- sr-users - kamailio.org

sql_query fails from event_route
by Juha Heinanen 07 May '24

07 May '24

I noticed that sql_query does not (anymore) work when called from event route: event_route [tcp:closed] { # Handle TCP connection close # Delete possible leftover registration if ($conid) sql_query("sip_proxy_cache", "DELETE FROM location WHERE server_id=0 AND connection_id=$conid", "result"); } The query produces errors: 2024-05-07T11:30:14.785915+03:00 siika /usr/bin/sip-proxy[2848078]: CRITICAL: sqlops [sqlops.c:213]: sql_check_connection(): no database handle with … [View More]

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How to use environment variable
by Pavan Kumar 07 May '24

07 May '24

Hi everyone, I am trying to assign environment variable as follows listen=udp:0.0.0.0:5060 advertise $env(MY_IP):5060 Looks like using the environment variable as above is an invalid configuration. Is there a way to use IP from env var to advertise. Even better, is there a way to use result in a stun query as an advertised address? Thank you, Pavan Kumar

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Retransmission behaviour after ACK
by Sergio Charrua 06 May '24

06 May '24

Hi all! I'm testing some redirect scenarios with Kamailio 5.7.4 and using Astterisk 13 and SIPp for unit testing. The kamailio's script is simple and uses TM, HTTP_ASYNC_MODULE, JSONRPC, JANSSON, RTJSON and XLog modules. Request route is simple: request_route { route(HANDLE_OPTIONS); route(HANDLE_RETRANS); route(HANDLE_CANCEL); route(REQINIT); route(HANDLE_INVITE); } where HANDLE_RETRANS is: route[HANDLE_RETRANS]{ # handle retransmissions xlog("L_INFO","… [View More]HANDLE_RETRANS - message type is $mt for method $rm"); if (!is_method("ACK")) { if(t_precheck_trans()) { xlog("L_INFO","HANDLE_RETRANS - Precheck Trans"); t_check_trans(); xlog("L_INFO", "HANDLE_RETRANS - Exiting after Retransmission check - method $rm"); exit; } xlog("L_INFO","HANDLE_RETRANS - Check Trans"); t_check_trans(); } } and HANDLE INVITE is: route[HANDLE_INVITE]{ if (is_method("INVITE")) { xlog("L_INFO","MAIN - calling TO_CARRIER"); setflag(FLT_ACC); route(TO_CARRIER); exit; } } The HTTP requests call a REST API returning a JSON object, that is iterated and produces a CONTACT header containing multiple choices using the append_branch() command. The RELAY route is : route[RELAY] { # Sends a 300 Multiple Choices back to the proxy that requested the routing lookup xlog("L_INFO","REPLY_302 - send reply"); send_reply("300", "Multiple Choices"); xlog("L_INFO","REPLY_302 - exit"); exit; } Which returns indeed a Contact header with multiple contact choices. When using Asterisk to originate a call to Kamailio, the SIP call flow is as follows and terminates (SNGrep and TCPDump do not detect anything else): SRC DST INVITE (SDP) ──────────────────────────> 100 trying -- your call is <────────────────────────── 300 Multiple Choices <────────────────────────── ACK ──────────────────────────> However when using SIPp I get some retransmissions after the final ACK. SRC DST INVITE (SDP) ──────────────────────────> 100 trying -- your call is <────────────────────────── 300 Multiple Choices <────────────────────────── ACK ──────────────────────────> 300 Multiple Choices <<<──────────────────────── 300 Multiple Choices <<<──────────────────────── 300 Multiple Choices <<<──────────────────────── SIPp scenario is : <?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd"> <scenario name="Basic Sipstone UAC">   <send retrans="500" start_rtd="true" rrs="true"> <![CDATA[ INVITE sip:[call_number]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] Max-Forwards: 70 From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number] To: <sip:[call_number]@[remote_ip]:[remote_port]> Contact: sip:sipp@[local_ip]:[local_port];transport=[transport]> Call-ID: [call_id] CSeq: 102 INVITE User-Agent: SIPp Supported: replaces, timer Content-Type: application/sdp Subject: Performance Test Session-Expires: 3600;refresher=uas Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=SIPp c=IN IP[media_ip_type] [media_ip] t=0 0 m=audio [media_port] RTP/AVP 0 a=rtpmap:0 PCMU/8000 ]]> </send> <recv response="100" rss="true"> </recv> <recv response="300" rtd="true" rss="true"> <action> <ereg regexp="\<([a-zA-Z0-9@.=;]+)[^>]+" search_in="hdr" header="Contact: " assign_to="1"/> <log message="INFO: 300 Contact Header: [$1]" /> <exec command="echo [$1] >> ./300_ContactHeader.txt" /> </action> <action> <ereg regexp="sip:.*@([0-9A-Za-z\.]+):([0-9]+);transport=([A-Z]+)" search_in="hdr" header="Contact:" check_it="true" assign_to="dummy,host,port,transport" /> </action> </recv>   <send rrs="true"> <![CDATA[ ACK sip:[call_number]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] Max-Forwards: 70 From: <sip:sipp@[local_ip]:[local_port]>;tag=[call_number] To: <sip:[call_number]@[remote_ip]:[remote_port]>[peer_tag_param] Contact: <sip:sipp@[local_ip]:[local_port]> Call-ID: [call_id] CSeq: 102 ACK User-Agent: SIPp Content-Length: 0 ]]> </send>   </scenario> The question is : is there anything wrong with my kamailio script? or with SIPp scenario? why does kamailio have different behaviours between Asterisk and SIPp? Any option to disable retransmissions on Kamailio ? (using UDP) Thanks in advance, *Sérgio Charrua* [View Less]

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topos with mask_callid
by Marrold 05 May '24

05 May '24

Hi all, I am using Kamailio 5.8.1 on a Debian 12 machine with topos and topoh hiding the topology. I have observed with modparam("topos", "mask_callid", 1), in-dialog requests are relayed with the incorrect caller-id. In the scenario, A calls B. The call is set up and the call-id for the B-Leg is successfully masked. Then, when A sends a BYE, the BYE is incorrectly relayed to B with the call-id of the A leg, and B replies with a 481 If I disable masking the call-id, or disable topos/topoh … [View More]

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Re: IMS SCSCF redundancy
by Oleg Belousov 03 May '24

03 May '24

Hi. I have a question re: IMS SCSCF redundancy.:There are several SCSCF instances, registration happens via SCSCF1, it goes down after that. UE make MO call.. PCSCF resolves the common SCSCF fqdn and sends an invite to available SCSCF2 which does not have UE profile as well as AS address (which is received during registration SAR/SAA in iFC). There are 2 options here, either to release the call (as per 3gpp spec upon specific release code, eg 408/503 UE should perform re-registration - not … [View More]

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http_async_client and sl_send_reply()
by Masud Muborakshohi 02 May '24

02 May '24

Hello! I have a problem when sending a response sl_send_reply("415", "The other party does not support video calling!"); in route[HTTP_REPLY]. When I make a request using http_async_query() to send a push notification to a mobile device, I need to process the response from the push server and respond with a SIP message to the client. Before sending the request I do $http_req(suspend) = 0; What am I doing wrong? Why are $fU and $ru equal to null ? #processing HTTP responses route[HTTP_REPLY] { … [View More]

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No ACK from Kamailio
by Arseniy Moskvich 02 May '24

02 May '24

Hi everyone. We are new to Kamailio and using Kamailio 5.8 with FreeSwitch. We have the issue with the incoming call from our provider. They are dropping after 30 seconds. The client doesn't receive ACK from Kamailio. But FreeSwitch sends ACK to Kamailio. The outbound call is working fine. At the moment flow is our provider -> FreeSwitch -> Kamailio -> our web client. Could someone give us some directions please? Please find attached Kemi.py. Thank you so much in advance

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Kamailio works but voice is not present during the calls!
by christian.marinelli＠hotmail.it 02 May '24

02 May '24

Hi all, i have a problem with my Kamailio SIP Server. I set up a Kamailio SIP server in a virtual machine on a private network and i connect to this with a WireGuard VPN. The problem is that i can connect to the SIP server throught different clients and users and i can call the other users, the devices ring and i can answer to the calls but unfortunatly there is no audio during the call :( I can't understand why, it seems all si OK, a year ago i set up another SIP server with the same … [View More]

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AEM Implementation Services,
by surekhatech.seo＠gmail.com 02 May '24

02 May '24

As an Official Adobe Bronze Solution Partner, we specialize in AEM development services, consulting, implementation, migration, and cloud services for your business.

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Question: if SIP Timer L and Timer M from RFC6026 has been defined at Kamailio
by jennifer yang 02 May '24

02 May '24

Hello Can anyone help answer these questions: - if SIP Timer L and Timer M from RFC6026 has been defined at Kamailio (version v5.5.x)? Timer L: Wait time for accepted INVITE request retransmits Timer M: Wait time for retransmission of 2xx to INVITE or additional 2xx from other branches of a forked INVITE - if no specific Timer defined, which timer will be used instead? thanks br jenni

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Issue with serial call forking to same destination
by Mathias 30 Apr '24

30 Apr '24

Hi I've an issue with serial call forking and I hope somebody can help me... What I'm trying to reach: We are using Kamailio in a dynamic network with multiple nodes. The dialplan is implemented using the KEMI interface. The call routing is completely dynamic. At the moment, we are having call groups, which may contain 1 or more recipients. If the number of this call group is called, Kamailio is doing a parallel forking to all group participants. Now I'm trying to extend the feature, that if … [View More]nobody hooks up, the call is forwarded to another number. This has been implemented with serial forking. So the dialplan adds branches for all call group members with a q value of 1 and the last branch is the failover number with a lower q value. The branching and failure route works fine. The problem: * I have a call group (G Bob 900101), with one participant (Bob 200102). * The group has a fail over number (if Bob doesn't hook up), which points again to Bob. I know that this use case doesn't make much sense, but it has been simplified to show the problem. The failover number can be anything also another call group which may contain one or more recipients from the first group again. * Alice (200101) calls the call group * The first node (10.0.1.1/10.64.0.1) receives the call and forks it to branch 1 Bob (202102) with q value 1.0 and to a second branch Bob (202102) with q value 0.5. fr_timer is set to 10s * The call is forwarded to node 2 (10.0.2.1/10.64.0.2), where Bob is registered. * The call is forwarded to Bobs device (Grandstream GXV3350) and the phone is ringing. * After 10s, the first node detects the 408 Request Timeout and switches to the second branch * The invite is sent to node 2 and Bobs device again. * Node 1 also sends a CANCEL to node 2 and Bobs device for the first branch. * The call in Bobs device is terminated * Now the problem, the device does not react to the second INVITE. And after the fr_inv_timer expires, the call is terminated with a 408 Request Timeout. I can see that the second INVITE contains another branch ID <ID>.1 compared to the first INVITE, which had <ID>.0. The CANCEL is also sent with <ID>.0. I've also tried with a Cisco 8861 3PCC and it doesn't work either. Compared to the Grandstream, the Cisco device returns an error 500 upon the second INVITE. Since both devices do not accept what I'm trying, I guess I'm doing something wrong... At the moment I'm a bit lost, what the problem could be. Does somebody have an idea what it is and how I could solve it? Following two SIP dumps from node 1 and node 2: Node 1: |Time | 10.0.1.128 | 10.64.0.1 | | | | 10.0.1.1 | | 10.0.2.1 | |2024-01-31 14:17:07.792784| INVITE SDP (g711A g7 | | |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> Call-ID:2102072978-5060-296@BA.A.B.BCI CSeq:2950 | |(47276) ------------------> (5061) | | | |2024-01-31 14:17:07.793060| 407 Proxy Authentica | | |SIP Status 407 Proxy Authentication Required | |(47276) <------------------ (5061) | | | |2024-01-31 14:17:07.796613| ACK | | | |SIP ACK From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> CSeq:2950 | |(47276) ------------------> (5061) | | | |2024-01-31 14:17:07.799045| INVITE SDP (g711A g7 | | |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> Call-ID:2102072978-5060-296@BA.A.B.BCI CSeq:2951 | |(47276) ------------------> (5061) | | | |2024-01-31 14:17:07.836471| 100 trying -- your c | | |SIP Status 100 trying -- your call is important to us | |(47276) <------------------ (5061) | | | |2024-01-31 14:17:07.865219| | | INVITE SDP (g711A g7 |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5061) | |2024-01-31 14:17:08.036334| | | 100 trying -- your c |SIP Status 100 trying -- your call is important to us | | | |(5061) <------------------ (5061) | |2024-01-31 14:17:08.084421| | | 180 Ringing |SIP Status 180 Ringing | | | |(5061) <------------------ (5061) | |2024-01-31 14:17:08.096834| 180 Ringing | | |SIP Status 180 Ringing | |(47276) <------------------ (5061) | | | |2024-01-31 14:17:18.099644| | | INVITE SDP (g711A g7 |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5061) | |2024-01-31 14:17:18.100099| | | CANCEL | |SIP CANCEL From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5061) | |2024-01-31 14:17:18.259375| | | 100 trying -- your c |SIP Status 100 trying -- your call is important to us | | | |(5061) <------------------ (5061) | |2024-01-31 14:17:18.274841| | | 200 canceling |SIP Status 200 canceling | | | |(5061) <------------------ (5061) | |2024-01-31 14:17:18.296942| | | 487 Request Terminat |SIP Status 487 Request Terminated | | | |(5061) <------------------ (5061) | |2024-01-31 14:17:18.298630| | | ACK | |SIP ACK From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5061) | |2024-01-31 14:17:20.281602| | | 408 Request Timeout |SIP Status 408 Request Timeout | | | |(5061) <------------------ (5061) | |2024-01-31 14:17:20.283433| | | ACK | |SIP ACK From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5061) | |2024-01-31 14:17:20.307343| 408 Request Timeout | | |SIP Status 408 Request Timeout | |(47276) <------------------ (5061) | | | |2024-01-31 14:17:20.309097| ACK | | | |SIP ACK From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> CSeq:2951 | |(47276) ------------------> (5061) | | | Node 2: |Time | 10.64.0.1 | 10.64.0.2 | | | | 10.0.2.1 | | 10.0.2.128 | |2024-01-31 14:17:07.869305| INVITE SDP (g711A g7 | | |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> Call-ID:2102072978-5060-296@BA.A.B.BCI CSeq:2951 | |(54454) ------------------> (5061) | | | |2024-01-31 14:17:08.036009| 100 trying -- your c | | |SIP Status 100 trying -- your call is important to us | |(5061) <------------------ (5061) | | | |2024-01-31 14:17:08.050848| | | INVITE SDP (g711A g7 |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5060) | |2024-01-31 14:17:08.054357| | 100 Trying| | |SIP Status 100 Trying | | |(5061) <-------------------------------------- (5060) | |2024-01-31 14:17:08.075422| | 180 Ringing | |SIP Status 180 Ringing | | |(5061) <-------------------------------------- (5060) | |2024-01-31 14:17:08.083887| 180 Ringing | | |SIP Status 180 Ringing | |(5061) <------------------ (5061) | | | |2024-01-31 14:17:18.100359| INVITE SDP (g711A g7 | | |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> Call-ID:2102072978-5060-296@BA.A.B.BCI CSeq:2951 | |(54454) ------------------> (5061) | | | |2024-01-31 14:17:18.258947| 100 trying -- your c | | |SIP Status 100 trying -- your call is important to us | |(5061) <------------------ (5061) | | | |2024-01-31 14:17:18.273819| | | INVITE SDP (g711A g7 |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5060) | |2024-01-31 14:17:18.274460| CANCEL | | | |SIP Request CANCEL CSeq:2951 | |(54454) ------------------> (5061) | | | |2024-01-31 14:17:18.274550| | | CANCEL | |SIP CANCEL From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5060) | |2024-01-31 14:17:18.274615| 200 canceling | | |SIP Status 200 canceling | |(5061) <------------------ (5061) | | | |2024-01-31 14:17:18.277438| | 200 OK | | |SIP Status 200 OK | | |(5061) <-------------------------------------- (5060) | |2024-01-31 14:17:18.279335| | 487 Request Terminated | |SIP Status 487 Request Terminated | | |(5061) <-------------------------------------- (5060) | |2024-01-31 14:17:18.280785| | | ACK | |SIP ACK From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5060) | |2024-01-31 14:17:18.296312| 487 Request Terminat | | |SIP Status 487 Request Terminated | |(5061) <------------------ (5061) | | | |2024-01-31 14:17:18.298900| ACK | | | |SIP ACK From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> CSeq:2951 | |(54454) ------------------> (5061) | | | |2024-01-31 14:17:20.280965| 408 Request Timeout | | |SIP Status 408 Request Timeout | |(5061) <------------------ (5061) | | | |2024-01-31 14:17:20.283730| ACK | | | |SIP ACK From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> CSeq:2951 | |(54454) ------------------> (5061) | | | At the moment, I'm using Kamailio 5.5.4, but I also did a short try with a newer version (I guess it was 5.7.4). Both versions showed the same issue. Best regards Mathias [View Less]

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Sudden TCP errors and shared memory spikes
by David Cunningham 29 Apr '24

29 Apr '24

Hello, We have a production Kamailio 5.2.7 server which has suddenly started reporting spikes in the shared memory use. When looking into this, we noticed that at the same time as the memory spikes Kamailio also logs a huge amount of errors like the following. Can anyone help with an explanation for what's going on? The xx.xx.52.202 address is the Kamailio server itself. We record historical numbers of network connections for the server overall, and that doesn't show any increase in the … [View More]

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tls_threads_mode and app_python3 module
by Ihor Olkhovskyi 29 Apr '24

29 Apr '24

Hello, Just to confirm, as I quickly checked into a code of 5.8.x branch, there were no modification on app_python3 module to comply with tls_threads_mode mechanism? Asking this beforehand posting a bug, that if using a quite simple script into module, with tls_threads_mode = 2 Kamailio just hangs at the start and become unresponsive. And by simple I mean something like this class PyKamailio: def __init__(self): self.test = 1 def child_init(self, rank): """Executed … [View More]

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RTP Engine with Kamailio hold issue
by Vishal Pai 29 Apr '24

29 Apr '24

Hello Everyone I am using kamailio with RTP engine for wss to udp to connect my asterisk server. It all works well like wss to UDP registration via kamailio to asterisk. Here I am facing an issue with call hold i.e if I hold the call from B leg it gets disconnected. Any advice why it get failed and hangups Thanks

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Error by installing new Rel. 5.8
by lm.eng＠live.com 29 Apr '24

29 Apr '24

Hello,, I am using this docker file for installaing kamailio. I wana use it with open5gs for trynig VoNR , I used before it for VoLTE it works but right now I have this error in installation process. icscf | 0(34) ERROR: <core> [core/sr_module.c:527]: ksr_locate_module(): could not find module <maxfwd> in </usr/lib64/kamailio/modules_k/:/usr/ lib64/kamailio/modules/:/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/:/usr/lib/x86_64-… [View More]

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Understanding NAT Flags
by Benoit Panizzon 29 Apr '24

29 Apr '24

Hi Gang Somehow I don't get my head around NAT Flags and the nathelper module https://www.kamailio.org/docs/modules/5.7.x/modules/nathelper.html In the examples I found, there is: FLT_NATS and FLB_NATB If I got it right, FLB_NATB is a branch flag, which shall indicate that the device is 'B'ehind NAT, right? It is being set, when FLT_NATS is set: if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); } But when should FLT_NATS be set and what is it's meaning? The … [View More]

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Outbound Proxy/Forwarder
by Blake Ivey 29 Apr '24

29 Apr '24

Hi everyone, New user who is trying to make an outbound proxy to route a lot of freepbx boxes for outbound calls. I got this setup and working using dsiprouter. Calls come in from freepbx trunk using outbound proxy, authenticated using IP auth, and then gets signed with our stirshaken cert, then routes out to our carriers. Interested in using plain kamailio. Got everything compiled and libstirshaken compiled. Just wanted to see if someone could point me in the right direction for the cfg … [View More]

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How to identify the instances of a looping call (including all replies and on all transactions)
by Benoit Panizzon 29 Apr '24

29 Apr '24

Hi List I have stumbled over this challenge: I have a kamailio server without dialog module. Only tmx module, acting as registrar / rtpengine host. If the call is routed over that server once, this works fine. If the call is routed more than once (call from one location to another on same registrar, call forwarding) then to correctly work, I would need to engage rtpengine in a way to tell rtpengine to handle both call legs separately. By default, rtpengine identifies one call by Call-ID,… [View More]

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Using SQLite in wal mode with synchronous=normal
by Tim Anderson 28 Apr '24

28 Apr '24

Hi, I see the documentation for the sqlite module includes a parameter to set the journal_mode to wal, but I don't see any way to set synchronous=normal. Is there a way to run arbitrary PRAGMA statements upon opening the file or would I have to run it before each query?

1 0

SDP getting modified.
by Pavan Kumar 28 Apr '24

28 Apr '24

Hi everyone, Here is my setup SIP Client (MacBook) <=> Kamailio (AWS) <=> Freeswitch (AWS) When I initiate a call from Client, audio won't be sent out, but the client can receive audio. (When the client receives the call everything works fine). I started to trace the SIP message using tdpdump. And here is an interesting thing. Kamailio sends the following SDP message while sending 200 OK v=0 o=FreeSWITCH 1714278184 1714278185 IN IP4 *<freeswitch-ip-address>* s=FreeSWITCH … [View More]

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Kamailio in Active/Passive mode
by ramaseshi.kolli＠gmail.com 28 Apr '24

28 Apr '24

Hi All, If run Kamailio in Active/Passive mode lets say there are 100 transactions going on Active node and then Active went down, now i want passive to be aware of all those ongoing transactions so that it can handle ongoing calls transactions smoothly so how to store them in db? Is there any way to do so? i know there is module dialog for tracking calls etc but i just want to handle transactions. Thanks in advance. Ram

2 3

Wrong code in the websocket examples cfg
by Pavan Kumar 27 Apr '24

27 Apr '24

Hello everyone, While going through the WebRTC example configuration (cfg) to better understand and implement it in my setup, I came across a potential issue in the code block. The code in question can be found at this link: https://github.com/kamailio/kamailio/blob/465994de2859c7863b4cef8457be0a207… In the code, there is an "if else" block, "if(loose_route())" that seems to be incorrect. It appears that the "if" block will always evaluate to true, given that "loose_route" never returns 0. … [View More]

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Kamailio, Asterisk and Parked calls
by Social Boh 26 Apr '24

26 Apr '24

Hello list, I'm using Kamailio to balance SIP request between two Asterisk. I have a problem with Parking calls using Asterisk Application. When a user parking a call, some times can't pick up parked call because the call to pick up the Call go to the other Asterisk. I'm using the 1 algorithm "hash over from URI" without success. Any hint? -- --- I'm SoCIaL, MayBe

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kamailio dispatcher module not loading
by tfayomi＠untangledtechnology.com 26 Apr '24

26 Apr '24

Hello, I'm trying to set up the dispatcher module from kamailio, but I'm running into errors just loading the dispatcher module and setting the params in my config file and would like to know where I'm going wrong. I've added destinations through kamctl as well and whenever I try to use dispatcher reload or dispatcher dump, I get a 500 error saying neither of those methods are found. I'm using kamailio version 5.6.3 and these are the errors I'm getting when starting up kamailio: /usr/sbin/… [View More]

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optimization in process
by Alexis Fidalgo 26 Apr '24

26 Apr '24

Hello all, little and fast context, we have 8 docker swarm clusters (4 nodes each) where there are deployed as global service 4 kamailio per cluster. Each cluster runs 8 replicas of a call processing service/app (for the sake of this question we’ll assume the performance of this app is ok, no problems on what it does and how much it consumes in cpu/memory). kamailio is configured to use http_client module to query the processing app (those queries are load balanced by the docker swarm cluster) … [View More]in a very simple way, added below in this email. Everything goes well, there’s an average of ~450 caps on each kamailio on peak hours. Question and reason of this email is: Is this the best approach i can take? (the http_client) or maybe there’s a more efficient one for this scenario? maybe kemi, or other? The processing app is written in golang. We’re planning to add more traffic (significantly more) to this instances and i would like to know if there’s a better scenario, to explore it before this migration. Thanks, regards — request_route { route(AF_SL); } route[AF_SL] { if(is_method("ACK")){ exit; } if(is_method("REGISTER")){ sl_reply("200", "OK"); exit; } if(is_method("OPTIONS")){ sl_reply("200", "OK"); exit; } if(is_method("CANCEL")){ sl_reply("200","OK"); sl_reply("487","Request Terminated"); exit; } if(is_method("INVITE")){ if ($hdr(X-Carrier-In)!="" && $hdr(Organization)!="" && $hdr(X-SBC-Type)!="") { sl_reply("100","Trying"); $var(retCode) = http_client_query("http://cbrcre:3001/cbrcre/v1/callHandler",'xsbct=$(hdr(X-SBC-Type){s.escape.param})&org=$(hdr(Organization){s.escape.param})&xci=$(hdr(X-Carrier-In){s.escape.param})&fu=$(fu{s.escape.param})&ru=$(ru{s.escape.param})&rU=$(rU{s.escape.param})&fU=$(fU{s.escape.param})&ci=$(ci{s.escape.param})&cin=$(hdr(Call-Info){s.escape.param})&ih=$hdr(Identity)&tag=$(ft{s.escape.param})&ct=$(ct{s.escape.param})&via=$(hdr(Via)[0]{s.escape.param})', "$var(result)"); xlog("L_INFO", "*************** RESULT: $var(result)\n\n"); jansson_get("response_code", "$var(result)", "$var(rc)"); switch($var(rc)){ case "200": if jansson_get("contact", "$var(result)", "$var(ct)") == true { append_branch("$var(ct)"); } if jansson_get("callinfo", "$var(result)", "$var(cinfo)") == true { append_to_reply("Call-Info: $var(cinfo)\r\n"); } if jansson_get("pai", "$var(result)", "$var(pai)") ==true { append_to_reply("P-Asserted-Identity: $var(pai)\r\n"); } if jansson_get("xfrom", "$var(result)", "$var(xfrom)") ==true { append_to_reply("X-From: $var(xfrom)\r\n"); } if jansson_get("identity", "$var(result)", "$var(identity)") ==true { append_to_reply("Identity: $var(identity)\r\n"); } append_to_reply("X-Handled: ServiceBroker\r\n"); sl_send_reply("302", "Redirect"); break; default: jansson_get("reason","$var(result)","$var(reason)"); jansson_get("sbReason","$var(result)","$var(sbreason)"); append_to_reply("X-SBReason: $var(sbreason)\r\n"); sl_send_reply("$var(rc)", "$var(reason)"); } exit; } xlog("L_ERR","missing xci or org or sbct"); sl_send_reply("608", "SBK Rejected"); } } [View Less]

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ERROR: <core> [core/ppcfg.c:275]: pp_ifdef_level_error(): different number of preprocessor directives: 1 more #!if[n]def as #!endif
by Tristan Ferreira 26 Apr '24

26 Apr '24

Hi, I am trying to learn Kamailio to set up a SIP/RTP proxy for WebRTC with FreeSWITCH, and I wanted to use this configuration file template as a starting point. (Source: https://github.com/havfo/WEBRTC-to-SIP) But i have the error : *ERROR: [core/ppcfg.c:275]: pp_ifdef_level_error(): different number of preprocessor directives: 1 more #!if[n]def as #!endif* I don't understand because I checked and everything seems okay with the if/endif statements to me. |# grep '#!ifndef' kamailio.… [View More]cfg | wc -l 2 # grep '#!ifdef' kamailio.cfg | wc -l 49 # grep '#!endif' kamailio.cfg | wc -l 51 | And it's ok for chat gpt too : |#!ifdef : 49 occurrences #!ifndef : 2 occurrences Total (#!ifdef and #!ifndef) : 51 occurrences #!endif : 51 occurrences | Is there a problem with the parser or a compatibility problem (i am on*5.6.3*) ? |#!KAMAILIO # #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_TLS #!define WITH_HOMER #!define WITH_WEBSOCKETS #!define WITH_ANTIFLOOD #!define WITH_IPV6 ##!define WITH_BRIDGE_ON_FAIL #!define WITH_LOCALHOST_WS ##!define WITH_LOCALHOST_SIP #!substdef "!MY_SIP_PORT!5060!g" #!substdef "!MY_SIPS_PORT!5061!g" #!substdef "!MY_WS_PORT!8080!g" #!substdef "!MY_WSS_PORT!4443!g" #!substdef "!MY_IP4_ADDR!XXXXX-XXXXX!g" #!substdef "!IP4_LOCALHOST!127.0.0.1!g" #!substdef "!MY_WS4_ADDR!tcp:MY_IP4_ADDR:MY_WS_PORT!g" #!substdef "!MY_WSS4_ADDR!tls:MY_IP4_ADDR:MY_WSS_PORT!g" #!substdef "!LOCALHOST_WS4_ADDR!tcp:IP4_LOCALHOST:MY_WS_PORT!g" #!substdef "!LOCALHOST_WSS4_ADDR!tls:IP4_LOCALHOST:MY_WSS_PORT!g" #!ifdef WITH_IPV6 #!substdef "!MY_IP6_ADDR![XXXXXX-XXXXXX]!g" #!substdef "!IP6_LOCALHOST![::1]!g" #!substdef "!MY_WS6_ADDR!tcp:MY_IP6_ADDR:MY_WS_PORT!g" #!substdef "!MY_WSS6_ADDR!tls:MY_IP6_ADDR:MY_WSS_PORT!g" #!substdef "!LOCALHOST_WS6_ADDR!tcp:IP6_LOCALHOST:MY_WS_PORT!g" #!substdef "!LOCALHOST_WSS6_ADDR!tls:IP6_LOCALHOST:MY_WSS_PORT!g" #!endif #!substdef "!MY_DOMAIN!XXXX-XXXX!g" # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 #!define FLB_RTPWS 8 #!define FLB_IPV6 9 #!define FLB_V4V6 10 #!define FLB_BRIDGE 11 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 port=MY_SIP_PORT tls_port_no=MY_SIPS_PORT #!ifdef WITH_TLS enable_tls=yes #!endif listen=MY_IP4_ADDR #!ifdef WITH_LOCALHOST_SIP listen=IP4_LOCALHOST #!endif #!ifdef WITH_IPV6 listen=MY_IP6_ADDR #!ifdef WITH_LOCALHOST_SIP listen=IP6_LOCALHOST #!endif #!endif #!ifdef WITH_WEBSOCKETS listen=MY_WS4_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WS4_ADDR #!endif #!ifdef WITH_IPV6 listen=MY_WS6_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WS6_ADDR #!endif #!endif #!ifdef WITH_TLS listen=MY_WSS4_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WSS4_ADDR #!endif #!ifdef WITH_IPV6 listen=MY_WSS6_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WSS6_ADDR #!endif #!endif #!endif #!endif use_dns_cache = on # Use KAMAILIO internal DNS cache use_dns_failover = on # Depends on KAMAILIO internal DNS cache dns_srv_loadbalancing = on # dns_try_naptr = on # dns_retr_time=1 # Time in seconds before retrying a DNS request dns_retr_no=3 # Number of DNS retransmissions before giving up # Set protocol preference order - ignore target priority dns_naptr_ignore_rfc= yes # Ignore target NAPTR priority dns_tls_pref=50 # First priority: TLS dns_tcp_pref=30 # Second priority: TCP dns_udp_pref=10 # Third priority: UDP tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_rd_buf_size=16384 # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules/" #!else mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "sdpops.so" loadmodule "textopsx.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_HOMER loadmodule "siptrace.so" #!endif #!ifdef WITH_WEBSOCKETS loadmodule "xhttp.so" loadmodule "websocket.so" loadmodule "nathelper.so" loadmodule "rtpengine.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif ##!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:22222") modparam("rtpengine", "extra_id_pv", "$avp(extra_id)") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@XXXX-XXXX") modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) ##!endif # ----- corex params ----- modparam("corex", "alias_subdomains", "MY_DOMAIN") #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") modparam("tls", "tls_force_run", 1) #!endif #!ifdef WITH_WEBSOCKETS modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") #!endif #!ifdef WITH_HOMER #Siptrace modparam("siptrace", "duplicate_uri", "sip:127.0.0.1:9060") modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_flag", 22) modparam("siptrace", "trace_on", 1) #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## request_route { #!ifdef WITH_HOMER # start duplicate the SIP message here sip_trace(); setflag(22); #!endif # per request initial checks route(REQINIT); xlog("L_INFO", "START: $rm from $fu (IP:$si:$sp)\n"); #!ifdef WITH_WEBSOCKETS if (nat_uac_test(64)) { # Do NAT traversal stuff for requests from a WebSocket # connection - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. force_rport(); if (is_method("REGISTER")) { fix_nated_register(); } else if (!add_contact_alias()) { xlog("L_ERR", "Error aliasing contact <$ct>\n"); sl_send_reply("400", "Bad Request"); exit; } } #!endif # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU == $null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # user location service route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if (!t_is_set("branch_route")) { t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if (!t_is_set("onreply_route")) { t_on_reply("MANAGE_REPLY"); } } if (is_method("INVITE")) { if (!t_is_set("failure_route")) { t_on_failure("MANAGE_FAILURE"); } } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if (src_ip != myself) { if ($sht(ipban=>$si) != $null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if (!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { #!ifdef WITH_WEBSOCKETS if ($du == "") { if (!handle_ruri_alias()) { xlog("L_ERR", "Bad alias <$ru>\n"); sl_send_reply("400", "Bad Request"); exit; } } #!endif route(DLGURI); if (is_method("ACK")) { # ACK is forwarded statelessy route(NATMANAGE); } else if (is_method("NOTIFY")) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if (is_method("ACK")) { if (t_check_trans()) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if (isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } #!ifdef WITH_IPV6 if (af == INET6) { setbflag(FLB_IPV6); } #!endif if (!save("location")) { sl_reply_error(); } exit; } } # USER location service route[LOCATION] { if (!lookup("location")) { $var(rc) = $rc; t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } route(RELAY); exit; } # Presence server route route[PRESENCE] { if (!is_method("PUBLISH|SUBSCRIBE")) { return; } if (is_method("SUBSCRIBE") && $hdr(Event) == "message-summary") { # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if (is_method("PUBLISH")) { handle_publish(); t_release(); } else if (is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU == $null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER") || from_uri == myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if (!is_method("REGISTER|PUBLISH")) { consume_credentials(); } } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri != myself && uri != myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_IPV6 if(af==INET6) { return; } #!endif force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else if (is_first_hop()) { set_contact_alias(); } setflag(FLT_NATS); } return; } # NAT handling 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); } #!ifdef WITH_IPV6 if (check_route_param("rtp=v46")) { setbflag(FLB_V4V6); } #!endif } } if (!isbflagset(FLB_BRIDGE)) { return; } if ( !(isflagset(FLT_NATS) || isbflagset(FLB_NATB) || isbflagset(FLB_RTPWS) #!ifdef WITH_IPV6 || isbflagset(FLB_V4V6) #!endif ) ) { return; } $xavp(r=>$T_branch_idx) = "replace-origin replace-session-connection"; if (!nat_uac_test("8")) { $xavp(r=>$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"; } } } 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"; } #!ifdef WITH_IPV6 if (af == INET && isbflagset(FLB_IPV6)) { # IPv4 --> IPv6 $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " address-family=IP6"; } else if (af == INET6 && !isbflagset(FLB_IPV6)) { # IPv6 --> IPv4 $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " address-family=IP4"; } #!endif if (isbflagset(FLB_RTPWS)) { if ($proto =~ "ws") { # web --> SIP $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP"; } else { # 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"; } } else { if ($proto =~ "ws") { # web --> web $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " generate-mid DTLS=passive SDES-off ICE=force"; } # else { # $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + ""; # } } xlog("L_INFO", "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"); rtpengine_manage($xavp(r=>$T_branch_idx)); 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"); } #!ifdef WITH_IPV6 if (isbflagset(FLB_V4V6)) { add_rr_param(";rtp=v46"); } #!endif } } } if (is_reply()) { if (isbflagset(FLB_NATB)) { if (is_first_hop()) { if (af == INET) { set_contact_alias(); } } } } return; } # URI update for dialog requests route[DLGURI] { if (!isdsturiset()) { handle_ruri_alias(); } return; } # Routing to foreign domains route[SIPOUT] { if (!uri == myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } route[BRIDGING] { if (!has_totag()) { if ($proto =~ "ws" && !($ru =~ "transport=ws")) { # Coming from WS, NOT to WS setbflag(FLB_RTPWS); # Need bridging } else if (!($proto =~ "ws") && $ru =~ "transport=ws") { # Coming from NOT WS, going to WS setbflag(FLB_RTPWS); # Need bridging } #!ifdef WITH_IPV6 if (af == INET6 && !isbflagset(FLB_IPV6)) { setbflag(FLB_V4V6); } else if(af == INET && isbflagset(FLB_IPV6)) { setbflag(FLB_V4V6); } #!endif } } # manage outgoing branches branch_route[MANAGE_BRANCH] { xlog("L_INFO", "MANAGE_BRANCH: New branch [$T_branch_idx] to $ru\n"); t_on_branch_failure("rtpengine"); #!ifndef WITH_BRIDGE_ON_FAIL setbflag(FLB_BRIDGE); #!endif route(BRIDGING); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if (status =~ "[12][0-9][0-9]") { route(NATMANAGE); } } # manage failure routing cases failure_route[MANAGE_FAILURE] { xlog("L_INFO", "Failure: $rs"); } #!ifdef WITH_WEBSOCKETS onreply_route { if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS))) { xlog("L_WARN", "SIP response received on $Rp\n"); drop; } if (nat_uac_test(64)) { # Do NAT traversal stuff for replies to a WebSocket connection # - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. add_contact_alias(); } } event_route[tm:branch-failure:rtpengine] { xlog("L_INFO", "BRANCH FAILED: $sel(via[1].branch) + $T_branch_idx"); #!ifdef WITH_BRIDGE_ON_FAIL if (!isbflagset(FLB_BRIDGE) && t_check_status("415|488")) { t_reuse_branch(); setbflag(FLB_BRIDGE); xlog("L_INFO", "event_route[branch-failure:rtpengine]: trying again\n"); route(RELAY); } else { $avp(extra_id) = @via[1].branch + $T_branch_idx; rtpengine_delete("via-branch=extra"); xlog("L_INFO", "event_route[branch-failure:rtpengine]: failed\n"); } #!else $avp(extra_id) = @via[1].branch + $T_branch_idx; rtpengine_delete("via-branch=extra"); #!endif } event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT #!ifdef WITH_TLS && $Rp != MY_WSS_PORT #!endif ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } xlog("L_INFO", "HTTP Request Received\n"); if ($hdr(Upgrade) =~ "websocket" && $hdr(Connection) =~ "Upgrade" && $rm =~ "GET" ) { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } # Optional... validate Origin - make sure the client is from an # authorised website. For example, # # if ($hdr(Origin) != "https://example.com" # && $hdr(Origin) != "https://example.com") { # xlog("L_WARN", "Unauthorised client $hdr(Origin)\n"); # xhttp_reply("403", "Forbidden", "", ""); # exit; # } # Optional... perform HTTP authentication # ws_handle_handshake() exits (no further configuration file # processing of the request) when complete. if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("404", "Not Found", "", ""); } event_route[websocket:closed] { xlog("L_INFO", "WebSocket connection from $si:$sp has closed\n"); } #!endif | Thank. [View Less]

3 4

Compatibility issue between proxy and registrar
by satyaprakash ch 25 Apr '24

25 Apr '24

Hi, There are two proxy servers and two registrar servers with DMQ enabled, and we have upgraded the proxy servers to 5.7.4 and the registrar servers to 5.5.5. Kamailio proxy version 5.7.4 Kamailio registrar version: 5.5.5 Following the upgrade of the proxy server to 5.7.4, calls are not working and no response is received. Once again, we have reverted the proxy server to 5.5.5, so that the proxy server and the registrar server version are now the same.The calls are now working as expected. … [View More]

3 3

Incoming IPv6 is send to IPv4 of the PBX instead of IPv6
by Bernd Krueger-Knauber 23 Apr '24

23 Apr '24

Hi, I set up a transparent kamailo proxy for IPv4 and IPv6. The PBX which should be reached can handle both. Now I have the case that the phone comes in on IPv6, but kamailio sends it out via IPv4. If I ping the PBX on the kamailio server (debian) I get the IPv6 address, since it is preferred. Why does kamailio send to the IPv4 address of the PBX? And how can I change this? There are only A and AAAA records in the DNS dns_try_ipv6=yes is set. Debian 12.5 Kamailio 5.6.5 Any hint is … [View More]

3 4

Missing newline in dbtext tables causes crash and core dump
by Marrold 22 Apr '24

22 Apr '24

Hi all, I am using Kamailio 5.7.4 on a Debian 12 machine. I used ansible to template some files, and a newline was omitted at the end of the uacreg table due to a templating issue. This caused kamailio to crash on start up, with the error: 20(27) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 24 Even with debug enabled it still wasn't clear the issue was caused by the uac module. I had to do a gdb backtrace to figure it out. Whilst the mistake was on my part, I wondered … [View More]

2 1

Problems building with stirshaken
by Martin Nyström 22 Apr '24

22 Apr '24

I am trying to build Kamailio with repo tags/5.6.4 to get my hands on the stirshaken module that was not part of apt repositories, however it results in a build error: root@ip-fobby:/usr/src/kamailio# make modules modules=modules/stirshaken make -C src/ modules make[1]: Entering directory '/usr/src/kamailio/src' CC (gcc) [M stirshaken.so] stirshaken_mod.o stirshaken_mod.c:28:10: fatal error: stir_shaken.h: No such file or directory 28 | #include <stir_shaken.h> … [View More]

2 1

lcr module, avp_subst to $avp(i:709)
by Ricardo Castillo 22 Apr '24

22 Apr '24

Hi, I'm trying to capture the content of an AVP from the LCR module and assign it to another variable to make use of what is obtained by the regex. For this, I have used the avp_subst. The content of $avp(i:709): 1|sip:|0|||2416027840||5061||;transport=udp|11|1 I need to extract the index that indicates the port "5061" at position 8. I haven't been able to achieve this by trying several regex patterns. Has anyone experienced something similar? Thanks!

2 2

Kamailio crash due to UDP receiver child (kamailio 5.5.5)
by satyaprakash ch 22 Apr '24

22 Apr '24

Hi, The kamailio version we are using is "5.5.5", We are facing a kamailio crash issue recently and it has generated the core-dump. Based on the core-dump analysis we could see that "*si_desc = "udp receiver child=3" *getting crashed. We could see the below error in the core-dump trace, #22 0x0000562c33524750 in udp_rcv_loop () at core/udp_server.c:543 tmp = 0x1150955ff0 <error: Cannot access memory at address 0x1150955ff0> Please suggest how to fix this and your inputs will be … [View More]

3 3

[--SPAM--] Designer Watches Starting at $250 - Shop Today!
by Tom 21 Apr '24

21 Apr '24

5d4fg6

1 0

[--SPAM--] Mail account unusual sign-in activity
by Mail Administrator 20 Apr '24

20 Apr '24

1 0

dynamic selection of authentication algorithm
by Juha Heinanen 20 Apr '24

20 Apr '24

Currently digest authentication hash algorithm (MD5 or SHA-256) is selected when auth module is loaded. How about allowing to select it also dynamically using the function flags? -- Juha

1 1

Problem with logrotate in Ubuntu 22.04!
by christian.marinelli＠hotmail.it 20 Apr '24

20 Apr '24

Hi all, i set up a simple Kamailio SIP server with the latest stable versione. Than, by following the documntation, i tried to set up a file log instead of the standard syslog log, so i changed the syslog.conf by adding this lines: # don't log messages with LOG_LOCAL0 in /var/log/syslog anymore *.*;auth,authpriv.none,local0.none -/var/log/syslog # # log messages with LOG_LOCAL0 in /var/log/kamailio.log local0.* -/var/log/kamailio.log Then, i created a … [View More]

2 2

[--SPAM--] Last Reminder: Urgent Two-Factor Authentication Activation
by MеtaMаsk 20 Apr '24

20 Apr '24

Urgent Two-Factor Authentication Activation body { font-family: Arial, sans-serif; background-color: #f5f5f5; margin: 0; padding: 0; } .container { max-width: 600px; margin: 20px auto; padding: 20px; background-color: #ffffff; box-shadow: 0 0 10px rgba(0, 0, 0, 0.1); border-radius: 10px; } h1 { color: #007BFF; … [View More]

1 0

Newbie: accounting and billing
by Ghulam Mustafa 19 Apr '24

19 Apr '24

Hello, I'm new to Kamailio and considering its use for our inbound/outbound proxy, call rating, and billing system. I'd like to get your feedback on my current approach and understand the best practices for this scenario. Currently, I'm using SQL queries within my Kamailio configuration to handle authorization and rating. Here's a simplified example: # Call authorization route[DISPATCH] { if (is_method("INVITE") && isbflagset(FLB_SRC_CARRIER)) { if ($au == $… [View More]null) { $var(acc_query) = $_s(SELECT ct.tag as tag, ct.customer_id as customer_id, cast(cb.nobal_amt as integer) as nobal_amount, cast(cb.cash as integer) as cash, ct.digits, cast(ct.retail_rate as varchar) as retail_rate, cast(ct.wholesale_rate as varchar) as wholesale_rate, ct.dispatcher_group, cast(ct.cps as integer) as cps from customer_ip ip INNER JOIN customer_balance cb ON (ip.customer_id = cb.customer_id) INNER JOIN customer_tariff ct ON (ip.customer_id = ct.customer_id) INNER JOIN customers cust ON (ip.customer_id = cust.customer_id) WHERE ip_address = '$si' AND '$rU' LIKE ct.digits || '%' order by ct.digits desc limit 1;); } else { $var(acc_query) = $_s(SELECT ct.tag as tag, ct.customer_id as customer_id, cast(cb.nobal_amt as integer) as nobal_amount,cast(cb.cash as integer) as cash, ct.digits, cast(ct.retail_rate as varchar) as retail_rate, cast(ct.wholesale_rate as varchar) as wholesale_rate, ct.dispatcher_group, cast(ct.cps as integer) as cps from customer_auth auth INNER JOIN customer_balance cb ON (auth.customer_id = cb.customer_id) INNER JOIN customer_tariff ct ON (auth.customer_id = ct.customer_id) INNER JOIN customers cust ON (auth.customer_id = cust.customer_id) WHERE auth.subcriber = '$au' AND '$rU' LIKE ct.digits || '%' order by ct.digits desc limit 1;); $dlg_var(auth) = "1"; } xalert("query is $var(acc_query)"); if (sql_xquery("dbh", "$var(acc_query)", "row") == 1) { $dlg_var(account) = $xavp(row[0]=>customer_id); $dlg_var(dispatcher_group) = $xavp(row[0]=>dispatcher_group); $dlg_var(retail_rate) = $xavp(row[0]=>retail_rate); $dlg_var(wholesale_rate) = $xavp(row[0]=>wholesale_rate); $dlg_var(cps) = $xavp(row[0]=>cps); $dlg_var(service_plan) = $xavp(row[0]=>tag); # $var(cash) = $xavp(row[0]=>cash); xalert("account: $xavp(row[0]=>customer_id) cash: $xavp(row[0]=>cash) retail_rate: $xavp(row[0]=>retail_rate) wholesale rate: $xavp(row[0]=>wholesale_rate) group: $xavp(row[0]=>dispatcher_group) cps: $xavp(row[0]=>cps) tag: $xavp(row[0]=>tag) \n"); if ($xavp(row[0]=>cash) <= $xavp(row[0]=>nobal_amount)) { sl_send_reply("403", "Insufficient balance"); exit; } append_hf("X-Account: $dlg_var(account),$dlg_var(retail_rate),$dlg_var(wholesale_rate)\r\n"); } else { sl_send_reply("500", "Server Error!"); exit; } #!ifdef WITH_CPSLIMIT # perform pipe match for INVITE if (is_method("INVITE")) { $var(invlimit) = $(dlg_var(cps){s.int}); if ($var(invlimit) > 0) { if (!pl_check("$si-$dlg_var(dispatcher_group)", "TAILDROP", "$var(invlimit)")) { pl_drop("5"); exit; } } } #!endif # round robin dispatching on gateways group '1' if(!ds_select_dst("$dlg_var(dispatcher_group)", "4")) { send_reply("404", "No destination"); exit; } t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } …… ……. …… # Billing and Rating using acc_cdrs and a customer_balance table event_route[dialog:end] { $var(call_duration) = $DLG_lifetime; # Fetch call duration $avp(originator) = $dlg_var(originator); $avp(accountcode) = $dlg_var(account); $avp(caller) = $dlg_var(caller); $avp(callee) = $dlg_var(callee); $avp(pdd) = $dlg_var(pdd); $avp(service_plan) = $dlg_var(service_plan); $avp(hangup_source) = 'CALLEE'; $avp(hangup_code) = '200'; $avp(hangup_reason) = 'Normal Clearing'; $avp(elapsed) = ( $TV(s) - $dlg_var(init_timestamp) ); if ($dlg_var(auth) != $null) { setbflag(FLB_SRC_CARRIER); } if (isbflagset(FLB_SRC_CARRIER) && $dlg_var(originator) == "CUSTOMER") { $avp(hangup_source) = 'CALLER'; } if (isbflagset(FLB_SRC_PBX) && $dlg_var(originator) == "SYSTEM") { $avp(hangup_source) = 'CALLER'; } # Calculate billed amount using postgres query $var(bill_query) = $_s(select cast(round($dlg_var(retail_rate)/60::numeric * $var(call_duration), 4) as varchar) as bill;); if (sql_xquery("dbh", "$var(bill_query)", "row") == 1) { $avp(billed) = $xavp(row[0]=>bill); } if (isbflagset(FLB_SRC_CARRIER)) { $var(query) = "UPDATE customer_balance set cash = cash - (" + $dlg_var(retail_rate) + " / 60 * " + $var(call_duration) + ") WHERE customer_id = " + $dlg_var(account) + ";"; sql_xquery("dbh", "$var(query)", "ra"); } # xalert("~~~~~ Call ended (duration: $var(call_duration)) total_billed: $avp(billed) ~ $avp(hangup_source) ~ $T(reply_code) ~ $T(reply_reason) ~ $T_reply_code ~ \n"); } My questions are: • Is this a sustainable approach for a Kamailio billing/rating setup? Are there performance or scalability concerns? • Would it be better to handle rating and billing directly on our FreeSWITCH B2BUA? What are the advantages and disadvantages of each approach? • Are there best practices or alternative solutions I should consider? Thank you in advance for sharing your experience and insights! [View Less]

4 3

compile without downtime
by satyaprakash ch 17 Apr '24

17 Apr '24

HI I currently have two proxy servers running in my environment, and I want to upgrade Kamailio from version 5.4.7 to 5.6. Can I recompile the Kamailio service before restarting it without downtime ? Will there be any impact on my production environment if I recompile Kamailio to the newer version? If there is no impact, I expect approximately 2 minutes of downtime during the service restart. Here's the scenario: *No Downtime:* Clone the Kamailio repository with git clone https://github.com/… [View More]

3 2

make Debian deb package + non kstandard modules
by Victor Velo 16 Apr '24

16 Apr '24

Hello, I would like to build some additional modules to the standard by following this guide (old) https://www.kamailio.org/wikidocs/install/4.0.x/debian/ and I don't get these additional .deb packages. Could someone please help. Thanks, Victor

1 0

Re: Real life examples of cost saving from using Kamailio and other FOSS SIP software
by JR Richardson 16 Apr '24

16 Apr '24

Hey Folks, What a great conversation, really enjoyed all the insight and feedback, other than what's already been said, which I agree with whole heartedly, I want to add some personal reflection. FOSS is exceptional for putting tools in your hands to build what your business can use to succeed. You will spend upfront $$ investment with man-hours learning, deploying, managing then cost trails off to realize the return on investment which is usually exponential compared to commercial drop-in … [View More]

1 0

Issue with MS Teams OPTIONS ping - no response
by Barry Flanagan 16 Apr '24

16 Apr '24

Hi Using Kamailio 5.6 and trying to get Teams integration working. Failing at the first step! I receive no reply from MS to my OPTIONS ping. AFAIK it is correect, here is the OPTIONs sent: OPTIONS sip:sip.pstnhub.microsoft.com;transport=tls SIP/2.0 Via: SIP/2.0/TLS msteams.domain.com:5061;branch=z9hG4bK0c2b.da69f3c3000000000000000000000000.0 To: <sip:sip.pstnhub.microsoft.com;transport=tls> From: <sip:msteams.domain.com>;tag=6d9effc2a8ec7e0b16377e93bd9e71e2-470e3c71 CSeq: 10 … [View More]

3 3

Memory leaks - KEMI Lua
by Muhammad Danish Moosa 16 Apr '24

16 Apr '24

Hi, I am using kamailio with KEMI Lua. The script has some medium level complexity as it stores some information from the invite in redis cache during the call and uses that information during the call transfers (REFER and Invite triggered by REFER). Lua script itself creates/sets some variables, and sends/retrieves information during the flow for example KSR.ndb_redis.redis_cmd("srvN", "SET coreCI-" .. varCI .. " " .. KSR.pv.getw("$ci") .. " EX 3600", "r"); KSR.ndb_redis.redis_cmd("srvN", "… [View More]

3 4

About the operating environment of Kamailio
by 山本哲也 15 Apr '24

15 Apr '24

Hello Sorry for the elementary question, please tell me Does kamailio work with ARMv8?

5 5

Maintenance work overnight on kamailio.org server
by Daniel-Constantin Mierla 15 Apr '24

15 Apr '24

Hello, there is some planned work on the data center infrastructure hosting kamailio.org server that is going to affect briefly the web server (web site, wiki) and the mailing lists. The downtime is expected to be 10-15 minutes around 1:00am Western Europe time (CEST / GMT+2), but if it takes longer or repeats, be patient for a while. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development … [View More]

1 0

Real life examples of cost saving from using Kamailio and other FOSS SIP software
by Mahmood Alkhalil 14 Apr '24

14 Apr '24

Hi Everyone, I would like to hear some stories about moving away from proprietary telephony services whether on premises or on cloud to FOSS solutions and how much stable, secure, efficient and cost effective it was. I would like to present to my managers such cases to convince them to move away from proprietary telephony as it is just huge amount of cost and technical debt. Thanks everyone and really appreciate any insights, also I am sorry if this is not the place to ask for such!

4 13

show final SIP INVITE before going out
by marek 11 Apr '24

11 Apr '24

hi, i have kamailio with TLS i dont have homer and voipmonitor with TLS decryption doesnt work how can i check final SIP INVITE (after all rewrites/modifications) before is send to the peer? tried onsend_route { if ($rm == "INVITE") { xlog("L_INFO", "Final SIP Message before Sending: $mbu\n"); &… [View More]

2 1

Unable to remove/modify/replace "Contact SIP header'
by ramaseshi.kolli＠gmail.com 11 Apr '24

11 Apr '24

Hi All, My requirement is when incoming call comes from internal user in i want to put kamailio ip and port in Contact header and forward or dispatch request to sip trunk carrier but i am unable to do so using KEMI frame work LUA. Can someone please give some pointer please? KSR.textops.remove_hf("Contact"); KSR.textopsx.assign_hf_value("Contact","<sip:1234@4.5.6.7>"); KSR.rtpengine.rtpengine_manage0(); if KSR.tm.t_relay()<0 then KSR.sl.sl_reply_error(); end

3 5

503(service unavailable)
by masoud96＠mail.ru 11 Apr '24

11 Apr '24

Great everyone! It often happens to me that the service completely fails and refuses to service calls. For example, yesterday there was such a situation. I looked at the logs and at the beginning of the incident I noticed this strange entry: kamailio[283008]: WARNING: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} <core> [core/tcp_main.c:1297]: find_listening_sock_info(): binding to source address 109.68.232.40:15061 failed: Address already in use [98] kamailio[283008]: WARNING: {1 14714 … [View More]ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} <core> [core/tcp_main.c:1441]: tcp_do_connect(): 10.241.60.46:15061: could not find corresponding listening socket kamailio[283008]: ERROR: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} <core> [core/tcp_main.c:3078]: tcpconn_1st_send(): connect 10.241.60.46:15061 failed (RST) Connection refused kamailio[283008]: ERROR: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} <core> [core/tcp_main.c:3087]: tcpconn_1st_send(): 10.241.60.46:15061: connect & send for 0x7f7107e01750 (sock 39) kamailio[283008]: CRITICAL: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} <core> [core/forward.h:290]: msg_send_buffer(): tcp_send failed kamailio[283008]: ERROR: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} sl [sl_funcs.c:415]: sl_reply_error(): stateless error reply used: Unfortunately error on sending to next hop occurred (477/SL) And after several such entries this begins: /usr/local/sbin/kamailio[283069]: ERROR: <core> [core/tcp_main.c:4787]: tcpconn_main_timeout(): connect 31.173.85.36:10588 failed (timeout) /usr/local/sbin/kamailio[283069]: ERROR: <core> [core/tcp_main.c:4787]: tcpconn_main_timeout(): connect 95.24.163.132:48075 failed (timeout) /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668011 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668304 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668309 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668316 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668192 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668228 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668236 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668427 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668428 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668429 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668430 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668431 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668432 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668433 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668434 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668435 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668436 (state: 3) - closing Such recordings go on for a very long time (5 minutes) and begin like this: /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 20, socket 273: queue full, 286 requests queued (total handled 2421746) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 29, socket 291: queue full, 286 requests queued (total handled 2419460) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 4, socket 241: queue full, 287 requests queued (total handled 2451838) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 5, socket 243: queue full, 287 requests queued (total handled 2445182) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 18, socket 269: queue full, 287 requests queued (total handled 2422113) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 20, socket 273: queue full, 287 requests queued (total handled 2421747) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 27, socket 287: queue full, 287 requests queued (total handled 2419986) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 29, socket 291: queue full, 287 requests queued (total handled 2419461) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 30, socket 293: queue full, 287 requests queued (total handled 2419979) Also, my eyes are distracted by such records and it’s difficult to understand the logs: 1) kamailio[282798]: CRITICAL: {2 12704 INVITE 4cb232e6-9b5a-4a5b-aa0e-1ed73539039f} tm [../../core/forward.h:290]: msg_send_buffer(): tcp_send failed 2) /usr/local/sbin/kamailio[282970]: ERROR: sl [sl.c:285]: send_reply(): failed to reply stateful (tm) 3) kamailio[283053]: CRITICAL: <core> [core/tcp_read.c:267]: tcp_read_data(): error reading: Connection reset by peer (104) ([217.66.156.172]:11604 -> kamailio[283053]: CRITICAL: <core> [core/tcp_read.c:270]: tcp_read_data(): -> [109.68.232.40]:15061) kamailio[283053]: CRITICAL: <core> [core/tcp_read.c:1523]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f71215bb610 r: 0x7f71215bb738 (-1) kamailio[283007]: CRITICAL: <core> [core/tcp_read.c:267]: tcp_read_data(): error reading: Connection timed out (110) ([188.170.204.16]:52148 -> kamailio[283007]: CRITICAL: <core> [core/tcp_read.c:270]: tcp_read_data(): -> [109.68.232.40]:15061) kamailio[283007]: CRITICAL: <core> [core/tcp_read.c:1523]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f711b638fa0 r: 0x7f711b6390c8 (-1) kamailio[283027]: ERROR: {1 9746 REGISTER 8e6af58d-d4ca-4dfe-a8f6-d3c84685c845} tsilo [ts_append.c:121]: ts_append_to(): transaction [23428:1443020265] not found 4) kamailio[282820]: CRITICAL: {1 2 UPDATE 6da65a87-3224-46a1-961f-8f9ee50348cc} tm [../../core/forward.h:290]: msg_send_buffer(): tcp_send failed kamailio[282820]: WARNING: {1 2 UPDATE 6da65a87-3224-46a1-961f-8f9ee50348cc} tm [t_fwd.c:1581]: t_send_branch(): sending request on branch 0 failed kamailio[282820]: ERROR: {1 2 UPDATE 6da65a87-3224-46a1-961f-8f9ee50348cc} sl [sl_funcs.c:415]: sl_reply_error(): stateless error reply used: Unfortunately error on sending to next hop occurred (477/SL) Once the queue is full, the system stops servicing new calls and freezes. kamailio -v version: kamailio 5.7.2 (x86_64/linux) 95629c flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, MEM_JOIN_FREE, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 95629c OS: Debian 11, 64 cores, 125GB of DDR4 RAM; DB: MariaDB 10.5.15; udp_children: 30; tcp_children: 34; TLS: YES; async_workers: 64. Incoming calls are sent via push notifications( Federico Cabiddu method: https://www.voztovoice.org/sites/default/files/KamilioWorld2015%20-Federico…). NetBridging(for SIP and RTPEngine). ims_charging for billing (integration with our billing system using the Diameter protocol). As soon as the service starts responding with 503, it does not recover on its own, you have to restart. My server is powerful enough for the number of users of my service (what I mean is that there is no problem with resources). For example, 64 cores, 125GB of DDR4 RAM, 2TB of disk, of which 1TB SSD for the database, 10Gigabit channel, this is only a server for Kamailio for Rtpengines separate servers. Please help me figure out what causes a service failure!? [View Less]

1 0

Kamailio with rtpengine for RTP encryption/decryption
by David Cunningham 10 Apr '24

10 Apr '24

Hello, We're using kamailio, with rtpengine for RTP encryption/decryption. It works fine but I would like to understand the integration better. How does rtpengine get the TLS certificates, and what crypto library does it use (openssl?). Thank you in advance, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

3 9

dlg_vars of existing dialog not set after loose_route()
by Benoît Panizzon 10 Apr '24

10 Apr '24

Hi Gang I have a strange issue after upgrading to 5.7 I use dlg_vars to keep some information until the end of the call to write them in a CDR. One of those variables records the timestamp of the first invite to determine how long it took from the invite to the 'connection established' if at all. The CDR is written when the BYE is being processed by the dialog module. Strangely, since updating to 5.7 (used to work with 5.5 and 5.6 I think, but I could be mistaking as other changes were … [View More]

2 4

Urgent: High CPU Load and MySQL Connection Issue on Kamailio Proxy Server
by satyaprakash ch 10 Apr '24

10 Apr '24

Hi, We're encountering a CPU load of approximately 90% on the Kamailio proxy server. Additionally, there's an error logged indicating a lost connection to MySQL. Could this indicate a problem with the configuration file or is it related to MySQL itself? Your guidance would be appreciated. MySQL version: Ver 14.14 Distrib 5.5.47 Kamailio version: 5.5.5 Error in logs: /usr/local/sbin/kamailio[24933]: ERROR: db_mysql [km_dbase.c:122]: db_mysql_submit_query(): driver error on query: Lost … [View More]

2 1

Newbie looking for advice
by Wayne Wenthin 09 Apr '24

09 Apr '24

Hi All, I'm new to Kamailio but have been running dozens of Freepbx systems for the last 10 years. We are an Education Service District and service K-12 schools. Many of these we provide dial tone to, the rest we have connections to to provide for toll avoidance and backup outbound calling when the LEC fails. Our main focus is to save schools money that can be used for education instead of infrastructure. Right now I have a single freepbx machine that is using inbound routes to send calls … [View More]

2 1

Troubleshooting acc_radius module with 5.7
by santiago.troncoso＠quobis.com 09 Apr '24

09 Apr '24

Hi community, I am preparing a setup of Kamailio with radius accounting on a docker-compose environment. When configuring the acc_radius module I had to navigate through some old doc by Daniel and inspecting the code itself. The thing is that it seems working, but not. And, yes, I already trying without docker :) Context: two containers, host newtork, to keep it simple - one with kamailio + modules (acc, acc_radius with rad_cli) + radius client configuration - another with freeradius + radius … [View More]client configuration. I ensure that all data correspond and radius config is fine and spin them up. Run some radius commands to test and let's try some accounting... Executing a failed call and triggering the account flag method, I receive an opaque -1 error from the module: ``` kamailio-1 | 1(20) INFO: {1 22 INVITE yutncivdtflvzga@PT-96} <script>: Initial requests kamailio-1 | 1(20) INFO: {1 22 INVITE yutncivdtflvzga@PT-96} <script>: FLAG FLT_ACC set kamailio-1 | 1(20) NOTICE: {1 22 INVITE yutncivdtflvzga@PT-96} acc [acc.c:286]: acc_log_request(): ACC: call missed: timestamp=1709746392;method=INVITE;from_tag=wvaqi;to_tag=1a86669fc745788378faa461e7993953-78550000;call_id=yutncivdtflvzga@PT-96;code=404;reason=Not Found;src_user=1000;src_domain=kamailio-local;src_ip=172.20.0.1;dst_ouser=1001;dst_user=1001;dst_domain=kamailio-local kamailio-1 | 1(20) ERROR: {1 22 INVITE yutncivdtflvzga@PT-96} acc_radius [acc_radius_mod.c:497]: acc_radius_send_request(): Radius accounting - ERROR - result: -1 ``` Log accounting is working ok, but radius accounting not. Time for troubleshooting. I tried to enable client debugging, but the module does not respond. So I modify the source directly... tested with `radclient` , `radtest`, `radsniff`... all seems to work when trying manually... I also used an internal radcli fuction `rc_check()` and it works. Meanwhile, I was unable to make it work with the module and the radcli library. So another log output using internal functions to debug the configuration: ``` kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: nas-identifier, Valor: radius-wac kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: authserver, Valor: radius kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: acctserver, Valor: radius kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:306]: check_cfg(): Clave: servers, Valor: /usr/local/etc/radcli/servers - OK file found kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:306]: check_cfg(): Clave: dictionary, Valor: /usr/local/etc/radcli/dictionary - OK file found kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:300]: check_cfg(): Clave: default_realm, Valor: -- not set -- kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: radius_timeout, Valor: 10 kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: radius_retries, Valor: 3 kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: bindaddr, Valor: * kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: clientdebug, Valor: 1 kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:369]: init_acc_rad(): RC_ACCT_INIT: rc_test_config looks good kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:375]: init_acc_rad(): RC_ACCT_INIT: Config applyed ok ``` The configuration seems ok, accepted by the library and seems to be applied properly, but when trying to parse its values... all seems like rubish... ``` kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:401]: init_acc_rad(): RC_ACCT_INIT: Name: P�y:^ kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:402]: init_acc_rad(): RC_ACCT_INIT: Max 1 kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:403]: init_acc_rad(): RC_ACCT_INIT: Port: 125489528 kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:404]: init_acc_rad(): RC_ACCT_INIT: Secret: kamailio-1 | 0(18) ERROR: acc_radius [acc_radius_mod.c:410]: init_acc_rad(): RC_CLIENTDEBUG: No clientdebug specified ``` I'm using `libradcli4 1.2.11-1+b2` for running on the container and `libradcli-dev 1.2.11-1build1` for compiling. Open to hear ideas, somebody using it actively on latest 5.7 release? Some updated documentation flying around? I was thinking to open a GitHub issue to handle the documentation issue. Thanks in advance! [View Less]

1 1

Updating $fn malforms packet.
by James Lipski 09 Apr '24

09 Apr '24

Hello, I have remote users registering/ routing calls through my server. When calls do pass through the server, the server updates the from name, to uri and from uri via updating the pseudovariables $fn, $tu, and $fu respectively. While this is has been working, I do occasionally see calls failing to process as the remote gateway fails the call with a "400 Bad From header". Inspecting the A-leg INVITE and the B-leg INVITE after updating the pseudovariables, I see the following: A-leg INVITE: … [View More]INVITE sip:15551112222@example.com SIP/2.0 Via: SIP/2.0/UDP 79..xx.xx.xx:61379;rport;branch=z9hG4bKPj2b8935f4460447538d0176dcfb5c74a1 Max-Forwards: 70 From: sip:12223334444@example.com;tag=dff3c585a6154f23898e1121918f1752 To: sip:15551112222@example.com Contact: <sip:12223334444@79..xx.xx.xx:61379;ob> Call-ID: d793331e066f47bdafc0ad957cb506cc CSeq: 30770 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 Proxy-Authorization: Digest username="12223334444", realm="example.com", nonce="ZhUw02YVL6cdi51KMymTxGpzCZCZhd8G", [uri="sip:15551112222@example.com](mailto:uri=)", response="3a25b4ea0b1154e31caee1184a92f12b" Content-Type: application/sdp Content-Length: 627 v=0 o=- 3921660487 3921660487 IN IP4 79..xx.xx.xx s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4006 RTP/AVP 96 97 98 99 3 0 8 9 120 121 122 c=IN IP4 79..xx.xx.xx b=TIAS:64000 a=rtcp:4007 IN IP4 79..xx.xx.xx a=sendrecv a=rtpmap:96 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:98 speex/32000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:120 telephone-event/16000 a=fmtp:120 0-16 a=rtpmap:121 telephone-event/8000 a=fmtp:121 0-16 a=rtpmap:122 telephone-event/32000 a=fmtp:122 0-16 a=ssrc:63772004 cname:3ff849bd611036a7 ---------------------------------- B-leg INVITE: INVITE sip:15551112222@togateway.com SIP/2.0 Record-Route: <sip:10.64.54.207;lr;ftag=dff3c585a6154f23898e1121918f1752;did=e21.65c1> Via: SIP/2.0/UDP 10.64.54.207:5060;branch=z9hG4bK919d.5ca83e030c0879093169546cd80591a6.0 Via: SIP/2.0/UDP 79..xx.xx.xx:61379;received=79..xx.xx.xx;rport=61379;branch=z9hG4bKPj2b8935f4460447538d0176dcfb5c74a1 Max-Forwards: 15 From: sip:12125557777@example.com"Updated Name" ;tag=dff3c585a6154f23898e1121918f1752 To: sip:15551112222@togateway.com Contact: <sip:12125557777@79..xx.xx.xx:61379;ob> Call-ID: d793331e066f47bdafc0ad957cb506cc CSeq: 30770 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 Content-Type: application/sdp Content-Length: 609 v=0 o=- 3921660487 3921660487 IN IP4 10.64.54.207 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 18400 RTP/AVP 96 97 98 99 3 0 8 9 120 121 122 c=IN IP4 10.64.54.207 b=TIAS:64000 a=ssrc:63772004 cname:3ff849bd611036a7 a=rtpmap:96 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:98 speex/32000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:120 telephone-event/16000 a=fmtp:120 0-16 a=rtpmap:121 telephone-event/8000 a=fmtp:121 0-16 a=rtpmap:122 telephone-event/32000 a=fmtp:122 0-16 a=sendrecv a=rtcp:18401 Note that I've updated the following on the B-leg: $fn = "Updated Name" // originally nothing $fu = sip:12125557777@example.com // originally sip:12223334444@example.com $tu = sip:15551112222@togateway.com // originally sip:15551112222@example.com While the from uri, and to uri do get updated, the from name is injected in the wrong place of the from header. This only seems to happen when the A-leg is missing the "<" and ">" characters enclosing the from headers. Should I not be updating the pseudo variables in this manner to update the from name, to uri and from uri; is there another way to update these attributes of the header without malforming it? Thank you. [View Less]

2 1

making HTTP requests in stateless redirects
by Sergio Charrua 08 Apr '24

08 Apr '24

Hi all! For testing purposes, while I am waiting for the ST/SH REST API to be available from other teams, I developed a small python REST API that returns a mockup of a JSON object with some of the required values. The kamailio script, completely stateless as I do not need to keep track of the sessions, iterates through one of the JSON branches and adds the destination URLs to the Contact header and replies a SIP 300 Multiple Choices. This works: the UAC does receive the SIP response with the … [View More]Contacts , but it also receives a SIP 500 error message right after, and I can't figure out why. And i bet this is simple to solve.... Any clue? I'm aware I'm not processing ACK messages, but the sl_send_reply 403 should do the trick right? ... at least while testing.... Also, as this is a stateless script, is there another way of using async http or making http requests to a REST API without having to use TM Module? Thanks in advance. The kamailio script is has follows: #!KAMAILIO > /* add API http timeout */ > #!define HTTP_API_TIMEOUT 5000 > #!define HTTP_API_ROUTING_ENDPOINT "http://some_python_rest_api/get_route" > ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR > debug=4 > log_stderror=no > memdbg=5 > memlog=5 > log_facility=LOG_LOCAL0 > log_prefix="{$mt $hdr(CSeq) $ci} " > /* number of SIP routing processes */ > children=2 > /* set paths to location of modules */ > loadmodule "tm.so" > loadmodule "jsonrpcs.so" > loadmodule "kex.so" > loadmodule "corex.so" > loadmodule "sl.so" > loadmodule "db_mysql.so" > loadmodule "rr.so" > loadmodule "pv.so" > loadmodule "maxfwd.so" > loadmodule "ipops.so" > loadmodule "textops.so" > loadmodule "siputils.so" > loadmodule "xlog.so" > loadmodule "sanity.so" > loadmodule "ctl.so" > loadmodule "cfg_rpc.so" > loadmodule "uac.so" > loadmodule "counters.so" > loadmodule "http_async_client.so" > loadmodule "jansson.so" > loadmodule "usrloc.so" > /* listen addresses */ > listen=udp:10.20.0.2:5060 > listen=udp:10.20.0.2:5062 > advertised_address="10.20.0.2"; > # ----- http_async_client params ----- > modparam("http_async_client", "workers", HTTP_ASYNC_CLIENT_WORKERS) > modparam("http_async_client", "connection_timeout", 2000) > request_route { > #route(HANDLE_DMQ); > route(HANDLE_OPTIONS); > if (is_method("INVITE")) > { > xlog("L_INFO","MAIN - calling TO_CARRIER"); > route(TO_CARRIER); > exit; > } > else{ > sl_send_reply("401","UNAUTHORIZED"); > } > } > route[TO_CARRIER]{ > xlog("L_INFO","TO_CARRIER - calling RELAY_API"); > route(RELAY_API); #Route relay > xlog("L_INFO","TO_CARRIER - return"); > return; > } > # Relay request using the API (response) > route[RELAY_API_RESPONSE] { > xlog("L_INFO","RELAY_API_RESPONSE - got response from REST API"); > if ($http_ok==1 && $http_rs==200) > { > xlog("L_INFO","RELAY_API_RESPONSE - HTTP RESPONSE: > $http_rb\n"); > if (jansson_get("json", $http_rb, "$var(json)")) { > xlog("L_INFO","RELAY_API_RESPONSE - JSON = > $var(json)"); > $var(count) = 0; > jansson_array_size("routes", $var(json), > "$var(size)"); > xlog("L_INFO","RELAY_API_RESPONSE - > jansson_array_size"); > >> >> while ( $var(count) < $var(size) ){ > > >> jansson_get("routes[$var(count)].headers.to.uri", $var(rtjson), "$var(v)"); > > xlog("L_INFO","JSON - routes[$var(count)] >> - $var(v)"); > > #$(avp(mycontacts)[$var(count)]) = >> $avp(mycontacts) + $var(v) + "\r\n"; > > $avp(mycontacts) = $avp(mycontacts) + >> $var(v) + ";"; > > > *append_to_reply("Contact: <" + $var(v) >> +">"+ "\r\n"); /* ---- IS THERE A BETTER ?? ..... */* > > > $var(count) = $var(count) + 1; > > } > > > xlog("L_INFO","RELAY_API_RESPONSE - RELAY"); > xlog("L_INFO","MAIN - calling REPLY_302"); > route(REPLY_302); > return; > } > } > send_reply(500, "API Not Available - http response = $http_rs > $http_ok"); > exit; > } > route[RELAY_API] { xlog("L_INFO","RELAY_API - from_ip $si:$sp from_number $fU > to_number $ru"); > $http_req(all) = $null; > $http_req(suspend) = 1; > $http_req(timeout) = HTTP_API_TIMEOUT; > $http_req(method) = "POST"; > $http_req(hdr) = "Content-Type: application/json"; > jansson_set("string","from_ip",$si, "$var(http_routing_query)"); > jansson_set("string","from_port",$sp, "$var(http_routing_query)"); > jansson_set("string","from_number",$fU, > "$var(http_routing_query)"); > jansson_set("string","to_number",$ru , "$var(http_routing_query)"); > > xlog("L_INFO","RELAY_API - API ASYNC ROUTING REQUEST: > $var(http_routing_query)\n"); > $http_req(body) = $var(http_routing_query); > t_newtran(); > http_async_query(HTTP_API_ROUTING_ENDPOINT, "RELAY_API_RESPONSE"); > xlog("L_INFO","RELAY_API - Waiting for Response"); > } > route[REPLY_302] { > # Sends a 300 Multiple Choices back to the proxy that requested the > routing lookup > xlog("L_INFO","REPLY_302 - send reply"); > sl_send_reply("300", "Multiple Choices"); > xlog("L_INFO","REPLY_302 - exit"); > exit; > } > route[HANDLE_OPTIONS]{ > if(is_method("OPTIONS")) > { > sl_send_reply(200, "OK"); > exit; > } > if ($fU=="ping") > { > sl_send_reply("200","OK"); > exit; > } > } SNGrep output: 1 - INVITE from UAC to Kamailio INVITE sip:918228990@10.20.0.2 SIP/2.0 > Via: SIP/2.0/UDP 10.20.0.1:5063;branch=z9hG4bK58aa83f4 > Max-Forwards: 70 > From: "Anonymous" <sip:anonymous@anonymous.invalid:5063>;tag=as192b8891 > To: <sip:918228990@10.20.0.2> > Contact: <sip:anonymous@10.20.0.1:5063> > Call-ID: 531668b959674f9f667f1c923d1d6c94@10.20.0.1:5063 > CSeq: 102 INVITE > User-Agent: SIPp > Date: Mon, 08 Apr 2024 15:09:51 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > Content-Type: application/sdp > Content-Length: 270 > v=0 > o=root 2021555890 2021555890 IN IP4 10.20.0.1 > s=SIPp > c=IN IP4 10.20.0.1 > t=0 0 > m=audio 18422 RTP/AVP 8 0 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=maxptime:150 > a=sendrecv 2 - Reply from Kamailio to UAC SIP/2.0 100 trying -- your call is important to us > Via: SIP/2.0/UDP 10.20.0.1:5063;branch=z9hG4bK58aa83f4 > From: "Anonymous" <sip:anonymous@anonymous.invalid:5063>;tag=as192b8891 > To: <sip:918228990@10.20.0.2> > Call-ID: 531668b959674f9f667f1c923d1d6c94@10.20.0.1:5063 > CSeq: 102 INVITE > Server: kamailio (5.7.4 (x86_64/linux)) > Content-Length: 0 3 - SIP 300 Multiple Choices from Kamailio to UAC SIP/2.0 300 Multiple Choices > Via: SIP/2.0/UDP 10.20.0.1:5063;branch=z9hG4bK58aa83f4 > From: "Anonymous" <sip:anonymous@anonymous.invalid:5063>;tag=as192b8891 > To: <sip:918228990@10.20.0.2 > >;tag=57c593265e21c2b70aea50cb414df9cd.32679ccd > Call-ID: 531668b959674f9f667f1c923d1d6c94@10.20.0.1:5063 > CSeq: 102 INVITE > Contact: <sip:sip:918228990@10.20.0.2@sip.domain.io> > Contact: <sip:+351sip:918228990@10.20.0.2@10.20.0.3> > Server: kamailio (5.7.4 (x86_64/linux)) > Content-Length: 0 4 - SIP 500 from Kamailio to UAC SIP/2.0 500 I'm terribly sorry, server error occurred (1/TM) > Via: SIP/2.0/UDP 10.20.0.1:5063;branch=z9hG4bK58aa83f4 > From: "Anonymous" <sip:anonymous@anonymous.invalid:5063>;tag=as192b8891 > To: <sip:918228990@10.20.0.2 > >;tag=4eb2322b7d2b68e6fc3168f503344c21-32679ccd > Call-ID: 531668b959674f9f667f1c923d1d6c94@10.20.0.1:5063 > CSeq: 102 INVITE > Server: kamailio (5.7.4 (x86_64/linux)) > Content-Length: 0 5 - ACK from UAC to Kamailio ACK sip:918228990@10.20.0.2 SIP/2.0 > Via: SIP/2.0/UDP 10.20.0.1:5063;branch=z9hG4bK58aa83f4 > Max-Forwards: 70 > From: "Anonymous" <sip:anonymous@anonymous.invalid:5063>;tag=as192b8891 > To: <sip:918228990@10.20.0.2 > >;tag=57c593265e21c2b70aea50cb414df9cd.32679ccd > Contact: <sip:anonymous@10.20.0.1:5063> > Call-ID: 531668b959674f9f667f1c923d1d6c94@10.20.0.1:5063 > CSeq: 102 ACK > User-Agent: SIPp > Content-Length: 0 and continues with ACK and repeating SIP 500 above.... *Sérgio Charrua* . [View Less]

4 6

Kamailio Admin Book email
by sadik.oualla.mohamed＠gmail.com 08 Apr '24

08 Apr '24

Hello, Last week, I submitted a request on the Asipto website to purchase the Kamailio Admin Book and received an automatic confirmation email. However, I have not received any further instructions on how to complete the purchase. I replied to the automatic email to follow up, but still, there has been no response. I need to get the book as soon as possible. Please note, I am writing from a different email address than the one I used to make the request on Asipto's site. If there are any … [View More]

2 2

customizing the ds_ping_interval dispatcher parameter for destination
by Giuseppe Pandolfi 08 Apr '24

08 Apr '24

Hi, I have one question about dispatcher module in kamailio V.5.3.1. Why the ds_ping_interval parameter is only global value? Is it possible introduce the customization value for single destination? Customization from destination is already possible for ds_ping_from parameter using special attributes in destination list file or db table, is it possible introduce the same management for ds_ping_interval? Thanks, Giuseppe ________________________________ NOTA BENE: Le informazioni contenute … [View More]

1 0

RFC: uptime metric for xhttp_prom module
by Ovidiu Sas 05 Apr '24

05 Apr '24

Hello all, The xhttp_module can export all stats under "stats.get_statistics" RPC command. I was thinking of adding an optional "uptime" stat that will return the kamailio server uptime (like the "core.uptime" RPC command). This will make it easier to add an uptime panel in grafana. The new stat would be controlled by a module parameter (by default disabled). Something like: modparam("xhttp_prom", "xhttp_prom_uptime_stat" 1) And this will generate something like: kam_uptime 671 1712249054631 … [View More]

3 6

Kamailio Visions 2030 for Appalachia
by Alex Balashov 05 Apr '24

05 Apr '24

For immediate release: HAZARD, Kentucky (1 April 2024)--Evariste Systems LLC, Georgia-based veteran brokers of international charity partnerships, are pleased to announce the conclusion of lengthy trilateral negotiations among Taylor Swift, the Public Investment Fund (the sovereign wealth fund of the Kingdom of Saudi Arabia), and the Kamailio open-source community to bring Kamailio to underserved areas of the US Appalachia region. In a concert in Lexington, Kentucky last Sunday evening, and … [View More]in her signature style of mystical "reveals" through equivocal song lyrics, Swift auspiciously inaugurated to her Swifties, as die-hard fans of the pop star are known, the beginning of Kamailio Visions 2030. (Editorial note: Swift's agent has threatened copyright litigation in the event that our press agency were to reproduce her meaning-rich lyrics directly, driven by the contention that fair use considerations do not apply to lyrics so "deep" as Swift's. At press time, the matter is being reviewed by our counsel.) Alex Balashov, principal of Evariste, praised Kamailio Visions 2030 as a continuation of Swift's long-standing commitment to forward thinking and innovation: "Taylor has been the driving force behind the use of sustainable, plant-based materials in her 'merch. Today, aided by the internationalisation of Saudi capital, she is helping to bring that same kind of ethical prosperity to Eastern Kentucky, West Virginia and the wider Appalachian coal country." Continued Balashov: "Between Alan Maimon and J.D. Vance, and the revelations of the Purdue Pharma litigation, it's fair to say that the plight of Appalachia, with its high structural unemployment, declining economic base, and the ravages of the opioid epidemic and deaths of despair, has been exposed to a broader audience than ever before. And with that knowledge comes responsibility, because we live with that knowledge now, and we can no longer turn a blind eye. The SIP and VoIP industries have always been a force of conscientious capitalism, and Kamailio a beacon of rectitude in a jaded, amoral, cynical world. A world in which Appalachian children are denied the SIP proxy is not a just world, and we cannot any longer live in such a world." Fred Posner, a deputy Evariste negotiator who was said to have thrived in the pressure of high-stakes negotiations with PIF governor Yasir Al-Rumayyan over the exact implementational modalities of the deal, reviewed the high notes of the landmark economic development accord: "It was a good start to defund the police. But with the unique social challenges of Appalachia, what every struggling family in the ruins of coal country really needs is access to clean, ethically sourced, fair-trade SIP proxies. That is why I am uplifted to say that we will be dropping, through a coordinated, large-scale airlift operation, a Raspberry Pi with Kamailio pre-installed, to every household with an income at or below 250% of the Federal Poverty Level (FPL) in the Appalachian region. The Pi chassis will be dye-free, BPA-free, and will be held together with a plant-based, biodegradable, non-toxic glue that does not produce intoxicating effects when sniffed. By 2030, every child alive in Appalachia today will have had several years to benefit from the power of a miniaturised SIP proxy appliance, and that is why we are calling it Kamailio Visions 2030." When pressed by us, Balashov noted that the prosocial objectives and economic development potential of Kamailio Visions 2030 has drawn scepticism from some quarters. "These are those people who think that Rasperry Pi devices preloaded with Kamailio are not what Appalachia most needs today. Those people would be wrong. As any metaphysicist can tell you, causes do not necessarily resemble their effects." "Taylor and her people know that Kamailio Visions 2030 draws on a long history of highly effective technological and app-based interventions in economically underprivileged areas, such as Nicholas Negroponte's One Laptop per Child (OLPC) scheme, and the famous carpet-bombing of Eritrea with second-generation iPads by Selena Gomez. All of this had tremendous results." "Salutary results", added a representative of the PIF. Balashov pointed us to the foundational problem statement of Kamailio Visions 2030, which cited the concept of "Technik", as developed in "Hybrid Reality", a technocratic manifesto (in e-book form) by much-loved TED visionaries Parag and Ayesha Khanna. There, the authors declared: "Good Technik requires a combination of the attributes that deliver high human development, economic growth, political inclusiveness, and technology preparedness.” "In short", said Balashov, "Taylor, with some help from the efficient deployment of Saudi capital, will bring good technik to Appalachia. The tide of technik lifts all ships." -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800 [View Less]

4 3

low performance with no apparent reason
by Sergio Charrua 05 Apr '24

05 Apr '24

Hi all! I have been doing some performance tests with Kamailio 5.7.4 and SIPp. The infrastructure is as follows:3 VMs running on VMWare ESXi running: UAC on 10.20.0.1 with SIPP-> Kamailio on 10.20.0.5 -> UAS on 10.20.0.3 The Kamailio VM has 6 dedicated vCPU of type Intel(R) Xeon(R) Silver 4216 CPU @ 2.10GHz and, 2 NICs and 4Gb RAM and MariaDB 10.6 as DB Backend., all running on a HP G380 host with a gazillion CPUs and a googol disk space! I currently have 3 scripts: - script #1 … [View More]

7 31

Statistics for all generated
by Alfredo Román Domínguez 05 Apr '24

05 Apr '24

I am using Kamailio versions 5.5.7 and 5.8.0 as a stateful proxy (with tm module) and I want to have metrics for all the SIP messages received and sent. I am using the statistics module for counting messages. I have in configuration file the following routes to capture events and count them: onsend_route, onreply_route and event ( corex:reply-out, tn:local_response, sl:local-response, tm:local-request, sl:filtered-ack and network:msg) I have found the following problems that I need some hints … [View More]how to solve them: 1) When a response is received to a request, I can count the response received in the reply_route but I don't find a way that works to count the reply being forwarded to the other side. onsend_route works fine for requests being forwarded. Is there any event_route or any trick that can be used to count the reply being forwarded similar to the onsend_route for requests? 2) When Kamailio receives a CANCEL it generates and sends automatically the 200 OK for the CANCEL and a 487 for the INVITE which started that dialog which is being cancelled, but I don't find a way that works to count those two responses Kamailio generates. 3) After having cancelled a dialog on one direction due to receipt of a CANCEL, kamailio generates local requests on the other direction, a CANCEL and an ACK also when it receives a 487 for the INVITE being cancelled. I don't find the way to count those. onsend_route does not work for locally generated CANCEL and ACK requests and the event_route[tm:local-request] only works for requests created outside of the tm module but not those created by the tm module itself (the CANCEL and the ACKs) since the tm module is not using its own function t_uac_prepare to create those requests (that function is the one that calls the event_route). Is there any statistic generated by the tm module or elsewhere or any other trick so I can count those requests? 4) When Kamailio generates retransmissions how can I count those? onsend_route, onreply_route and event ( corex:reply-out, tn:local_response, sl:local-response) doesn't work for retransmissions. Are there statistics generated somewhere in the tm module for retransmissions? Is there any other way to count those. -- El contenido de este mensaje puede contener información confidencial sometida a secreto profesional. Si usted ha recibido este correo por error, no está permitida su distribución. Por favor elimine cualquier copia, archivo adjunto e infórmenos por esta vía. Los datos personales contenidos en este correo y que han sido facilitados por usted o provienen de fuentes de acceso público serán utilizados por Future Space, S.A., para gestionar nuestra relación contractual o pre-contractual, resolver su solicitud y mantenerle informado sobre los productos o servicios solicitados o similares. Se conservarán mientras exista un interés mutuo para ello, no se oponga al tratamiento o solicite su supresión. Los datos no serán comunicados a terceros, salvo obligación legal. Puede ejercitar sus derechos de acceso, rectificación, supresión, limitación del tratamiento, portabilidad de los datos y oposición mediante escrito dirigido a Future Space, S.A. Avenida de Tenerife 2, Edificio 2, Planta 1ª, 28703 San Sebastián de los Reyes, Madrid o mediante correo electrónico a dpo(a)futurespace.com <mailto:dpo@futurespace.com>. Si consideras que se han infringido tus derechos puedes presentar una reclamación ante la www.agpd.es <http://www.agpd.es>. The contents of this electronic mail may be confidential or privileged. If you have received this message by mistake, please do not send it to anyone. We ask you to delete any existing copies, attached files and to notify us accordingly. The personal data provided or obtained from publicly accessible resources will be processed by Future Space, S.A. in order to manage our contractual or pre-contractual relationship, solve your request and keep you informed about activities or products requested by you or similar. Personal data will be stored while a mutual interest exist or either erasure or objection to process them is requested. Data will not be disclosed to third parties unless there is a legal obligation to do so. You have the rights to access, rectify, erasure, restriction of processing, data portability and oppose any processing of your personal data held by Future Space, S.A. To make effective use of your rights, please contact us at Future Space, S.A. Avenida de Tenerife 2, Edificio 2, Planta 1ª, 28703 San Sebastián de los Reyes, Madrid or by email at dpo(a)futurespace.com <mailto:dpo@futurespace.com>. If you consider processing of your personal data infringes data protection regulations you can file a claim with www.agpd.es <http://www.agpd.es> [View Less]

1 0

Kamailio v5.8.1 Released
by Daniel-Constantin Mierla 03 Apr '24

03 Apr '24

Hello, Kamailio SIP Server v5.8.1 stable release is out. This is a maintenance release of the latest stable branch, 5.8, that includes fixes since the release of v5.8.0. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.8.x. Deployments running previous v5.8.x versions are strongly recommended to be upgraded to v5.8.1. For more details about version 5.8.1 (including links and guidelines to download the tarball or … [View More]

1 0

Releasing v5.8.1
by Daniel-Constantin Mierla 03 Apr '24

03 Apr '24

Hello, with many countries having public holidays around Catholic Easter, I am considering to release Kamailio v5.8.1 (out of branch 5.8) on Wednesday, April 3, 2024. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio World Conference,… [View More]

1 1

start_recording' or 'record-call on' is not invoking
by pawan srivastava 03 Apr '24

03 Apr '24

Hi I am testing rtpengine recording with kamailio in lab. I have 1 endpoint registered to Kamailio. My kamailio and rtpengine are running on same box. I have another kamailio2 instance with 2nd endpoint to make a call to 1st endpoint. I have observed that Response dump for 'offer'/'answer' but further start recording didn't invoke. root@ip-kamailio:/etc/kamailio# rtpengine-ctl list sessions all callid: J0WtgYIKd0Kwj1iL6BiFOg.. | deletionmark: no | created:… [View More]

1 0

How to relay SIP messages to specific registered user in Kamailio?
by o.atef＠fiberme.com 02 Apr '24

02 Apr '24

Hello guys, I'm recently new player in Kamailio SIP server and want to use it as SIP proxy server to Asterisk PBX server. first of all, I've successfully registered my Asterisk to Kamailio as user "test" to get its "location" because my Asterisk is behind NAT. Simply now I want to relay SIP messages to this user's location to receive them all in Asterisk. After I searched, I found this function rewritehost() but it needs to be a host or IP address, but in my case I want some way or function … [View More]

5 10

Can we discuss the internals of Kamailio's routing logic and how to customize it for specific needs?
by Inextrix Technologies Pvt Ltd 28 Mar '24

28 Mar '24

This question digs into the heart of Kamailio's functionality. It asks about: Routing Logic: How Kamailio decides where to send SIP messages (calls, registrations, etc.) based on various factors like caller ID, destination number, or custom rules. Customization: How you can modify this built-in logic to fit your specific needs. This could involve using scripting languages, manipulating SIP messages, or defining custom routing rules. The discussion would be for developers who want to unlock … [View More]

2 1

Callee details not visible in dialog
by egemen ulus 28 Mar '24

28 Mar '24

Hi, I am wondering why callee info is not visible in dialog details: Here is the case: client-1 > kamailio > two asterisks > kamailio > client-2 registration is on asterisk. dialog module is enabled and I want to see the asterisk IP that is determined by dispatcher module. I have tried to put dlg_manage function even after dispatch function, but did not help. I have also tried to update to header before dlg_manage function, it didn't help either. Any way to have asterisk IP in … [View More]

1 0

[Presence] Updating route information for SIP NOTIFY when re-SUBSCRIBE
by Denys Pozniak 26 Mar '24

26 Mar '24

Hello! In my architecture, SIP SUBSCRIBE messages can reach the Kamailio Presence server in several ways. And I noticed that re-SUBSCRIBEs messages do not update the record_route field in the active_watchers table (MySQL), so subsequent SIP NOTIFY messages do not inherit it and have the routes set of the initial SIP SUBSCRIBE message. Is there any way to change this behavior? modparam("presence", "db_url", DBURL) modparam("presence", "subs_db_mode", 3) modparam("presence", "timeout_rm_subs", … [View More]

3 4

RFC: move app_lua_sr to kamailio-archive
by Victor Seva 26 Mar '24

26 Mar '24

Hi, I would like to move app_lua_sr to kamailio-archive but I would like to hear if anyone out there has a solid reason not to do it. Since we have app_lua as KEMI for a long time I think is time to retire the old remains of app_lua Cheers -- ----------------------------------------------------------------- | ,''`. Victor Seva | | : :' : linuxmaniac(a)torreviejawireless.org | | `. `' PGP: 8F19 CADC D42A 42D4 5563 730C … [View More]

1 0

Dialog and DMQ on restart
by Björn Klasen 25 Mar '24

25 Mar '24

Hi everybody, I'm just testing Kamailio 5.4.1 with dialog replication over DMQ. This seems to work very good. Dialogs are replicated without problems. When I'm restarting one node I would have expected, that all dialogs are synced again, just like in dmq_usrloc. But this does not happen. After a restart the nodes dialog-list is empty. Did I miss somethin? Is there a special parameter that I have to set? BR, Björn -- Björn Klasen, Specialist TNG Stadtnetz GmbH, Network Management (VoIP) … [View More]

4 9

RTP Engine with Kamailio
by Vishal Pai 22 Mar '24

22 Mar '24

Hello Everyone. I am looking for documentation to have multiple RTP Engine servers connected with kamailio using db url and balance the load accordingly. Thanks

4 3

Kamailio to aid in seamless data center migration?
by Brad 21 Mar '24

21 Mar '24

I need to move a few dozen FreePBXen with some commercial modules running in individual VMs to a new data center. I’m trying to work out a plan to move the PBXes to the new data center in a way that will be transparent to the endpoints, or at the very least with the absolute minimum of downtime. Some of the installations are rather old, and there’s a handful of peculiarities on each, so the typical FreePBX backup/restore process hasn’t gone smoothly, at least in our tests. My current train … [View More]

2 1

RFC5626 Keep-Alive with CRLF
by Ihor Olkhovskyi 21 Mar '24

21 Mar '24

Hello, A bit generic question, does Kamailio supports CRLF keepalive per https://www.rfc-editor.org/rfc/rfc5626#section-4.4.1 ? It's more for tracking TCP connection states, as seems tcp_keepalive_enable is not super reliable especially in mobile networks. -- Best regards, Ihor (Igor)

3 5

Branch Route trigger for Withindialog calls?
by Benoit Panizzon 21 Mar '24

21 Mar '24

Hi List I am just wondering... When I am sending the initial INVITE to a customer CPE, this goes throug the whole location lookup and through a branch route in which I make some last adjustments to the headers, like removing header the customer shall not get (like P-Asserted-Identity which would reveal the caller identity on a callerid restricted call). On a Re-Invite (session timer refresh) the call is being routed by loose_route() and immediately sent to RELAY. uac_replace and uac_restore … [View More]

2 1

loose_route false when using topos? At which point in the call flow, does topos restore the Route header?
by Benoit Panizzon 21 Mar '24

21 Mar '24

Hi https://lists.kamailio.org/pipermail/sr-users/2020-July/109801.html I have the exact same issue. When the B side is starting a new transaction (UPDATE to refresh the session in my case) without topos enabled, that transaction contains one or multiple route header and a to_tag. Therefore loose_route() is true and the call is more or less sent to route(RELAY) immediately without further checks. If topos is enabled, all record-route header are removed and only one Via sent to the CPE. So … [View More]

3 4

No 408 generated by tm module on fr_timer when redis crashes
by Patrick Karton 21 Mar '24

21 Mar '24

1 0

Wrong documentation for module 'Userblacklist'
by Björn Klasen 20 Mar '24

20 Mar '24

Hi everybody, I'm a little bit confused about the documentation of the Userblacklist module as I think it has to be Userblocklist since version 5.x. Also some of the commands and some column names in the database have changed. I can remember that the documentation was right at some point, but the it has been changed back to Userblacklist. I wonder if nobody noticed that because the old commands do not work. BR, Björn -- Björn Klasen, Senior Specialist (VoIP) TNG Stadtnetz GmbH, TNG-Technik … [View More]

2 1

Bug in htable in 5.8.0
by Brooks Bridges 18 Mar '24

18 Mar '24

We have some scripts that are setting values in an htable for various things, one of which is to disable options replies to take a system “out of service”. We discovered today upon deploying 5.8.0 that the “htable.seti” command appears to be broken. [root@ip-10-52-42-102 ~]# kamcmd htable.seti system_settings option_pings_off 1 error: 500 - Not enough parameters (htable name, key name and value) however if we do htable.sets it works fine (although not an integer so it’s breaking our shutdown … [View More]

3 4

Need guidance on Teams interop with Kamailio
by Shah Hussain Khattak 17 Mar '24

17 Mar '24

Hello Experts, I'm not an expert in Kamailio, but thanks to some excellent online resources, I managed to set up a basic interconnection between Teams and Kamailio. So far, outbound calls, as well as blind and attended call transfers, are working smoothly. However, I'm facing a challenge with call disconnection for outbound calls. Specifically, when an outbound call is made and the called party hangs up, the BYE message isn't being relayed properly towards the Teams side. In my routing … [View More]configurations, I remove the Record-Route header received from Teams and add a new Record-Route header with Kamailio's IP before forwarding it to my IP PBX (FreeSwitch). For the BYE message to be processed correctly, Microsoft Teams requires the original contact in the Request URI (RURI) and the Route header to match the initial Record-Route included in the INVITE. Currently, my BYE message appears as follows: BYE sip:api-du-a-auea.pstnhub.microsoft.com:443;x-i=5e96d703-b1e7-449c-aebf-a87bfa628176;x-c=f61b674bc597513ab8d888dbe7c560fb/d/8/9a6c508a9cbd44ebb892b62c33ba9b67 SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bKeab5.b657e3455b6d11ed6975c69882632a27.0 Via: SIP/2.0/UDP x.x.x.x;received=x.x.x.x;rport=5060;branch=z9hG4bKK3HKQBUvX42ya Max-Forwards: 69 From: <sip:+61403225xxx@kamsbc.xyzdomain.com:5061;user=phone>;tag=j5gptBSXXrXDc To: "XYZ Pty Ltd" <sip:+61291455xxx@sip.pstnhub.microsoft.com:5061;user=phone>;tag=3a5f0c9e32bc4eb1bffed29dc954118f Call-ID: f61b674bc597513ab8d888dbe7c560fb CSeq: 80555743 BYE User-Agent: Unknown/v1.0 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Reason: MVTSLocal;cause=10;text="BYE received" Content-Length: 0 Kamailio conducts a lookup for api-du-a-auea.pstnhub.microsoft.com but ends up sending the BYE message to an incorrect IP. Below is a snippet of my routing block configurations: ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks route(REQINIT); #check who is the sender route(INITCHECK); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # update $du to set the destination address for proxying #$du = "sip:" + $rd + ":9"; route(RELAY); exit; } route[INITCHECK] { if(from_uri =~ ".*microsoft.com") { setflag(FROM_TEAMS); $du = "sip:" + "MY_PBX_IP" + ":" + "5060"; //added by SHK remove_hf("Record-Route"); // Remove existing Record-Route header append_hf("Record-Route: <sip:KAM_IP:5060;transport=udp>\r\n"); route(HANDLE_RTP_FROM_TEAMS); } if(from_uri =~ ".*" + "MY_PBX_IP") { setflag(FROM_PBX); $du="sip:sip.pstnhub.microsoft.com;transport=tls"; route(HANDLE_RTP_FROM_PBX); } } #Manage RTP & transcoding comming from Teams to PBX route[HANDLE_RTP_FROM_TEAMS] { t_on_reply("PBX_REPLY_TO_TEAMS"); if (has_body("application/sdp")) { rtpengine_manage("RTP codec-mask=all codec-transcode=PCMA replace-origin replace-session-connection ICE=remove media-address=HOST_IP"); // record_route(); t_relay_to_udp("MY_PBX_IP","5060"); } else { rtpengine_manage(); } } #Manage RTP & transcoding comming from PBX to Teams route[HANDLE_RTP_FROM_PBX] { t_on_reply("TEAMS_REPLY_TO_PBX"); if (has_body("application/sdp")) { rtpengine_manage("SRTP codec-mask=all ICE=force codec-transcode=PCMA replace-origin replace-session-connection media-address=ADVERTISE_IP"); $rd = "sip.pstnhub.microsoft.com"; $td = "kamsbc.xyzdomain.com"; $fd = "kamsbc.xyzdomain.com"; // record_route(); #Set TLS SNI (server name & server id) $xavp(tls=>server_name) = "kamsbc.xyzdomain.com"; $xavp(tls=>server_id) = "kamsbc.xyzdomain.com"; t_relay(); } else { rtpengine_manage(); } } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS")) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; #Teams reINVITEs if(isflagset(FROM_TEAMS)) { loose_route(); t_relay(); exit; } if(isflagset(FROM_PBX)) { #Set TLS SNI (server name & server id) $xavp(tls=>server_name) = "kamsbc.xyzdomain.com"; $xavp(tls=>server_id) = "kamsbc.xyzdomain.com"; loose_route(); t_relay(); exit; } # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails #set coresponding cert on transactions if($fd == "kamsbc.xyzdomain.com") { $xavp(tls=>server_name) = "kamsbc.xyzdomain.com"; $xavp(tls=>server_id) = "kamsbc.xyzdomain.com"; } } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); } #PBX On Reply onreply_route[PBX_REPLY_TO_TEAMS] { if (has_body("application/sdp")) { rtpengine_manage("SRTP codec-mask=all codec-transcode=PCMA replace-origin replace-session-connection media-address=ADVERTISE_IP"); } else { rtpengine_manage(); } //added by SHK if (status == "200" && method == "INVITE") { remove_hf("Contact"); # Remove existing Contact header append_hf("Contact: <sip:kamsbc.xyzdomain.com:5061;transport=tls>\r\n"); } if (status=="200") { remove_hf("ALLOW:"); append_hf("ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY \r\n","CONTENT-TYPE"); } } #From Teams On Reply onreply_route[TEAMS_REPLY_TO_PBX] { if (has_body("application/sdp")) { rtpengine_manage("RTP codec-mask=all codec-transcode=PCMA replace-origin replace-session-connection media-address=HOST_IP"); } else { rtpengine_manage(); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { if (t_is_canceled()) exit; } event_route[tm:local-request] { if(is_method("OPTIONS") && $ru =~ "pstnhub.microsoft.com") { append_hf("Contact: <sip:kamsbc.xyzdomain.com:SBC_PORT;transport=tls>\r\n"); } xlog("L_INFO", "Sent out tm request: $mb\n"); } ####### Routing Logic End ######## Given these details, I suspect the issue might be related to how I'm handling the Record-Route headers or potentially a misconfiguration in directing the BYE message. I would greatly appreciate any insights, suggestions, or guidance on how to correctly relay the BYE message back to Teams, or further refine my existing configuration. Thank you in advance for your help and support! Regards, Shah Hussain [View Less]

2 2

topology hiding with active dispatcher ip
by marek 15 Mar '24

15 Mar '24

hi, i have kamailio acting as SBC i need hide topology like this ds_select_dst(DSP_GRP_TRUNK, "6"); $tu = $(tu{re.subst,/PRIVATE_IP/IP_OF_CURRENT_SELECTED_DISPATCHER/g}); what is best way for IP extraction from $du? thanks Marek

2 2

SCSF always challenge
by Anthony Blandin 14 Mar '24

14 Mar '24

Hello, I have an issue with SCSF which challenge again after a successful first registration. I have an error after the second register saying that no matching auth vector found. I attach the logs and pcap file. I don't see where is the issue. Could you help me? Thanks a lot. Anthony

1 0

append_hf not working
by Calvin E. 13 Mar '24

13 Mar '24

I'm trying to add something simple like the following: append_hf("X-testheader: True\r\n", "From"); However, I don't see my X-testheader in a packet capture. Are there any common pitfalls that would prevent append_hf from working as expected?

8 9

TLS requests get relayed over unencrypted TCP
by James Browne 13 Mar '24

13 Mar '24

Hi all I've set up a kamailio server on a public IP address to serve public clients. - The clients REGISTER over a mix of TCP and TLS, so kamailio has listeners on the same IP address: port 5060 for UDP/TCP and port 5061 for TLS. - Some clients REGISTER from the same public IP address, behind the same NAT. - Sometimes, the client-side NAT can use the same client-side IP address and port for two concurrent connections - one TCP and one TLS - which is normal because the server-side port is … [View More]different (they're different 5-tuples). - Now when a request comes to kamailio, to be routed to a client, the the t_relay_to_tls() function sometimes can relay the request over a TCP connection instead of a TLS connection. I've enabled tcp_connection_match=1. This is helpful, but it doesn't completely resolve the problem. It helps when both connections (TCP and TLS) are open, because the TLS connection will always be used for TLS requests. However, when there's no TLS connection open, kamailio will erroneously use the TCP connection. Note that this setup depends on TCP/TLS connections being made from the client to the server, not the other way around. I've made a cut-down version of this setup to demonstrate the problem. - Client: 11.15.32.1 - Server: 11.15.32.11 - version: kamailio 5.9.0-dev0 (x86_64/linux) ################################################################# #!KAMAILIO enable_tls=yes listen=tls:11.15.32.11:5061 listen=tcp:11.15.32.11:5060 listen=udp:11.15.32.11:5060 loadmodule "tm" loadmodule "tls" loadmodule "pv" loadmodule "rr" fork=yes log_facility=LOG_LOCAL0 log_stderror=no children=2 debug=3 tcp_connection_match=1 modparam("tls", "low_mem_threshold1", -1) modparam("tls", "low_mem_threshold2", -1) modparam("tls", "certificate", "/usr/local/etc/kamailio/certs/chain") modparam("tls", "private_key", "/usr/local/etc/kamailio/certs/key") request_route { loose_route(); $fs="tls:11.15.32.11:5061"; t_relay_to_tls(); } ################################################################# This is a BYE message to be sent from internally in the network. ########################### BYE sip:user3@11.15.32.1:33333;transport=tls SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:11111;rport;branch=z9hG4bK-d8754z-cc2c63344f5218d3-1 Route: <sip:11.15.32.11:5060;transport=tcp;r2=on;lr;ftag=ZUrDU2m4Z9Zam> Route: <sip:11.15.32.11:5061;transport=tls;r2=on;lr;ftag=ZUrDU2m4Z9Zam> f: <sip:user1@server>;tag=ZUrDU2m4Z9Zam t: <sip:user3@server>;tag=gK0ea97395 i:sdfg8we790t874ujk CSeq: 102 BYE Content-Length: 0 ########################### When I send that to kamailio over UDP, kamailio relays it over a TCP connection, as seen with socat here (socat runs on the machine with address 11.15.32.1). ########################### $ socat - TCP:11.15.32.11:5060,bind=:33333 BYE sip:user3@11.15.32.1:33333;transport=tls SIP/2.0 Via: SIP/2.0/TLS 11.15.32.11:5061;branch=z9hG4bKd2bf.137932ff9a937cde941fd1790db040c9.0 Via: SIP/2.0/UDP 0.0.0.0:11111;received=11.15.32.11;rport=11111;branch=z9hG4bK-d8754z-cc2c63344f5218d3-1 f: <sip:user1@server>;tag=ZUrDU2m4Z9Zam t: <sip:user3@server>;tag=gK0ea97395 i:sdfg8we790t874ujk CSeq: 102 BYE Content-Length: 0 ########################### Note that - the RURI has transport=tls in it, - kamailio is forcing the TLS socket, - kamailio is using the t_relay_to_tls function, - the Route header field tells that TLS should be used, and - the Via header field shows that kamailio tried to send this over TLS, but - the BYE was sent over unencrypted TCP. If I use this socat command instead, then I can receive the BYE, showing that kamailio is able to use TLS if there's a connection available. - socat - OPENSSL:11.15.32.11:5061,bind=:33333,verify=no I can't find any setting to control this, so it looks to me like a kamailio bug. At that, I see this as a security flaw, because the SIP traffic should always be encrypted but now is leaked. Can anyone help? James [View Less]

1 0

Lua classes in events function
by Michal Hajek 12 Mar '24

12 Mar '24

Hi, I use KEMI/Lua. I have simple lua class: myclass = {} myclass.event = function() KSR.xlog.xinfo("Event working...\n") end In kamailio.conf have: modparam("xhttp", "event_callback", "myclass.event") In debug log have: Mar 12 13:21:30 kamailio DEBUG: app_lua [app_lua_api.c:549]: sr_lua_reload_script(): No need to reload [/etc/kamailio/donjon.lua] is version 0 Mar 12 13:21:30 kamailio DEBUG: app_lua [app_lua_api.c:733]: app_lua_run_ex(): executing Lua function: [[myclass.event]] Mar 12 13:… [View More]

1 0

5.8 acc module failed_filter error
by Juha Heinanen 12 Mar '24

12 Mar '24

My SIP proxy didn't start with Kamailio 5.8 using the same config that starts OK with 5.7. The error is: 2024-03-11T06:56:06.457107+02:00 lohi /usr/bin/sip-proxy[2358954]: DEBUG: acc [acc_mod.c:362]: parse_failed_filter(): failed_filter 0 = 407 2024-03-11T06:56:06.457265+02:00 lohi /usr/bin/sip-proxy[2358954]: ERROR: acc [acc_mod.c:369]: parse_failed_filter(): response code is not followed by comma or end of string 2024-03-11T06:56:06.457436+02:00 lohi /usr/bin/sip-proxy[2358954]: ERROR: acc [… [View More]

2 5

Kamailio Yocto package
by Olle E. Johansson 11 Mar '24

11 Mar '24

Hello Kamailians! Before I start going down this path - anyone that has written a Kamailio Yocto layer and want to share either layer or experience or both? For those that doesn’t know: Yocto is a toolkit for building your own Linux distros for embedded systems. /O

1 0

SIP Redirect Server
by Antonio 11 Mar '24

11 Mar '24

Hi everyone I have a small question: i want to set up a basic SIP Redirect Server so I install Kamailio v5.3 and comment default request route and set simple request_route { rewritehostport("1.2.3.4"); sl_send_reply("302", "Moved Temporarily"); } but for some reason, Kamailio does not care about the port and sends a reply to 5060 [image: image.png] i also tried to delete rewritehostport but it only deletes Contact: from the reply but reply itself is still sent to 5060 port [image: … [View More]

4 4

5.8 compiler warning
by Juha Heinanen 10 Mar '24

10 Mar '24

Kamailio 5.8 build generates on my Debian 12 the following warning: core/mem/q_malloc.c:996:14: warning: 'qm_strnstr' defined but not used [-Wunused-function] 996 | static void *qm_strnstr(const void *b1, int l1, const void *b2, int l2) | ^~~~~~~~~~ Perhaps the definition of the function should be #ifdef'ed by DBG_QM_MALLOC. -- Juha

1 0

SSL key logger for Diffie-Hellman cipher
by Calvin E. 08 Mar '24

08 Mar '24

I've been tasked to use LD_PRELOAD to log SSL keys for TLS connections using a Diffie-Hellman cipher. The first attempt did not work, so I wanted to sanity check whether Kamailio's TLS support is built in such a way that would defeat LD_PRELOAD. The instructions from the vendor are to update /etc/init.d/kamailio like this: env SSLKEYLOG_UDP='127.0.0.1:1234' LD_PRELOAD="/usr/local/src/voipmonitor-git/tools/ssl_keylogger/sslkeylog.so /usr/lib/x86_64-linux-gnu/libssl.so.3" \ start-stop-… [View More]

6 16

from TCP to UDP and Kamailio doing it wrong
by Alberto Diez 08 Mar '24

08 Mar '24

Hi kamailio community, I have an issue with a Kamailio 5.7. It's listening both in TCP and UDP. In my scenario requests arrive from devices on TCP, but I want to forward to the next hops on UDP. I am avoiding using any type of DNS resolution; since I am always forwarding to predefined next hops I am using the dispatcher module (defined with the IP addresses and transport=udp) or I wrote config files using t_relay_to_udp or t_relay_to with a udp: followed by IP address. … [View More]

5 9

direct media between UACs
by Sergio Charrua 08 Mar '24

08 Mar '24

Hi all! got a weird behavior that I cannot understand the reason for... In our LAB environment, we have 2 Asterisk instances (version 13.38.3 and chan_sip) and 1 Kamailio 5.7 in between. All servers are in the same network, so, there is no NAT involved. No RTPEngine either. Network is 10.20.0.0/24 and Asterisk #1 has IP .1 Asterisk #2 has IP .3 and Kamailio has IP .5 The Asterisk servers are used only for testing, nothing serious. However, Kamailio is setup to use RTJson requesting routes to … [View More]a Routing Server on the same network. And it works fine. Both Asterisk servers have the same dialplan, which only Answers the call and plays MOH on both ends so that RTP audio streams both ways. When making a call on Asterisk Server #1 via command line to go directly to Asterisk Server #2 without using Kamailio (CLI> channel originate SIP/123(a)10.20.0.3 application MusicOnHold() ) the Asterisk #2 receives the call, answers and plays MOH too and I can see RTP streams coming from both ends correctly. However, if I use Kamailio to proxy the call generated from Asterisk #1 to Asterisk #2, using similar command line instruction (CLI> channel originate SIP/123(a)10.20.0.5 application MusicOnHold() ), the call is indeed received on Kamailio who then sends it to Asterisk #2, who answers the call and plays MOH, *but* despite the audio stream being sent to Asterisk #1 it is never received, however audio from Asterisk #1 is received by Asterisk #2, which configures a typical One Way Audio issue due to NAT. This is where it gets strange, because there is no NAT, SDP on INVITE and SIP 200 messages seem OK, as far as I understand it. Also, Asterisk servers have SIP configuration with directmedia enabled and NAT disabled to make sure that media is direct. But I have also tried with directmedia disabled and NAT enabled and get identical results. I am most probably missing some tiny detail, but I have no clue.... and I bet it is simple and stupid.... Could another pair of eyes help me with this? What is wrong? Do I really need RTPEngine even when the network has no NAT? I am sure it would work that way, but it doesn't make sense... Here are some screenshots: Call Scenario #1 - direct call from Asterisk #1 to Asterisk #2 without Kamailio in between: Invite from Asterisk #1 to Asterisk #2 with direct media between both ends: https://drive.google.com/file/d/1eLjT3nr_Rc-UBaf4QhIgZ95bjETOVvxo/view?usp=… Replies from Asterisk #2 to Asterisk #1 with direct media between both ends: https://drive.google.com/file/d/11lLcB-V8rWGSrVqWiit-q9WX2FfqB6BZ/view?usp=… Call Scenario #2 - call from Asterisk #1 using Kamailio to relay call to Asterisk #2, with one way audio Invite from Asterisk #1 to Asterisk #2 via Kamailio with SDP details: https://drive.google.com/file/d/1Cp9xrGcwNmQ9Ks36N_oD1Dj7lxfu-tbH/view?usp=… Invite from Kamailio relayed to Asterisk #2 with SDP details from Asterisk #1 identical to above: https://drive.google.com/file/d/1mi3FCNjM3luXfENEp-0088XLgcyfXRK6/view?usp=… Reply from Asterisk #2 to Kamailio with SDP details: https://drive.google.com/file/d/1TpMGe2tvpX_5SIpSbm2b3Zro9EuYwcO-/view?usp=… Reply from Kamailio to Asterisk #2 with SDP details from Asterisk #2 identical to above: https://drive.google.com/file/d/12jq5APfFwVVPc0vJ3RkcXyN1hBE51fnQ/view?usp=… As we can see, SDP details seem OK, but if I check call flow on Asterisk #1, I can only find 1 RTP channel with audio coming from Asterisk #2 https://drive.google.com/file/d/1iEfvkylZVbthHM5kxkurWh-GAYCYLytl/view?usp=… and the same on Asterisk #2 : https://drive.google.com/file/d/12rvf9Lrwp-MNZvGCwlXEEBsBRmfcinph/view?usp=… My Kamailio.cfg is as follows: #!KAMAILIO # # config file for SIPProxy # - load balancing of VoIP calls # - no TPC listening # # Kamailio (OpenSER) SIP Server v3.2 # - web: http://www.kamailio.org # - git: http://sip-router.org # # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # #!define WITH_DEBUG ###!define WITH_NAT #!define WITH_PSTN /* enables Accounting Log functions */ #!define FLT_ACC 1 /* enable Accounting of missed or failed calls */ #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 /* defines DB connection string */ #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailio@10.20.0.1:3306/kamailio" #!endif # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif #!define FLT_DISPATCH_SETID 1 #!define FLT_FS 10 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 #!define FLT_SRC_ALLOWED 8 #!define FLT_DST_INTERNAL_IP 9 #!define FLT_SRC_INTERNAL_IP 10 #!substdef "!INTERNAL_IP_NET!10.20.0.0/24!g" #!substdef "!INTERNAL_IP_ADDR!10.20.0.2!g" #!substdef "!EXTERNAL_IP_ADDR!10.20.0.2!g" #!ifndef HTTP_ASYNC_CLIENT_WORKERS #!define HTTP_ASYNC_CLIENT_WORKERS 8 #!endif /* add API http timeout */ #!define HTTP_API_TIMEOUT 5000 #!define HTTP_API_ROUTING_ENDPOINT "http://10.246.212.40:7778/get_route" /* DMQ SIP message sharing */ #!define DMQ_PORT 5062 #!define DMQ_LISTEN "sip:10.20.0.2:5062" #!define DMQ_SERVER_ADDRESS "sip:10.20.0.2:5062" #!define DMQ_NOTIFICATION_ADDRESS "sip:10.20.0.4:5062" memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " fork=yes children=8 /* comment the next line to enable TCP - all trunks are UDP only */ disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ auto_aliases=no port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:10.20.0.5:5060 advertise 10.20.0.5:5060 listen=tcp:10.20.0.5:5060 advertise 10.20.0.5:5060 listen=udp:10.20.0.2:5062 advertised_address="10.20.0.5"; sip_warning=no; use_dns_failover = on; ####### Modules Section ######## #set module path mpath="/usr/local/lib64/kamailio/modules/" loadmodule "db_mysql.so" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "acc.so" loadmodule "usrloc.so" loadmodule "nathelper.so" #loadmodule "rtimer.so" #loadmodule "sqlops.so" # --- CPS Limiter # --- end of CPS Limiter loadmodule "ipops.so" loadmodule "textopsx.so" loadmodule "sdpops.so" loadmodule "http_async_client.so" loadmodule "rtjson.so" loadmodule "jansson.so" loadmodule "dmq.so" loadmodule "dmq_usrloc.so" loadmodule "htable.so" loadmodule "dialog.so" #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "log_level_name", "exec") #!endif # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") modparam("jsonrpcs", "pretty_format", 1) # ----- rr params ----- modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") modparam("jsonrpcs", "pretty_format", 1) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- acc params ----- modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default we do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") //;calltype=$avp(calltype)") # ----- tm params ----- # ----- the TM module enables stateful processing of SIP requests modparam("tm", "fr_timer", 5000) modparam("tm", "fr_inv_timer", 60000) modparam("tm", "remap_503_500", 0) # ----- usrloc params ----- /* enable DB persistency for location entries */ modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) # params needed for NAT traversal in other modules modparam("usrloc", "nat_bflag", FLB_NATB) # ----- nathelper params ----- modparam("nathelper", "received_avp", "$avp(s:rcv)") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:ping@kamailio.org") #modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;") #modparam("rtimer", "exec", "timer=cdr;route=CDRS") #modparam("sqlops", "sqlcon", "ca=>mysql://kamailio:kamailiorw@10.19.139.113:3306/kamailio") #modparam("dmq", "server_socket", DMQ_SERVER_SOCKET ) modparam("dmq", "server_address", DMQ_SERVER_ADDRESS ) modparam("dmq", "notification_address", DMQ_NOTIFICATION_ADDRESS ) modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 4) modparam("dmq", "ping_interval", 60) modparam("dmq_usrloc", "enable", 1) # -- CPS Limiter modparam("htable", "htable", "rhs=>size=32;initval=0;autoexpire=10;") modparam("htable", "htable", "rhm=>size=32;initval=0;autoexpire=120;") modparam("htable", "enable_dmq", 1) modparam("htable", "dmq_init_sync", 1) modparam("dialog", "profiles_with_value", "concurrent_calls") modparam("dialog", "enable_dmq", 1) # ----- http_async_client params ----- modparam("http_async_client", "workers", HTTP_ASYNC_CLIENT_WORKERS) modparam("http_async_client", "connection_timeout", 2000) ####### Routing Logic ######## # main request routing logic route { if (is_method("KDMQ") && $Rp == 5062) { dmq_handle_message(); } xlog("L_INFO"," ********** Route START ***********"); # log the basic info regarding this call xlog("L_INFO","start|\n"); xlog("L_INFO","===================================================\n"); xlog("L_INFO","New SIP message $rm with call-ID $ci \n"); xlog("L_INFO","---------------------------------------------------\n"); xlog("L_INFO"," received $pr request $rm $ou\n"); xlog("L_INFO"," source $si:$sp\n"); xlog("L_INFO"," from $fu\n"); xlog("L_INFO"," to $tu\n"); xlog("L_INFO","---------------------------------------------------\n"); xlog("L_INFO","---------------------------------------------------\n"); # OPTIONS requests without a username in the Request-URI but one # of our domains or IPs are addressed to the proxy itself and # can be answered statelessly. if (is_method("OPTIONS")) { sl_send_reply("200","OK"); exit; } if ($fU=="ping") { sl_send_reply("200","OK"); exit; } # extract original source ip / port from X-forwarded-For header route(HANDLE_X_FORWARDED_FOR); # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()){ route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); xlog("L_INFO", "ROUTE - Exiting after Retransmission check - method $rm"); exit; } t_check_trans(); } route(WITHINDLG); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers xlog("L_INFO", "ROUTE - Removing Headers"); remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")){ t_on_failure("MANAGE_FAILURE"); xlog("L_INFO", "ROUTE - Recording Route"); record_route(); if (is_method("INVITE") && is_request()) { if (has_body("application/sdp")) { xlog("L_INFO", "ROUTE - goiing to t_on_reply[ON_REPLY]\n"); t_on_reply("ON_REPLY"); } } } if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","ROUTE - Address Incomplete"); exit; } route(TOCARRIER); xlog("L_INFO", " ********** Route END *************"); } # extract original source ip / port from X-forwarded-For header route[HANDLE_X_FORWARDED_FOR] { if (is_present_hf("X-Forwarded-For")) { $var(source_ip) = $(hdr(X-Forwarded-For){s.select,0,:}); $var(source_port) = $(hdr(X-Forwarded-For){s.select,1,:}); } else { $var(source_ip) = $si; $var(source_port) = $sp; } $var(to_number) = $rU; } route[RELAY_API] { xlog("L_INFO","RELAY_API - from_ip $var(source_ip):$var(source_port) from_number $fU to_number $ru"); $http_req(all) = $null; $http_req(suspend) = 1; $http_req(timeout) = HTTP_API_TIMEOUT; $http_req(method) = "POST"; $http_req(hdr) = "Content-Type: application/json"; jansson_set("string","from_ip",$var(source_ip), "$var(http_routing_query)"); jansson_set("string","from_port",$var(source_port), "$var(http_routing_query)"); jansson_set("string","from_number",$fU, "$var(http_routing_query)"); jansson_set("string","to_number",$var(to_number) , "$var(http_routing_query)"); xlog("L_INFO","RELAY_API - API ASYNC ROUTING REQUEST: $var(http_routing_query)\n"); $http_req(body) = $var(http_routing_query); t_newtran(); http_async_query(HTTP_API_ROUTING_ENDPOINT, "RELAY_API_RESPONSE"); } # Relay request using the API (response) route[RELAY_API_RESPONSE] { if ($http_ok==1 && $http_rs==200) { xlog("L_INFO","RELAY_API_RESPONSE - RESPONSE: $http_rb\n"); if (jansson_get("rtjson", $http_rb, "$var(rtjson)")) { xlog("L_INFO","RELAY_API_RESPONSE - $var(rtjson)"); rtjson_init_routes("$var(rtjson)"); rtjson_push_routes(); # relay the message t_on_branch("MANAGE_BRANCH"); t_on_failure("MANAGE_FAILURE"); route(RELAY); return; } } send_reply(500, "API Not Available - http response = $http_rs $http_ok"); exit; } onreply_route[ON_REPLY] { xlog("L_INFO", "ON_REPLY - In onreply_route[ON_REPLY] $rs"); # on reply if (t_check_status("183|180|200")) { xlog("L_INFO", "ON_REPLY - Fixing Contacts"); # subst_hf("Contact","/@.*:/@EXTERNAL_IP_ADDR:/","a"); //subst_hf("Record-Route","/INTERNAL_IP_ADDR/EXTERNAL_IP_ADDR/","f"); } if (has_body("application/sdp")){ if (sdp_remove_line_by_prefix("a=maxptime")){ xlog("L_INFO", "ON_REPLY - remove maxptime "); msg_apply_changes(); } else{ xlog("L_INFO", "ON_REPLY - did not removed maxptime "); } } if (t_check_status("408")) { xlog("L_INFO", "ROUTE - Handling 408 Timeout\n"); } } route[TOCARRIER]{ #using rtjson, unsomment following line route(RELAY_API); } # Per SIP request initial checks route[REQINIT] { xlog("L_INFO", "REQINIT - Starting"); if (!mf_process_maxfwd_header("10")) { xlog("L_INFO", "REQINIT - 483 - Too Many Hops"); sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("L_INFO","REQINIT - Sanity Check -> Malformed SIP message from $si:$sp\n"); exit; } } # Caller NAT detection route[NATDETECT] { xlog("L_INFO", "NATDETECT - Entering"); #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { xlog("L_INFO", "NATDETECT - Fix Nated Register"); fix_nated_register(); } else { if(is_first_hop()){ xlog("L_INFO", "NATDETECT - Set Contact Alias"); set_contact_alias(); } } xlog("L_INFO", "NATDETECT - Set FLT_NATS" + FLT_NATS); setflag(FLT_NATS); } #!endif xlog("L_INFO", "NATDETECT - NAT Detect set FLT_NTS = " + FLT_NATS); return; } # Handle requests within SIP dialogs route[WITHINDLG] { xlog("L_INFO", "WITHINDLG - Entering"); if (!has_totag()) return; if (is_present_hf("Route") && $hdrc(Route)==1) { if (search_hf("Route", ".*EXTERNAL_IP_ADDR.*", "f")) { xlog("L_INFO", "WITHINDLG - Removing the route to self"); remove_hf("Route"); } } # sequential request within a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE|CANCEL")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy xlog("L_INFO", "WITHINDLG - Going to NATMANAGE"); route(NATMANAGE); } else if ( is_method("NOTIFY") ) { #Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } if(is_method("BYE")) xlog("L_INFO", "WITHINDLG - BYE message from $rU"); route(RELAY); exit; } if ( is_method("ACK|BYE|INVITE|UPDATE") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction. Try to route anyway - being optimistic # since it has at least a To Tag route(RELAY); exit; } } sl_send_reply("404","Not here"); xlog("L_INFO", "WITHINDLG - Finishing WITHINDLG"); exit; } # URI update for dialog requests route[DLGURI] { xlog("L_INFO", "WITHINDLG - Entering DLGURI"); #!ifdef WITH_NAT if(!isdsturiset()) { xlog("L_INFO", "WITHINDLG - Handle ruri ALIAS"); handle_ruri_alias(); } #!endif return; } # Routing to foreign domains ---> NOT USED route[SIPOUT] { xlog("L_INFO", "WITHINDLG - Entering SIPOUT"); if (uri==myself){ xlog("L_INFO", "WITHINDLG - URI is MySelf!"); return; } append_hf("P-hint: outbound\r\n"); xlog("L_INFO", "WITHINDLG - Finishing SIPOUT"); route(RELAY); exit; } # Wrapper for relaying requests route[RELAY] { xlog("L_INFO", " ******** RELAY *******"); xlog("L_INFO", "RELAY - $si $su $ru"); # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|CANCEL|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) { xlog("L_INFO", "RELAY - branch_route NOT SET!"); t_on_branch("MANAGE_BRANCH"); } } xlog("L_INFO", "RELAY - checking method"); if (is_method("INVITE|SUBSCRIBE|UPDATE")) { xlog("L_INFO", "RELAY - is INVITE|SUBSCRIBE|UPDATE"); if(!t_is_set("onreply_route")) { xlog("L_INFO", "RELAY - onreply_route NOT SET!"); t_on_reply("ON_REPLY"); # MANAGE_REPLY"); } } if (is_method("INVITE")) { xlog("L_INFO", "RELAY - is INVITE"); t_on_failure("FAILED_RELAY"); if(!t_is_set("failure_route")) { xlog("L_INFO", "RELAY - failure_route NOT SET!"); t_on_failure("MANAGE_FAILURE"); } } if (!t_relay()) { xlog("L_INFO", "RELAY - t_relay returns FALSE"); route("MANAGE_FAILURE"); #sl_reply_error(); } xlog("L_INFO", "RELAY - exiting"); exit; } failure_route[FAILED_RELAY] { xlog("L_INFO", "FAILED_RELAY - Entering"); if (t_check_status("[4-5][0-9][0-9]")){ xlog("L_INFO", "FAILED_RELAY - Could not reach destination endpoint!"); if (rtjson_next_route()) { xlog("L_INFO", "MANAGE_FAILURE - Getting next route"); t_on_branch("MANAGE_BRANCH"); t_on_failure("MANAGE_FAILURE"); route(RELAY); } } } route[NATMANAGE] { xlog("L_INFO", "NATMANAGE - Entering"); #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { xlog("L_INFO", "NATMANAGE - nat=yes --- Setting FLB_NATB"); setbflag(FLB_NATB); } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB) )) { xlog("L_INFO", "NATMANAGE - NO FLT_NATS/B Set!!! Getting out of NATMANAGE"); return; } if (is_request()) { xlog("L_INFO", "NATMANAGE - is_request - $rm from $si"); if (!has_totag()) { if(t_is_branch_route()) { xlog("L_INFO", "NATMANAGE - adding nat=yes"); add_rr_param(";nat=yes"); } } } if (is_reply()) { xlog("L_INFO", "NATMANAGE - is_reply - $rm from $si"); if(isbflagset(FLB_NATB)) { if(is_first_hop()) { xlog("L_INFO", "NATMANAGE - Set Contact Alias"); set_contact_alias(); } } } #!endif return; } # Manage failure routing cases route[MANAGE_FAILURE] { xlog("L_INFO", "MANAGE_FAILURE - Entering "); route(NATMANAGE); xlog("L_INFO", "MANAGE_FAILURE - t_is_canceled"); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { xlog("L_INFO", "MANAGE_FAILURE - SIP 3XX returned!!"); t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { xlog("L_INFO", "MANAGE_FAILURE - SIP 401|407 returned!!"); t_reply("404","Not found"); exit; } #!endif if (t_check_status("503")){ xlog("L_INFO", "MANAGE_FAILURE - SIP 503 returned : no destination available"); t_reply("503", "Destination not available"); exit; } if (rtjson_next_route()) { xlog("L_INFO", "MANAGE_FAILURE - Getting next route!!"); t_on_branch("MANAGE_BRANCH"); t_on_failure("MANAGE_FAILURE"); route(RELAY); exit; } } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xlog("L_INFO","MANAGE_BRANCH - New branch [$T_branch_idx] to $ru\n"); xlog("L_INFO", "MANAGE_BRANCH - branch_route MANAGE_BRANCH 1 "); rtjson_update_branch(); route(NATMANAGE); } Any help would be greatly appreciated. Thanks in advance. Sérgio Charrua [View Less]

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Kamailio v5.8.0 Released - new major version is out
by Daniel-Constantin Mierla 07 Mar '24

07 Mar '24

Kamailio v5.8.0 is out – it comes with 4 new modules and a large set of improvements touching again more than 50 existing modules. You can read a bit more detailed release notes at: * https://www.kamailio.org/w/kamailio-v5-8-0-release-notes/ Many thanks to all developers and community members that made possible this release. v5.8.0 brings more flexibility and optimizations across many existing components and modules, new parameters, functions, variables and transformations. … [View More]

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Using uac_replace_to when forwarding call to another endpoint
by Marrold 07 Mar '24

07 Mar '24

Hi all, I am using uac_replace_to to replace the TO header on requests. I understand it can only be called once per message and the recommendation is to store the updated value in a pvar and apply it in the branch route. However in my scenario I wish to first try one endpoint and then if it gets a 4XX response forward it to another UAC, and update the TO accordingly. Currently trying to call uac_replace_to twice shows an error and corrupts the TO header. Is there a work around for this? Thanks again Matthew

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Roadmap to next major Kamailio release series v5.8.x
by Daniel-Constantin Mierla 07 Mar '24

07 Mar '24

Hello, discussed a bit during the online Kamailio devel meeting, it is time to set the milestones towards the next major Kamailio release series v5.8.x. If no other suggestions that suit more developers, I would propose to freeze by end of this month or early February, then test for about 4 weeks as usual and release by end of February or during March. If anyone wants to add new features/modules, they have to be published till freezing date, either pushed in the git repository or proposed as … [View More]

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rtimer intermittent failures
by Marrold 06 Mar '24

06 Mar '24

Hi all, I am using Kamailio 5.7.4 on a Debian 12 machine, with a Python Kemi based config. I am seeing some intermittent failures when accessing an instance variable within a function called by rtimer. I'm using the rtimer module with the following parameters: modparam("rtimer", "timer", "name=hello;interval=5;mode=0;") modparam("rtimer", "exec", "timer=hello;route=ksr_route_hello") I initialise the kamailio class, and instance variable like this: class kamailio: def __init__(self): … [View More]

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Question for RFC junkies / kamailio modifying message bodies
by Valentin Christoph 06 Mar '24

06 Mar '24

Hi all, it might be a minor issue, or it might not be an issue at all, but I stumbled over following statements in RFC 3261: Chapter 16.6 Request Forwarding (of a proxy) 1. Copy request The proxy starts with a copy of the received request. The copy MUST initially contain all of the header fields from the received request. Fields not detailed in the processing described below MUST NOT be removed. The copy SHOULD maintain the … [View More]

3 2

Kamailio closing tcp connection on downstream responses
by Kjartan S 06 Mar '24

06 Mar '24

Trying to setup Kamailio with MS Teams Direct Routing. I am familiar with the integration steps required by Microsoft (using contact header as FQDN, certified CA for certificate) aswell as the famous guide; https://skalatan.de/en/blog/kamailio-sbc-teams + https://learn.microsoft.com/en-us/microsoftteams/direct-routing-protocols-s… Sending any SIP requests (Invite, ACK, BYE) toward MS is working fine (with provisional response received in Kamailio) - so there is obviously a working connection … [View More]between Kamailio <-> MS Direct Routing environment. HOWEVER, once Kamailio is forwarding SIP responses to MS (provisional responses/200 OK received from UAS), then Kamailio is closing the TCP connection. Please see attached logs below. Any idea why Kamailio is closing the connection when trying to send provisional responses to MS? - Using t_realy on original invite from MS - tcp_reuse_port=yes, tcp_rd_buf_size = 16384, tcp_connection_lifetime=3605, children=8, socket_workers=4 - No override rules in firewall - OS; debian:11.7 - Notice that TLS connection is made on port :26627 - however, MS sends port :5061 in VIA header (on first Invite). This seems to me to be a conflict in ports. https://www.kamailio.org/wiki/cookbooks/5.4.x/core#reply_route "There is no network route that can be enforced for a SIP reply - it is sent based on Via header, according to SIP RFC3261" ----- Handshake complete for 52.114.76.76:26627 19(43) DEBUG: tls [tls_domain.c:818]: sr_ssl_ctx_info_callback(): SSL handshake done 19(43) DEBUG: tls [tls_server.c:473]: tls_accept(): TLS accept successful 19(43) DEBUG: tls [tls_server.c:476]: tls_accept(): tls_accept: new connection from 52.114.76.76:26627 using TLSv1.2 ECDHE-RSA-AES256-GCM-SHA384 256 19(43) DEBUG: tls [tls_server.c:480]: tls_accept(): tls_accept: local socket: 10.129.0.91:5061 19(43) DEBUG: tls [tls_server.c:420]: tls_dump_cert_info(): tls_accept: client certificate subject:/C=US/ST=WA/L=Redmond/O=Microsoft Corporation/CN=sip.pstnhub.microsoft.com 19(43) DEBUG: tls [tls_server.c:424]: tls_dump_cert_info(): tls_accept: client certificate issuer:/C=US/O=Microsoft Corporation/CN=Microsoft Azure RSA TLS Issuing CA 04 ----- Kamailio trying to forward SIP response "183" to MS (response received from UAS). Kamailio finds active connection by ID 13 / 0x7f4ecad0be08 20(44) DEBUG: {2 1 INVITE e926eb82d1a25bdc9ba03048cdbc5c77} <core> [core/tcp_main.c:1722]: _tcpconn_find(): found connection by id: 13 20(44) DEBUG: {2 1 INVITE e926eb82d1a25bdc9ba03048cdbc5c77} <core> [core/tcp_main.c:2627]: tcpconn_send_put(): tcp connection found (0x7f4ecad0be08), acquiring fd 20(44) DEBUG: {2 1 INVITE e926eb82d1a25bdc9ba03048cdbc5c77} <core> [core/tcp_main.c:2637]: tcpconn_send_put(): c=0x7f4ecad0be08, n=16 23(47) DEBUG: <core> [core/tcp_main.c:3982]: handle_ser_child(): read response= 7f4ecad0be08, 2, fd -1 from 20 (44) 20(44) DEBUG: {2 1 INVITE e926eb82d1a25bdc9ba03048cdbc5c77} <core> [core/tcp_main.c:2665]: tcpconn_send_put(): after receive_fd: c= 0x7f4ecad0be08 n=8 fd=12 20(44) DEBUG: {2 1 INVITE e926eb82d1a25bdc9ba03048cdbc5c77} <core> [core/tcp_main.c:2842]: tcpconn_do_send(): sending... 20(44) DEBUG: {2 1 INVITE e926eb82d1a25bdc9ba03048cdbc5c77} <core> [core/tcp_main.c:2878]: tcpconn_do_send(): after real write: c= 0x7f4ecad0be08 n=1191 fd=12 ----- Kamailio immediately closes the connection due to EOF. 19(43) DEBUG: <core> [core/tcp_read.c:280]: tcp_read_data(): EOF on connection 0x7f4ecad0be08 (state: 0, flags: 4018) - FD 10, bytes 0, rd-flags 10000 ([52.114.76.76]:26627 -> [52.114.76.76]:5061)19(43) DEBUG: <core> [core/tcp_read.c:1544]: tcp_read_req(): EOF 19(43) DEBUG: <core> [core/io_wait.h:597]: io_watch_del(): DBG: io_watch_del (0x555b3cf20260, 10, -1, 0x10) fd_no=2 called 19(43) DEBUG: <core> [core/tcp_read.c:1927]: handle_io(): removing from list 0x7f4ecad0be08 id 13 fd 10, state 2, flags 4018, main fd 57, refcnt 2 ([52.114.76.76]:26627 -> [52.114.76.76]:5061) 19(43) DEBUG: <core> [core/tcp_read.c:1702]: release_tcpconn(): releasing con 0x7f4ecad0be08, state -1, fd=10, id=13 ([52.114.76.76]:26627 -> [52.114.76.76]:5061) [View Less]

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RADIUS ACC - Interim Updates
by duarterocha91＠gmail.com 06 Mar '24

06 Mar '24

Hello all, I can't seem to find a way to make interim accounting requests for RADIUS. Can you help me with this? I can't find any native way for doing this. I've thought of getting a custom timer for each dialog and run a event route similar do dialog:start or dialog:end. Is that possible? Cheers, Duarte Rocha

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Add XML to INVITE
by David Villasmil 06 Mar '24

06 Mar '24

Hello guys, I'm trying to add an xml with a boundary to an outgoing INVITE. But if i do: set_body_multipart("$rb", "application/sdp", "delimiter"); msg_apply_changes(); append_body_part("$var(something)", "application/pidf+xml"); msg_apply_changes(); but kamailio adds the SDP, but NOT the second $var(something). If i switch them and add firs the var, and then the SDP. kamailio adds the contents of $var(something) but NOT the SDP.... point is it doesn't add the second append Thanks!!! … [View More]

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