- sr-users - kamailio.org

evapi:message-received and state
by Alex Balashov 07 Nov '22

07 Nov '22

Hi, I am t_continue()ing a transaction inside of event_route[evapi:message-received]. The $evapi(msg) has quite a bit of payload that will be needed by the resumed transaction. Is there a good way to inject this data into the transaction scope externally? No [X]AVPs created in the event_route persist, of course, since the route is scoped outside of the transaction. I have been using htable to pass data into the transaction, keyed by the index and label. But, this strikes me as a kind of clumsy memory synchronisation anti-pattern. Is there a better way? Cheers, — 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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Kamailio Not Responding to Connections
by Carlos Escalona 07 Nov '22

07 Nov '22

Hello! I hope all of you are having a nice day. I would like some help. I recently had some troublesome events in my Kamailio setup, at a particular time of the day specifically after lunch when my platform's traffic spikes Kamailio suddenly stops responding to any connection and restarting it didn't fix the problem basically I have to restart the hardware to get out of that failed state. The only unusual log is the following: Oct 28 16:49:10 ip-172-32-16-97 /usr/sbin/kamailio[952]: ERROR: topos [tps_storage.c:615]: tps_storage_record(): failed to store Someone recently posted a problem that occurred when doing stress tests with the siptrace module active and the service that collects the traces was failing, I wonder if this may be the case because it is possible that something similar is happening in my case. Unfortunately, I don't have much more information since I had to implement a quick fix that mitigates the problem for now, but I would like to be able to go back to the previous configuration that seemed more appropriate. My plan is to test now that I have solved the problem I had with the traces service, but I wanted to know if anyone had experienced a similar situation. Atenciosamente, [image: photo] <https://www.techer.com.br/> *Carlos Escalona* IT - Development 41 3073-0091 | R 1011 www.techer.com.br -- O conteúdo desta mensagem é confidencial. Se você não se encontra na lista de destinatários ou tenha recebido por engano, não a copie, imprima, envie, ou utilize, de qualquer forma, seu conteúdo. Neste caso, destrua a mensagem e, por favor, notifique o remetente.

2 1

Serial Forking on BYE message
by Ilie Soltanici 05 Nov '22

05 Nov '22

Hello, I have a challenge here to do serial forking, not when the call is failed on the first destination, but when the call is ended. So, the first will go to one destination, will be answered, and when it's ended the BYE message should be dropped and forward the initial INVITE to another location. Is that possible?

2 1

Re: [SR-Users] $Rp variable seems to contain wrong port
by Luis Azedo 04 Nov '22

04 Nov '22

have you tried to use $T_rpl($Rp) ?

1 0

$Rp variable seems to contain wrong port
by Ross McKillop 03 Nov '22

03 Nov '22

Hi All, At the risk of possibly dragging up an old thread (https://lists.kamailio.org/pipermail/sr-users/2021-June/112791.html <https://lists.kamailio.org/pipermail/sr-users/2021-June/112791.html>) I’m seeing unusual behaviour where $Rp is still showing what I think is the ‘wrong’ behaviour, but given Daniel’s reply on the earlier thread I want to be sure this isn’t the intended behaviour. The thread above confirms that $Rp is not taken from the IP layer but is intended to give the attribute of the listen socket associated with the SIP traffic, which is fine and what I’m expecting, but in my case it doesn’t seem to be working this way. (I’ve tried with and without tcp_reuse_port=yes and tcp_connection_match=1 too just in case one of the above parameters has changed this behaviour) If Kamailio listening on multiple ports e.g. listen=tcp:10.30.22.73:7701 listen=tcp:10.30.22.73:7702 listen=tcp:10.30.22.73:7703 listen=tls:10.30.22.73:7801 listen=tls:10.30.22.73:7802 listen=tls:10.30.22.73:7803 And I use $fs to force the socket of an outbound message to one of the above before t_relay() it works as expected (with tcp_reuse_port it originates from the given port number above, without that set it will originate from a different ephemeral port even to the same destination, so $fs is working as expected) Then, in reply_route{} I have a simple line to add the port it was received on to the headers (I’ve also tried with $RAp) append_hf("X-Rcvd-Port: $Rp\r\n”); In every case, the reply seems to contain the *first* port of the protocol type (so in the above example every TCP reply irrespective of whether it is to 7701/02/03 will show $Rp 7701, and every TLS reply will show 7801) Any advice or insight anyone can offer would be much appreciated. Thanks, Ross

1 0

Siptrace and Kamailio "stress" test
by Igor Olhovskiy 03 Nov '22

03 Nov '22

Hello, Found an interesting scenario, maybe something could be done here. I have a siptrace module configured like modparam("siptrace", "duplicate_uri", "sip:<MY_HOMER_SERVER>:9060") modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_on", 1) modparam("siptrace", "hep_version", 3) modparam("siptrace", "trace_mode", 1) When Kamailio is under stress test (sipflood via OPTIONS/TLS), with this settings it transfers all flood to HOMER server. At some point Kamailio just stops accepting any new TLS connections. Turning off siptrace solves this issue. As I got, siptrace module is not changing protocol of the message (means not converting TCP/UDP) or so. Could it be possible, that at some point siptrace "eats" all TCP connections (especially if remote HOMER is down or not answering) which leads Kamailio to stop processing any new connections? And is it possible to send HEP traffic only via UDP to prevent this? Yes, sure other option is to trace only "legal" traffic -- Best regards, Ihor (Igor)

3 5

Pipelimit needs restart to reload values
by Duarte Rocha 02 Nov '22

02 Nov '22

Greetings, I've been using the pipelimit module and the pl_check function to control users rates in my proxy. if (!pl_check("$var(UserID)", "TAILDROP", "$var(UserRate)")) { pl_drop(); exit; } In this case $var(UserID) is the pipe and $var(UserRate) is the maximum rate. The value of the rate is retrieved on every call via query to a DB. If I change the UserRate value to 0 CPS on the DB calls are not rejected unless they go over 1 CPS. However, if I restart Kamailio all calls are correctly rejected. Why does this happen? Can I change this behaviour? Cheers, Duarte

3 3

Appending new branches via external means
by Alex Balashov 01 Nov '22

01 Nov '22

Hi, Is there a means of appending new branches from config script for a transaction which is already proceeding in the background, without using `tsilo`? That is to say, I have an async transaction which is resumed in an async worker and is t_relay()’d away to some destination. Meanwhile, I have learned about new destinations for it to try via an incoming registration, and would like to append additional parallel forking branches to that transaction, even though it is already proceeding. ts_append() does this for me; the issue I have is that it is mediated through a registrar, but I am not a registrar. My registrar is upstream, and my proxy is just a front-end to it. So, in essence, I require something like `tsilo` at a more general level, where contacts can be populated by some other mechanism rather than being learned via callbacks out of `usrloc`. Is there a way to accomplish this? Cheers, — 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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consume_credentials not working on PRACK?
by Benoit Panizzon 01 Nov '22

01 Nov '22

Hi List I noticed, that one of our CPE copies the Proxy-Authorization HF in almost all messages sent. As PRACK were not authenticated, those headers were potentially sent on to the destination disclosing the authentication username and realm. So assuming, if credentials are present, the client wishes them to be validated, I added: if (has_credentials("$fd")) { xlog("L_INFO", "$cfg(route): got $rm with credentials. Validate them!\n"); route(AUTH); } and in route[AUTH] I call: pv_auth_check() which returns 1 thus success upon which I use: if(!is_method("REGISTER|PUBLISH")) consume_credentials(); If the method is INVITE: Proxy-Authorization HF is removed by consume_credentials() if the method is PRACK: Proxy-Authorization HF is still present on the outbound leg. Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

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Authentication: Is it possible to ignore realm? How to avoice unnecessary challenge because of changed realm?
by Benoit Panizzon 01 Nov '22

01 Nov '22

Hi As mentioned in the last email, we have a CPE which adds credentials to any request so I would like to validate them. We use $rd as realm. Let's use example.com as example. During the initial INVITE, this works as expected. I try to only show the headers I think are relevant to the issue I face. INVITE sip:1234@example.com SIP/2.0 Proxy-Authorization: [...] realm="example.com" Contact: <sip:9876@192.168.1.1:5060> Authentication user in realm "example.com" == $rd is validated and accepted. The connection is establised: 200 OK From: <sip:9876@example.com>;tag=1 To: <sip:1234@example.com>;tag=2 Contact: <sip:1234@192.168.7.7:5060> The caller terminates the connection: BYE: sip:1234@192.168.7.7:5060 SIP/2.0 From: <sip:9876@example.com>;tag=1 To: <sip:1234@example.com>;tag=2 Proxy-Authorization: [...] realm="example.com" Performing: $var(authres) = pv_auth_check("$rd", "$avp(authsecret)", "0", "1"); fails wirh result -5 as there is no authsecret for realm "192.168.7.7" So an unnecessarey challenge is issued: SIP/2.0 407 Proxy Authentication Required To: <sip:1234@example.com> From: <sip:9876@example.com> CSeq: 3 BYE Proxy-Authenticate: Digest realm="192.168.7.7" To which the client replies with the changed realm: Proxy-Authorization: Digest username="****",realm="157.161.7.7",[...] and succeeds authentication. Is there a way to accept 'any' realm? Or to avoid this unnecessary change of realm just because the remote site submitted a domain realm in the contact header? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 2

SIP routing per MSISDN
by Anthony Blandin 30 Oct '22

30 Oct '22

Hello, I need to route SIP INVITE towards a specific node based on the MSISDN of the Request URI. I saw that the dispatcher module should be able to make this with the function ds_select_dst and the algorithm "hash over request-URI user" but I don't understand how the routing is done if we can't associate the request-URI user with the destination address in the dispatcher DB. Maybe my understanding of this module is wrong. Could you help me to achieve this? Thanks Anthony

3 2

Kamailio sending response to source IP and port from Via ?
by Chaigneau, Nicolas 27 Oct '22

27 Oct '22

Hello, I'm using Kamailio in stateless mode to send redirect responses. I've noticed that the responses are sent to the packet source IP address, and to the port found in the Via header. I'm a bit confused. I was expecting the response to be sent to the address and port found in the Via header. Is it the expected behaviour ? I'm using Kamailio 5.5.2. Thanks for your help ! Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

2 4

Issue with debug 3 after upgrade to kamailio 5.6.2+
by Julia Bo 27 Oct '22

27 Oct '22

Hello, After upgrading kamailio 5.6.1 to 5.6.2 (from source) debug 3 doesn't print messages related to calls to the log file. I only see OPTIONS transactions in debug. Does anyone have the same problem? Best regards, Julia.

2 3

About the WITH_VOICEMAIL Subscriber message 407 issue
by SparkleZou 26 Oct '22

26 Oct '22

Hi Sir/Madam, I'm trying to set a Fresswitch as the Voicemail server. I set the #!define WITH_VOICEMAIL in the kamailio.cfg file to enable the default VoiceMail configuration. Set the voicemail server address as following : voicemail.srv_ip = "10.120.100.179" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" And the kamailio server address is 10.120.100.180. Here face the issue, the Subscribe message could NOT be authorized. The SUBSCRIBER message with token could NOT be forworded to Freeswitch. The log is also attached. It's resllay strange. Please help check. Thanks! BR, Sparkle Zou

3 4

ims_auth with AKAv1-MD5
by Oleg Belousov 24 Oct '22

24 Oct '22

Hello, I'm wondering if ims_auth is tested with actual SIM-card/AKAv1-MD5 authentication? Bit concerned with that since as per authorize.c source code can see for AUTH_AKAV1_MD5/AUTH_AKAV2_MD5/AUTH_MD5 response is counted in the same way, though algorithms are different. -- obelousov.tel

2 2

No REGISTER sent to third party AS from scscf
by Anthony Blandin 24 Oct '22

24 Oct '22

Hi Community, I am using scscf from Kamailio and I need that scscf sent a REGISTER to a third party AS during the registration phase. When scscf received a new registration, no REGISTER is sent to the AS. SAR-SAA is well processed but only 200 OK is sent for the REGISTER to the pcscf. REGISTER is well sent to the AS only for Re-Registration or De-registration. Do you have any idea to handle this case? Regards Anthony

2 3

Dialog Module: How to get two different dialogues if a call is looped through kamailo (call forwarding scenario)
by Benoit Panizzon 21 Oct '22

21 Oct '22

Hi Using the dialog module to generate CDR for billing. We have this situation: Kamailio 'routing' instance, where dialog is used and billing happens. Kamailio 'register' instance. IC is connected to the routing instance. Customers register to the register instance. Call originates from 'A' via IC through routing instance to register instance and rings customer 'B'. This times out and 'call forwarding' is engaged by sending the call back to the routing instance with a new RURI containing the call forwarding destination 'C'. An appropriate 'no-answer' Diversion Header is added. Now we need two different CDR for this scenario (add more CDR if the call is forwarded multiple times) One from the A to B leg to bill termination fee to the IC carrier. One from B to C to bill the forwarded call with destination 'C' to customer B. The issue I now face is that I only get the kamailio routing instance to generates one dialogue from B to C despite handling two separate connections. How could that challenge be solved? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 2

relay sip-tls vs sips-tcp
by beer Ll 21 Oct '22

21 Oct '22

Hi everyone, I'm using Kamailio as TLS gateway/filter for an internal Asterisk server the network schema is : UAC (tls) --- INTERNET --- (tls) KAMAILIO (sip udp) --- LAN --- (sip udp) ASTERISK with kamailio in multi-homed mode WAN network interface for sip tls LAN network interface for sip udp to asterisk server UAC address 80.0.0.1 KAMAILIO Wan address 80.0.0.2 KAMAILIO Lan address 172.16.0.2 ASTERISK Lan address 172.16.0.3 SIP-TLS call example If the UAC use tls(sip) all works good [image: sip-ok-small.jpeg] SIPS call example If the same UAC use his default settings tls(sips) , there are problems with ACK and BYE packet [image: sip-ko-small.jpeg] the SIP OK SDP packet from kamailio to UAC is 2022/10/10 09:28:47.854721 80.0.0.2:5061 -> 80.0.0.1:49992 SIP/2.0 200 OK Via: SIP/2.0/TLS 192.168.0.1:49992 ;rport=49992;received=80.0.0.1;branch=z9hG4bKM01j360VrBdH5VSV Record-Route: <sip:172.16.0.1:5060 ;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F> Record-Route: <sip:80.0.0.2:5061 ;transport=tls;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F> Call-ID: 1EC2AB679C1EA1BAB60FD03B09F878020B12D3E7 From: <sips:200@pbx.voip.com>;tag=F798336AA08EF9FCFA89D3BDFE0C8C8F To: <sips:*43@pbx.voip.com>;tag=961d0e22-a4f0-453c-9870-6a41578afc96 CSeq: 2 INVITE Contact: <sip:172.16.0.2:5060> P-Asserted-Identity: "xxxxxxxxx" <sips:*43@pbx.voip.com> Content-Type: application/sdp and the UAC send the ACK and BYE from a different tcp port and to: sips:172.16.0.2:5060;transport=tcp 2022/10/10 09:28:48.495365 80.0.0.1:49996 -> 80.0.0.2:5061 ACK sips:172.16.0.2:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TLS 192.168.0.1:49996;branch=z9hG4bKppftdQze20lnwT41;rport Route: <sip:80.0.0.2:5061 ;transport=tls;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F> Route: <sip:172.16.0.1:5060;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F> Max-Forwards: 70 To: <sips:*43@pbx.voip.com>;tag=961d0e22-a4f0-453c-9870-6a41578afc96 From: <sips:200@pbx.voip.com>;tag=F798336AA08EF9FCFA89D3BDFE0C8C8F Call-ID: 1EC2AB679C1EA1BAB60FD03B09F878020B12D3E7 CSeq: 2 ACK kamailio error log WARNING: <core> [core/forward.c:229]: get_send_socket2(): protocol/port mismatch (forced udp:172.16.0.2:5060, to tls:172.16.0.3:5060) How can I solve this ? Best Regards Leo

3 4

After changing instance type for kamailio server in aws log file is corrupted
by Sammy Dominguez 19 Oct '22

19 Oct '22

Hello guys, after changing instance type from t2 to t3a for kamailio server in aws log(/var/log/kamailio.log) file and some other files are corrupted and I can not touch,delete,create or anything and when kamailio start receiving traffic it crashed. Is there any particular reason for this? Basically on t2, they are under /dev/sda1 and /dev/xvda. Under t3 they are nvme disks (biggest part of the speed increase), so they are under /dev/nvme1n1. Note: once I rollback to t2 everything is working as expected.

1 0

Issue with duplicite SIP messages in traffic loop
by Michal Popovič 19 Oct '22

19 Oct '22

Hello, I am trying to configure kamailio to use external antifraud protection using so called "traffic loop" when call is forwarded to external SIP proxy that would response even with same INVITE back or with error message if number is fraud. schema: PBX -> kamailio -> traffic loop -> kamailio -> carrier Kamailio in second and fourth step is the same. Issue is that if kamailio receives same INVITE back to I would like to forward to next carrier it is possible, but kamailio struggles with transaction handling. If ACK message come from PBX, kamailio will correctly forward it to traffic loop but when it comes back, function t_check_trans is failed and I cant relay it. Only workaround is to use send() message, but that causes deny of BYE message from carrier. My question is following: Is it possible to register two different transactions states for same message? If yes how? If no, would it help to send and receive messages on another network port or not? Thank you, Michal

2 1

Recommended openSSL version
by Igor Olhovskiy 19 Oct '22

19 Oct '22

Hello! Due to I still experience irregular Kamailio 5.4 crashes (like 1/month) related to SSL (using websockets and SIPS) I'm wondering, could openSSL upgrade change the situation? As of now in CentOS 7 I have 1.0.2k version. Does anyone have experience to fix crash-related to TLS problems with openSSL upgrade? Or maye some tuneup of TCP parameters can help here?My current setup is quite simple: children=4 enable_tls=yes tcp_accept_no_cl=yes tcp_connection_lifetime=600 tcp_max_connections=998976 # 1000000 - 1024, so we're leaving 1k for system reserve tls_max_connections=998976 Number of clients ~ 200 constantly connected to websocket. -- Best regards, Igor

5 7

IMC module
by Giovanni Iamonte 19 Oct '22

19 Oct '22

HI, I am facing a problem related to the IMC module. I have installed kamailio 5.2.x and I tried to list the members of a room-qsipchat. In my test environment the kamailio.cfg file it is very basic, the kamailio server listen on tcp=192.168.100.192:15006 the client run on a PC with IP address 192.168.100.156 the imc outbound_proxy is set modparam("imc", "db_url", DBURL) modparam("imc", "imc_cmd_start_char", "#") modparam("imc", "outbound_proxy", "sip:sip-test.quintetto.it:15006;transport=tcp") The client start by sending a #join message to kamailio and looking the log file I can see that the room-qsipchat has been create, and also I received the message 200 OK. By using the tcpdump I can see the join message from the Client to kamailio server and the reply from kamailio to the client. When I send the #list message to the room-qsipchat , I expecting to receive the message relate to room's members and the 200 OK message. When I tcp dumping I saw the following messages ******************************************************************** Request from client to list the members - from client to server ******************************************************************** TX 538 bytes Request msg MESSAGE/cseq=57652 (tdta0x77b26460a8) to TCP 192.168.100.192:15006: MESSAGE sip:room-qsipchat@sip-test.quintetto.it SIP/2.0 Via: SIP/2.0/TCP 192.168.100.156:48129;rport;branch=z9hG4bKPjRzgOBgY6vfKZ8Fh8GA4CWheCO840Xzud;alias Max-Forwards: 70 From: <sip:quintetto00@sip-test.quintetto.it>;tag=1is.Ng6ZKJmCT1uAJ9WuJhe3hiQysj9f To: <sip:room-qsipchat@sip-test.quintetto.it> Call-ID: GwOQMLFfIq7o7jvF4OQlJ-jBb2OBaAKY CSeq: 57652 MESSAGE Accept: text/plain, application/im-iscomposing+xml Route: <sip:sip-test.quintetto.it:15006;transport=tcp;lr> Content-Type: text/plain Content-Length: 5 #list ******************************************************************** The server ask for the authentication - from server to client ******************************************************************** 2022-10-17 15:45:25.837 4700-10610/it.quintetto.qsipchat D/pjlib(pjsua_1): RX 574 bytes Response msg 407/MESSAGE/cseq=57652 (rdata0x77b3337330) from TCP 192.168.100.192:15006: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/TCP 192.168.100.156:48129;rport=48129;branch=z9hG4bKPjRzgOBgY6vfKZ8Fh8GA4CWheCO840Xzud;alias;received=192.168.100.156 From: <sip:quintetto00@sip-test.quintetto.it>;tag=1is.Ng6ZKJmCT1uAJ9WuJhe3hiQysj9f To: <sip:room-qsipchat@sip-test.quintetto.it>;tag=1e5fbdd4b14eff95c400342f4634da91.e417 Call-ID: GwOQMLFfIq7o7jvF4OQlJ-jBb2OBaAKY CSeq: 57652 MESSAGE Proxy-Authenticate: Digest realm="sip-test.quintetto.it", nonce="Y01doWNNXHVEyRtiAyN28xJdMjNIG8Ju" Server: kamailio (5.2.8 (x86_64/linux)) Content-Length: 0 ********************************************************************* The client send the authenticated request - from client to server ******************************************************************** TX 755 bytes Request msg MESSAGE/cseq=57653 (tdta0x77b26460a8) to TCP 192.168.100.192:15006: MESSAGE sip:room-qsipchat@sip-test.quintetto.it SIP/2.0 Via: SIP/2.0/TCP 192.168.100.156:48129;rport;branch=z9hG4bKPjj6uuy25r0kKlnICHKQWEaIweOex-sNyq;alias Max-Forwards: 70 From: <sip:quintetto00@sip-test.quintetto.it>;tag=1is.Ng6ZKJmCT1uAJ9WuJhe3hiQysj9f To: <sip:room-qsipchat@sip-test.quintetto.it> Call-ID: GwOQMLFfIq7o7jvF4OQlJ-jBb2OBaAKY CSeq: 57653 MESSAGE Accept: text/plain, application/im-iscomposing+xml Route: <sip:sip-test.quintetto.it:15006;transport=tcp;lr> Proxy-Authorization: Digest username="quintetto00", realm="sip-test.quintetto.it", nonce="Y01doWNNXHVEyRtiAyN28xJdMjNIG8Ju", uri="sip:room-qsipchat@sip-test.quintetto.it", response="d495ec20cb1be8f1b605d4e5b1a4c357" Content-Type: text/plain Content-Length: 5 #list ************************************************************************** The server send the 200 OK message - from server to client ******************************************************************** RX 447 bytes Response msg 200/MESSAGE/cseq=57653 (rdata0x77b3337330) from TCP 192.168.100.192:15006: SIP/2.0 200 ok Via: SIP/2.0/TCP 192.168.100.156:48129;rport=48129;branch=z9hG4bKPjj6uuy25r0kKlnICHKQWEaIweOex-sNyq;alias;received=192.168.100.156 From: <sip:quintetto00@sip-test.quintetto.it>;tag=1is.Ng6ZKJmCT1uAJ9WuJhe3hiQysj9f To: <sip:room-qsipchat@sip-test.quintetto.it>;tag=1e5fbdd4b14eff95c400342f4634da91.365f Call-ID: GwOQMLFfIq7o7jvF4OQlJ-jBb2OBaAKY CSeq: 57653 MESSAGE Server: kamailio (5.2.8 (x86_64/linux)) Content-Length: 0 ********************************************************************************* Till here everything is OK. But regarding the members when a TCP dumping I saw a message with the room's members that go from the kamailio server to the kamailio server ********************************************************************************* MESSAGE sip:quintetto00@sip-test.quintetto.it SIP/2.0 Via: SIP/2.0/TCP 192.168.100.192:15006;branch=z9hG4bKf20a.be253b92000000000000000000000000.0 To: <sip:quintetto00@sip-test.quintetto.it> From: <sip:room-qsipchat@sip-test.quintetto.it>;tag=03e4c67b0f95c0aeb1f633479b4aa5cd-72f2 CSeq: 10 MESSAGE Call-ID: 37db44aa523ea24a-6567(a)192.168.100.192 Max-Forwards:.70 Content-Length: 47 User-Agent: kamailio (5.2.8 (x86_64/linux)) Content-Type: text/plain Members: *sip:quintetto00@sip-test.quintetto.it ************************************************************************************** a reply from the kamailio server to the kamailio server, with an authenticated request ************************************************************************************** SIP/2.0.407.Proxy Authentication.Required Via: SIP/2.0/TCP 192.168.100.192:15006;branch=z9hG4bKf20a.be253b92000000000000000000000000.0 To: <sip:quintetto00@sip-test.quintetto.it>;tag=1e5fbdd4b14eff95c400342f4634da91.cfc3 From: <sip:room-qsipchat@sip-test.quintetto.it>;tag=03e4c67b0f95c0aeb1f633479b4aa5cd-72f2 CSeq: 10 MESSAGE Call-ID: 37db44aa523ea24a-6567(a)192.168.100.192 Proxy-Authenticate: Digest realm="sip-test.quintetto.it", nonce="Y01fQmNNXhbsHdM1THmENyJn1MH/VD/8" Server: kamailio (5.2.8 (x86_64/linux)) Content-Length:.0 ************************************************************************************** bottom of line the outbound_proxy request for an authenticated request, but kamailio do not reply to this request. Could you please point me in the right direction? Thanks a lot for any help. Bye -- ------------------------------------------------------------------------ Ing. Giovanni Iamonte Developments and technologies area Quintetto Srl Via Monte Navale, 1 10015 - Ivrea (TO)  mobile: +39 393 9196310  tel: +39 0165 1845290  e-mail: giovanni.iamonte(a)quintetto.it  web: www.quintetto.it

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ClueCon and ... FreeSWITCH 1.10.8/RHEL8 packages!
by Richard Chan 18 Oct '22

18 Oct '22

Hi sr-users, Since ClueCon is going on right now, and some Kamailio community members are involved I hope you will forgive a little promo here: I have a RHEL8 build of the just released 1.10.8 on Copr: https://copr.fedorainfracloud.org/coprs/beaveryoga/FreeSWITCH-1.10.8/ Hope you will find it useful in your kamailio + Class 5 scenarios! Cheers Shih-Ping

1 0

Testing dialog module (and acc with dialog cdrs)
by Daniel-Constantin Mierla 18 Oct '22

18 Oct '22

Hello, as I am not a user of dialog variables, I am turning to community to ask for help testing the current master branch with configurations that make use of dialog variables and acc dialog-based cdr generation. With a few reports of issues related to dialog modules and unexpected crashes, I looked over the code and noticed that the access of the value for dialog variables was not protected, making them vulnerable of invalid memory access in case of the variable was updated by another process or dialog was terminated. I introduced a couple of new functions to try to cover different use cases of getting the dlg variable values, dialog management code was not affected, but given that these commits need to be backported to stable branch (5.6), I want to get proper feedback from community that things work fine. A previous attempt of a simpler fix was not enough, having side effects to acc module for dialog-based cdr generation, because it was keeping referenced to many dlg variables at the same time. In short, it would be appreciated any feedback on testing dialog and acc with dialog-based cdr generation using git master branch. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online Nov 7-10, 2022 (Europe Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

3 13

kamailio & rtpengine question
by Gerry Kernan 15 Oct '22

15 Oct '22

Hi Looking for advise if anyone could point out where im going wrong. My setup is Kamailio Ver 5.6.0 running in azure , rtpengine Version: git-master-dc395d5, behind kamailio I have asterisk ver 18.12.1 Kamailio conf I have interface set as below listen=udp:10.3.1.6:5060 advertise WANIP:5060 and rtpengine.conf set as below interface = eth0/10.3.1.5 but when I can make a call I don't get any audio, rtpengine.log extract below shows that rtp is going to the private IP rather that the Public IP of the phone that initiated the call, do I need to set rtpengine_manage() to fix this? Oct 7 09:38:58 kamailio-proxy rtpengine[8127]: INFO: [NmE0MjdhMzA4NjUzYWVjMmRjYzIxNGJjMTY4ZTc2MWE. port 30278]: [core] Confirmed peer address as 10.3.1.5:12890 Oct 7 09:38:58 kamailio-proxy rtpengine[8127]: INFO: [NmE0MjdhMzA4NjUzYWVjMmRjYzIxNGJjMTY4ZTc2MWE. port 30278]: [core] Kernelizing media stream: 10.3.1.5:12890 -> 10.3.1.6:30278 | 10.3.1.6:30298 -> 192.168.0.123:40042 Oct 7 09:39:00 kamailio-proxy rtpengine[8127]: INFO: [NmE0MjdhMzA4NjUzYWVjMmRjYzIxNGJjMTY4ZTc2MWE. port 30279]: [core] Confirmed peer address as 10.3.1.5:12891 Kind regards gerry Gerry Kernan | Technical Director Mobile: +353(0)86-170 9790 [Graphical user interface, website Description automatically generated]<https://www.infinityit.ie/> [A picture containing text, clipart Description automatically generated,Icon Description automatically generated]

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