default>: require_certificate=1 INFO: tls [tls_domain.c:340]: ksr_tls_fill_missing(): TLSc<default>: cipher_list='(null)' INFO: tls [tls_domain.c:347]: ksr_tls_fill_missing(): TLSc<default>: private_key='(null)' INFO: tls [tls_domain.c:351]: ksr_tls_fill_missing(): TLSc<default>: verify_certificate=1 INFO: tls [tls_domain.c:354]: ksr_tls_fill_missing(): TLSc<default>: verify_depth=9 INFO: tls [tls_domain.c:692]: set_verification(): TLSc<default>: Server MUST present valid certificate ... ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417C086:SSL routines:tls_process_client_certificate:certificate verify failed Which looks like verification is being enabled when I add require? Would someone be kind enough to point out what I am missing please? (Assuming it’s not a bug :-) Thanks Mark -- Mark Boyce Dark Origins Ltd

2 8

About routing calls
by ahmed moghazy 07 Jul '20

07 Jul '20

Excuse me i already made integration for kamailio with asterisk but i want to make sip calls through kamailio and media fuctions through asterisk , i want kamailio to make calls away from asterisk , i just wanted asterisk to use its dahdi module .’ Thank you.

2 1

Kamailio Push Notification for iOS13
by william anderson 07 Jul '20

07 Jul '20

Hello, Yesterday, there was a new release of Linphone IOS app, but i need help from some kamailio expert how to add PUSH notification to the application to receive incoming calls when the phone is not in use (wake up required) from a kamailio server. I have read on internet old tutorials (tsilo module) but most of them are incomplete or do not explain in an easy way how to do it (for an average kamailio user). Anyone could do an easy to understand tutorial ? Regards, William Anderson

2 1

Looking for a VoIP & Kamailio Dev-Ops
by Henning Westerholt 06 Jul '20

06 Jul '20

Dev-Ops VoIP & Kamailio GILAWA is looking for a new colleague to join our team! Tasks: Deploying, developing, and integrating our software based on Kamailio Testing and validation of our customer solutions Solve complex customer problems in the VoIP field Fix Bugs and create extensions in the Kamailio project Requirements: Good background in VoIP telephony and Kamailio Stack Experience with signalling standards (SIP, RTP etc..) and troubleshooting Knowledge and experience in one of the following languages: C/C++, Bash, PHP Experience with Debian or RedHat/CentOS Linux, knowledge of Ansible would be a plus Fluent in English Based in European or neighbouring time zone (e.g. EMEA area) Self-organized and strong communication skills Given the current situation this is a remote position. If you are interested, please send me a (direct) e-mail with an CV or Linkedin link, please include your availability and salary expectations as well. Best regards, Henning Westerholt -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

3 3

Kamailio 5.3.5, usrloc: Error inserting location (integer out of range)
by Stefan 03 Jul '20

03 Jul '20

Hi Maillist Kamailio-Version: 5.3.5 built from source with postgres-db. I get an Error if I try to add a static location with kamctl, see my example: kamctl ul add 1234 sip:1234@10.10.10.10:5060 NULL NULL Log output: Jul 1 09:28:20 kamailio-srv02 /sbin/kamailio[3892]: ERROR: db_postgres [km_dbase.c:266]: db_postgres_submit_query(): 0x7fbe5398c148 PQsendQuery Error: ERROR: integer out of range#012 Query: insert into location (username,contact,expires,q,callid,cseq,flags,cflags,user_agent,path,methods,last_modified,ruid,reg_id,server_id,connection_id,keepalive,partition ) values ('1234','sip:1234@10.10.10.10:5060','1970-01-01 00:00:00',1.000000 ,'dfjrewr12386fd6-343(a)kamailio.rpc',1,0,0,'kamailio SIP Router - RPC Server','NULL',2147483648,'1970-01-01 00:00:00','ulcx-5ef99724-f37-1',0,0,0,0,22) truncated output of kamctl ul show: "AoRs": [{ "Info": { "AoR": "1234", "HashID": 924651761, "Contacts": [{ "Contact": { "Address": "sip:1234@10.10.10.10:5060", "Expires": "permanent", "Q": 1, "Call-ID": "dfjrewr12386fd6-343(a)kamailio.rpc", "CSeq": 1, "User-Agent": "kamailio SIP Router - RPC Server", "Received": "[not set]", "Path": "NULL", "State": "CS_NEW", "Flags": 0, "CFlags": 0, "Socket": "[not set]", "Methods": -2147483648, "Ruid": "ulcx-5ef99724-f37-1", "Instance": "[not set]", "Reg-Id": 0, "Server-Id": 0, "Tcpconn-Id": 0, "Keepalive": 0, "Last-Keepalive": 0, "Last-Modified": 0 } }] } }, { So the log tells me: integer out of range. I think it's this one: "Methods": -2147483648, This let's me thinking that this is a bug. Someone else? Regards Stefan

2 8

403 Forbidden error You must register first with a S-CSCF
by Pavithra Mohanraja 03 Jul '20

03 Jul '20

Hi, I am facing below error when i try to make a call using pjsip .. Registration is successful. I am following the open 5gs VolTE setup. I have not changed any configurations in any of the config files. But I am facing below issue .. Can you please help me what could be the reason. *reason=403 (Forbidden - You must register first with a S-CSCF)* Could anyone please help me .. I am ready to provide anyother info if required . Thanks, Pavithra

1 0

Error in SCSCF
by Pavithra Mohanraja 02 Jul '20

02 Jul '20

Hi , I am facing below error in s-cscf when i send a BYE from alice to bob in kamailio IMS. The problem is in only call termination .Kindly help . Error log in scscf : Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/mem/q_malloc.c:526]: qm_free(): freeing frag. 0x7f4746f89148 alloc'ed from core: core/rvalue.c: rval_new_empty(224) Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/socket_info.c:630]: grep_sock_info(): checking if host==us: 11==11 && [10.244.0.57] == [10.244.0.54] Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/socket_info.c:637]: grep_sock_info(): checking if port 4080 (advertise 0) matches port 5060 Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/name_alias.h:62]: grep_aliases(): matching (0:10.244.0.57:5060) vs. (0:sip.example.com:0) Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/name_alias.h:62]: grep_aliases(): matching (0:10.244.0.57:5060) vs. (0:sip.example.com:5060) Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/name_alias.h:62]: grep_aliases(): matching (0:10.244.0.57:5060) vs. (0: scscf.sip.example.com:4080) Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/forward.c:423]: check_self(): *host (0:10.244.0.57:5060 <http://10.244.0.57:5060>) != me* Jun 30 08:40:22 scscf kamailio[557]: ERROR: <script>: sip:bob at sip.example.com Jun 30 08:40:22 scscf kamailio[557]: ERROR: <script>: *Domain not served 403* Jun 30 08:40:22 scscf kamailio[557]: ERROR: <script>: *BYE *( sip:alice@sip.example.com (10.244.0.52:4060) to sip:bob@sip.example.com, -ARSjM.TTZgxhzwDjKRN0DtENWD4bKl0) Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/mem/q_malloc.c:374]: qm_malloc(): qm_malloc(0x7f4746dd7010, 30) called from core: core/ut.c: as_asciiz(161) I am not getting what it checks .. 10.244.0.57 -> UE bob 10.244.0.52 -> pcscf 10.244.0.53 -> icscf 10.244.0.54 -> scscf 10.244.0.56 -> UE alice Kindly help Thanks, Pavithra

2 4

Sql_result_free() not cleaning properly
by Duarte Rocha 01 Jul '20

01 Jul '20

Greetings, I'm trying to clean a SQL result using "sql_result_free()" but i'm not having success. Can you please verify if i'm doing anything wrong here? This is my code : xerr("Pre clean [0] = Iterator 0 = $xavp(Cli_PICKER_Default_Search_Result[0]=>Code)"); sql_result_free("Cli_PICKER_Default_Search_Result"); xerr("Pos clean [0] = Iterator 0 = $xavp(Cli_PICKER_Default_Search_Result[0]=>Code)"); The result is the following : ERROR: <script>:Pre clean [0] = Iterator 0 = 111111111111 ERROR: <script>:Pos clean [0] = Iterator 0 = 111111111111 Am i using sql_result_free correctly or doing something wrong? Best Regards,

4 4

Re: [SR-Users] [EXT] RE: kamailio 522 S-CSCF failing to start -UPDATE
by Martin W Woscek 30 Jun '20

30 Jun '20

Problem: The mysql db issues with the scscf db include: table version, disappearing impu and/or UE data retained in tables when none are registered Workarround: In order to get over those various mysql issues listed causing the S-CSCF not to come up properly as boot up, 1. create a fresh new scscf database (kamadbctl [drop|create] scscf and import the msql script from the scscf configs) 2. back up the new scscf db 3. modify the init.d startup script for the scscf to restore the fresh db upon each start/restart of the S-CSCF /etc/init.d/kamailio_scscf .. case "$1" in start|debug) mysql -u root --database=scscf < /etc/kamailio/ims_databases/scscf.sql The S-CSCF has been stable and dependable on each boot. _Martin From: Henning Westerholt <hw(a)skalatan.de> Sent: Monday, June 8, 2020 4:50 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Cc: Martin W Woscek <mwoscek(a)mitre.org> Subject: [EXT] RE: kamailio 522 S-CSCF failing to start Hello, the error message below is from the DB startup version check. So do you have sometimes DB errors on this virtual machine? Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Martin W Woscek Sent: Monday, June 8, 2020 2:46 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> Subject: [SR-Users] kamailio 522 S-CSCF failing to start Hello, There are times that our S-CSCF module just wont start. I’ve tried correlating the issue with some operation such as not de-registering an IMS client (Boghe or imsdroid) prior to shutting down the Kamailio server. But this isn’t always the case. Being that we are running this an a Ubuntu Virtual Machine, simply overwriting the image with a backed up VM is the temporary solution. I’ve dropped and created the db for S (kamdbctl) and rearranged the order of loading the presence module with same result. Status of the S-CSCF daemon on failure to start: # systemctl status kamailio_scscf ● kamailio_scscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_scscf; generated) Active: inactive (dead) Docs: man:systemd-sysv-generator(8) Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 08 08:22:55 ubuntu kamailio_scscf[16495]: * already running Jun 08 08:22:55 ubuntu kamailio_scscf[16495]: ...done. Jun 08 08:22:55 ubuntu systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Jun 08 08:24:03 ubuntu systemd[1]: Stopping LSB: Start the Kamailio SIP proxy server... Jun 08 08:24:03 ubuntu kamailio_scscf[16517]: * Stopping Kamailio IMS S-CSCF: kamailio_scscf Jun 08 08:24:03 ubuntu kamailio_scscf[16517]: ...done. Jun 08 08:24:03 ubuntu systemd[1]: Stopped LSB: Start the Kamailio SIP proxy server. root@ubuntu:~/IMS_manual_scripts# systemctl start kamailio_scscf root@ubuntu:~/IMS_manual_scripts# systemctl status kamailio_scscf ● kamailio_scscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_scscf; generated) Active: active (exited) since Mon 2020-06-08 08:27:23 EDT; 58s ago Docs: man:systemd-sysv-generator(8) Process: 16543 ExecStart=/etc/init.d/kamailio_scscf start (code=exited, status=0/SUCCESS) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: <core> [db.c:457]: db_check_table_version(): invalid version 0 for table presentity found, expected 5 (check table structure and ta Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: presence [presence.c:367]: mod_init(): error during table version check Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: <core> [core/sr_module.c:898]: init_mod(): Error while initializing module presence (/usr/local/lib64/kamailio/modules/presence.so) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: WARNING: <core> [core/mem/q_malloc.c:480]: qm_free(): WARNING: free(0) called from cdp_avp: cdp_avp_mod.c: cdp_avp_destroy(226) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 08 08:27:23 ubuntu kamailio_scscf[16543]: * already running Jun 08 08:27:23 ubuntu kamailio_scscf[16543]: ...done. Jun 08 08:27:23 ubuntu systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Any insight is appreciated. Thanks, _Martin

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default config and options for Kamailio 5.4.x
by Daniel-Constantin Mierla 30 Jun '20

30 Jun '20

Hello, with the release of Kamailio 5.4.0 approaching, I opened an item in the tracker to collect suggestions about what should be tuned in the default config file. If you feel like something should be there, add your comment to: * https://github.com/kamailio/kamailio/issues/2381 Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

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Re: [SR-Users] RTP voice failover
by Yufei Tao 30 Jun '20

30 Jun '20

Hi, RTP FO is possible, e.g. with keepalived+rtpengine+Redis as people suggested. Here's the guide for configuring it which I find very useful: https://github.com/sipwise/rtpengine/wiki/Redis-keyspace-notifications A possible way is to have a pair of rtpengine instances in master-slave setup. When master goes down, RTP continues on slave (new master) after a few seconds gap, when the virtual IP(s) migrate over. Another useful link: https://www.kamailio.org/events/2016-KamailioWorld/Day2/20-Pawel.Kuzak-High… Good luck! Yufei <https://github.com/sipwise/rtpengine/wiki/Redis-keyspace-notifications>

1 0

kamailio and rtpengine in docker
by David Villasmil 29 Jun '20

29 Jun '20

Hello guys, I've got a kamailio on a docker and kamailio on another docker on different boxes. The issue i'm having is that when kamailio tries to enable rtpengine it tries pinging it. The ping arrives at the actual container where rtpengine is running, but for some reason i haven't figured out, rtpengine (or the container) is not responding to the ping. I'm writing here in the hopes that someone has already implemented this an come across this and a solved (for sure) I'd appreciate some help. Many thanks! David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 3

Valid URL characters in the RURI
by Sam Ware 29 Jun '20

29 Jun '20

I'm using Sanity Checks. Is an underscore (_) not allowed in the host portion of the RURI? I'm on version 5.3. I reviewed the pase_uri function in the core parse_uri.c but I am not super proficient in C. case URI_HOST_P: switch(*p) { check_host_end; default: if(!isalnum(*p) && (*p != '.') && (*p != '-') && !uri_host_char_allowed(*p)) { goto error_bad_host; } } break; I see it calls out periods (.) and dashes (-) but not underscores. I'm wondering why, being underscores are allowed in URLs. -- Sam D Ware

2 1

Changing Debug Level on the fly
by Sam Ware 29 Jun '20

29 Jun '20

I am trying to change the debug level without restarting the service. My research says this is right but I get an error message. $ sudo -u kamailio kamcmd dbg.set_mod_level core 2 error: 500 - cannot store parameter -- Sam D Ware

2 1

SIP OPTIONS between UE and P-CSCF
by BALL SUN 29 Jun '20

29 Jun '20

Hi We have setup UE with Kamailio IMS, and after the registration is successful. The 2nd SIP REGISTER is sent from UE to P-CSCF through IPSEC tunnel. But we found that the SIP OPTIONS (I believed it is a health check) from P-CSCF to UE is not through the IPSEC tunnel. our question is 1. Should SIP OPTIONS run inside the IPSEC tunnel or it should be outside? 2. if it can be configured to run inside the IPSEC tunnel, what configuration can we change to achieve it? - RBK

2 3

How to add parameter ;user=phone in RURI
by Minh Phan 28 Jun '20

28 Jun '20

Hi List, I'm having trouble adding the parameter user=phone in the RURI. I have tried to use the function add_uri_param from siputils module but it doesn't work (see below). Here is the code I use to add user=phone parameter to FROM, TO and RURI uac_replace_from("", "sip:$var(normalized_number)@sip.domain.com ;user=phone"); uac_replace_to("", "sip:$var(normalized_number)@sip.domain.com;user=phone"); $rU = $var(normalized_number); *add_uri_param*("user=phone"); Thanks in advance for any hints, Best regards, Minh Phan

2 1

RTP voice failover
by Evgeniy 27 Jun '20

27 Jun '20

Is that possible to implement RTP voice fail-over? In case RTP node goes down - the call should not be interrupted. This is kinda strange - but client want to have super redundant system. For example use 2 RTP nodes and simultaneously send 2 RTP stream over 2 nodes (yes - network traffic will increase) Or maybe is a way to recover UDP connections after a crash and replace node state. Maybe use a Kubernetes tools for that purpose ? Here what i found in mailing-list - https://lists.kamailio.org//pipermail/sr-users/2015-December/091006.html Many thanks for any hints to this topic.

6 9

Kamailio IMS with ENUM
by BALL SUN 26 Jun '20

26 Jun '20

Hi Is there any information that can help to setup ENUM query in Kamailio IMS? what we wanted to achieve is to when A in domain A (the local domain) makes an INVITE to B, it will check the domain of B from ENUM backend. if it is not on the local domain, it will route to remote IMS core. Thanks - RBK

2 7

Does kamailio sip domain makes sense if i defined some alias
by PICCORO McKAY Lenz 26 Jun '20

26 Jun '20

in kamctlrc i defined an DOMAIN but later that ip was updated.. so my question: it make sense inclusivelly if already defined more allias in kamailiorc.cfg file? in case of more complicated response,, anybody can explain me about that why? -- Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

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Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) -RESOLVED
by Martin W Woscek 26 Jun '20

26 Jun '20

Hello, Just wanted to update what fixed our issue. The "contact_user" which is "asterisk" that we were using for the Kamailio context for outgoing SIP trunk/channel in the pjsip.conf file of the Asterisk 16 server was not a valid user in the IMS HSS database (FHoSS implementation). However not being an expert in the IMS, IMHO, the IMS should have kicked out this INVITE as soon as credentials were checked at the front door of the IMS, but rather Kamailio sent it from I-CSCF to S-CSCF to P-CSCF, which is the correct flow, until the P-CSCF sent it back to I-CSCF which then resulted in a 604 HSS user unknown being sent thru the reverse direction back to Asterisk. Anyway, its always good to end Friday on a high note. Thanks, _Martin From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Martin W Woscek Sent: Wednesday, June 24, 2020 1:32 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>; miconda(a)gmail.com Subject: Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) Just to clarify what we are seeing, here is a packet from the INVITE from the S-CSCF (6060) to the P-CSCF(4060). I have eliminated the S-CSCF as this packet looks correct. The P-CSCF is not handling the request URI correctly, and simply sends it back to the I-CSCF where I-CSCF cannot find the user in the HSS based on the ruri updated by the S-CSCF with the local IP address of the UE (192.168.20.1). The P-CSCF should send this INVITE to the UE (703222888(a)192.168.20.1<mailto:703222888@192.168.20.1>) No. Arrival Time Source Destination Source Port Destination Port SIP to tag SIP from tag Method Record-Route Userinfo Request-URI Media Port Protocol Length Request-URI Host Port Owner Username Media Type Media Format Origin-Host Type slice_type SIP from address User Part SIP from address Host Part SIP to address User Part SIP to address Host Part Contact URI User Part Contact URI Host Part Request-URI User Part Request-URI Host Part Request-URI Host Port Info 1553 Jun 17, 2020 09:31:21.722451703 Eastern Daylight Time 192.168.20.128 192.168.20.128 6060 4060 d490a30c-42a8-4fc0-9fe9-b7a21537386d INVITE mt sip:7032228888@192.168.20.1:63871;transport=udp 10626,13370 SIP/SDP 2263 63871 - audio,video ITU-T G.711 PCMU,DynamicRTP-Type-101,0,101,101,DynamicRTP-Type-99,99,99 7033331111 asterisk.irisims.org 7032228888 ims522.irisims.org asterisk 192.168.20.131 7032228888 192.168.20.1 63871 Request: INVITE sip:7032228888@192.168.20.1:63871;transport=udp | Frame 1553: 2263 bytes on wire (18104 bits), 2263 bytes captured (18104 bits) on interface any, id 0 Linux cooked capture Internet Protocol Version 4, Src: 192.168.20.128, Dst: 192.168.20.128 User Datagram Protocol, Src Port: 6060, Dst Port: 4060 Session Initiation Protocol (INVITE) Request-Line: INVITE sip:7032228888@192.168.20.1:63871;transport=udp SIP/2.0 Method: INVITE Request-URI: sip:7032228888@192.168.20.1:63871;transport=udp Request-URI User Part: 7032228888 Request-URI Host Part: 192.168.20.1 Request-URI Host Port: 63871 [Resent Packet: False] Message Header Record-Route: <sip:mt@192.168.20.128:6060;lr=on;ftag=d490a30c-42a8-4fc0-9fe9-b7a21537386d;did=7d2.e0f1> Route: <sip:term@pcscf522.irisims.org;lr> Via: SIP/2.0/UDP 192.168.20.128:6060;branch=z9hG4bK285b.2a57abaad57cfd55bc18a23d4e9cefc8.0 Via: SIP/2.0/UDP 192.168.20.128;branch=z9hG4bK285b.793ba7dcf9fcd18ddef225b83896340e.1 Via: SIP/2.0/UDP 192.168.20.131:5060;received=192.168.20.131;rport=5060;branch=z9hG4bKPjef451437-0a4f-4732-ac54-318740956335 From: <sip:7033331111@asterisk.irisims.org>;tag=d490a30c-42a8-4fc0-9fe9-b7a21537386d To: <sip:7032228888@ims522.irisims.org> Contact: <sip:asterisk@192.168.20.131:5060> Call-ID: 39c15c42-b0b7-48b5-b4bf-ac7a99fbe660 [Generated Call-ID: 39c15c42-b0b7-48b5-b4bf-ac7a99fbe660] CSeq: 19314 INVITE Route: <sip:ims522.irisims.org:5060;lr> Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 Max-Forwards: 68 User-Agent: Asterisk PBX 16.6.0 Content-Type: application/sdp Content-Length: 1121 Message Body From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Martin W Woscek Sent: Wednesday, June 24, 2020 11:21 AM To: miconda(a)gmail.com<mailto:miconda@gmail.com>; Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> Subject: Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) Hi, The issue is the P should be sending to the UE not back to the I like the failed flow: Asterisk16-[phone] --->ICSCF--->S-CSCF--->P-CSCF--->I-CSCF ---> [604 HSS user unknown] to Asterisk16[phone] The successful call should look like: Asterisk16-[phone] --->ICSCF--->S-CSCF--->P-CSCF--->Kamailio-[UE] Does anyone have a example P-cscf scripts including the mt.cfg or other script file that make this work? The reverse calls work just fine. Thanks, _Martin From: Daniel-Constantin Mierla <miconda(a)gmail.com<mailto:miconda@gmail.com>> Sent: Wednesday, June 24, 2020 4:53 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>; Martin W Woscek <mwoscek(a)mitre.org<mailto:mwoscek@mitre.org>> Subject: [EXT] Re: [SR-Users] Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) Hello, Kamailio is not generating the 200ok for INVITE (calls), it just sends out what it received from end point, after consuming own Via header (of course, other headers can be changed based on config rules, but Kamailio is not in charge of setting To-tag). Cheers, Daniel On 19.06.20 21:19, Martin W Woscek wrote: Hello all, SIP/UE (boghe or imsdroid) client registered to Kamailio makes call to an Asterisk registered SIP phone is successful: [UE]-Kamailio--->INVITE--->Asterisk16-[phone] [UE]-Kamailio--->200 OK with to_tag--->Asterisk16-[phone] But in reverse direction for a call, Kamailio does not return the SIP OK so no to_tag is sent so call fails to ring and complete: [phone]-Asterisk16 --->INVITE--->Kamailio Where the UE device never receives the INVITE, Asterisk never gets and 200 OK message with the to_tag from the I-CSCF, the call flow itself gets lost in Kamailio, where the P-CSCF sends final INVITE to I-CSCF and and ultimately a 604 HSS user unknown message is sent back to Asterisk from the I-CSCF. Basically Im using the default "sample" configs for both the P and the I-CSCF. Our sauce is in the S-CSCF for out going calls that originate by a registered UE. Any insight or sample Kamailio configuration that Im lacking? Has anyone done this and could share the asterisk and Kamailio script snippets that make it possible. Thanks, _Martin _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- www.asipto.com<http://www.asipto.com> www.twitter.com/miconda<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda> Funding: https://www.paypal.me/dcmierla

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