- sr-users - kamailio.org

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

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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

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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

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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

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