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DMQ Not Discovering all Nodes
by John Petrini 16 Aug '17

16 Aug '17

Hello, I've just started doing some testing with dmq but I'm having trouble getting it to discover all of the nodes in my cluster. There are 7 kamailio instances with the following in their configs: Example from the notification server (10.0.10.211): listen=udp:10.0.10.211:5080 loadmodule "dmq.so" # ----- dmq params ----- modparam("dmq", "server_address", "sip:10.0.10.211:5080") modparam("dmq", "notification_address", "sip:10.0.10.211:5080") Example from one of the nodes (10.0.10.216). (Only the listen and server_address is changed to match the local ip of each node.) listen=udp:10.0.10.216:5080 loadmodule "dmq.so" # ----- dmq params ----- modparam("dmq", "server_address", "sip:10.0.10.216:5080"); modparam("dmq", "notification_address", "sip:10.0.10.211:5080") Output from kamcmd dmq.list_nodes { host: 10.0.10.211 port: 5080 resolved_ip: 10.0.10.211 status: 2 last_notification: 0 local: 0 } { host: 10.0.10.216 port: 5080 resolved_ip: 10.0.10.216 status: 2 last_notification: 0 local: 1 } I'm expecting to see all 7 nodes in the output but only see the local node and the notification node. I imagine I could use an SRV record for the notication address to add all of the nodes but I thought the notification node was supposed to share the rest of the nodes in the cluster. Can anyone explain what it is I'm doing wrong? Thank You, John Petrini

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Kamailio not forces TLS socket
by Yuriy Gorlichenko 16 Aug '17

16 Aug '17

Hi! I using uac for send MESSAGE request from kamailio to remote destination endpoint Before sending i setting up $fs seudovariable as $fs = "tls:12.3.4.5:443"; kamailio listeinig this interface and succesfully receives requests and sending responses on it When i call uac_req_send();There is error in log file: ERROR: tm [ut.h:327]: uri2dst2(): no corresponding socket found for "2.3.26.11" af 2 (udp:2.3.26.11:57050) ERROR: tm [uac.c:272]: t_uac_prepare(): t_uac: no socket found If I setting up udp interface for listening then kamailio ognores my $fs set up and sending message through udp socket. kamailio version 4.4.6 Now i trying to roll back to 4.4.4 version but not sure if it will be helpfull.

1 0

Permission denied for /var/run/kamailio
by Ryan Wagoner 16 Aug '17

16 Aug '17

I'm running Kamailio 4.4.6-2.1 installed from RPM on CentOS 7, where /var/run is a tmpfs (by default). After every reboot Kamailio fails to start with the following error: Aug 16 00:27:39 sbc1 /usr/sbin/kamailio[7135]: ERROR: mi_fifo [fifo_fnc.c:72]: mi_init_fifo_server(): Can't create FIFO: Permission denied (mode=432) I have the following line in my kamailio.cfg modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo") At this point the /var/run/kamailio directory looks like this drwx------. 2 root root 80 Aug 16 00:30 . drwxr-xr-x. 26 root root 880 Aug 16 00:30 .. srw-------. 1 kamailio kamailio 0 Aug 16 00:30 kamailio_ctl If I run "chown kamailio /var/run/kamailio" and "systemctl start kamailio" I am good to go. Am I missing something in my config? It appears Kamailio is creating the /var/run/kamailio folder as root, writing the kamailio_ctl file, dropping root privileges, and then trying to write the kamailio_fifo file. Thanks, Ryan

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sipcapture/HEP msg sizes>1500
by Daniel Tryba 16 Aug '17

16 Aug '17

While looking into HOMER I saw that large INVITEs got truncated in the webinterface, looking at the HEP traffic these (UDP) messages got fragmented in transport (asterisk/res_hep) and I started to think these were not getting reassembled correctly. Only to find out the database schemas have set the msg size set to varchar(1500). Is there any good reason for not changing this to a bigger default?

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RTP loops
by Mauro Costantini 16 Aug '17

16 Aug '17

Hi everyone, we have an installation with: - 1 kamailio instance - with 2 interface, one with public and one with private IP (for local communication to the db and asterisk ). We use keepalived service for failover and so, every interface has 2 IPs: the real one and the “virtual” one for the keepalived service - some asterisk instances for transcoding and billing - with one interface with private IP - 2 RTP (rtpengine) instances - with 2 interfaces, one with public and one with private IP A few times happens a loop on the RTPengine and we could see a lot of lines like this in the log: “Too many packets in UDP receive queue (more than 50), aborting loop. Dropped packets possible” This loop doesn't stop until all the resources are exhausted and it results in the freeze of the machine. We don’t understand the reason why these loops are generated. We have 2 cases: 1) we send an INVITE to the CPE (a Bria on iOS) The CPE answers with a "100 Trying", a "180 Ringing" and a few seconds later sends us 3 packets of "200 ok" with SDP like this: SIP/2.0 200 OK Via: SIP/2.0/UDP 178.250.x.y;branch=z9hG4bKb842.29952898c30d6d06b75b916618357ccd.2 Via: SIP/2.0/UDP 192.168.x.y:5090;rport=5090;branch=z9hG4bK7fd100fb Record-Route: <sip:178.250.x.y;r2=on;lr=on;ftag=as26b82ac7;did=cb5.0ce;nat=yes> Record-Route: <sip:192.168.x.z;r2=on;lr=on;ftag=as26b82ac7;did=cb5.0ce;nat=yes> Contact: <sip:6051018****@158.148.x.y:3180;rinstance=ffdff5e0142dfd2d> To: <sip:6051018****@192.168.x.z>;tag=29859d3c From: <sip:051438****@192.168.x.y>;tag=as26b82ac7 Call-ID: 698200851849894c5874186163abe668@192.168.x.y:5090 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, OPTIONS, UPDATE, PRACK, MESSAGE, SUBSCRIBE Content-Type: application/sdp Supported: replaces User-Agent: Bria iOS release 3.9.7 stamp 38887.38893 Content-Length: 230 v=0 o=- 1085831961 3 IN IP4 100.81.231.195 s=Cpc session c=IN IP4 100.81.231.195 t=0 0 m=audio 65454 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Could the IP of the media part (100.81.231.195 in the CGNAT 100.64.0.0/10) be the cause of the loop? If that’s the case, how can we prevent the loop? 2) in the second case the CPE is an Avaya The Avaya send us an INVITE. (Also in this case in the SDP there is a private IP) INVITE sip:04818****@178.250.x.y SIP/2.0 From: <sip:0432415453@185.55.x.y>;tag=4500808-c0a8077d-13c4-50022-313081f-200d750c-313081f To: <sip:04818****@178.250.x.y> Call-ID: 44e8f20-c0a8077d-13c4-50022-313081f-31e99586-313081f CSeq: 1 INVITE Via: SIP/2.0/UDP 185.55.x.y:5060;rport;branch=z9hG4bK-313081f-257bc48-1472e4a5 Privacy: none Max-Forwards: 70 User-Agent: OfficeServ 7200 Contact: <sip:0432415453@185.55.z.y:5060;transport=udp> Allow: REGISTER,INVITE,ACK,BYE,REFER,NOTIFY,CANCEL,INFO,OPTIONS,PRACK,SUBSCRIBE,UPDATE Supported: 100rel Content-Type: application/sdp Content-Length: 289 v=0 o=SAMSUNG_SIP_GATEWAY 39304265 0 IN IP4 185.55.x.y s=SIP_CALL c=IN IP4 192.168.7.126 t=0 0 m=audio 30012 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv We reply with a “183 Session Progress” and the Avaya send US an “INFO” packet (dtmf) we replay with a “200 OK”; and then, after another “INFO” packet we replay with a “200 OK” with SDP. After that the Avaya replies with an ACK 2017/08/02 10:17:32.101121 185.55.x.y:5060 -> 178.250.x.y:5060 ACK sip:04818****@178.250.x.y SIP/2.0 From: <sip:0432415453@185.55.x.y>;tag=4500808-c0a8077d-13c4-50022-313081f-200d750c-313081f To: <sip:04818****@178.250.x.y>;tag=as783106e4 Call-ID: 44e8f20-c0a8077d-13c4-50022-313081f-31e99586-313081f CSeq: 1 ACK Via: SIP/2.0/UDP 185.55.x.y:5060;rport;branch=z9hG4bK-3130831-257ffea-84906e4 Max-Forwards: 70 User-Agent: OfficeServ 7200 Route: <sip:178.250.x.y;lr=on;r2=on;ftag=4500808-c0a8077d-13c4-50022-313081f-200d750c-313081f;did=adc.2f61> Route: <sip:192.168.x.y;lr=on;r2=on;ftag=4500808-c0a8077d-13c4-50022-313081f-200d750c-313081f;did=adc.2f61> Contact: <sip:043241****@185.55.x.y:5060;transport=udp> Content-Length: 0 And after that the chaos… Has anyone had a problem like this? Can somebody help us in order to prevent this loop? In the attached images you can see: [0] a call, our machines IPs, an ACK from public interface of our SBC to its private one; and than a lot of ACKs from the keepalived IP to the private IP of our SBC. [1] one of these strange ACKs, they are all different in content, but similar in structure: multiple Record-Route / Via, truncating too long message; [2] a "never ending" BYEs. After modifying the configuration adding some "listen=IP_ADDR:PORT" (and of course excluding the keepalive IP) we had no loops but is this enough? Why Kamailio generated the loops? Is there something else we should review in our config? Any advice would help a lot. Thanks. [0] https://goo.gl/rWh5db [1] https://goo.gl/MNShq9 [2] https://goo.gl/6mxepG

1 0

UAC reg_random_delay issue
by Ivo Vastert 16 Aug '17

16 Aug '17

Hi, I'm using the UAC module in Kamailio to setup about 500 outbound registration towards a single host. The issue I'm currently experiencing is that when we start Kamailio all this 500 registrations are initiated at once towards the host which can't handle all the requests at the same time. Therefor I tried to spread the registrations using the reg_random_delay function: modparam("uac", "reg_random_delay", 60) I would expect that this would spread the registrations over about a minute but in reality the registrations are spread as follow: Starttime / Registrations 21:59:14 151 21:59:15 166 21:59:16 127 21:59:18 60 21:59:22 35 So Kamailio is started at 21:59:14 and initiates 151 registrations, 1 second after start another 166 2 seconds after start another 127 4 seconds after start another 60 8 seconds after start the last 35 In reality it looks like some increment is applied instead of calculating a random starting offset for each outbound registration? Is there any solutions to this issue available? Best regards, Ivo Vastert

2 6

rtpengine : question about RTP/RTCP multiplexing (RFC 5761) and demultiplexing
by Abdoul Osséni 16 Aug '17

16 Aug '17

Hello All, I read rtpengine can multiplex and demultiplex RTP/RTCP. Cf.https://github.com/sipwise/rtpengine Example: UAC -> Kamailio + rtpengine -> Asterisk UAC and Asterisk don't support RFC 5761. Only rtpengine does. Does this mean that uac must be compliance with RFC 5761 if I want multiplex and demultiplex RTP/RTCP between UAC and rtpengine? Thanks. Abdoul.

2 2

log file
by Uzma Anjum 16 Aug '17

16 Aug '17

the log statements written in acc.c accounting module source code are not entered into syslog file.what can be wrong?

1 0

functions in accounting module
by Uzma Anjum 16 Aug '17

16 Aug '17

when the call answered,which methods are executed from acc.c source code and which method is inserted into database while call answer and call terminate.

1 0

Kamailio not listening on TLS port (5061)
by Deepak Jaiswal 16 Aug '17

16 Aug '17

Hi, I am new to kamailio and unable to debug the issue. Kamailio is not listening on TLS port. Could you please suggest some workaround? [root@uk-vap-002 ~]# ss -tnlp | grep kamailio LISTEN 0 1024 127.0.0.1:5060 *:* users:(("kamailio",29400,18)) LISTEN 0 1024 xxx.xxx.xxx.134:5060 *:* users:(("kamailio",29400,17)) LISTEN 0 1024 xxx.xxx.xxx.135:5060 *:* users:(("kamailio",29400,16)) LISTEN 0 1024 xxx.xxx.xxx:5060 *:* users:(("kamailio",29400,15)) Kamailio.cfg config done: listen=tls:xxx.xxx.xxx.133:5061 listen=tls:xxx.xxx.xxx.135:5061 listen=tls:xxx.xxx.xxx.134:5061 listen=tls:127.0.0.1:5061 2. #!ifdef WITH_TLS enable_tls=yes 3. #!ifdef WITH_TLS loadmodule "tls.so" #!endif 4. #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif Regards Deepak Jaiswal

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