- sr-users - kamailio.org

allow inbound calls from any ip
by Logeshwaran G 21 Aug '17

21 Aug '17

Hi, I need to allow the inbound calls from any ip i kamailio. Input please Thanks & Kind Regards, Logeshwaran G

2 1

Take call counts from kamailio
by Logeshwaran G 21 Aug '17

21 Aug '17

Hi, How to take call counts from the kamailio? Thanks & Kind Regards, Logeshwaran G

2 12

sdp extra \r\n in multipart body
by Diego Nadares 21 Aug '17

21 Aug '17

Hi guys, I have the following multipart body Content-Type: multipart/mixed;boundary=uniqueBoundary Remote-Party-ID: <tel:1112222;phone-context=+54> Content-Length: 364 --uniqueBoundary Content-Type: application/sdp v=0 o=user1 53655765 2353687637 IN IP4 111.111.111.111 s=- c=IN IP4 111.111.111.111 t=0 0 m=audio 6000 RTP/AVP 8 *a=rtpmap:8 PCMA/8000* *--uniqueBoundary--* Content-Type: application/gtd Content-Disposition: signal;handling=optional IAM, CPC,20 GCI,885sdfasdf1123 --uniqueBoundary-- As you can see *--uniqueBoundary *before *a=rtpmap:8 PCMA/8000* has not an extra '\r\n'. Is this correct? Or should have it? The thing is that rtpengine is receiving sdp with no final '\r'n' from kamailio: Dump for 'offer' from 127.0.0.1:37388: { "sdp": "v=0#015#012o=user1 53655765 2353687637 IN IP4 111.111.111.111#015#012s=-#015#012c=IN IP4 111.111.111.111#015#012t=0 0#015#012m=audio 6000 RTP/AVP 8#015#012*a=rtpmap:8 PCMA/8000"*, "ICE": "remove", "direction": [ "pub", "priv" ], "replace": [ "origin", "session-connection" ], "transport-protocol": "RTP/AVP", "call-id": "1-17755(a)111.111.111.111", "received-from": [ "IP4", "111.111.111.111" ], "from-tag": "1", "command": "offer" } And the resulting sdp from rtpengine is this: --uniqueBoundary Content-Type: application/sdp v=0 o=user1 53655765 2353687637 IN IP4 172.16.213.16 s=- c=IN IP4 172.16.213.16 t=0 0 m=audio 35066 RTP/AVP 8 *a=rtpmap:8 PCMA/8000a=sendrecv* a=rtcp:35067 The problem is related with that '\r\n' but Is this a malformed message or a possible bug? Thanks in advance! Diego

2 1

Documenation of functions exported for KEMI
by Sebastian Damm 19 Aug '17

19 Aug '17

Hi, I was wondering whether there is some kind of documentation of all functions currently exported for KEMI by Kamailio modules. I know I can find them somehow via grep, but I could not find a page in the Kamailio wiki listing all of them. Is there any? Best Regards, Sebastian

2 1

Converting to kemi
by James Cloos 19 Aug '17

19 Aug '17

[I see I posted this to -dev instead of -users; fixing that -JimC] I'm working on converting some configs to kemi lua and have a couple of questions: First: Does KSR.pv.get() support multiple pseudos in a single call? The examples such as misc/examples/kemi/kamailio-basic-kemi-lua.lua make several calls, but also do things like KSR.pv.get("$T_branch_idx]") to avoid having the append "]" to the string. That makes me wonder why that file has code like this: KSR.dbg("new branch [".. KSR.pv.get("$T_branch_idx]") .. " to ".. KSR.pv.get("$ru") .. "\n"); instead of like: KSR.pv.get("[$T_branch_idx] to $ru\n") Second: Is it still the case that, eg, event_route[xhttp:request] cannot be in the kemi lua code? There was another. Perhaps I'll recall what it was... :( -JimC -- James Cloos <cloos(a)jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6

2 1

Kamailio_428 memory usage
by Evgeniy G. 18 Aug '17

18 Aug '17

Good afternoon! Can you please tell us how the Kamailio works with shared memory. The fact is that there was an accident in which Kamailio reported an error - little common memory. This happened a day later as I added the dialplan module. After that, I began to monitor the values: used_size real_used_size max_used_size free And I saw that I can not properly assess. I note that the server does not process calls, it is idle. After the server is started, the indicators are as follows: used_size: 63 real_used_size: 146 max_used_size: 146 free 106 After rebooting the module (for example, drouting, in which the data has not changed): used_size: 63 real_used_size: 193 max_used_size: 231 free 62 I found, a description, the reasons for the increase in memory after rebooting the modules, I quote "As suggested by Daniel, for carrierroute reload, we need the same amount of memory as it used to be. Block when reloading the table. " But why: 1. after increasing the memory the server frees only real_used_size 2. That that is specified as real_used_size on my observations can not be released in current of several days (probably weeks). 3.If in this state again to reboot the module (the same one that led to an increase in memory), the memory does not increase. Question: during what time should kamailio release the memory specified as real_used_size? Also what is considered as real_used_size? -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-428-memory-usage-tp161327.… Sent from the Users mailing list archive at Nabble.com.

1 0

syslog CDR time format
by Yu Boot 18 Aug '17

18 Aug '17

Hello. Can't get 'date-time' format timestamps in syslog CDRs, while SQL CDRs are OK. According to acc module documentation: modparam("acc", "cdr_enable", 1) modparam("acc", "time_format", "%Y-%m-%d %H:%M:%S") modparam("acc", "time_mode", 3) modparam("acc", "cdr_facility", "LOG_DAEMON") syslog: NOTICE: acc [acc_cdr.c:352]: log_write_cdr(): start_time=1503068984.627; end_time=1503068991.250; duration=6.623; src_user=100; src_domain=a.b.c.d; src_ip=e.f.g.h; dst_ouser=200; dst_user=200; dst_domain=k.l.m.n

1 0

On the fly Change
by Logeshwaran G 18 Aug '17

18 Aug '17

Hi, Is there any possibilities of "on the fly" option in kamailio.cfg. For Example: I need to change my carrier IP address in the lcr_gw table, kamailio.cfg should takes it on the fly. Without restarting the kamailio. Thanks & Kind Regards, Logeshwaran G

4 18

Acc cdr duration
by yu＠yu-boot.ru 18 Aug '17

18 Aug '17

Hello. How to remove milliseconds from duration field in CDR? -- Отправлено из myMail для Android

1 0

dispatch_rpc
by Logeshwaran G 18 Aug '17

18 Aug '17

Hi, Is there any other module need to be added for xmlrpc for using dispatch_rpc(). Because I am getting an below error: 0(22485) ERROR: <core> [core/cfg.y:3274]: yyparse(): cfg. parser: failed to find command dispatch_rpc (params 0) 0(22485) CRITICAL: <core> [core/cfg.y:3414]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio_final.cfg, line 640, column 17: unknown command, missing loadmodule? I have loaded xmlrpc as well as xhttp modules. Do I need to enable some other modules as well?? Please Guide me. Thanks & Kind Regards, Logeshwaran G

1 0

SIP_DOMAIN in /etc/kamailio/kamctlrc
by Trent 17 Aug '17

17 Aug '17

Hello. I have not been able to find anything in the wiki, or documentation which covers this. The only item I have found is related to a module. >From the information, I have gleaned on this, it is clear an IP address, or a domain name is to be set here. What is not clear, is if the server is on multiple IP address whether they are on a WAN, LAN, or both, if multiple IP addresses are need, or can be added. I have a situation where I have a public WAN address, and a LAN address. Thanks a bunch!

1 0

kamailio restart:errors
by Uzma Anjum 17 Aug '17

17 Aug '17

Hi,I am new to kamailio. when i restart kamailio,I found errors on log.please help through this. 6(5633) ERROR: mymodule [my_mod.c:373]: outbound_call_process(): [outbound_call_process] URI : sip:11111111112@107.170.69.191:1008 received ip 183.82.115.246 in network ips:107.170.69.191:6090|c1|c2 6(5633) ERROR: mymodule [my_mod.c:379]: outbound_call_process(): [outbound_call_process] Request URI : 11111111112 6(5633) ERROR: mymodule [my_mod.c:395]: outbound_call_process(): outbound_call_process, obcalls privilges :select u.ob_call_privileges, a.phone_number, replace(a.name,'.','-') from account_info a, users u, subscriber s where a.id=u.account_id and u.account_id=s.account_id and u.extension=s.extension and s.username='dyan-dayan' 6(5633) ERROR: mymodule [my_mod.c:425]: outbound_call_process(): outbound_call_process query from user:dyan-dayan, phone_type:1 acc_name:dayan 6(5633) ERROR: mymodule [my_mod.c:501]: outbound_call_process(): [outbound_call_process]NEW URI : sip:ivr-1-dayan@107.170.69.191:6090 6(5633) ERROR: <core> [udp_server.c:448]: udp_rcv_loop(): recvfrom:Received encrypted packet len:4, v▒..

1 0

ClueCon Slides: Industrial Grade FreeSWITCH, Scaling, Balancing, High Availability, SIP, WebRTC
by Giovanni Maruzzelli 17 Aug '17

17 Aug '17

*Industrial Grade FreeSWITCH Scaling, Balancing and High Availability for SIP and WebRTC* Scaling your FreeSWITCH platform to serve a growing user base is a critical challenge. We'll go through the best techniques, practices, and implementations for Voice and Video Calls, Conferencing, WebRTC, SIP, Chatting, Presence and Instant Messaging Slides can be downloaded from http://opentelecom.it/cluecon2017.pdf Enjoy ClueCon! -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

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Re: [SR-Users] DMQ Not Discovering all Nodes
by Charles Chance 16 Aug '17

16 Aug '17

Hello, Can you share the rest of your config (at least, any dmq modparams and the main route block)? Cheers, Charles On 16 Aug 2017 22:48, "John Petrini" <jpetrini(a)coredial.com> wrote: 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 _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Faraday Wharf, Innovation Birmingham Campus, Holt Street, Birmingham Science Park, Birmingham B7 4BB.

2 1

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

1 0

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

2 1

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?

1 0

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

2 1

Siremis Install: Unable to write in the session save path (CentOS 6)
by Trent 16 Aug '17

16 Aug '17

Hello. Doing a new install, and I am in the stage of accessing the web interface for Sermis which is located at /var/www/html/siremis Full error: Unable to write in the session save path [/var/www/html/siremis/session]Unable to write in the session save path [/var/www/html/siremis/session] I would think this is a single access problem, but I have checked ownership in httpd.conf, and the files, and directories. All have the apache ownership assigned to them. I have even given the "session" folder access of 777 with still no change in the error. -rw-r--r--. 1 apache apache 10176 May 18 05:18 ChangeLog drwxr-xr-x. 4 apache apache 4096 May 18 05:18 misc drwxr-xr-x. 7 apache apache 4096 May 18 05:18 openbiz -rw-r--r--. 1 apache apache 1019 May 18 05:18 README.md drwxr-xr-x. 14 apache apache 4096 May 18 05:18 siremis In addition, there is nothing about this appearing in the httpd error_log, nor access_log for an access problem. Any suggestions? Thanks!

3 3

UAC random_delay not that random
by Ivo Vastert 15 Aug '17

15 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 numbers are sprad as follow: 2017/08/09 21:59:14: 301 2017/08/09 21:59:15: 332 2017/08/09 21:59:16: 253 2017/08/09 21:59:18: 120 2017/08/09 21:59:22: 69

1 0

KSR.pv.get("$(hdr(P-Asserted-Identity){uri.user})") returns unable to get pv value
by Muhammad Zaka 15 Aug '17

15 Aug '17

Hi We are experiencing issue with python script KSR.pv.get using branch 5.0.2 as it fails "unable to get pv value" Example Code: if KSR.pv.get("$ai"): fromid = KSR.pv.get("$(hdr(P-Asserted-Identity){uri.user})") Error: Aug 14 18:38:23 sbc99-kam-dev /usr/sbin/kamailio[30459]: PYT {INVITE}: ERROR: app_python [apy_kemi.c:637]: sr_apy_kemi_f_pv_get(): unable to get pv value for [$(hdr(P-Asserted-Identity){uri.user})] Is anybody have any idea how to workaround? Many Thanks Regards Muhammad Zaka

2 4

Kamailio 4.3.4: receive_msg(): no via found in reply
by Grant Bagdasarian 15 Aug '17

15 Aug '17

Hello, We're testing kamailio 4.3.4 with two kamailio's communicating with each other. One of them is a gateway with rtpengine at the carrier side, the other is a basic sip router. When calling a destination and cancelling the INVITE a 487 is received from the carrier and directly ACK'd by the first kamailio and in turn is forwarded to the second. But the 487 response received by the 2nd kamailio doesn't have a Via header which results in kamailio printing the error: receive_msg(): no via found in reply. Any ideas why the forwarded 487 doesn't contain a Via header and how to fix this? Note: IP's are changed. Carrier: 10.0.0.1 Kamailio 1: 10.14.0.1 Kamailio 2: 10.14.0.2 U 10.0.0.1:5060 -> 10.14.0.1:5060 SIP/2.0 487 Request Cancelled. Via: SIP/2.0/UDP 10.14.0.1;branch=z9hG4bK8e28.642c68037d059634308d4be8fe3f0cdc.0. From: "+31612345678" <sip:+31612345678@10.14.0.3:5060>;tag=109ac722. To: <sip:003112345678@10.14.0.3:5060>;tag=sbcsipuas_1_C22196_20160314080307791_b59sb10. Call-ID: 78100MjY3OGExZmMyODU2MzAyMzM5ODA3MmNhMDdlMzg0MzU. Contact: <sip:10.0.0.1:5060>. CSeq: 2 INVITE. Server: sbc_5. Content-Length: 0. . # U 10.14.0.1:5060 -> 10.0.0.1:5060 ACK sip:003112345678@10.0.0.1:5060 SIP/2.0. Via: SIP/2.0/UDP 10.14.0.1;branch=z9hG4bK8e28.642c68037d059634308d4be8fe3f0cdc.0. Max-Forwards: 67. To: <sip:003112345678@10.14.0.3:5060>;tag=sbcsipuas_1_C22196_20160314080307791_b59sb10. From: "+31612345678"<sip:+31612345678@10.14.0.3:5060>;tag=109ac722. Call-ID: 78100MjY3OGExZmMyODU2MzAyMzM5ODA3MmNhMDdlMzg0MzU. CSeq: 2 ACK. Content-Length: 0. . # U 10.14.0.1:5060 -> 10.14.0.2:5060 SIP/2.0 487 Request Cancelled. From: "+31612345678" <sip:+31612345678@10.14.0.3:5060>;tag=109ac722. To: <sip:003112345678@10.14.0.3:5060>;tag=sbcsipuas_1_C22196_20160314080307791_b59sb10. Call-ID: 78100MjY3OGExZmMyODU2MzAyMzM5ODA3MmNhMDdlMzg0MzU. Contact: <sip:10.0.0.1:5060>. CSeq: 2 INVITE. Server: sbc_5. Content-Length: 0.

4 16

Reinvite, rewrite reply SDP?
by Yu Boot 15 Aug '17

15 Aug '17

Finally I find a root of a problem with my T.38 reinvites. A->(NAT)->Kami->PSTN When NATted caller does reinvite, and remote side answers with 200 OK+SDP, Kami (why?) doesn't rewrite SDP for its own IP (so NATted caller can receive RTP) and leaves remote's IP as media address, so of course there is no RTP/UDPTL. All other NAT related stuff works fine. Even if remote side sends me T.38 reinvite first, everything works properly. Config file is default from Kami sources. Any ideas?

2 2

RTPEngine recording
by Alex Balashov 14 Aug '17

14 Aug '17

I have gathered that RTPEngine has recently, or perhaps some time ago, evolved a recording feature set: https://kamailio.org/docs/modules/5.0.x/modules/rtpengine.html#rtpengine.f.… Does anyone have any experience using it with Kamailio? How does it work? Any gotchas or pitfalls? It looks like recording is done via a separate daemon specifically for that purpose. Does it emit directly in a playable format, or more or less just dump the raw, RTP-encapsulated frames? -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 8

DNS Geo Location Kamailio servers, registration and routing
by Sadeesh Radhakrishnan 14 Aug '17

14 Aug '17

We have kamailio servers that do DNS-Geo-Location on different locations, using same domain name. So the users from a country will be preferred to his nearby server by DNS routing. Similarly users from different countries will be registering to their nearby kamailio sip servers. All the servers share a common user location table on database. Now the users are getting registered fine, with no issue. We want the users at one location should be able to call user@other-location using the user location contact information. When we tried, the INVITE is sent on common domain name and that reverse resolves dns to same server and gets a 404 not found. I need some suggestion / ideas in order to make it happen. Thanks

2 1

set digest mime
by wsotest.512 14 Aug '17

14 Aug '17

2 1

Inconsistency in sems v1.6 installation: sems.conf is missing
by leslie mekiyess 14 Aug '17

14 Aug '17

For some reason, the installation creates the etc directory under the /usr/local/etc/sems/ : make[2]: Entering directory `/usr/src/sems-1.6.0/apps/conf_auth' mkdir -p **/usr/local/etc/sems/**etc**/** installing conf_auth.conf make[2]: Leaving directory `/usr/src/sems-1.6.0/apps/conf_auth' Then at the make install ends the following is printed: *** install complete. Run SEMS with *** *** /usr/local/sbin/sems -f /**usr/local/etc/sems/sems.conf** At bottom libe, while following the spain document instructions, I figured out that the sems.conf wasn;t created at the /usr/local/etc/sems/ path. I can create a new one or even copy this one found : /usr/src/sems-1.6.0/apps/mobile_push/load_test/sems_cfg/core/etc/sems.conf but I am afraid of further files/directory inconsistencies. Any assistance is very appreciated.

2 1

acc billsec
by Aidar Kamalov 14 Aug '17

14 Aug '17

Hello, is there any way to log in cdr duration and billsec? -- Aydar A. Kamalov

3 3

UAC RPC Commands problem
by Volkan Oransoy 14 Aug '17

14 Aug '17

Hi, I use Kamailio 5.0. I try to run RPC commands for UAC module but some of them doesn’t work like documented. kamcmd uac.reg_dump kamcmd uac.reg_reload These commands work as expected. But commands with extra arguments give error. kamcmd> uac.reg_refresh 908508850000 error: 400 - Invalid Parameters kamcmd> uac.reg_info l_uuid 908508850000 error: 404 - Record not found Do I miss something? Thanks, /Volkan

2 3

Carrierroute, fallback, reinvites
by yu＠yu-boot.ru 12 Aug '17

12 Aug '17

Hello. Trying to implement fallback route for carrierroute and got sick to make reinvites working correctly. Until fallback route triggers, everything is OK. Anyone has working config for that purposes (NAT traversal, reinvites, carrierroute ), please? And abstract question - is it generally a good idea to implement a simple softswitch (not PBX like Asterisk) like Yate or Freeswitch with Kami? Now I just want to learn Kami and learn SIP deeper in general.

2 1

Route calls from Kamailio to Asterisk behind NAT
by vs＠cell.tel 12 Aug '17

12 Aug '17

3 5

Re: [SR-Users] [OpenSIPS-Users] ClueCon Slides: Industrial Grade FreeSWITCH, Scaling, Balancing, High Availability, SIP, WebRTC
by Abdul Basit 11 Aug '17

11 Aug '17

nice presentation!!! -- regards, abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 On 10 August 2017 at 23:12, Giovanni Maruzzelli <gmaruzz(a)gmail.com> wrote: > *Industrial Grade FreeSWITCH Scaling, Balancing and High Availability for > SIP and WebRTC* > > Scaling your FreeSWITCH platform to serve a growing user base is a > critical challenge. We'll go through the best techniques, practices, and > implementations for Voice and Video Calls, Conferencing, WebRTC, SIP, > Chatting, Presence and Instant Messaging > Slides can also be downloaded from http://opentelecom.it/cluecon2017.pdf > > Enjoy ClueCon! > > -giovanni > > -- > > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > _______________________________________________ > Users mailing list > Users(a)lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >

1 0

Dispatcher call id hash for server side invite
by Ryan Wagoner 11 Aug '17

11 Aug '17

For a call invite from a phone to an load balanced asterisk server farm I can use ds_select_dst with hash over callid for the algorithm. What I don't understand is what happens for a server side invite. Say user A calls user B. The server will send and invite to user B's device. User B's device will reply and ds_select_dst won't have the call id hash so it will choose a random server, which might not be the server sending the invite. How do I mark these server side invites so the call hash is known by kamailio? Or am I thinking about this the wrong way? Thanks, Ryan

3 6

Issue in Dialplan Module
by Logeshwaran G 10 Aug '17

10 Aug '17

Hi, I am facing an Problem in Dialplan Module. Scenario: ======== I am Dialling 00091xxxxxxxxxx number, I need to remove only the prefix 000, then the call placement will be 91xxxxxxxxxx. My dialplan rule: ============== MariaDB [kamailio]> select * from dialplan; +----+------+----+----------+-----------+-----------+-----------+----------+-------+ | id | dpid | pr | match_op | match_exp | match_len | subst_exp | repl_exp | attrs | +----+------+----+----------+-----------+-----------+-----------+----------+-------+ | 1 | 1 | 1 | 1 | ^000$ | 0 | ^000$ | | | +----+------+----+----------+-----------+-----------+-----------+----------+-------+ But the prefix 000 is not replaced. Please Guide me to Resolve this issue. Do I need to change somet rules?? Additional info: ============= But, With additional to the above dialplan data rule I used the below rule. +----+------+----+----------+-----------+-----------+-----------+----------+-------+ | id | dpid | pr | match_op | match_exp | match_len | subst_exp | repl_exp | attrs | +----+------+----+----------+-----------+-----------+-----------+----------+-------+ | 1 | 1 | 1 | 1 | ^000$ | 0 | ^000$ | | | 111 +----+------+----+----------+-----------+-----------+-----------+----------+-------+ The above rule replacing only the 000, If we Dial 000xxxx its omitting, While Dialing 000 its replacing that with 111. Thanks & Regards, Logeshwaran

4 23

BUG: invalid branch
by Yasin CANER 10 Aug '17

10 Aug '17

2 1

Kamailio: how to route RTP media directly to endpoint
by wsotest.512 10 Aug '17

10 Aug '17

Hi all, We have usual config Kamailio + Asterisk where Kamailio play as sip and rtp proxy. Kamailio have public IP, asterisk - no. All calls between clients now going like that: UserA ---sip--> Kamailio --> Asterisk --> UserB \-rtp--> Kamailio (rtpproxy) --> Asterisk --> UserB All clients of course from Internet and behind Nat. Main problem is amount of traffic going through Kamailio and Asterisk. We need to pay for every additional GB behind limit in tariff plan to hosting provider. So we decided to try route all rtp traffic between users directly. UserA ---sip--> Kamailio --> Asterisk --> UserB \-rtp--> --> --> UserB Is it possible at all? Maybe someone already did it . -- BR, Alex

3 2

IMPORTANT: acc module - SQL define removed and diameter code relocated
by Daniel-Constantin Mierla 09 Aug '17

09 Aug '17

Hello, I just pushed a bit of clean up to acc module, therefore be aware of following changes: 1) the define conditions on SQL_ACC were removed -- this was enabled for more than 10 years and only made the code look complex and hard to follow up its logic. 2) the code related to DIAMETER accounting was relocated to acc_diameter (new) module. It was a consistent size of code that was not enabled for sooo... long. It is now a dedicated module, similar to acc_radius. The diameter accounting code, even a new module now, is in the same stage, compiling but not tested, in pair with auth_diameter module, it may work, but very likely not. In summary, what's important for those using the acc module -- it offers the same functionality as it was enabled by default in the past 10 years or more, only the unused code was relocated. It offers the functionality of writing accounting records to syslog and sql databases. The acc module is now slimmer, only with the code that it needs, therefore easier to maintain and enhance for the future. Hopefully, there was no side effect with this update -- anyhow, if you find any issue, just open a bug report on github project. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

Kamailio: how to route RTP media directly to endpoint
by Aleksey Hodin 09 Aug '17

09 Aug '17

Hi all, We have usual config Kamailio + Asterisk where Kamailio play as sip and rtp proxy. Kamailio have public IP, asterisk - no. All calls between clients now going like that: UserA ---sip--> Kamailio --> Asterisk --> UserB \-rtp--> Kamailio (rtpproxy) --> Asterisk --> UserB All clients of course from Internet and behind Nat. Main problem is amount of traffic going through Kamailio and Asterisk. We need to pay for every additional GB behind limit in tariff plan to hosting provider. So we decided to try route all rtp traffic between users directly. UserA ---sip--> Kamailio --> Asterisk --> UserB \-rtp--> --> --> UserB Is it possible at all? Maybe someone already did it ... -- BR, Alex

1 0

Registrar limit
by Yu Boot 09 Aug '17

09 Aug '17

How to limit one user login with one AOR (IP) and always use IP from last registration attempt?

2 1

RFC: Moving misc/obsolete/ to its own git repository
by Daniel-Constantin Mierla 09 Aug '17

09 Aug '17

Hello, trying to make the git repository a bit slimmer and avoid unuseful indexing of unused modules, I am considering to move misc/obsolete to its own github repository (planned to be named kamailio-obsolete). The initial idea was to keep there modules that were duplicated or no longer maintained, just in case someone will come later and wants to revive it. However, iirc, it is more than 4 years and nobody touched them, so very likely it requires a bit of work to revive any of them, so a fresh re-import will be the same from an external repository. If anyone still wants to keep them in the main kamailio repository, respond on the mailing list and present the benefits. If not, I will do the operation in the next several days. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 1

"kamctl restart" fails
by Yu Boot 09 Aug '17

09 Aug '17

# kamctl restart INFO: Stopping Kamailio : INFO: stopped INFO: Starting Kamailio : 2292 pts/0 S+ 0:00 /bin/egrep kamailio -rw-r--r--. 1 kamailio kamailio 5 Aug 8 11:15 /var/run/kamailio.pid INFO: PID file exists (/var/run/kamailio.pid)! Kamailio already running? "service kamailio restart" works fine. What's wrong? Centos 6.9 x64, Kamailio 4.3 installed from official repo.

3 3

how to remove a line in sdp
by Sunil More 09 Aug '17

09 Aug '17

Hello All, I have the following SDP coming in from a carrier and would like to remove the following line. Cant figure how to go about it. line to remove : *a=cdsc: 1 image udptl t38* sdp is as below Content-Type: application/sdp. Content-Length: 293. . v=0. o=hiQ9200 5859020170708143552 1175715965 IN IP4 X.Y.Z.A. s=Phone Call via hiQ9200 SIPCA. c=IN IP4 X.Y.Z.A. t=0 0. m=audio 16648 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sqn: 0. *a=cdsc: 1 image udptl t38.* a=sendrecv. a=ptime:20. This reply comes from a upstream carrier in session progress and 200ok causing one-way audio for me. Thanking You, Sunil More Ph : 9503338275

2 1

Billing plan
by Logeshwaran G 09 Aug '17

09 Aug '17

Hi, Is there any billing application or modules to use with kamailio , for example if i want to check the credits of the user on every passing minute , and when he's out of credits i stop the call. Thanks & Kind Regards, Logeshwaran G

2 1

sqlops error
by Logeshwaran G 09 Aug '17

09 Aug '17

modparam("sqlops","sqlcon",DBURL) Whether the above modparam implementation is wrong ? I am not sure Its saying some error as 0(2519) ERROR: sqlops [sql_api.c:669]: sql_parse_param(): invalid sqlops parameter

3 5

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