- sr-users - kamailio.org

Automating installation
by John Tuxies 14 Dec '18

14 Dec '18

Hi all. I am new to this and i would like some help please. I am looking to deploy Kamailio and Asterisk in my systems and i would like to automate the process. I have found a bash script but it it outdated from here: https://github.com/gurumelo/asteriskkamailio i did some changes and modifications to adapt it to newer version of software, but it fails to start Kamailio and continue to Asterisk. I would appreciate if someone could help me with this script and make it work. The systems i am planning to execute it is in Debian 9 and Ubuntu 18.04 server, both 64bit. Sincerely yous, John

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FreeBSD Kamailio pkg
by Sergiu Pojoga 14 Dec '18

14 Dec '18

Did anyone notice that when installing Kamailio from FreeBSD package, by default, it accepts registration with whatever account credentials? Not sure if it's like that with other OS packages (usually I install it from source), but this behavior kind of doesn't go with the popular saying that Kamailio by default "does nothing", lol. Regards, --Sergiu

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Kamailio ERROR: _reply_light(): can't generate xxx reply when a final yyy was sent out
by Olli Attila 13 Dec '18

13 Dec '18

Hello, I'm getting these error messages on kamailio log from time to time: Dec 11 19:18:30 /usr/sbin/kamailio[23913]: ERROR: tm [t_reply.c:482]: _reply_light(): can't generate 487 reply when a final 200 was sent out Dec 8 04:44:56 /usr/sbin/kamailio[14960]: ERROR: tm [t_reply.c:482]: _reply_light(): can't generate 487 reply when a final 486 was sent out Dec 5 07:46:04 /usr/sbin/kamailio[14961]: ERROR: tm [t_reply.c:482]: _reply_light(): can't generate 487 reply when a final 480 was sent out Kamailio is configured to act as a proxy between SBC and a softswitch. According to the signaling we can see that these errors occure when the SBC sends CANCEL message towards Kamailio when at the same time final reply is received from the Softswitch side. The call seems to be teared down nicely and finally Kamailio replies to SBC by sending "200 ok, no more pending brances". I guess Kamailio treats this situation as an error because the sip processing engine has sent the final reply out and at the same time SBC sends the CANCEL to Kamailio. Is there any way we could handle this situation better on Kamailio end? The three example ERROR messages are from cases where B subscriber has answered the call, the B has been busy and the last one is from a case where B subscriber has been temporarily unavailable. Cheers, Olli

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is_e164 logic to detect valid number
by Patrick Wakano 11 Dec '18

11 Dec '18

Hello list, Hope you are all doing fine! I am using the is_e164() function to validate the number we receive, and I come to see that the number +555 was accepted.... After some googling it looks like(it is not very clear though) that 7 digits are the minimum we could have for e164 numbers but after checking the source code, I saw it accepts anything starting with + and having between 2 and 16 numbers. So is it really valid to have a number with just 2 digits? What is the case? Thank you, Kind regards, Patrick Wakano

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Performance tests of native vs. kemi scripting for SIP routing
by Daniel-Constantin Mierla 11 Dec '18

11 Dec '18

Hello, during the preparations of Kamailio v5.2.0 and the days after, I got the time to run some tests in order to see the differences of running similar SIP routing logic when using native configuration file scripting versus KEMI languages. So far I did it for Lua and Python (v2), hopefully I will find time to run also for the rest of supported scripting languages: Python (v3), JavaScript, Ruby, Squrrel and Lua with LuaJIT. The pleasant surprise was to see that this time, the results of using Python were similar to native scripting and Lua. When I did some basic tests during development of KEMI, using Python seemed slower. More details about how tests were performed, the config files/scripts, used tools and results were published in a wiki page: - https://www.kamailio.org/wiki/kemi/performance-tests/5.2.x A news article with further details is on kamailio website: - https://www.kamailio.org/w/2018/12/performance-tests-for-kemi-scripting-lan… Should anyhow of you have some time to run the tests in own environment and share the results (on sr-users mailing list), it would be appreciated and would help to validate the tests and discover if I forgot something in any of the configs. Note that the goal of the tests is not measuring the capacity of Kamailio in your environment, but see the differences between using different scripting languages in the same environment (so you can do the tests on raspberry pi, if you like). I would also be interested in knowing if you are using Kamailio with KEMI or you are planning to, if yes, which scripting language. If already doing it, is already in production? These are useful to plan where to allocate developer resources in the future... And to start, I am running Kamailio with Lua (4 instances) and Python (1) scripting in production deployments, with very good results so far. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com Kamailio Advanced Training -- www.asipto.com

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core at kamailio 5.1.6
by David Escartín 11 Dec '18

11 Dec '18

hello all we are seeing in a new kamailio instance some crashes. We are not sure the root cause, here you have the last core generated https://pastebin.com/xMRrug2X COuld you please take a look to see if there might be any issue with the last message that caused the core (to me seems normal) or there might be some issues related to other system libraries, etc? thanks a lot and regards david

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DTLS HA redis-pacemaker-corosync
by Жан Базаров 10 Dec '18

10 Dec '18

Hello! Are there any ways to implement using DTLS in redis HA scenario? When I reboot the active RTPEngine server the system correctly fails over to the other server and, after only a small pause (~1s), incoming audio continues to arrive. However, approximately 15 seconds after this I hear nothing but silence. Thanks.

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Kamailio World 2019 - Call for Presentations
by Daniel-Constantin Mierla 10 Dec '18

10 Dec '18

Hello, I would like to announce that Call for Presentations at Kamailio World 2019 is now open. You can submit your proposal or see more details at: - https://www.kamailioworld.com/k07/call-for-speakers/ The 7th edition of the event takes place again in Berlin, Germany, during May 6-8, 2019. Expect over 150 participants, developers and community members as well as representatives from other popular open source VoIP projects such as Asterisk or FreeSwitch. Looking forward to meeting many of you there! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com Kamailio Advanced Training -- www.asipto.com

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SIP SIMPLE TO XMPP AND VICE VERSA
by sagar malam 08 Dec '18

08 Dec '18

Hello, We have a system which using FreeSWITCH for calls and Ejabberd( XMPP) for chat.Now we want to support SIP SIMPLE as well.But we are not going to move old users to SIP SIMPLE but only new users will use SIP SIMPLE.This leads to requirement of having SIP-XMPP gateway for which are are considering Kamailio server. I have gone through documentation about Kamailio XMPP module, it is clear that Kamailio can convert SIP MESSAGE to XMPP so SIP to XMPP will work but what about XMPP to SIP MESSAGE ? Does that work as well ? Thanks in advance. -- Thanks, Sagar

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Dialplan Module limits
by Enrico Bandiera 07 Dec '18

07 Dec '18

Hi guys, does anyone know which are the limits of the diaplan module? After how many entries I will encounter serious performance issues? Thanks, Enrico.

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question about replace_body_all
by Wilkins, Steve 07 Dec '18

07 Dec '18

Hello, I am attempting to change the Message Body of an SIP message using replace_body_all; I want to replace all attributes that start with rtcp. Here is my code => replace_body_all("rtcp:{1}[0-9]{5,}", newstring), and I call msg_apply_changes() after this call. Note that I verified that the regex is working using search. The problem is that nothing gets changed. However, what I do get is a new string at the bottom of the Message Body with my newstring concatenated three times, which happens to be the number of times that rtcp occurs in the Message Body. Note, I am doing this so that I can add the IP address to the rtcp attribute because this is required by a Provider. Any ideas? Thank you, -Steve

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Dialog getting disconnected as soon as call is ended
by Subham Singhal 07 Dec '18

07 Dec '18

Hello, I am new to kamailio. I have been trying to integrate kamailio with asterisk. I am able to send INVITE and get RTP packets as well, but I am unable to receive BYE from asterisk. Call is getting terminated through following websocket error. "WebSocket connection to 'wss://172.16.16.169:5061/' failed: One or more reserved bits are on: reserved1 = 1, reserved2 = 0, reserved3 = 0." -- Thanks and Regards Subham Singhal

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Auth_xkeys module - freeing already freed pointer
by José Seabra 06 Dec '18

06 Dec '18

Hello there, I'm constantly getting the following message when i'm using module auth_xkeys. CRITICAL: <core> [core/mem/q_malloc.c:514]: qm_free(): BUG: freeing already freed pointer (0x7f6310987400), called from core: core/data_lump.c: free_lump(466), first free core: core/data_lump.c: free_lump(466) - ignoring Apparently kamailio only throw this message but i would like to know if its a normal protection from kamailio or it should be fixed? Thank you for the support Regards -- José Seabra

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Assistance with re.subst & newline
by Mark Hall 06 Dec '18

06 Dec '18

Hi, I am using rtpengine to proxy audio to/from media servers. The outbound calls are being rejected by a couple of my carriers (although accepted by others), and the closest that I have been able to find for the reason is the inclusion of "a=rtcp:xxxxx" in the SDP generated by rtpengine. To test further, I am trying to remove this line. Rtpengine itself doesn't have any options to remove it, so I am left with manipulating the generated SDP in Kamailio. A sample of generated SDP is: a=maxptime:150 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-16 a=sendrecv a=rtcp:30939 s=My Media Server I am using rtpengine_manage("RTP/AVP ICE=remove"); to generate the SDP to a variable ($avp(sdp)) Now, I am trying to just remove the whole a=rtcp line with re.subst $avp(sdp)=$(avp(sdp){re.subst,|a=rtcp:[0-9]*\R||s}); *** Tried / as delimiter also, but | looks cleaner *** This should match the line and any CR/LF characters - BUT in results, nothing is changed (various online Regex testers show this match to be correct - I tried \r\n also). If I do the same but without the \R, then the line is removed but not the CR/LF, leaving a gap in the SDP - which is invalid. $avp(sdp)=$(avp(sdp){re.subst,|a=rtcp:[0-9]*||s}); Gives: a=maxptime:150 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-16 a=sendrecv s=My Media Server Can anyone see where I am going wrong, or suggest an alternative? Kind Regards, Mark Hall

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New features: dialog - list dialogs via rpc using matching expressions
by Daniel-Constantin Mierla 06 Dec '18

06 Dec '18

Hello, being discussed and requested couple of times via lists, irc channel and discussions at events world wide, I just want to send a short note to announce that yesterday I pushed a new rpc command to dialog module that returns dialog records that match on conditions applied to request uri, from uri, to uri or callid fields. The filtering can be done with string comparison, regular expression or starts-with (prefix) matching. See more details at: - https://www.kamailio.org/docs/modules/devel/modules/dialog.html#dlg.r.list_… It should enable listing dialogs only for specific users, caller or callee, without dumping all dialogs. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

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set_advertised_address
by Kjeld Flarup 06 Dec '18

06 Dec '18

Hello I have a PBX behind NAT. Thus I advertise the public IP, and forwards the port to my PBX. listen=LOCALIP:5070 advertise EXTERNALIP:5070 Now clients can connect to the PBX from the Internet. And also inside the LAN, because I have enabled NAT loopback. However some customers sysadmins complains that NAT loopback is a security risk. I have not been able to find any exploits of this, but the sales and support people asks if it is possible to remove this NAT loopback requirement. I could look at $rd and if it is local, then I could advertise LOCALIP. I found set_advertised_address("LOCALIP"); set_advertised_address however only seems to modify the latest Via header, not the Record-route, and audio neither works. Could I do something to make this work, or is it a dead end? -- -------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

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Dispatcher clear dst_avp
by Grant Bagdasarian 05 Dec '18

05 Dec '18

Hello, I'm trying to use multiple destination sets which may include the same destination. For instance: SET ID: 1 sip:10.0.0.1:5060 sip:10.0.0.2:5060 SET ID: 2 sip:10.0.0.1:5060 Whenever a new SIP INVITE is sent to the Kamailio instance it will use SET ID 1. If then for some reason this call attempt fails, failure_route will trigger and SET ID 2 will be used. At first glance this might not make any sense, but there is some business logic behind this, but it's not relevant for this case. For some reason, the dst_avp variable still contains the destinations of SET ID 1 in failure_route. Is there a built-in way to reset this variable for every call to ds_select_domain(), such that only the destinations of passed SET ID will be loaded into the avp? I tried setting the avp to $null before calling ds_select_domain, but that had some undesirable effects. The reason I'm asking this is the destinations the call is being sent to are Freeswitch instances, which will return a 482 Request Merged when the same INVITE is sent to the same Freeswitch within 4 seconds (default T4 timer). I'm trying to match the previously selected destination in failure_route with the next destination and skip if they're the same, or stop processing when they are the same and no more destinations are available. I know I can mess with the T4 timer in Freeswitch and perhaps even sleep in failure route to deal with the 482 failure, but I rather fix this in a normal way. Regards, Grant Bagdasarian Senior Developer

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xlog issue
by Soltanici Ilie 04 Dec '18

04 Dec '18

Hello, I have a strange situation by using xlog module. I don't know for what reason I'm not receiving all logs generated by kamailio. This is the configuration which I'm using in kamailio.cfg: log_facility=LOG_LOCAL0 log_name="kamailio" log_prefix="{$rm ($mt) | Seq=$cs | Source IP=$si ($proto) | Call ID=$ci} " # ----- xlog ----- modparam("xlog", "buf_size", 8192) modparam("xlog", "long_format", 0) modparam("xlog", "force_color", 1) modparam("xlog", "log_colors", "L_ERR=cr;L_WARN=px,L_INFO=px") This is rsyslog configuration file: local0.* /var/log/kamailio/kamailio.log Kamailio is running under kamailio user, permission for the log file are as shown below: -rwxrwxr-x 1 kamailio kamailio 252885132 Dec 2 14:51 /var/log/kamailio/kamailio.log In request_route block this is the first line: xlog("L_INFO","[START ROUTING]-(Source IP=$si:$sp/Destination IP=$Ri:$Rp)\n"); The problem is that not every request is logged in the log-file. For example, some "INVITES" requests I can find in the log file, but some of them - I cannot, even that in sngrep I see the request and the call is successfully processed by kamailio. Also, for some requests i can see only partial data, not full call-flow as it supposed to be (for ex. i see only BYE requests, or ACK response instead of full call flow). Does someone have the same issue? If you don't - how are you dealing with kamailio log files? I'm thinking to send them to the central ELK stack, but if I have such problems by saving them locally - I don't see any reason to send them elsewhere. I may think that the problem could be in rsyslog itself, but how can I troubleshoot that? The traffic on the server is not very high - 30-50 concurrent calls. As a storage i'm using an SSD disk and xfs filesystem. Load on the disk - according to iostat/iotop - is minimum. OS: CentOS Linux 7 (Core), Kamailio: 5.2.0 (x86_64/linux) 535e13 Thank You.

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presence/mwi issue
by Andreas Granig 04 Dec '18

04 Dec '18

Hi, Currently playing with presence_mwi in kamailio 5.1, and it appears the issue reported by Juha two years ago at https://lists.kamailio.org//pipermail/sr-dev/2016-September/036609.html seems to have appeared again somewhere along the way. I’m sending a PUBLISH with message-summary event type, however I’m getting this error: ERROR: presence [presentity.c:283]: check_if_dialog(): failed to parse xml document As said in the old thread, it’s not an XML document, so it shouldn’t try to treat it as such. Any ideas? Best, Andreas

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Dispatcher with a naptr address as destination
by José Seabra 04 Dec '18

04 Dec '18

Hello all, I'm trying to use a naptr address as a dispatcher destination and i'm facing some issues here. When i configure a naptr address in dispatcher table, like: *1:1:sip\:proxy-registrar-ci1.fuze.domain:0:0:'':''* I see kamailio making a dns query of type A, but since that this name doesn't have any A record configured in dns, kamailio didn't get any positive reply from it. Should't kamailio make dns query by this order? 1. NAPTR 2. SRV 3. A I noticed also that if the name configured in dispatcher table is resolved successfully with A record from dns, then kamailio tries to make a query to NAPTR and then SRV services for the exactly same name used for query previously done(type A), but only if it gets a positive answer from dns to the A type query. I'm using kamailio version 5.1.0 with the following dns parameters configured: ####### DNS Parameters ######### dns = no rev_dns = no dns_try_ipv6 = no dns_retr_time = 1 dns_retr_no = 1 dns_cache_del_nonexp = no dns_cache_flags = 1 dns_cache_gc_interval = 120 dns_cache_init = 1 dns_cache_mem = 1000 dns_cache_negative_ttl = 60 dns_try_naptr = yes use_dns_cache = on use_dns_failover = on dns_srv_lb = yes Is there something wrongly configured here that can make kamailio behaves like that? Thank you. Regards -- Cumprimentos José Seabra

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Issue with ACK's
by Dan Quinney 30 Nov '18

30 Nov '18

Hi all, I’m having some issues with ACK’s. Specifically, when an ACK reaches Kamailio from either Asterisk or an endpoint, Kamailio is adding a second Via header - which isn’t valid as per RFC 3261 (17.1.1.3); “The ACK MUST contain a single Via header field, and this MUST be equal to the top Via header field of the original request." Endpoints are registered with Kamailio but INVITES are routed via Asterisk for dial plan and media reasons. I’m running fairly bog-standard Kamailio and Asterisk installs. I’ve attached a sample PCAP for if anyone has any suggestions. Thanks. Dan

2 2

Issue with ACK's
by Dan Quinney 30 Nov '18

30 Nov '18

Hi all, I’m having some issues with ACK’s. Specifically, when an ACK reaches Kamailio from either Asterisk or an endpoint, Kamailio is adding a second Via header - which isn’t valid as per RFC 3261 (17.1.1.3); “The ACK MUST contain a single Via header field, and this MUST be equal to the top Via header field of the original request." Endpoints are registered with Kamailio but INVITES are routed via Asterisk for dial plan and media reasons. I’m running fairly bog-standard Kamailio and Asterisk installs. I’ve attached a sample PCAP for if anyone has any suggestions. Thanks. Dan

1 0

Kamailio v5.2.0 Released - new major version is out
by Daniel-Constantin Mierla 30 Nov '18

30 Nov '18

Kamailio v5.2.0 is out – it comes with 6 new modules and a consistent group of improvements touching more than 100 existing modules. You can read detailed release notes at: * https://www.kamailio.org/w/kamailio-v5-2-0-release-notes/ Many thanks to all developers and community members that made possible this release. v5.2.0 brings more flexibility and optimizations for KEMI interpreters (including two new interpreters for Python3 and Ruby), major enhancements to load balancer and tls implementations, new variables and lots of other new features. Enjoy Kamailio v5.2.0! Thank you for flying Kamailio! Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference 2019 - http://www.kamailioworld.com

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Response VIA Header
by Soltanici Ilie 29 Nov '18

29 Nov '18

Hi, We have a Kamailio Instance running on Public IP Address, one of our ISP cannot send ACK back to us because we are sending 100 Trying - without "received" parameter. Is there any way in Kamailio to force a "received" parameter in the via header for 100 Trying Message? SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK8tgchsfjaKJdf.1;rport=5060 Thank you.

2 1

Forward() not handles Route headers
by Yuriy Gorlichenko 29 Nov '18

29 Nov '18

We are building some stateless solution and found that in case of using forward() kamailio sends traffic only via URI or $du variable. It never removes itself from route headers. Is this is the truth? Or I need to add some other module to handle it? (rr already added)

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KAMAILIO 4.2 - RTPPROXY config confirmation - error message from systemctl status kamailio.service -l (ERROR: <script>: NATMANAGE coee)
by yboujraf＠by-research.be 28 Nov '18

28 Nov '18

Dear All, I am still using Kamailio 4.2 with Debian Jessie without any issues. RTPPROXY - CONFIG > # Defaults for rtpproxy > # The control socket. > #CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" > # To listen on an UDP socket, uncomment this line: > CONTROL_SOCK=udp:127.0.0.1:7722 > # Additional options that are passed to the daemon. > > # log in /var/log/kamailio.log > EXTRA_OPTS="-l ip_public/192.168.40.101 -m 11000 -M 12000 -d DBUG:LOG_LOCAL0" WHEN I MAKE AN INVITE FROM 801 TO 803, I HAVE THIS ERROR MESSAGE : KAMAILIO SERVICE (ERROR: <SCRIPT>: NATMANAGE COEE) root@sipsecure:~# systemctl status kamailio.service -l ● kamailio.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio) Active: active (running) since Tue 2018-11-20 22:34:31 CET; 40s ago Process: 1820 ExecStop=/etc/init.d/kamailio stop (code=exited, status=0/SUCCESS) Process: 1830 ExecStart=/etc/init.d/kamailio start (code=exited, status=0/SUCCESS) CGroup: /system.slice/kamailio.service ├─1846 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1848 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1849 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1850 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1851 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1852 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1853 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1854 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1855 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1856 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1858 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1860 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1863 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1865 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1867 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1869 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1871 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio ├─1873 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio └─1875 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio Nov 20 22:35:00 sipsecure /usr/sbin/kamailio[1853]: ERROR: <script>: NATMANAGE coee Nov 20 22:35:00 sipsecure /usr/sbin/kamailio[1854]: ERROR: <script>: NATMANAGE M=INVITE OU=<null> RURI=<null> RD=<null> F=sip:801@192.168.40.101;transport=UDP T=sip:803@192.168.40.101;transport=UDP NH=<null> IP=192.168.0.207 ID=fjpUwRGm0u--3RvvB4fZ7Q.. Nov 20 22:35:00 sipsecure /usr/sbin/kamailio[1854]: ERROR: <script>: NATMANAGE coee Nov 20 22:35:00 sipsecure /usr/sbin/kamailio[1855]: ERROR: <script>: NATMANAGE M=INVITE OU=<null> RURI=<null> RD=<null> F=sip:801@192.168.40.101;transport=UDP T=sip:803@192.168.40.101;transport=UDP NH=<null> IP=192.168.0.207 ID=fjpUwRGm0u--3RvvB4fZ7Q.. Nov 20 22:35:00 sipsecure /usr/sbin/kamailio[1855]: ERROR: <script>: NATMANAGE coee Nov 20 22:35:03 sipsecure /usr/sbin/kamailio[1852]: ERROR: <script>: NATMANAGE M=INVITE OU=<null> RURI=<null> RD=<null> F=sip:801@192.168.40.101;transport=UDP T=sip:803@192.168.40.101;transport=UDP NH=<null> IP=192.168.0.207 ID=fjpUwRGm0u--3RvvB4fZ7Q.. Nov 20 22:35:03 sipsecure /usr/sbin/kamailio[1852]: ERROR: <script>: NATMANAGE coee Nov 20 22:35:03 sipsecure /usr/sbin/kamailio[1852]: NOTICE: acc [acc.c:317]: acc_log_request(): ACC: transaction answered: timestamp=1542749703;method=INVITE;from_tag=4d488551;to_tag=2006d156;call_id=fjpUwRGm0u--3RvvB4fZ7Q..;code=200;reason=OK;src_user=801;src_domain=192.168.40.101;src_ip=192.168.0.229;dst_ouser=803;dst_user=803;dst_domain=192.168.0.207;user_agent=Z 5.2.19 rv2.8.99 Nov 20 22:35:03 sipsecure /usr/sbin/kamailio[1853]: ERROR: <script>: NATMANAGE M=ACK OU=sip:803@192.168.0.207:5060;alias=192.168.0.207:5060~1 RURI=sip:803@192.168.0.207:5060 RD=192.168.0.207 F=sip:801@192.168.40.101;transport=UDP T=sip:803@192.168.40.101;transport=UDP NH=192.168.0.207 IP=192.168.0.229 ID=fjpUwRGm0u--3RvvB4fZ7Q.. Nov 20 22:35:03 sipsecure /usr/sbin/kamailio[1853]: ERROR: <script>: NATMANAGE coee Don't understand why the error??? RTPPROXY SERVICE : (RTPPROXY IS DOING HIS JOB PERFECTLY AND GET AUDIO ON BOTH DIRECTION) root@sipsecure:~# systemctl status rtpproxy.service -l ● rtpproxy.service - LSB: RTP Proxy Loaded: loaded (/etc/init.d/rtpproxy) Active: active (running) since Tue 2018-11-20 22:13:58 CET; 22min ago Process: 443 ExecStart=/etc/init.d/rtpproxy start (code=exited, status=0/SUCCESS) CGroup: /system.slice/rtpproxy.service └─1137 /usr/bin/rtpproxy -s udp:127.0.0.1 7722 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -l 212.76.231.102 192.168.40.101 -m 11000 -M 12000 -d DBUG LOG_LOCAL0 Nov 20 22:34:31 sipsecure rtpproxy[1137]: DBUG:doreply: sending reply "1875_11 1 " Nov 20 22:35:00 sipsecure rtpproxy[1137]: DBUG:handle_command: received command "1853_8 USEEc106,9,3,111,0,8,97,110,112,98,101,100,99,102 fjpUwRGm0u--3RvvB4fZ7Q.. 192.168.0.229 11000 4d488551;1" Nov 20 22:35:00 sipsecure rtpproxy[1137]: INFO:handle_command: new session fjpUwRGm0u--3RvvB4fZ7Q.., tag 4d488551;1 requested, type strong Nov 20 22:35:00 sipsecure rtpproxy[1137]: INFO:handle_command: new session on a port 11496 created, tag 4d488551;1 Nov 20 22:35:00 sipsecure rtpproxy[1137]: INFO:handle_command: pre-filling caller's address with 192.168.0.229:11000 Nov 20 22:35:00 sipsecure rtpproxy[1137]: DBUG:doreply: sending reply "1853_8 11496 192.168.40.101 " Nov 20 22:35:03 sipsecure rtpproxy[1137]: DBUG:handle_command: received command "1852_8 LSEEc106,9,3,111,0,8,97,110,112,102,98,101,100,99 fjpUwRGm0u--3RvvB4fZ7Q.. 192.168.0.207 11112 4d488551;1 2006d156;1" Nov 20 22:35:03 sipsecure rtpproxy[1137]: INFO:handle_command: lookup on ports 11496/11402, session timer restarted Nov 20 22:35:03 sipsecure rtpproxy[1137]: INFO:handle_command: pre-filling callee's address with 192.168.0.207:11112 Nov 20 22:35:03 sipsecure rtpproxy[1137]: DBUG:doreply: sending reply "1852_8 11402 192.168.40.101 " ??? Question : for rtpproxy config, do I need to add the natted ip address (NAT 1:1) or the real public address (where ip phone needs to assign as domain)? I am confused. Thanks a lot in advance Best Regards, Youssef

3 6

Issue sending SIP OPTIONS request to multiple contacts
by Dan Quinney 28 Nov '18

28 Nov '18

Hi, When using Kamailio v5.2.0-rc2 (x86_64/linux) 9e729b and setting the below nathelper module parameters it appears only the first contact is sent a SIP OPTIONS request. modparam("nathelper", "natping_interval", 15) modparam("nathelper", "ping_nated_only", 0) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:ping@mydomain <sip:ping@mydomain>.com") modparam("nathelper", "keepalive_timeout", 90) Reading the release notes for v5.2 what I’m doing *should* be possible, but as the functionality has been recently added I’m assuming that it might be a bug. Any assistance on this would be great. Thanks, Dan

2 1

Releasing Kamailio v5.2.0
by Daniel-Constantin Mierla 28 Nov '18

28 Nov '18

Hello, I am consider to release v5.2.0 (the first stable version out of branch 5.2) next week, likely on Wednesday, Nov 28, 2018. It still allows a bit more than a week of testing as well as well time to prepare the online resources for it (documentation, wiki pages, upgrade guidelines, etc...). If there is any issue you are of and not yet reported to github bug tracker, do it as soon as possible to give it a chance to be fixed in time for the next major release. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

1 2

Error opening Kamailio's FIFO
by Markus 28 Nov '18

28 Nov '18

Hi list, when I run "/usr/sbin/kamctl address reload" I get: ERROR: Error opening Kamailio's FIFO /tmp/kamailio_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")' in your config ERROR: and also have loaded the mi_fifo module. In /etc/kamailio/kamailio.cfg I have: loadmodule "mi_fifo.so" # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") But there's no fifo in /tmp. What am I doing wrong? kamailio-4.4.4-1.1.x86_64 on CentOS Linux release 7.2.1511 (Core). Thank you! Markus

3 8

[CFP] FOSDEM 2019, RTC devroom, speakers, volunteers neeeded
by FOSDEM RTC Team 28 Nov '18

28 Nov '18

FOSDEM is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2019 takes place 2-3 February 2019 in Brussels, Belgium. https://fosdem.org This email contains information about: - Real-Time Communications dev-room and lounge, - speaking opportunities, - volunteering in the dev-room and lounge, - social events (the legendary FOSDEM Beer Night and Saturday night dinners provide endless networking opportunities), - the Planet aggregation sites for RTC blogs Call for participation - Real Time Communications (RTC) ======================================================= The Real-Time dev-room and Real-Time lounge is about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, peer-to-peer, privacy and encryption. The dev-room is a successor to the previous XMPP and telephony dev-rooms. We are looking for speakers for the dev-room and volunteers and participants for the tables in the Real-Time lounge. The dev-room is only on Sunday, 3rd of February 2019. The lounge will be present for both days. To discuss the dev-room and lounge, please join the Free RTC mailing list: http://lists.freertc.org/mailman/listinfo/discuss To be kept aware of major developments in Free RTC, without being on the discussion list, please join the Free-RTC Announce list: http://lists.freertc.org/mailman/listinfo/announce Speaking opportunities ---------------------- Note: if you used FOSDEM Pentabarf before, please use the same account/username Real-Time Communications dev-room: deadline 23:59 UTC on 2nd of December. Please use the Pentabarf system to submit a talk proposal for the dev-room. On the "General" tab, please look for the "Track" option and choose "Real Time Communications devroom". https://penta.fosdem.org/submission/FOSDEM19/ Other dev-rooms and lightning talks: some speakers may find their topic is in the scope of more than one dev-room. It is encouraged to apply to more than one dev-room and also consider proposing a lightning talk, but please be kind enough to tell us if you do this by filling out the notes in the form. You can find the full list of dev-rooms at https://www.fosdem.org/2019/schedule/tracks/ and apply for a lightning talk at https://fosdem.org/submit Main track: the deadline for main track presentations is 23:59 UTC 3rd of November. Leading developers in the Real-Time Communications field are encouraged to consider submitting a presentation to the main track at https://fosdem.org/submit First-time speaking? -------------------- FOSDEM dev-rooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the dev-room administrators personally if you would like to ask any questions about it. Submission guidelines --------------------- The Pentabarf system will ask for many of the essential details. Please remember to re-use your account from previous years if you have one. In the "Submission notes", please tell us about: - the purpose of your talk - any other talk applications (dev-rooms, lightning talks, main track) - availability constraints and special needs You can use HTML and links in your bio, abstract and description. If you maintain a blog, please consider providing us with the URL of a feed with posts tagged for your RTC-related work. We will be looking for relevance to the conference and dev-room themes, presentations aimed at developers of free and open source software about RTC-related topics. Please feel free to suggest a duration between 20 minutes and 55 minutes but note that the final decision on talk durations will be made by the dev-room administrators based on the number of received proposals. As the two previous dev-rooms have been combined into one, we may decide to give shorter slots than in previous years so that more speakers can participate. Please note FOSDEM aims to record and live-stream all talks. The CC-BY license is used. Volunteers needed ================= To make the dev-room and lounge run successfully, we are looking for volunteers: - FOSDEM provides video recording equipment and live streaming, volunteers are needed to assist in this - organizing one or more restaurant bookings (dependending upon number of participants) for the evening of Saturday, 2nd of February - participation in the Real-Time lounge - helping attract sponsorship funds for the dev-room to pay for the Saturday night dinner and any other expenses - circulating this Call for Participation to other mailing lists Social events and dinners ========================= The traditional FOSDEM beer night occurs on Friday, 1st of February. On Saturday night, there are usually dinners associated with each of the dev-rooms. Most restaurants in Brussels are not so large so these dinners have space constraints and reservations are essential. Please subscribe to the Free-RTC mailing list for further details about the Saturday night dinner options and how you can register for a seat: http://lists.freertc.org/mailman/listinfo/discuss Spread the word and discuss =========================== If you know of any mailing lists where this CfP would be relevant, please forward this email. If this dev-room excites you, please blog or microblog about it, especially if you are submitting a talk. If you regularly blog about RTC topics, please send details about your blog to the planet site administrators: All projects http://planet.freertc.org planet(a)freertc.org XMPP http://planet.jabber.org ralphm(a)ik.nu SIP http://planet.sip5060.net planet(a)sip5060.net (Español) http://planet.sip5060.net/es/ planet(a)sip5060.net Please also link to the Planet sites from your own blog or web site as this helps everybody in the free real-time communications community. Contact ======= For any private queries, contact us directly using the address fosdem-rtc-admin(a)freertc.org and for any other queries please ask on the Free-RTC mailing list: http://lists.freertc.org/mailman/listinfo/discuss The dev-room administration team: Saúl Ibarra Corretgé <s(a)saghul.net> Ralph Meijer <ralphm(a)ik.nu> Daniel-Constantin Mierla <miconda(a)gmail.com> Daniel Pocock <daniel(a)pocock.pro> Guus der Kinderen <guus.der.kinderen(a)gmail.com>

2 1

Dialiog and dispatcher
by David Villasmil 27 Nov '18

27 Nov '18

Hello guys, I'm testing the dispatcher with the dialog module. After making several test calls, which are properly routed, the number of dialogs stay there for a while, preventing new calls from going to those destinations: "SET": { "ID": 101, "TARGETS": [{ "DEST": { "URI": "sip:1.2.3.4:5080", "FLAGS": "AP", "PRIORITY": 0, "ATTRS": { "BODY": "weight=100;maxload=10;duid=bedd39c7-6c73-4477-ae7d-77f8cdc96797", "DUID": "bedd39c7-6c73-4477-ae7d-77f8cdc96797", "MAXLOAD": 10, "WEIGHT": 100, "RWEIGHT": 0, "SOCKET": "" }, "RUNTIME": { "DLGLOAD": 10 } } }] The calls have been down for several seconds, but the "DLGLOAD" parameter doesn't go down, even though /usr/local/kamailio5/sbin/kamctl', 'dialog', 'show Output the correct amount of dialogs. I set the dialog handler just before relaying out: t_on_failure("RTF_DISPATCH"); set_dlg_profile("gateways","$du"); route(RELAY); And the module is supposed to remove them automatically. Am i doing something wrong? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 2

dispatcher: Get gateway current calls
by David Villasmil 27 Nov '18

27 Nov '18

Hello guys, is there any way of printing out the selected gw current calls? i.e.: if(ds_select_dst("1", "4")) { xlog("L_ERR", "[DISPATCH] $ci: ds_select_routes was succesful (du: <$du>)\n"); ****** PRINT GWs CALLS ******* } else { xlog("L_ERR", "[DISPATCH] $ci: did NOT find a gateway to use!\n"); send_reply("404", "No destination"); exit; } xlog("L_DBG", "[DISPATCH] $ci: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 1

Re: [SR-Users] "kamdbctl create" failing with MySQL 8
by Jonathan Tyler 26 Nov '18

26 Nov '18

I just had the same thing happen to me. I installed Kamailio via Apt, and I'm using a dedicated MySQL server, with non-root admin credentials. Previously today I installed it using a local MySQL 5.7 database with no issues. Kamailio Host: Kamailio 5.1.2 (Ubuntu 18.04) MySQL Host: MySQL 8.0.13 (Windows Server 2012 R2) It creates the database, then fails at granting permissions. Accounts were created and privileges flushed before executing. > root@kamailio:~# kamdbctl create > INFO: creating database kamailio ... > mysql: [Warning] Using a password on the command line interface can be insecure. > INFO: granting privileges to database kamailio ... > mysql: [Warning] Using a password on the command line interface can be insecure. > ERROR 1064 (42000) at line 1: You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'IDENTIFIED BY '<password>'' at line 1 I turned on query logging on the database host but didn't see any errors there, or in the error log itself. > 2018-11-21T01:58:15.045586Z 8 Connect <user>@kamailio on using SSL/TLS > 2018-11-21T01:58:15.046239Z 8 Query select @@version_comment limit 1 > 2018-11-21T01:58:15.046764Z 8 Query CREATE DATABASE kamailio CHARACTER SET utf8mb4 > 2018-11-21T01:58:15.189109Z 8 Quit > 2018-11-21T01:58:15.208476Z 9 Connect <user>@kamailio on using SSL/TLS > 2018-11-21T01:58:15.209019Z 9 Query select @@version_comment limit 1 > 2018-11-21T01:58:15.210114Z 9 Quit I've tried with the default MySQL 5.7 Client on Ubuntu, and upgrading it to 8.0.13, but got the same result. Unfortunately I don't know how to see the output of the SQL Statement being sent (without installing 5.7), or I'd try to run it manually. - Jonathan

3 5

issue tracking dialogs with db_redis in kamailio 5.2
by David Escartín 26 Nov '18

26 Nov '18

hello all using version 5.2.0-rc0 of kamailio, we are trying to track the dialogs of dialog module using db_redis module with this setup #!define DBURL_DIALOG "redis://127.0.0.1:6390/7" # ----- dialog params ----- modparam("dialog", "db_url", DBURL_DIALOG) modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 16384) modparam("dialog", "dlg_flag", 31) modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "db_mode", 1) modparam("dialog", "timer_procs", 1) modparam("dialog", "db_update_period", 1) # ----- db_redis params ----- modparam("db_redis", "schema_path", "/usr/local/kamailio/share/kamailio/db_redis/kamailio") modparam("db_redis", "keys", "dialog=entry:hash_entry,hash_id,callid") modparam("db_redis", "keys", "dialog_vars=entry:hash_entry,hash_id,dialog_key,dialog_value") and we see it works and if a kamailio instance resets in the middle of a dialog it recovers all the dlg_vars and values because we have the db_skip_load to 0 in dialog module. But we have noticed that the redis is being written with the information of the dlg_var assigned until that moment when we get a provisional response with SDP, and we dom see a set on the redis with the dialog entry and id untill call is connected. Nov 16 13:22:55 proxy-1 /usr/local/kamailio/sbin/kamailio[22424]: INFO: Call-ID 1-1511(a)9.7.8.1: Status 180 Nov 16 13:22:55 proxy-1 /usr/local/kamailio/sbin/kamailio[22424]: WARNING: db_redis [redis_dbase.c:1489]: db_redis_perform_update(): performing full table scan on table 'dialog' while performing update Nov 16 13:22:55 proxy-1 /usr/local/kamailio/sbin/kamailio[22424]: WARNING: db_redis [redis_dbase.c:1492]: db_redis_perform_update(): scan key 0 is 'hash_entry' Nov 16 13:22:55 proxy-1 /usr/local/kamailio/sbin/kamailio[22424]: WARNING: db_redis [redis_dbase.c:1492]: db_redis_perform_update(): scan key 1 is 'hash_id' on the redis (we only see hmset for dialog_vars key) 1542374575.721816 [7 127.0.0.1:56858] "HMSET" "dialog_vars:entry::13725:16494:_uac_funew:sip:1231215423@voda.interconnect.manxtelecom.im" "hash_entry" "13725" "hash_id" "16494" "dialog_key" "_uac_funew" "dialog_value" "sip:1231215423@voda.interconnect.manxtelecom.im" 1542374575.722001 [7 127.0.0.1:56858] "HMSET" "dialog_vars:entry::13725:16494:_uac_fu:sip:anonymous@voda.interconnect.manxtelecom.im" "hash_entry" "13725" "hash_id" "16494" "dialog_key" "_uac_fu" "dialog_value" "sip:anonymous@voda.interconnect.manxtelecom.im" .... Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22425]: INFO: Call-ID 1-1511(a)9.7.8.1: Status 200 Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22425]: WARNING: db_redis [redis_dbase.c:1489]: db_redis_perform_update(): performing full table scan on table 'dialog_vars' while performing update Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22425]: WARNING: db_redis [redis_dbase.c:1492]: db_redis_perform_update(): scan key 0 is 'hash_entry' Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22425]: WARNING: db_redis [redis_dbase.c:1492]: db_redis_perform_update(): scan key 1 is 'hash_id' Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22425]: WARNING: db_redis [redis_dbase.c:1492]: db_redis_perform_update(): scan key 2 is 'dialog_key' Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22426]: WARNING: db_redis [redis_dbase.c:1489]: db_redis_perform_update(): performing full table scan on table 'dialog' while performing update Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22426]: WARNING: db_redis [redis_dbase.c:1492]: db_redis_perform_update(): scan key 0 is 'hash_entry' Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22426]: WARNING: db_redis [redis_dbase.c:1492]: db_redis_perform_update(): scan key 1 is 'hash_id' Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22426]: WARNING: db_redis [redis_dbase.c:1489]: db_redis_perform_update(): performing full table scan on table 'dialog' while performing update Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22426]: WARNING: db_redis [redis_dbase.c:1492]: db_redis_perform_update(): scan key 0 is 'hash_entry' Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22426]: WARNING: db_redis [redis_dbase.c:1492]: db_redis_perform_update(): scan key 1 is 'hash_id' Nov 16 13:23:05 proxy-1 /usr/local/kamailio/sbin/kamailio[22426]: INFO: Call-ID 1-1511(a)9.7.8.1: ACK received in A-Leg on the redis 1542374585.810520 [7 127.0.0.1:56882] "SCAN" "0" "MATCH" "dialog:entry::*" "COUNT" "1000" 1542374585.810672 [7 127.0.0.1:56882] "EXISTS" "dialog:entry::13725:16494:1-1511@9.7.8.1" 1542374585.810683 [7 127.0.0.1:56882] "HMGET" "dialog:entry::13725:16494:1-1511@9.7.8.1" "hash_entry" "hash_id" 1542374585.810776 [7 127.0.0.1:56882] "HMSET" "dialog:entry::13725:16494:1-1511@9.7.8.1" "state" "4" "timeout" "1542377585" "caller_cseq" "1" "callee_cseq" "0" "caller_contact" "sip:0000123456@9.7.8.1:5084" "callee_contact" "sip:50622959898@9.70.1.52:5060;transport=udp" 1542374585.810899 [7 127.0.0.1:56882] "SCAN" "0" "MATCH" "dialog:entry::*" "COUNT" "1000" 1542374585.810969 [7 127.0.0.1:56882] "EXISTS" "dialog:entry::13725:16494:1-1511@9.7.8.1" 1542374585.810977 [7 127.0.0.1:56882] "HMGET" "dialog:entry::13725:16494:1-1511@9.7.8.1" "hash_entry" "hash_id" 1542374585.811059 [7 127.0.0.1:56882] "HMSET" "dialog:entry::13725:16494:1-1511@9.7.8.1" "state" "4" "timeout" "1542377585" "caller_cseq" "1" "callee_cseq" "0" "caller_contact" "sip:0000123456@9.7.8.1:5084" "callee_contact" "sip:50622959898@9.70.1.52:5060;transport=udp" i don't see a redis set order to update the dialog entry key before the 200OK is received, to create and update state 1, etc. when created with dlg_manage command could you please check if we might be missing something? thanks a lot and regards david

2 3

Kamailio "Presence Server" configuration
by denislav.tsonev＠musala.com 26 Nov '18

26 Nov '18

Hello, I'm trying to configure Kamailio to be "Presence Server". Simple scheme: two phones (one "hard phone" Grandstream GXP1620 and "Soft phone" MicroSIP) and Kamailio server with default configuration. So when "MicroSIP" send SIP SUBSCRIBE to Kamailio, server response with SIP NOTIFY but without XML part. Then when hard phone change his status from "Ready" to "Ringing" and then "On-The-Phone" Kamailio is not sending any SIP NOTIFY messages. Is this normal behavior or I missed something? (In DB I have records in "active_watchers" and "watchers" tables but not in "presentity" table) ====================================================================== ====================== Module configuration: ============================= ====================================================================== #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) modparam("presence", "server_address", "sip:10.0.6.123:5060") # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif ====================================================================== ============================================================== ======================== Route: ================================ ============================================================== route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!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; } ================================================================ Thanks in advance Denislav Raychev Tsonev | Integration and Infrastructure Engineer | Musala Soft JSC www.musala.com | t: +359 2 969 58 21 | m: +359 878 270 965 | f: +359 2 969 58 22 World Trade Center, 36 Dragan Tsankov blvd., Sofia 1057, Bulgaria

2 1

Dispatcher on 5.2
by Igor Olhovskiy 26 Nov '18

26 Nov '18

Hi! Am I missing something or it’s a bug? Idea is following. I’m using dispatcher to load-balance calls. Using same weight destinations, order like in a file. So, dispatcher.list looks like 1 sip:172.28.11.6:5060 1 sip:172.28.11.4:5060 I’m using round-robin algo with ds_select_dst(«1», «4») But when it selects 172.28.11.4 and it for a reason not working, it will never come back to 172.28.11.6. Log: … Relaying packet du: sip:172.28.11.4:5060 … BYE/INVITE timeout with no reply … 49(59) WARNING: dispatcher [dispatch.c:2314]: ds_update_dst(): no xavp uri field in next destination record ((nil)) modparam("dispatcher", "list_file", "/tmp/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "ds_hash_size", 9) Kamailio: version: kamailio 5.2.0-pre0 Regards, Igor

2 3

database connections per child process
by Karthik Srinivasan 23 Nov '18

23 Nov '18

Hello, Is there a formula/algorithm that determines how many TCP connections are made per child process to a database server defined by sqlops' sqlcon parameter? ex: if i have 1 database on a database server defined by sqlcon, and i have 1 UDP child process, i notice that the child process has made 2 TCP connections to the database server. ex: if i have 3 databases on the same database server defined by 3 sqlcon statements (1 pertaining to each database) and i have 4 UDP child processes, i notice that each child process makes 4 TCP connections to the database server. ex: if i have 3 databases on the same database server defined by 3 sqlcon statements (1 pertaining to each database) and i have 6 TCP child processes, i notice that each child process makes 4 TCP connections to the database server. Basically wanting to know what governs the number of tcp connections a child process makes to a particular database server. Thank you, Karthik

3 3

IMS: SMS over IP - how to get the originator's tel address?
by Ron McLeod 22 Nov '18

22 Nov '18

I am working on a VoLTE/IMS scenario to transfer text messages as SMS over IP (+g.3gpp.smsip feature) between UE's. This is a high-level view of the message flows between the endpoints (not showing CSCFs): UE1 IP-SM-GW |---- MESSAGE (SUBMIT) --------------->| |<------------------- 202 ACCEPTED ----| | | |<-------- MESSAGE (SUBMIT_REPORT) ----| |---- 200 OK ------------------------->| | UE2 | | | |<-------------- MESSAGE (DELIVER) ----| |---- 200 OK ------------------------->| The messages does get transferred from UE1 to UE2 successfully. Since this is GSM-style SMS, the OA (originating address) and DA (destination address) are specified using telephone numbers, not sip addresses. This is not a problem for the DA, since it is specified in the SUBMIT TPDU from the originating handset, but when the IP-SM-GW sends the DELIVER TPDU it needs to specify the OA so that the recipient knows who sent the SMS, and this was not provided in the MESSAGE with the SMS-SUBMIT. According to 3GPP TS 24.341 (http://www.qtc.jp/3GPP/Specs/24341-b20.pdf): "The IP-SM-GW will have to use an address of the SM-over-IP sender that the SC can process (i.e. an E.164 number). This address will come from a tel URI in a P-Asserted-Identity header (as defined in RFC 3325 [13]) placed in the SIP MESSAGE request by the P-CSCF or S-CSCF." Looking at the MESSAGE request sent from the S-CSCF to the IP-SM-GW, it looks like neither the P-CSCF nor the S-CSCF is providing the tel address in a P-Asserted-Identity header: MESSAGE tel:+3108012345680 SIP/2.0 Route: <sip:defaultapp@tas.core.ims1.test;lr>, <sip:iscmark@scscf.core.ims1.test:6060;lr;s=1;h=0;d=0;a=7369703a303031303330 31323334353637383940696d732e6d6e633030332e6d63633030312e336770706e6574776f72 6b2e6f7267> P-Served-User: <sip:001030123456789@ims.mnc003.mcc001.3gppnetwork.org>;sescase=orig;regstat e=reg f: <sip:001030123456789@ims.mnc003.mcc001.3gppnetwork.org>;tag=2703358633 t: <tel:+3108012345680> CSeq: 555874977 MESSAGE i: 2703358625_2324045032@2001:470:eb88:b021:27f:9507:589d:c201 Via: SIP/2.0/UDP [2001:470:EB88:150:0:0:0:22]:6060;branch=z9hG4bKb857.bd3345ba81ec10b1d11639d 9c7414606.0;i=a Via: SIP/2.0/TCP [2001:470:EB88:150:5054:FF:FE44:BFC9];branch=z9hG4bKb857.be53cc1c583b9c5f9dc ebf3306f2b7c1.0 v: SIP/2.0/UDP [2001:470:eb88:b021:27f:9507:589d:c201]:8906;rport=8013;branch=z9hG4bK355262 2488 Max-Forwards: 68 P-Access-Network-Info: 3GPP-E-UTRAN-FDD; utran-cell-id-3gpp=0010300030000001 c: application/vnd.3gpp.sms Allow: MESSAGE d: no-fork User-Agent: Xiaomi_Redmi 3S_Android6.0.1_9 l: 27 P-Charging-Vector: icid-value=49565300000000511100004B01000000; icid-generated-at=2001:470:EB88:150:5054:FF:FE44:BFC9 P-Asserted-Identity: <sip:001030123456789@ims.mnc003.mcc001.3gppnetwork.org> P-Visited-Network-ID: core.ims1.test Message Body GSM A-I/F RP - RP-DATA (MS to Network) Message Type RP-DATA (MS to Network) RP-Message Reference RP-Message Reference: 0x05 (5) RP-Originator Address RP-Destination Address - (3108012345680) RP-User Data Length: 14 TPDU (not displayed) GSM SMS TPDU (GSM 03.40) SMS-SUBMIT 0... .... = TP-RP: TP Reply Path parameter is not set in this SMS SUBMIT/DELIVER .0.. .... = TP-UDHI: The TP UD field contains only the short message ..1. .... = TP-SRR: A status report is requested ...0 0... = TP-VPF: TP-VP field not present (0) .... .0.. = TP-RD: Instruct SC to accept duplicates .... ..01 = TP-MTI: SMS-SUBMIT (1) TP-MR: 92 TP-Destination-Address - (8000) TP-PID: 0 TP-DCS: 0 TP-User-Data-Length: (5) depends on Data-Coding-Scheme TP-User-Data SMS text: HELLO Is this something that in configurable at the P-CSCF or the S-CSCF, or would it require now development? Thanks, Ron

2 3

issues with the debian repo for kamailio 5.2
by Aleksandar Sosic 22 Nov '18

22 Nov '18

Hi guys, I was trying to install the last 5.2 kamailio version editing the `/etc/apt/sources.list`: ``` deb http://deb.kamailio.org/kamailio52-nightly xenial main deb-src http://deb.kamailio.org/kamailio52-nightly xenial main ``` But as you can see there's only the 5.3 and 4.3 available: ``` root@e7845b3041eb:/# apt-cache madison kamailio kamailio | 5.3.0~dev0+0~20181114003826.5+xenial | http://deb.kamailio.org/kamailio52-nightly xenial/main amd64 Packages kamailio | 4.3.4-1.1ubuntu2 | http://archive.ubuntu.com/ubuntu xenial/universe amd64 Packages kamailio | 4.3.4-1.1ubuntu2 | http://archive.ubuntu.com/ubuntu xenial/universe Sources kamailio | 5.3.0~dev0+0~20181114003826.5+xenial | http://deb.kamailio.org/kamailio52-nightly xenial/main Sources ``` Any hints on how to solve this are very welcome. :) Thanks, -- Aleksandar Sosic linkedin.com/in/alexsosic/ mail: alex.sosic(a)evosip.cloud

5 6

t_set_retr and NOTIFY send after handle_publish
by Vasiliy Ganchev 21 Nov '18

21 Nov '18

Hi, list! I have a tricky problem, need advise from the community. The problem I need to solve: - have a SIP device, with a big number of BLFs configured, in some moment of time the device is slow to handle incoming SIP messages (a lot of NOTIFYs received), kamailio start retransmitting NOTIFYs, and the device get stuck. My idea is to change retransmits timeout for NOTIFYs that are sent out after handle_publish, so if get stuck, not to make "DoS" I have. What I tried: - t_set_retr(5000, 10000); before handle_publish => does not work - my tm is compiled with DTM_DIFF_RT_TIMEOUT, and t_set_retr works for e.g. INVITEs I have read the mailing list and also studied sources of presence and tm modules. I see that at the moment there is no way to solve what I want from config. My next steps are going to make some hacks in C code... at least to understand the code better and find some new insights Any other ideas? Comments/ideas are highly welcomed! cheers -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

2 3

Closing all ws connections
by Andrey Deykunov 21 Nov '18

21 Nov '18

Hi guys, I've implemented an rpc call that closes all WS connections: void ws_rpc_close_all(rpc_t *rpc, void *ctx) { ws_connection_t **list = NULL, **list_head = NULL; ws_connection_t *wsc = NULL; int ret; list_head = wsconn_get_list(); if(!list_head) return; list = list_head; wsc = *list_head; while(wsc) { LM_WARN("Closing connection\n"); ret = close_connection(&wsc, LOCAL_CLOSE, 1000, str_status_normal_closure); wsc = *(++list); } wsconn_put_list(list_head); } Please take a look. Can I call close_connection() and perform list iterations in this function not using WSCONN_LOCK? Thanks, Andrey

3 2

from Asterisk to Kamailio
by Valter Nogueira 21 Nov '18

21 Nov '18

Today, I use Asterisk as a SIP/RTP PROXY I proxy from customers Asterisks to a VOIP provider, in a multi-homed server. Now, I want to move to Kamailio without any rupture in customer's configuration. As anyone can imagine I am kind of lost. USER ACCOUNTS In Asterisk, I create a dynamic host account named ACCOUNT1 and I receive in *FROM HEADER sip:ACCOUNT1@customer_ip_address* In Kamailio, I have to define the account's domain like *kamctl add ACCOUNT1(a)mydomain.com <ACCOUNT1(a)mydomain.com> password. *Kamailio just accepts a REGISTER/INVITE from *ACCOUNT1(a)mydomain.com <ACCOUNT1(a)mydomain.com>* SIP/RTP PROXY In Asterisk, I just dialout to the VOIP PROVIDER like *dial (SIP/VOIP_ACCOUNT/${EXTENSION})* Asterisk does all the magic (it is a B2BUA). It bridges the new call and media to the original call. Moreover, user don't know anything about how call are completed, nor how credentials are setup and soon. In Kamailio, I guess that I have to use nat, tm and rtpproxy modules and maybe uac. I am not sure how to setup it. Can someone send me a clue? Thank you, Valter

4 7

Kamailio-5.1.6 Crash
by Hamid Hashmi 20 Nov '18

20 Nov '18

I have been using the following version of kamailio Version: kamailio 5.1.6 (x86_64/linux) Default config: /usr/local/etc/kamailio/kamailio.cfg Default paths to modules: /usr/local/lib64/kamailio/modules Compile flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES MAX_RECV_BUFFER_SIZE=262144 MAX_URI_SIZE=1024 BUF_SIZE=65535 DEFAULT PKG_SIZE=8MB DEFAULT SHM_SIZE=64MB ADAPTIVE_WAIT_LOOPS=1024 TCP poll methods: poll, epoll_lt, epoll_et, sigio_rt, select Source code revision ID: unknown Compiled with: gcc 4.8.5 Compiled on: 11:48:33 Oct 11 2018 and I noticed today that the Kamailio service is crashing. [root@kamailiostandby /]# gdb core.17805 GNU gdb (GDB) Red Hat Enterprise Linux 7.6.1-100.el7_4.1 Copyright (C) 2013 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html> This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-redhat-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... [New LWP 17805] Core was generated by `/usr/local/sbin/kamailio -P /run/kamailio/kamailio.pid -f /usr/local/SIP'. Program terminated with signal 11, Segmentation fault. #0 0x00007f636559e54b in ?? () "/core.17805" is a core file. Please specify an executable to debug. (gdb) list No symbol table is loaded. Use the "file" command. (gdb) bt #0 0x00007f636559e54b in ?? () #1 0x00007f63a9571810 in ?? () #2 0x0000000000000000 in ?? () Regards Hamid R. Hashmi

2 3

registrar::save() does not save 'received' field of the 'location' table
by Andrey Deykunov 20 Nov '18

20 Nov '18

Hi, We deploy kamailio in the cluster containing two nodes (Active and Passive). When Active node becomes Passive, our websocket clients continue sending REGISTER to this node. Passive node routes these requests to the Active node through the UDP, that causes the following warning message: 2018-11-14T04:43:31.070005-08:00 qapbx-ca1-1 kamailio[25597]: WARNING: <core> [core/forward.c:231]: get_send_socket2(): protocol/port mismatch (forced udp:64.78.52.88:5060, to ws:10.9.3.116:20033) Every registered client should be stored in the 'location': $var(save_val) = save("location", 0, "sip:$au@$td"); if ($var(save_val) <= 0) { xlog("L_ERR", "[$fU,R,$ci]: Registration saving failed, contact $ct"); sl_reply_error(); } else if ($var(save_val) == 3) { xlog("L_INFO", "[$fU,R,$ci]: Unregistration accepted. Login: $fU; ua: $ua; s-ip: $si; s-port: $sp; 3d parm: sip:$au@$td"); } else { xlog("L_INFO", "[$fU,R,$ci]: Registration accepted. Login: $fU; ua: $ua; s-ip: $si; s-port: $sp; 3d parm: sip:$au@$td"); } Log output of the Avtive node: 2018-11-14T04:43:31.063376-08:00 qapbx-ca1-2 kamailio[24722]: INFO: <script>: [54004221,R,ibca3ujmr0q81nn0l5puen]: REGISTER received with R-URI: sip:64.78.52.90, ct: <sip:vu2o757j@si7ofk7i8cs7.invalid;transport=ws>;+sip.ice; reg-id=1;+sip.instance="<urn:uuid:00ff4c8d-2a5e-4614-b80a-d7c14c0e41e3>";expires=600, cseq=1, s-ip: 64.78.52.88, s-port: 5060 2018-11-14T04:43:31.063947-08:00 qapbx-ca1-2 kamailio[24722]: INFO: <script>: Our node is active 2018-11-14T04:43:31.064323-08:00 qapbx-ca1-2 kamailio[24722]: INFO: <script>: [54004221,R,ibca3ujmr0q81nn0l5puen]: Authorization headers are empty in REGISTER command. Send 401 Unauthorized to 64.78.52.88 2018-11-14T04:43:31.285967-08:00 qapbx-ca1-2 kamailio[24716]: INFO: <script>: [54004221,R,ibca3ujmr0q81nn0l5puen]: REGISTER received with R-URI: sip:64.78.52.90, ct: <sip:vu2o757j@si7ofk7i8cs7.invalid;transport=ws>;+sip.ice; reg-id=1;+sip.instance="<urn:uuid:00ff4c8d-2a5e-4614-b80a-d7c14c0e41e3>";expires=600, cseq=2, s-ip: 64.78.52.88, s-port: 5060 2018-11-14T04:43:31.286564-08:00 qapbx-ca1-2 kamailio[24716]: INFO: <script>: Our node is active 2018-11-14T04:43:31.286943-08:00 qapbx-ca1-2 kamailio[24716]: INFO: <script>: [54004221,R,ibca3ujmr0q81nn0l5puen]: Registration accepted. Login: 54004221; ua: JsSIP 3.2.11; s-ip: 64.78.52.88; s-port: 5060; 3d parm: sip:54004221@qapbx-ca1.qaserverdata.net but in this scenario the save() method stores null value in the 'received' row. How can I fix it? Thank you, Andrey

2 1

RTPengine: shouldn't SDP <sess-version> be incremented when SDP is updated in-dialog?
by George Diamantopoulos 20 Nov '18

20 Nov '18

Hello all, I have the following issue (I think) with RTP engine. An INVITE comes in, and rtpengine will rewrite the SDP accordingly, as configured in kamailio.cfg. After some time a reINVITE is sent out in the opposite direction, for session refresh purposes. As I use rtcp-mux-offer in kamailio.cfg for this direction, RTPengine will inject the rtcp-mux parameter, and this reINVITE is forwarded to the UAC that sent the original INVITE. However, the SDP in the reINVITE is exactly the same as the SDP in the 200 OK to the original INVITE, with the exception of the rtcp-mux parameter. Since the SDP offered from the same end has changed, shouldn't session version be incremented as well? Does this sound like something that should be reported as a bug to rtpengine? Or am I missing something here? Thanks! Best regards, George Diamantopoulos

3 4

Number of sql connections
by Duarte Rocha 19 Nov '18

19 Nov '18

Greetings, I have my core parameter listen like this : listen=udp:eth0 children=12 Like this and the other settings, my kamailio has 21 threads and 20 Sql connections. If I add more listens, listen=udp:eth0 listen=udp:eth0:5072 listen=udp:eth0:5074 Now I have 45 threads and 45 Sql connections. Why does this happen and can i set a a number of threads and/or sql connections that doesn't change with the number of "listen" parameters that i have ? Best Regards, Duarte Rocha

3 2

recording audio stream
by bill＠telopar.net 18 Nov '18

18 Nov '18

Hi All: I have a project that requires that I record the audio streams from an ongoing phone call. It seems that rtp_proxy module should do that, but I can't seem to get it to work. My config file: *if(nat_uac_test("8")) { //rtpproxy_manage("co"); xlog("L_INFO", "rtpproxy Call is going to private IPv4 Media Server Engage RTPProxy Now\n"); rtpproxy_manage("crwie","10.1.1.2"); } else { //rtpproxy_manage("cor"); xlog("L_INFO", "rtpproxy Call is coming from a private IPv4 Media Server Engage RTPProxy Now\n"); rtpproxy_manage("crwei","10.1.1.2"); } *There appears to be conversations going on with rtpproxy on both legs of the call. But I seem to be missing the step that records the audio*. *Anybody have an example of how to do this. TIA, Bill **

4 3

Which algorithm to use as load-balance
by David Villasmil 16 Nov '18

16 Nov '18

Hello, Is it possible to set an algorithm like the following? fs1 - Priority 10 - maxload 20 calls fs2 - Priority 10 - maxload 20 calls fs3 - Priority 9 - maxload 20 calls fs4 - Priority 8 - maxload 20 calls fs5 - Priority 7 - maxload 20 calls So that, whenever any new call comes in, fs1 and fs2 would be used, meaning the fs with the highest priority would be use UNLESS there's 20 calls on that server, in which case, the next server with the highest priority would be used. Thanks!, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 7

Freeradiusclient Interim-Update Support
by Hamid Hashmi 16 Nov '18

16 Nov '18

Dear All Is module acc_radius supports interim updates? If yes then how to configure it. Regards Hamid R. Hashmi

2 3

textops module get_body_part function gives boundary error even if not multipart
by YAS0 CANER 16 Nov '18

16 Nov '18

Hello, i try to get content from Invite request but it gives boundary error. it is not multipart or mixed content but it checks boundary and gives error. Thanks. My message INVITE Content-Type: application/resource-lists+xml <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"> <list> <entry uri="100?Call-ID=$calli&From=$fromtag&To=$toTag&Session-ID=1"/> <entry uri="103?Call-ID=$calli2&From=$fromtag2&To=$toTag2&Session-ID=2"/> </list> </resource-lists> static int ki_get_body_part_helper(sip_msg_t* msg, str* ctype, pv_spec_t *dst, int mode) { ... if(get_boundary(msg, &boundary)!=0) { LM_ERR("Cannot get boundary. Is body multipart?\n"); return -1; } .... int get_boundary(struct sip_msg* msg, str* boundary) params.s = memchr(msg->content_type->body.s, ';',msg->content_type->body.len); if (params.s == NULL) { LM_INFO("Content-Type hdr has no params <%.*s>\n", msg->content_type->body.len, msg->content_type->body.s); return -1; }

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