- sr-users - kamailio.org

Setup dmq_usrloc
by Gertjan Wolzak 17 Dec '18

17 Dec '18

Hello Kamailions, Im trying to setup registration replication with the use of the dmq and dmq_usrlog modules. I have configured two servers, configured kamailio on both. When I register a sip client on the one I get a nice KDMQ message on the second server, but the second server reacts with a 404 not found: U 2018/12/17 05:08:38.974495 10.10.0.1:6000 -> 10.10.0.2:6000 #3 KDMQ sip:usrloc@10.10.0.2:6000 SIP/2.0. Via: SIP/2.0/UDP 10.10.0.1:6000;branch=z9hG4bK00ed.e1666b85000000000000000000000000.0. To: <sip:usrloc@10.10.0.2:6000>. From: <sip:usrloc@10.10.0.1:6000>;tag=a2652bb825a097f3b7285d4c70edd51a-0513. CSeq: 10 KDMQ. Call-ID: 35571d0247178282-3780(a)10.88.77.172. Content-Length: 360. User-Agent: kamailio (5.1.6 (x86_64/linux)). Max-Forwards: 1. Content-Type: application/json. . {"action":2,"aor":"test01(a)10.88.77.172","ruid":"uloc-5c177363-ec4-1","c":"sip:test01@10.88.77.140:59293;ob","received":"","path":"","callid":"zjucvmzUT2hNJCIfqeUXEsOaEzyHCs-4","user_agent":"CSipSimple_G3121-26/r2457","instance":"","expires":1545041786,"cseq":24435,"flags":0,"cflags":0,"q":-1,"last_modified":1545040886,"methods":8159,"reg_id":0,"server_id":0} U 2018/12/17 05:08:38.979062 10.10.0.2:6000 -> 10.10.0.1:6000 #4 SIP/2.0 404 Not Found. Via: SIP/2.0/UDP 10.10.0.1:6000;branch=z9hG4bK00ed.e1666b85000000000000000000000000.0;rport=6000. To: <sip:usrloc@10.10.0.2:6000>;tag=02f7a7b91eadf5e07797231cb2f64326-b794. From: <sip:usrloc@10.10.0.1:6000>;tag=a2652bb825a097f3b7285d4c70edd51a-0513. CSeq: 10 KDMQ. Call-ID: 35571d0247178282-3780(a)10.88.77.172. Server: kamailio (5.1.6 (x86_64/linux)). Content-Length: 0. . I think I have to configure a bit more than just the parameters of the dmq and dqm_usrloc modules. Ive googled, looked in the Sip routing with kamailio book, but no luck. Did not see any examples in the kamailio source, with regards to the dmq_usrloc module. I am not looking for direct answers, have not provided you with enough information for that, but I would appreciate information on how to correctly configure registration replication using the dmq/dmq_usrloc module... Thank you. With regards, Gertjan

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Reminding again about private/direct emails from mailing lists discussions
by Daniel-Constantin Mierla 17 Dec '18

17 Dec '18

I see that still a lot of emails are sent directly to me after a discussion started on public mailing lists (subject with mailing lists prefix, but the sender header is not the mailing list daemon), I would like to remind that, at least in my case, writing emails directly without cc-ing the mailing list is highly not recommended and there is practically 0 chance to get an answer from myself. The rule is also suggested in the mailing lists presentation page: - https://www.kamailio.org/w/mailing-lists/ Unless I asked explicitly for a private email with some specific details, the message will not be noticed in decent time and therefore not replied. I use this email address for many public mailing lists and it is intended only for that usage, I am reacting based on the filters I created, unmatched messages getting the lowest priority. Therefore if you wrote me and expect a response, re-send the message to the appropriate public mailing list. Checking and answering the messages on Kamailio mailing lists are among the top of priorities on my daily activity, so it is high chance you get the response faster when you write there. Note that you can always mask sensitive data (e.g., replace passwords, IP addresses) before sending logs or config file. There are several common sense as well as technical reasons for keeping the discussion on public mailing lists. Here are some along with other suggestions to improve the likeliness of getting an answer: - we are not answering questions on mailing lists to help only one person, but also other people that may have similar issues in the future -- they can find the answers in the archive with a search engine. Also, other people subscribed to the list can learn from conversation or contribute to the discussion, resulting in better outcome - the amount of unexpected messages is very high -- it is practically a zero chance to get to the folder with messages that didn't match any filter (thus your message is not going to be replied) - whenever I find time, the first emails I answer are those coming on mailing lists. They are many there as well, people are traveling or having other personal or business projects, so sometime is good to send a reminder if a question does not get an answer - starting with negative approach or no technical content, like you are going to use something else if you don't get a reply instantly, or this application is missing what so ever tutorial giving exactly the solution you need, or simply asking for full config of complex requirements, is not helping at all, but actually making people to ignore your messages. Start with what you tried and where you got stuck. Everything you get is for free, have appropriate expectation and behavior. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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P-CSCF without Rx interface
by Tsvetan Filev 17 Dec '18

17 Dec '18

Hi All. May be this has been asked before but I was not able to find an answer. I setup IMS with 3 instances of kamailio: P-CSCF, I-CSCF and S-CSCF (I got the config files from here https://github.com/kamailio/kamailio/tree/master/misc/examples/ims). I would like to register two clients to the IMS and make a VoLTE like call. So I want to setup my P-CSCF to send SIP messages to my I-CSCF without Rx interface (without using PCRF). I'm not able to find how to tell P-CSCF where to direct the sip traffic. It obviously should happen in route[REGISTER]. Right now P-CSCF sends all REGISTER requests to itself and goes to a loop. Regards.

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Sending to an announcement server from a failure_route
by Boden Garman 17 Dec '18

17 Dec '18

Hi list, I'm trying to play an announcement for my users when my PSTN gateway responds with a 404 Not Found (Reason: Q.850;cause=3), indicating a wrong number. I'm a bit lost and would appreciate any advice. What I am seeing is two parallel SIP invites, one to my announcement server (hooray!) and one to the PSTN gateway again (which I don't want). How do I stop the original branch from starting again? I've tried changing the order of rewritehostport and append_branch without any difference. I've also tried to use t_relay() instead of the RELAY route. I'm using Kamailio 5.1 and have modified the deafult kamailio.cfg Thanks! Boden I think this is the relevant parts of the config: # PSTN GW routing route[PSTN] { [PSTN regex logic omitted] # classify as an outbound call set_dlg_profile("outbound_calls"); # select a PSTN gateway from the dispatcher if (!ds_select_dst("66", "8")) { send_reply("404", "No destination"); exit; } # failover if the selected PSTN gateway is down t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Try next destinations in failure route failure_route[RTF_DISPATCH] { xlog("inside RTF_DISPATCH"); if (t_is_canceled()) { exit; } # wrong number from PSTN if (t_check_status("404")) { append_branch(); append_hf("X-Application: announcement\r\n"); append_hf("X-Announcement-Playback: wrong-number\r\n"); route(TO_ANNOUNCEMENT); exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { xlog("selecting next destination"); if (ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } # Routing to announcement server route[TO_ANNOUNCEMENT] { if (!is_method("INVITE")) { return; } rewritehostport('ANNOUNCEMENT_SERVER:5060'); route(RELAY); exit; }

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Problem with selection Transport Protocol - MultiHomed Problem
by YAS0 CANER 17 Dec '18

17 Dec '18

Hello , i have trouble about selection transport protocol. i have tried so many thing but couldnt find TM selected which IPv? and protocol type (TCP or UDP) For example; i have 2 interface ipv6 and ipv4. i will route my message (INVITE-Register etc.) another node that use TCP and IPV6 , domain name is test1.test. TM module or some module resolve this domain to IPV6:UDP but i want to sent it over IPV6:TCP socket. so i cannot know which protocol is resolved and cannot route the right socket. $du variable show only sip:ali@test1.test Conclusion : There isn't any pseudo value about which destination of IP version will use. Thanks Flow: UAC (IPv4:UDP) ----------> Kamailio (MultiHomed) ----> (TCP:IPv6)Kamailio ---- Kamailo (multiHomed ) cfg ------ auto_bind_ipv6=1 dns_try_ipv6=yes dns_retr_time=1 dns_retr_no=1 dns_cache_flags=4 mhomed=1 fork=yes tcp_source_ipv6 = [fd61:c23c:b3b6::5] listen=tcp:[fd61:c23c:b3b2::5]:5060 listen=udp:[fd61:c23c:b3b2::5]:5060 listen=udp:192.168.1.107:5060 listen=tcp:192.168.1.107:5060 route []{ if($dP=="TCP"){ xlog("L_INFO","[route] [TCP] 1\n"); if (!t_relay_to_tcp()) { xlog("L_ERR", "Error t_relay [route] \n"); t_reply("500","Server - ERROR "); exit; } } else if($dP=="TCP"){ xlog("L_INFO","[route] [TCP] 2\n"); if (!t_relay_to_tcp()) { xlog("L_ERR", "Error t_relay [route] \n"); t_reply("500","Server - ERROR "); exit; } }else if($dP=="UDP"){ xlog("L_INFO","[route] [UDP] 1\n"); if (!t_relay_to_udp()) { xlog("L_ERR", "Error t_relay [route] \n"); t_reply("500","Server - ERROR "); exit; } }else{ xlog("L_INFO","[route] [??] 1\n"); if (!t_relay()) { xlog("L_ERR", "Error t_relay [route] \n"); t_reply("500","Server - ERROR "); exit; } } }

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Re: [SR-Users] sr-users Digest, Vol 163, Issue 15
by Toufic Mobarak 17 Dec '18

17 Dec '18

Tnhxtx Regards, Toufic Mobarak > On Dec 15, 2018, at 6:00 AM, sr-users-request(a)lists.kamailio.org wrote: > > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: Routing of Audio and Video Stream. (Daniel-Constantin Mierla) > 2. Problem with selection Transport Protocol - MultiHomed > Problem (YAS0 CANER) > 3. Re: Problem with selection Transport Protocol - MultiHomed > Problem (YAS0 CANER) > 4. Re: Routing of Audio and Video Stream. (Sergiu Pojoga) > 5. Re: Problem with selection Transport Protocol - MultiHomed > Problem (Daniel-Constantin Mierla) > 6. Issue on kamailio 4.1.9 of database reconnection (Emmanuel BUU) > 7. Re: Routing of Audio and Video Stream. (Abhijit Dutta) > 8. Re: Routing of Audio and Video Stream. (Abhijit Dutta) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 14 Dec 2018 12:54:44 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.kamailio.org>, Abhijit Dutta <duttaa(a)hotmail.com> > Subject: Re: [SR-Users] Routing of Audio and Video Stream. > Message-ID: <26c51f12-5242-4794-5124-f37bc523c7ae(a)gmail.com> > Content-Type: text/plain; charset="windows-1252" > > Hello, > > > I guess you still need to send the INVITE to Asterisk, right? > > > Cheers, > Daniel > > >> On 14.12.18 11:15, Abhijit Dutta wrote: >> Hi All, >> >> Is it possible to redirect only the audio stream towards Asterisk from >> Kamailio (Proxy) via RTPEngine and not to send the video stream to >> Asterisk. I am yet to get the complete reason why my client wants this >> to happen though. >> >> Thanks >> Abhijit >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users(a)lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > -- > 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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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.

3 9

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

2 8

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

4 8

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

2 2

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)

1 1

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; }

1 0

signal 15 received and .. on Kamailio 5.1.6
by AL RSM 16 Nov '18

16 Nov '18

Hi, I just installed Kamailio 5.1.6 on Ubuntu 18.04. Starting Kamailio with the default config will give us the following in the log. Is this normal? Is there a way to fix this issue? Thanks, AL Nov 15 14:01:59 lab-proxy03 kamailio[13544]: * Stopping Kamailio SIP Server: kamailio Nov 15 14:01:59 lab-proxy03 /usr/local/sbin/kamailio[5182]: NOTICE: <core> [main.c:692]: handle_sigs(): Thank you for flying kamailio!!! Nov 15 14:01:59 lab-proxy03 /usr/local/sbin/kamailio[5213]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received Nov 15 14:01:59 lab-proxy03 /usr/local/sbin/kamailio[5212]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received .... Nov 15 14:01:59 lab-proxy03 /usr/local/sbin/kamailio[5182]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Nov 15 14:02:00 lab-proxy03 kamailio[13544]: ...done. Nov 15 14:02:00 lab-proxy03 kamailio[13556]: * Starting Kamailio SIP Server: kamailio Nov 15 14:02:00 lab-proxy03 kamailio: INFO: <core> [core/sctp_core.c:74]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module Nov 15 14:02:00 lab-proxy03 kamailio: INFO: <core> [core/tcp_main.c:4721]: init_tcp(): using epoll_lt as the io watch method (auto detected) .... Nov 15 14:02:00 lab-proxy03 /usr/local/sbin/kamailio[13596]: INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded Nov 15 14:02:00 lab-proxy03 /usr/local/sbin/kamailio[13596]: INFO: rr [rr_mod.c:175]: mod_init(): outbound module not available Nov 15 14:02:00 lab-proxy03 /usr/local/sbin/kamailio[13596]: INFO: <core> [core/udp_server.c:153]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Nov 15 14:02:00 lab-proxy03 /usr/local/sbin/kamailio[13596]: INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Nov 15 14:02:00 lab-proxy03 /usr/local/sbin/kamailio[13596]: INFO: <core> [core/udp_server.c:153]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Nov 15 14:02:00 lab-proxy03 /usr/local/sbin/kamailio[13596]: INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Nov 15 14:02:00 lab-proxy03 /usr/local/sbin/kamailio[13618]: INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/13618 Nov 15 14:02:00 lab-proxy03 /usr/local/sbin/kamailio[13619]: INFO: ctl [io_listener.c:214]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) Nov 15 14:02:00 lab-proxy03 kamailio[13556]: ...done.

2 1

priority load balance
by David Villasmil 15 Nov '18

15 Nov '18

Hello, Trying to do priority load balancing (8) but i'm getting an error `unable to update destination load - classic dispatching` anyone knows why? i've got ```modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("dispatcher", "attrs_avp", "$avp(dsattrs)")``` Looking at the code, dstid_avp would be null, but not sure why? I have set in the table the priorities as 10 for 2 gws (edited) Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 1

"signal 15 received" and ... on Kamailio 5.1.6
by AL RSM 15 Nov '18

15 Nov '18

Hi, I just installed Kamailio 5.1.6 on Ubuntu 18.04. Starting Kamailio with the default config will give us the following in the log. Is this normal? Is there a way to fix this issue? Thanks, AL Nov 15 14:01:59 lab-proxy03 kamailio[13544]: * Stopping Kamailio SIP Server: kamailio Nov 15 14:01:59 lab-proxy03 /usr/local/sbin/kamailio[5182]: NOTICE: <core> [main.c:692]: handle_sigs(): Thank you for flying kamailio!!! Nov 15 14:01:59 lab-proxy03 /usr/local/sbin/kamailio[5213]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received Nov 15 14:01:59 lab-proxy03 /usr/local/sbin/kamailio[5212]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received .... Nov 15 14:01:59 lab-proxy03 /usr/local/sbin/kamailio[5182]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Nov 15 14:02:00 lab-proxy03 kamailio[13544]: ...done. Nov 15 14:02:00 lab-proxy03 kamailio[13556]: * Starting Kamailio SIP Server: kamailio Nov 15 14:02:00 lab-proxy03 kamailio: INFO: <core> [core/sctp_core.c:74]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module Nov 15 14:02:00 lab-proxy03 kamailio: INFO: <core> [core/tcp_main.c:4721]: init_tcp(): using epoll_lt as the io watch method (auto detected) .... Nov 15 14:02:00 lab-proxy03 /usr/local/sbin/kamailio[13596]: INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded Nov 15 14:02:00 lab-proxy03 /usr/local/sbin/kamailio[13596]: INFO: rr [rr_mod.c:175]: mod_init(): outbound module not available Nov 15 14:02:00 lab-proxy03 /usr/local/sbin/kamailio[13596]: INFO: <core> [core/udp_server.c:153]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Nov 15 14:02:00 lab-proxy03 /usr/local/sbin/kamailio[13596]: INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Nov 15 14:02:00 lab-proxy03 /usr/local/sbin/kamailio[13596]: INFO: <core> [core/udp_server.c:153]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Nov 15 14:02:00 lab-proxy03 /usr/local/sbin/kamailio[13596]: INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Nov 15 14:02:00 lab-proxy03 /usr/local/sbin/kamailio[13618]: INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/13618 Nov 15 14:02:00 lab-proxy03 /usr/local/sbin/kamailio[13619]: INFO: ctl [io_listener.c:214]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) Nov 15 14:02:00 lab-proxy03 kamailio[13556]: ...done.

1 0

REFER packet - Kamailio gives request_uri instead request-uri -
by YAS0 CANER 15 Nov '18

15 Nov '18

Hello, i test on refer packet on Kamailio 5.3-dev . it gives route_uri (@route_uri) instead of request-uri ($ru). Could you give me help about it please. Thanks. Packet 2018/11/15 10:51:20.522512 192.168.1.109:5080 -> 192.168.1.120:5060 REFER sip:100@192.168.1.120 SIP/2.0 CSeq: 1 REFER From: <sip:105@192.168.1.120;transport=UDP>;tag=10215SIPpTag001 To: <sip:100@192.168.1.120;transport=UDP>;tag=7e3fb504 Call-ID: 1-10215(a)192.168.1.109 Max-Forwards: 70 Route: <sip:192.168.1.120;lr> Via: SIP/2.0/UDP 192.168.1.136:5080;branch=z9hG4bK-10215-1-7 Contact: <sip:105@192.168.1.136:5080> Refer-To: <sip:103@192.168.1.120> Referred-By: <sip:105@192.168.1.120;transport=UDP> Content-Length: 0 1(48532) WARNING: {1 1 REFER 1-10215(a)192.168.1.109} <script>: REFER in WITHINDLG [$ru sip:192.168.1.120;lr ] 1(48532) WARNING: {1 1 REFER 1-10215(a)192.168.1.109} <script>: REFER MYROUTE [ru sip:192.168.1.120;lr ] [route_uri sip:192.168.1.120;lr ] [ rU <null> ] Scenairo SIPP UAC -> Kamailio - > UAC 2 After call answered , SIPP send REFER request as above.

1 2

mod siptrace - option ping from nathelper do not get duplicated
by Roman Dissauer 14 Nov '18

14 Nov '18

Hi, we are using kamailio 5.1 with siptrace module in trace_mode on. Everything works perfect except that option pings from nathelper module do not get duplicated. Is there a reason for that? Thanks, Roman

2 1

Core dump on diameter CCR request
by Venkatesh Macha 14 Nov '18

14 Nov '18

Hi all, I am trying to implement Diameter charging with Kamailio. I am following below tutorial https://www.kamailio.org/events/2016-KamailioWorld/Day2/15-Carsten.Bock-Dia… But whenever I try to do the Ro_CCR from the INVITE block, Kamailio is crashing and generating the core dump. I am new to diameter. I might be doing something wrong here. please someone help me here. Please find the *diameter.xml* and *Diameter_core_dump_full_debug.txt* file. diameter.xml <http://sip-router.1086192.n5.nabble.com/file/t6374/diameter.xml> Diameter_core_dump_full_Debug.txt <http://sip-router.1086192.n5.nabble.com/file/t6374/Diameter_core_dump_full_…> Here is my kamailio.cfg : loadmodule "cdp" loadmodule "cdp_avp" loadmodule "ims_charging" loadmodule "ims_dialog" modparam("cdp","config_file","/etc/kamailio/diameter.xml") modparam("cdp", "debug_heavy", 1) # ----- ims_diameter_ro params ----- #modparam("ims_charging", "single_ro_session_per_dialog", 1) modparam("ims_charging", "origin_host", "10.195.0.38"); modparam("ims_charging", "origin_realm", "10.195.0.38"); modparam("ims_charging", "destination_realm", "mydomain.com"); modparam("ims_charging","interim_update_credits",30); modparam("ims_charging","timer_buffer",5); # -- ims_dialog params -- modparam("ims_dialog", "dlg_flag", 8) While starting the kamailio, i am getting following errors: Nov 11 11:40:03 ip-10-195-0-38 /usr/local/sbin/kamailio[25653]: ERROR: cdp [peerstatemachine.c:675]: add_peer_application(): Too many applications for this peer (max 0), not adding Application 4:0. Before crashing kamailio showing following error: Nov 11 11:39:49 ip-10-195-0-38 /usr/local/sbin/kamailio[25571]: INFO: {1 18325 INVITE Nf6-dl2WeXF.LJhXotiRsuMhijsNx45o} <script>: Entered into INVITE Block Nov 11 11:39:49 ip-10-195-0-38 /usr/local/sbin/kamailio[25571]: ERROR: {1 18325 INVITE Nf6-dl2WeXF.LJhXotiRsuMhijsNx45o} ims_dialog [dlg_handlers.c:264]: dlg_get_iuid_shm_clone(): ***** Requesting 8 Nov 11 11:39:49 ip-10-195-0-38 /usr/local/sbin/kamailio[25571]: CRITICAL: {1 18325 INVITE Nf6-dl2WeXF.LJhXotiRsuMhijsNx45o} <core> [core/mem/q_malloc.c:144]: qm_debug_check_frag(): BUG: qm: prev. fragm. tail overwritten(0, abcdefed)[0x7f467d438768:0x7f467d4387a0]! Memory allocator was called from cdp: diameter_avp.c:157. Fragment marked by cdp: peerstatemachine.c:602. Exec from core/mem/q_malloc.c:384. Nov 11 11:39:49 ip-10-195-0-38 /usr/local/sbin/kamailio[25599]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 16 Nov 11 11:39:52 ip-10-195-0-38 /usr/local/sbin/kamailio[25567]: ALERT: <core> [main.c:740]: handle_sigs(): child process 25571 exited by a signal 6 Nov 11 11:39:52 ip-10-195-0-38 /usr/local/sbin/kamailio[25567]: ALERT: <core> [main.c:743]: handle_sigs(): core was generated Nov 11 11:39:52 ip-10-195-0-38 /usr/local/sbin/kamailio[25567]: INFO: <core> [main.c:766]: handle_sigs(): terminating due to SIGCHLD Thanks you in Advance, Venkey -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

4 9

WebRTC video call terminated after ~30 seconds
by Mathias Brodala - Pagemachine AG 14 Nov '18

14 Nov '18

Hi, I am currently testing a setup to do video calls via WebRTC and have set up Kamailio as SIP server. So far the calls are working fine, but after around 30 seconds the calls are terminated. I can reproduce this easily both with the JsSIP demo [0] as well as the SIPjs demo [1]. The browser console using JsSIP log [2] shows me this: > JsSIP:WebSocketInterface WebSocket wss://sip.example.org closed +60s > jssip-3.2.10.js:23490 JsSIP:WebSocketInterface WebSocket abrupt disconnection +1ms Please also have a look at my kamailio.cfg [3]. I tried to follow the docs of the websocket module [4] as close as possible to get everything up and running. Kamailio is currently running on a private DigitalOcean droplet and I already tried to disable all firewall rules without change. I also asked the DO staff if something could lead to UDP packets getting dropped after some time but I was told that there is no such logic in the DO network. Please let me know if you can find anything unusual which could cause this behavior or if you need more information. Thank you, Mathias Brodala [0] https://tryit.jssip.net/ [1] https://sipjs.com/demo-phone/ [2] https://gist.github.com/mbrodala/0f81aaa174be87096cf20502633f3ac3 [3] https://gist.github.com/mbrodala/066faa455da46362dab3dd5326ba8ad1 [4] https://www.kamailio.org/docs/modules/5.1.x/modules/websocket.html -- Dipl.-Inf. (FH) Mathias Brodala Tel.: 069.260.99.70.45 Fax: .31 mbrodala(a)pagemachine.de <https://www.pagemachine.de/> Pagemachine AG Solmsstraße 6a, 60486 Frankfurt am Main <https://maps.google.com/?q=Solmsstra%C3%9Fe+6a,+60486+Frankfurt+am+Main&ent…> Vorstand: Volker Neuenhaus, Stefan Schütt, Miklos Weiszhaupt Vorsitzender des Aufsichtsrats: Holger Kleinschmidt Registergericht: Amtsgericht Frankfurt Registernummer: HRB 56436 Umsatzsteuer-ID: DE226286966 <https://www.kununu.com/de/pagemachine> <https://www.kununu.com/de/pagemachine> <http://www.frankfurt-main.ihk.de/> <https://typo3.com/>

2 1

topos without db?
by Giuseppe Pandolfi 13 Nov '18

13 Nov '18

Hi, Is it possible using topos module without db support? I'm trying to use topos module with sqlite db backend, but I have any concurrency problems during write procedure (I use WALL pragma). When execute speed calls traffic sqlite emits an error trace telling it's locked. Is it possible to use a sort of hash tabel to save topos informations and don't use the db? Best regards, Giuseppe

5 4

variable $rb not updated on failure route after a 302 on the same PID
by David Escartín 13 Nov '18

13 Nov '18

Hello all we are seeing what we think is quite strange behaviour We have expecienced this on 5.0 and 5.1 version. this is the call flow INV--> <-- 302 (SDP) INV--> <-- 503 (SDP) (or another 302) When we receive a 302 and we load contacts and create a new branch, when we receive the second error response, and we receive it on the same udp receiver process, if we read the content of $rb on the the onreply_route, we get the proper value of the response SDP, but when we read $T_rpl($rb) on the failure_route, we get the value of the first error transaction. If the response (second) is received in another PID, it's ok. I'm not sure if i'm explaining myself, but we would think that the value should be updated, no matter which PID is used to process the response for example, running the kamailio with only 1 child. For the first 302 we have onreply route checking $rb 1(7698) INFO: <script>: LAL {"sdp_data":"3466666666","sdp_data_2":null,"nrn":null} failure_route checking $T_rpl($rb) 1(7698) INFO: <script>: LOOOOOOOOL {"sdp_data":"3466666666","sdp_data_2":null,"nrn":null} and for the next error 1(7698) INFO: <script>: LAL {"sdp_data":"34619796636","sdp_data_2":null,"nrn":null} 1(7698) INFO: <script>: LOOOOOOOOL {"sdp_data":"3466666666","sdp_data_2":null,"nrn":null} If we run it with 16 children and the PID don't match. For the first 302 we have onreply route checking $rb 13(8360) INFO: <script>: LAL {"sdp_data":"3466666666","sdp_data_2":null,"nrn":null} failure_route checking $T_rpl($rb) 13(8360) INFO: <script>: LOOOOOOOOL {"sdp_data":"3466666666","sdp_data_2":null,"nrn":null} and for the next error after loading the contacts 15(8362) INFO: <script>: LAL {"sdp_data":"34619796636","sdp_data_2":null,"nrn":null} 15(8362) INFO: <script>: LOOOOOOOOL {"sdp_data":"34619796636","sdp_data_2":null,"nrn":null} i attach you the logs in both cases could you please check if you see anything wrong? best regards david 16_children.log 11(8358) DEBUG: <core> [core/udp_server.c:492]: udp_rcv_loop(): received on udp socket: (106/100/634) [[INVITE sip:34606147628@10.100.10.67:5069 SIP/2.0 0D 0A Via: SIP/2.0/UDP 10.100.10.67:5061;branch=z9hG4bK-]] 11(8358) DEBUG: <core> [core/parser/msg_parser.c:604]: parse_msg(): SIP Request: 11(8358) DEBUG: <core> [core/parser/msg_parser.c:606]: parse_msg(): method: <INVITE> 11(8358) DEBUG: <core> [core/parser/msg_parser.c:608]: parse_msg(): uri: <sip:34606147628@10.100.10.67:5069> 11(8358) DEBUG: <core> [core/parser/msg_parser.c:610]: parse_msg(): version: <SIP/2.0> 11(8358) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-9680-1-0>; state=16 11(8358) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 11(8358) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=2 11(8358) DEBUG: <core> [core/parser/msg_parser.c:494]: parse_headers(): this is the first via 11(8358) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=10 11(8358) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [44]; uri=[sip:34606147628@10.100.10.67:5069] 11(8358) DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): to body ["Javi" <sip:34606147628@10.100.10.67:5069> ] 11(8358) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <1> <INVITE> 11(8358) DEBUG: <core> [core/receive.c:205]: receive_msg(): --- received sip message - request - call-id: [1-9680(a)10.100.10.67] - cseq: [1 INVITE] 11(8358) DEBUG: <core> [core/receive.c:248]: receive_msg(): preparing to run routing scripts... 11(8358) INFO: <script>: hereeeeeeeeeee 11(8358) DEBUG: tm [t_lookup.c:1320]: t_newtran(): msg id=1 , global msg id=0 , T on entrance=0xffffffffffffffff 11(8358) DEBUG: <core> [core/parser/msg_parser.c:183]: get_hdr_field(): content_length=217 11(8358) DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header 11(8358) DEBUG: tm [t_lookup.c:477]: t_lookup_request(): start searching: hash=13774, isACK=0 11(8358) DEBUG: tm [t_lookup.c:435]: matching_3261(): RFC3261 transaction matching failed 11(8358) DEBUG: tm [t_lookup.c:659]: t_lookup_request(): no transaction found 11(8358) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=9680SIPpTag001 11(8358) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 11(8358) DEBUG: <core> [core/md5utils.c:67]: MD5StringArray(): MD5 calculated: 56115a2d09f62d466f1d45a5f5ff7cc2 11(8358) DEBUG: tm [t_funcs.c:324]: t_relay_to(): new INVITE 11(8358) DEBUG: <core> [core/msg_translator.c:159]: check_via_address(): (10.100.10.67, 10.100.10.67, 0) 11(8358) DEBUG: tm [t_reply.c:596]: _reply_light(): reply sent out. buf=0x7f334b5f4ea0: SIP/2.0 100 trying -..., shmem=0x7f3346638240: SIP/2.0 100 trying - 11(8358) DEBUG: tm [t_reply.c:606]: _reply_light(): finished 11(8358) INFO: <script>: LEL 11(8358) DEBUG: <core> [core/msg_translator.c:159]: check_via_address(): (10.100.10.67, 10.100.10.67, 0) 11(8358) INFO: <script>: hereeeeeeeeeee 11(8358) DEBUG: tm [t_funcs.c:372]: t_relay_to(): new transaction forwarded 11(8358) DEBUG: <core> [core/receive.c:289]: receive_msg(): request-route executed in: 550 usec 11(8358) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 11(8358) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 11(8358) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 11(8358) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 11(8358) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 11(8358) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 11(8358) DEBUG: <core> [core/xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) 11(8358) DEBUG: <core> [core/receive.c:378]: receive_msg(): cleaning up 11(8358) DEBUG: <core> [core/udp_server.c:492]: udp_rcv_loop(): received on udp socket: (106/100/397) [[SIP/2.0 100 Trying 0D 0A Via: SIP/2.0/UDP 52.88.180.226:5069;branch=z9hG4bKec53.56115a2d09f62d466f1d45a5f]] 11(8358) DEBUG: <core> [core/parser/msg_parser.c:614]: parse_msg(): SIP Reply (status): 11(8358) DEBUG: <core> [core/parser/msg_parser.c:616]: parse_msg(): version: <SIP/2.0> 11(8358) DEBUG: <core> [core/parser/msg_parser.c:618]: parse_msg(): status: <100> 11(8358) DEBUG: <core> [core/parser/msg_parser.c:620]: parse_msg(): reason: <Trying> 11(8358) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKec53.56115a2d09f62d466f1d45a5f5ff7cc2.0>; state=9 11(8358) DEBUG: <core> [core/parser/parse_via.c:2681]: parse_via(): parsing via: next via 11(8358) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-9680-1-0>; state=16 11(8358) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 11(8358) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=2 11(8358) DEBUG: <core> [core/parser/msg_parser.c:494]: parse_headers(): this is the first via 11(8358) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=10 11(8358) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [44]; uri=[sip:34606147628@10.100.10.67:5069] 11(8358) DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): to body ["Javi" <sip:34606147628@10.100.10.67:5069> ] 11(8358) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <1> <INVITE> 11(8358) DEBUG: <core> [core/receive.c:205]: receive_msg(): --- received sip message - reply - call-id: [1-9680(a)10.100.10.67] - cseq: [1 INVITE] 11(8358) DEBUG: tm [t_lookup.c:1019]: t_check_msg(): msg (0x7f334b5f3768) id=2 global id=1 T start=0xffffffffffffffff 13(8360) DEBUG: <core> [core/udp_server.c:492]: udp_rcv_loop(): received on udp socket: (106/100/479) [[SIP/2.0 302 Moved Temporarily 0D 0A Via: SIP/2.0/UDP 52.88.180.226:5069;branch=z9hG4bKec53.56115a2d09f62d]] 11(8358) DEBUG: tm [t_lookup.c:897]: t_reply_matching(): t_reply_matching: hash 13774 label 0 branch 0 11(8358) DEBUG: tm [t_lookup.c:951]: t_reply_matching(): reply (0x7f334b5f3768) matched an active transaction (T=0x7f334663b040)! 11(8358) DEBUG: tm [t_lookup.c:1088]: t_check_msg(): msg (0x7f334b5f3768) id=2 global id=2 T end=0x7f334663b040 11(8358) DEBUG: tm [t_reply.c:2209]: reply_received(): org. status uas=100, uac[0]=0 local=0 is_invite=1) 11(8358) DEBUG: <core> [core/parser/msg_parser.c:183]: get_hdr_field(): content_length=0 11(8358) DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header 11(8358) INFO: <script>: LAL 11(8358) DEBUG: tm [t_reply.c:1265]: t_should_relay_response(): ->>>>>>>>> T_code=100, new_code=100 11(8358) DEBUG: tm [t_reply.c:1788]: relay_reply(): branch=0, save=0, relay=-1 icode=0 11(8358) DEBUG: <core> [core/receive.c:362]: receive_msg(): reply-route executed in: 149 usec 11(8358) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 11(8358) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 11(8358) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 11(8358) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 11(8358) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 11(8358) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 11(8358) DEBUG: <core> [core/xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) 11(8358) DEBUG: <core> [core/receive.c:378]: receive_msg(): cleaning up 13(8360) DEBUG: <core> [core/parser/msg_parser.c:614]: parse_msg(): SIP Reply (status): 13(8360) DEBUG: <core> [core/parser/msg_parser.c:616]: parse_msg(): version: <SIP/2.0> 13(8360) DEBUG: <core> [core/parser/msg_parser.c:618]: parse_msg(): status: <302> 13(8360) DEBUG: <core> [core/parser/msg_parser.c:620]: parse_msg(): reason: <Moved Temporarily> 13(8360) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKec53.56115a2d09f62d466f1d45a5f5ff7cc2.0>; state=9 13(8360) DEBUG: <core> [core/parser/parse_via.c:2681]: parse_via(): parsing via: next via 13(8360) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-9680-1-0>; state=16 13(8360) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 13(8360) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=2 13(8360) DEBUG: <core> [core/parser/msg_parser.c:494]: parse_headers(): this is the first via 13(8360) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=5 13(8360) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 13(8360) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [50]; uri=[sip:34606147628@10.100.10.67:5069] 13(8360) DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): to body ["Javi" <sip:34606147628@10.100.10.67:5069>] 13(8360) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <1> <INVITE> 13(8360) DEBUG: <core> [core/receive.c:205]: receive_msg(): --- received sip message - reply - call-id: [1-9680(a)10.100.10.67] - cseq: [1 INVITE] 13(8360) DEBUG: tm [t_lookup.c:1019]: t_check_msg(): msg (0x7f334b5f3768) id=1 global id=0 T start=0xffffffffffffffff 13(8360) DEBUG: tm [t_lookup.c:897]: t_reply_matching(): t_reply_matching: hash 13774 label 0 branch 0 13(8360) DEBUG: tm [t_lookup.c:951]: t_reply_matching(): reply (0x7f334b5f3768) matched an active transaction (T=0x7f334663b040)! 13(8360) DEBUG: tm [t_lookup.c:1088]: t_check_msg(): msg (0x7f334b5f3768) id=1 global id=1 T end=0x7f334663b040 13(8360) DEBUG: tm [t_reply.c:2209]: reply_received(): org. status uas=100, uac[0]=100 local=0 is_invite=1) 13(8360) DEBUG: <core> [core/parser/msg_parser.c:183]: get_hdr_field(): content_length=63 13(8360) DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header 13(8360) INFO: <script>: LAL {"sdp_data":"3466666666","sdp_data_2":null,"nrn":null} 13(8360) DEBUG: tm [t_reply.c:1265]: t_should_relay_response(): ->>>>>>>>> T_code=100, new_code=302 13(8360) DEBUG: <core> [core/parser/msg_parser.c:614]: parse_msg(): SIP Reply (status): 13(8360) DEBUG: <core> [core/parser/msg_parser.c:616]: parse_msg(): version: <SIP/2.0> 13(8360) DEBUG: <core> [core/parser/msg_parser.c:618]: parse_msg(): status: <302> 13(8360) DEBUG: <core> [core/parser/msg_parser.c:620]: parse_msg(): reason: <Moved Temporarily> 13(8360) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKec53.56115a2d09f62d466f1d45a5f5ff7cc2.0>; state=9 13(8360) DEBUG: <core> [core/parser/parse_via.c:2681]: parse_via(): parsing via: next via 13(8360) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-9680-1-0>; state=16 13(8360) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 13(8360) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=2 13(8360) DEBUG: <core> [core/parser/msg_parser.c:494]: parse_headers(): this is the first via 13(8360) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=5 13(8360) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 13(8360) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [50]; uri=[sip:34606147628@10.100.10.67:5069] 13(8360) DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): to body ["Javi" <sip:34606147628@10.100.10.67:5069>] 13(8360) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <1> <INVITE> 13(8360) DEBUG: <core> [core/parser/msg_parser.c:183]: get_hdr_field(): content_length=63 13(8360) DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header 13(8360) INFO: <script>: LOOOOOOOOL {"sdp_data":"3466666666","sdp_data_2":null,"nrn":null} 13(8360) DEBUG: tm [t_lookup.c:1019]: t_check_msg(): msg (0x7f3346637088) id=0 global id=0 T start=0x7f334663b040 13(8360) DEBUG: tm [t_lookup.c:1091]: t_check_msg(): T (0x7f334663b040) already found for msg (0x7f3346637088)! 13(8360) DEBUG: tmx [t_var.c:534]: pv_get_tm_reply_code(): reply code is <302> 13(8360) INFO: <script>: Call: 302 received, trying to load contacts 13(8360) DEBUG: uac_redirect [rd_funcs.c:72]: get_redirect(): resume branch=0 13(8360) DEBUG: uac_redirect [rd_funcs.c:79]: get_redirect(): checking branch=0 (added=0) 13(8360) DEBUG: uac_redirect [rd_funcs.c:83]: get_redirect(): branch=0 is a redirect (added=0) 13(8360) DEBUG: uac_redirect [rd_funcs.c:159]: sort_contacts(): sort_contacts: <sip:34666988297@10.100.10.67:5060> q=10 13(8360) DEBUG: uac_redirect [rd_funcs.c:295]: shmcontact2dset(): adding contact <sip:34666988297@10.100.10.67:5060> 13(8360) DEBUG: <core> [sruid.c:183]: sruid_next(): new sruid is [rdir-5be4236b-20a8-1] (1 / 20) 13(8360) DEBUG: tm [t_serial.c:191]: ki_t_load_contacts(): nr_branches is 1 13(8360) DEBUG: tm [t_serial.c:194]: ki_t_load_contacts(): nothing to do - only one contact! 13(8360) INFO: <script>: LEL 13(8360) DEBUG: <core> [core/msg_translator.c:159]: check_via_address(): (10.100.10.67, 10.100.10.67, 0) 13(8360) INFO: <script>: hereeeeeeeeeee 13(8360) DEBUG: tm [t_reply.c:1788]: relay_reply(): branch=0, save=0, relay=-1 icode=0 13(8360) DEBUG: <core> [core/receive.c:362]: receive_msg(): reply-route executed in: 783 usec 13(8360) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 13(8360) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 13(8360) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 13(8360) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 13(8360) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 13(8360) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 13(8360) DEBUG: <core> [core/xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) 13(8360) DEBUG: <core> [core/receive.c:378]: receive_msg(): cleaning up 14(8361) DEBUG: <core> [core/udp_server.c:492]: udp_rcv_loop(): received on udp socket: (106/100/397) [[SIP/2.0 100 Trying 0D 0A Via: SIP/2.0/UDP 52.88.180.226:5069;branch=z9hG4bKec53.56115a2d09f62d466f1d45a5f]] 15(8362) DEBUG: <core> [core/udp_server.c:492]: udp_rcv_loop(): received on udp socket: (106/100/482) [[SIP/2.0 503 Service Unavailable 0D 0A Via: SIP/2.0/UDP 52.88.180.226:5069;branch=z9hG4bKec53.56115a2d09f6]] 14(8361) DEBUG: <core> [core/parser/msg_parser.c:614]: parse_msg(): SIP Reply (status): 14(8361) DEBUG: <core> [core/parser/msg_parser.c:616]: parse_msg(): version: <SIP/2.0> 14(8361) DEBUG: <core> [core/parser/msg_parser.c:618]: parse_msg(): status: <100> 14(8361) DEBUG: <core> [core/parser/msg_parser.c:620]: parse_msg(): reason: <Trying> 15(8362) DEBUG: <core> [core/parser/msg_parser.c:614]: parse_msg(): SIP Reply (status): 15(8362) DEBUG: <core> [core/parser/msg_parser.c:616]: parse_msg(): version: <SIP/2.0> 15(8362) DEBUG: <core> [core/parser/msg_parser.c:618]: parse_msg(): status: <503> 15(8362) DEBUG: <core> [core/parser/msg_parser.c:620]: parse_msg(): reason: <Service Unavailable> 14(8361) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKec53.56115a2d09f62d466f1d45a5f5ff7cc2.1>; state=9 14(8361) DEBUG: <core> [core/parser/parse_via.c:2681]: parse_via(): parsing via: next via 14(8361) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-9680-1-0>; state=16 14(8361) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 14(8361) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=2 14(8361) DEBUG: <core> [core/parser/msg_parser.c:494]: parse_headers(): this is the first via 15(8362) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKec53.56115a2d09f62d466f1d45a5f5ff7cc2.1>; state=9 15(8362) DEBUG: <core> [core/parser/parse_via.c:2681]: parse_via(): parsing via: next via 15(8362) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-9680-1-0>; state=16 15(8362) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 14(8361) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=10 15(8362) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=2 14(8361) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [44]; uri=[sip:34606147628@10.100.10.67:5069] 15(8362) DEBUG: <core> [core/parser/msg_parser.c:494]: parse_headers(): this is the first via 14(8361) DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): to body ["Javi" <sip:34606147628@10.100.10.67:5069> ] 14(8361) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <1> <INVITE> 14(8361) DEBUG: <core> [core/receive.c:205]: receive_msg(): --- received sip message - reply - call-id: [1-9680(a)10.100.10.67] - cseq: [1 INVITE] 15(8362) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=5 15(8362) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 15(8362) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [50]; uri=[sip:34606147628@10.100.10.67:5069] 15(8362) DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): to body ["Javi" <sip:34606147628@10.100.10.67:5069>] 14(8361) DEBUG: tm [t_lookup.c:1019]: t_check_msg(): msg (0x7f334b5f3768) id=1 global id=0 T start=0xffffffffffffffff 14(8361) DEBUG: tm [t_lookup.c:897]: t_reply_matching(): t_reply_matching: hash 13774 label 0 branch 1 14(8361) DEBUG: tm [t_lookup.c:951]: t_reply_matching(): reply (0x7f334b5f3768) matched an active transaction (T=0x7f334663b040)! 14(8361) DEBUG: tm [t_lookup.c:1088]: t_check_msg(): msg (0x7f334b5f3768) id=1 global id=1 T end=0x7f334663b040 15(8362) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <1> <INVITE> 14(8361) DEBUG: tm [t_reply.c:2209]: reply_received(): org. status uas=100, uac[1]=0 local=0 is_invite=1) 15(8362) DEBUG: <core> [core/receive.c:205]: receive_msg(): --- received sip message - reply - call-id: [1-9680(a)10.100.10.67] - cseq: [1 INVITE] 15(8362) DEBUG: tm [t_lookup.c:1019]: t_check_msg(): msg (0x7f334b5f3768) id=1 global id=0 T start=0xffffffffffffffff 15(8362) DEBUG: tm [t_lookup.c:897]: t_reply_matching(): t_reply_matching: hash 13774 label 0 branch 1 15(8362) DEBUG: tm [t_lookup.c:951]: t_reply_matching(): reply (0x7f334b5f3768) matched an active transaction (T=0x7f334663b040)! 14(8361) DEBUG: <core> [core/parser/msg_parser.c:183]: get_hdr_field(): content_length=0 15(8362) DEBUG: tm [t_lookup.c:1088]: t_check_msg(): msg (0x7f334b5f3768) id=1 global id=1 T end=0x7f334663b040 15(8362) DEBUG: tm [t_reply.c:2209]: reply_received(): org. status uas=100, uac[1]=100 local=0 is_invite=1) 14(8361) DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header 14(8361) INFO: <script>: LAL 14(8361) DEBUG: tm [t_reply.c:1265]: t_should_relay_response(): ->>>>>>>>> T_code=100, new_code=100 14(8361) DEBUG: tm [t_reply.c:1788]: relay_reply(): branch=1, save=0, relay=-1 icode=0 14(8361) DEBUG: <core> [core/receive.c:362]: receive_msg(): reply-route executed in: 199 usec 14(8361) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 14(8361) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 14(8361) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 14(8361) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 14(8361) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 14(8361) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 14(8361) DEBUG: <core> [core/xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) 14(8361) DEBUG: <core> [core/receive.c:378]: receive_msg(): cleaning up 15(8362) DEBUG: <core> [core/parser/msg_parser.c:183]: get_hdr_field(): content_length=64 15(8362) DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header 15(8362) INFO: <script>: LAL {"sdp_data":"34619796636","sdp_data_2":null,"nrn":null} 15(8362) DEBUG: tm [t_reply.c:1265]: t_should_relay_response(): ->>>>>>>>> T_code=100, new_code=503 15(8362) DEBUG: <core> [core/parser/msg_parser.c:614]: parse_msg(): SIP Reply (status): 15(8362) DEBUG: <core> [core/parser/msg_parser.c:616]: parse_msg(): version: <SIP/2.0> 15(8362) DEBUG: <core> [core/parser/msg_parser.c:618]: parse_msg(): status: <503> 15(8362) DEBUG: <core> [core/parser/msg_parser.c:620]: parse_msg(): reason: <Service Unavailable> 15(8362) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKec53.56115a2d09f62d466f1d45a5f5ff7cc2.1>; state=9 15(8362) DEBUG: <core> [core/parser/parse_via.c:2681]: parse_via(): parsing via: next via 15(8362) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-9680-1-0>; state=16 15(8362) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 15(8362) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=2 15(8362) DEBUG: <core> [core/parser/msg_parser.c:494]: parse_headers(): this is the first via 15(8362) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=5 15(8362) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 15(8362) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [50]; uri=[sip:34606147628@10.100.10.67:5069] 15(8362) DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): to body ["Javi" <sip:34606147628@10.100.10.67:5069>] 15(8362) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <1> <INVITE> 15(8362) DEBUG: <core> [core/parser/msg_parser.c:183]: get_hdr_field(): content_length=64 15(8362) DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header 15(8362) INFO: <script>: LOOOOOOOOL {"sdp_data":"34619796636","sdp_data_2":null,"nrn":null} 15(8362) DEBUG: tm [t_lookup.c:1019]: t_check_msg(): msg (0x7f3346637088) id=0 global id=0 T start=0x7f334663b040 15(8362) DEBUG: tm [t_lookup.c:1091]: t_check_msg(): T (0x7f334663b040) already found for msg (0x7f3346637088)! 15(8362) DEBUG: tmx [t_var.c:534]: pv_get_tm_reply_code(): reply code is <503> 15(8362) DEBUG: tm [t_lookup.c:1019]: t_check_msg(): msg (0x7f3346637088) id=0 global id=0 T start=0x7f334663b040 15(8362) DEBUG: tm [t_lookup.c:1091]: t_check_msg(): T (0x7f334663b040) already found for msg (0x7f3346637088)! 15(8362) DEBUG: tmx [t_var.c:534]: pv_get_tm_reply_code(): reply code is <503> 15(8362) INFO: <script>: LEL 15(8362) DEBUG: <core> [core/msg_translator.c:159]: check_via_address(): (10.100.10.67, 10.100.10.67, 0) 15(8362) INFO: <script>: hereeeeeeeeeee 15(8362) DEBUG: tm [t_reply.c:1788]: relay_reply(): branch=1, save=0, relay=-1 icode=0 15(8362) DEBUG: <core> [core/receive.c:362]: receive_msg(): reply-route executed in: 676 usec 15(8362) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 15(8362) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 15(8362) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 15(8362) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 15(8362) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 15(8362) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 15(8362) DEBUG: <core> [core/xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) 15(8362) DEBUG: <core> [core/receive.c:378]: receive_msg(): cleaning up 15(8362) DEBUG: <core> [core/udp_server.c:492]: udp_rcv_loop(): received on udp socket: (106/100/404) [[SIP/2.0 180 Ringing 0D 0A Via: SIP/2.0/UDP 52.88.180.226:5069;branch=z9hG4bKec53.56115a2d09f62d466f1d45a5]] 15(8362) DEBUG: <core> [core/parser/msg_parser.c:614]: parse_msg(): SIP Reply (status): 15(8362) DEBUG: <core> [core/parser/msg_parser.c:616]: parse_msg(): version: <SIP/2.0> 15(8362) DEBUG: <core> [core/parser/msg_parser.c:618]: parse_msg(): status: <180> 15(8362) DEBUG: <core> [core/parser/msg_parser.c:620]: parse_msg(): reason: <Ringing> 15(8362) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKec53.56115a2d09f62d466f1d45a5f5ff7cc2.2>; state=9 15(8362) DEBUG: <core> [core/parser/parse_via.c:2681]: parse_via(): parsing via: next via 15(8362) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-9680-1-0>; state=16 15(8362) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 15(8362) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=2 15(8362) DEBUG: <core> [core/parser/msg_parser.c:494]: parse_headers(): this is the first via 15(8362) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=5 15(8362) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 15(8362) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [50]; uri=[sip:34606147628@10.100.10.67:5069] 15(8362) DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): to body ["Javi" <sip:34606147628@10.100.10.67:5069>] 15(8362) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <1> <INVITE> 15(8362) DEBUG: <core> [core/receive.c:205]: receive_msg(): --- received sip message - reply - call-id: [1-9680(a)10.100.10.67] - cseq: [1 INVITE] 15(8362) DEBUG: tm [t_lookup.c:1019]: t_check_msg(): msg (0x7f334b5f3768) id=2 global id=1 T start=0xffffffffffffffff 6(8353) DEBUG: <core> [core/udp_server.c:492]: udp_rcv_loop(): received on udp socket: (106/100/595) [[SIP/2.0 200 OK 0D 0A Via: SIP/2.0/UDP 52.88.180.226:5069;branch=z9hG4bKec53.56115a2d09f62d466f1d45a5f5ff7]] 15(8362) DEBUG: tm [t_lookup.c:897]: t_reply_matching(): t_reply_matching: hash 13774 label 0 branch 2 15(8362) DEBUG: tm [t_lookup.c:951]: t_reply_matching(): reply (0x7f334b5f3768) matched an active transaction (T=0x7f334663b040)! 15(8362) DEBUG: tm [t_lookup.c:1088]: t_check_msg(): msg (0x7f334b5f3768) id=2 global id=2 T end=0x7f334663b040 15(8362) DEBUG: tm [t_reply.c:2209]: reply_received(): org. status uas=100, uac[2]=0 local=0 is_invite=1) 15(8362) DEBUG: <core> [core/parser/msg_parser.c:183]: get_hdr_field(): content_length=0 15(8362) DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header 15(8362) INFO: <script>: LAL 15(8362) DEBUG: tm [t_reply.c:1265]: t_should_relay_response(): ->>>>>>>>> T_code=100, new_code=180 15(8362) DEBUG: tm [t_reply.c:1788]: relay_reply(): branch=2, save=0, relay=2 icode=0 15(8362) DEBUG: <core> [core/msg_translator.c:2270]: generate_res_buf_from_sip_res(): old size: 404, new size: 318 15(8362) DEBUG: <core> [core/msg_translator.c:2288]: generate_res_buf_from_sip_res(): copied size: orig:112, new: 26, rest: 292 msg= SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.100.10.67:5061;branch=z9hG4bK-9680-1-0 From: "BTS" <sip:34917019888@10.100.10.67:5061>;tag=9680SIPpTag001 To: "Javi" <sip:34606147628@10.100.10.67:5069>;tag=5 Call-ID: 1-9680(a)10.100.10.67 CSeq: 1 INVITE Contact: <sip:10.100.10.67:5060;transport=UDP> Content-Length: 0 15(8362) DEBUG: <core> [core/receive.c:362]: receive_msg(): reply-route executed in: 71 usec 15(8362) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 15(8362) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 15(8362) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 15(8362) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 15(8362) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 15(8362) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 15(8362) DEBUG: <core> [core/xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) 15(8362) DEBUG: <core> [core/receive.c:378]: receive_msg(): cleaning up 6(8353) DEBUG: <core> [core/parser/msg_parser.c:614]: parse_msg(): SIP Reply (status): 6(8353) DEBUG: <core> [core/parser/msg_parser.c:616]: parse_msg(): version: <SIP/2.0> 6(8353) DEBUG: <core> [core/parser/msg_parser.c:618]: parse_msg(): status: <200> 6(8353) DEBUG: <core> [core/parser/msg_parser.c:620]: parse_msg(): reason: <OK> 6(8353) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKec53.56115a2d09f62d466f1d45a5f5ff7cc2.2>; state=9 6(8353) DEBUG: <core> [core/parser/parse_via.c:2681]: parse_via(): parsing via: next via 6(8353) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-9680-1-0>; state=16 6(8353) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 6(8353) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=2 6(8353) DEBUG: <core> [core/parser/msg_parser.c:494]: parse_headers(): this is the first via 6(8353) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=5 6(8353) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 6(8353) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [50]; uri=[sip:34606147628@10.100.10.67:5069] 6(8353) DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): to body ["Javi" <sip:34606147628@10.100.10.67:5069>] 6(8353) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <1> <INVITE> 6(8353) DEBUG: <core> [core/receive.c:205]: receive_msg(): --- received sip message - reply - call-id: [1-9680(a)10.100.10.67] - cseq: [1 INVITE] 6(8353) DEBUG: tm [t_lookup.c:1019]: t_check_msg(): msg (0x7f334b5f3768) id=1 global id=0 T start=0xffffffffffffffff 6(8353) DEBUG: tm [t_lookup.c:897]: t_reply_matching(): t_reply_matching: hash 13774 label 0 branch 2 6(8353) DEBUG: tm [t_lookup.c:951]: t_reply_matching(): reply (0x7f334b5f3768) matched an active transaction (T=0x7f334663b040)! 6(8353) DEBUG: tm [t_lookup.c:1088]: t_check_msg(): msg (0x7f334b5f3768) id=1 global id=1 T end=0x7f334663b040 6(8353) DEBUG: tm [t_reply.c:2209]: reply_received(): org. status uas=180, uac[2]=180 local=0 is_invite=1) 6(8353) DEBUG: <core> [core/parser/msg_parser.c:183]: get_hdr_field(): content_length=161 6(8353) DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header 6(8353) INFO: <script>: LAL v=0 o=user1 53655765 2353687637 IN IP4 10.100.10.67 s=- c=IN IP4 10.100.10.67 t=0 0 m=audio 6000 RTP/AVP 0 18 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 6(8353) DEBUG: tm [t_reply.c:1265]: t_should_relay_response(): ->>>>>>>>> T_code=180, new_code=200 6(8353) DEBUG: tm [t_reply.c:1788]: relay_reply(): branch=2, save=0, relay=2 icode=0 6(8353) DEBUG: <core> [core/msg_translator.c:2270]: generate_res_buf_from_sip_res(): old size: 595, new size: 509 6(8353) DEBUG: <core> [core/msg_translator.c:2288]: generate_res_buf_from_sip_res(): copied size: orig:107, new: 21, rest: 488 msg= SIP/2.0 200 OK Via: SIP/2.0/UDP 10.100.10.67:5061;branch=z9hG4bK-9680-1-0 From: "BTS" <sip:34917019888@10.100.10.67:5061>;tag=9680SIPpTag001 To: "Javi" <sip:34606147628@10.100.10.67:5069>;tag=5 Call-ID: 1-9680(a)10.100.10.67 CSeq: 1 INVITE Contact: <sip:10.100.10.67:5060;transport=UDP> Content-Type: application/sdp Content-Length: 161 v=0 o=user1 53655765 2353687637 IN IP4 10.100.10.67 s=- c=IN IP4 10.100.10.67 t=0 0 m=audio 6000 RTP/AVP 0 18 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 6(8353) DEBUG: tm [t_reply.c:1631]: cleanup_uac_timers(): RETR/FR timers reset 6(8353) DEBUG: <core> [core/receive.c:362]: receive_msg(): reply-route executed in: 283 usec 6(8353) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 6(8353) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 6(8353) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 6(8353) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 6(8353) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 6(8353) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 6(8353) DEBUG: <core> [core/xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) 6(8353) DEBUG: <core> [core/receive.c:378]: receive_msg(): cleaning up 6(8353) DEBUG: <core> [core/udp_server.c:492]: udp_rcv_loop(): received on udp socket: (106/100/323) [[ACK sip:34606147628@10.100.10.67:5060 SIP/2.0 0D 0A Via: SIP/2.0/UDP 10.100.10.67:5061;branch=z9hG4bK-968]] 6(8353) DEBUG: <core> [core/parser/msg_parser.c:604]: parse_msg(): SIP Request: 6(8353) DEBUG: <core> [core/parser/msg_parser.c:606]: parse_msg(): method: <ACK> 6(8353) DEBUG: <core> [core/parser/msg_parser.c:608]: parse_msg(): uri: <sip:34606147628@10.100.10.67:5060> 6(8353) DEBUG: <core> [core/parser/msg_parser.c:610]: parse_msg(): version: <SIP/2.0> 6(8353) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-9680-1-0>; state=16 6(8353) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 6(8353) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=2 6(8353) DEBUG: <core> [core/parser/msg_parser.c:494]: parse_headers(): this is the first via 6(8353) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=5 6(8353) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 6(8353) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [50]; uri=[sip:34606147628@10.100.10.67:5069] 6(8353) DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): to body ["Test" <sip:34606147628@10.100.10.67:5069>] 6(8353) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <1> <ACK> 6(8353) DEBUG: <core> [core/receive.c:205]: receive_msg(): --- received sip message - request - call-id: [1-9680(a)10.100.10.67] - cseq: [1 ACK] 6(8353) DEBUG: <core> [core/receive.c:248]: receive_msg(): preparing to run routing scripts... 6(8353) DEBUG: sl [sl_funcs.c:387]: sl_filter_ACK(): too late to be a local ACK! 6(8353) INFO: <script>: hereeeeeeeeeee 6(8353) DEBUG: tm [t_lookup.c:1320]: t_newtran(): msg id=2 , global msg id=1 , T on entrance=0xffffffffffffffff 6(8353) DEBUG: <core> [core/parser/msg_parser.c:183]: get_hdr_field(): content_length=0 6(8353) DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header 6(8353) DEBUG: tm [t_lookup.c:477]: t_lookup_request(): start searching: hash=13774, isACK=1 6(8353) DEBUG: tm [t_lookup.c:435]: matching_3261(): RFC3261 transaction matching failed 6(8353) DEBUG: tm [t_lookup.c:659]: t_lookup_request(): no transaction found 6(8353) DEBUG: tm [t_funcs.c:284]: t_relay_to(): forwarding ACK statelessly 6(8353) DEBUG: <core> [core/md5utils.c:67]: MD5StringArray(): MD5 calculated: 56115a2d09f62d466f1d45a5f5ff7cc2 6(8353) DEBUG: <core> [core/msg_translator.c:159]: check_via_address(): (10.100.10.67, 10.100.10.67, 0) 6(8353) DEBUG: <core> [core/forward.c:557]: forward_request(): Sending: ACK sip:34606147628@10.100.10.67:5060 SIP/2.0 Via: SIP/2.0/UDP 52.88.180.226:5069;branch=z9hG4bKec53.56115a2d09f62d466f1d45a5f5ff7cc2.0 Via: SIP/2.0/UDP 10.100.10.67:5061;branch=z9hG4bK-9680-1-0 From: "BTS" <sip:34917019888@10.100.10.67:5061>;tag=9680SIPpTag001 To: "Test" <sip:34606147628@10.100.10.67:5069>;tag=5 Call-ID: 1-9680(a)10.100.10.67 CSeq: 1 ACK Max-Forwards: 70 Subject: 1 Content-Length: 0 . 6(8353) DEBUG: <core> [core/forward.c:559]: forward_request(): orig. len=323, new_len=414, proto=1 6(8353) INFO: <script>: hereeeeeeeeeee 6(8353) DEBUG: <core> [core/receive.c:289]: receive_msg(): request-route executed in: 232 usec 6(8353) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 6(8353) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 6(8353) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 6(8353) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 6(8353) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 6(8353) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 6(8353) DEBUG: <core> [core/xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) 6(8353) DEBUG: <core> [core/receive.c:378]: receive_msg(): cleaning up 3(8350) DEBUG: <core> [core/udp_server.c:492]: udp_rcv_loop(): received on udp socket: (106/100/338) [[BYE sip:34606147628@10.100.10.67:5060 SIP/2.0 0D 0A Via: SIP/2.0/UDP 10.100.10.67:5061;branch=z9hG4bK-968]] 3(8350) DEBUG: <core> [core/parser/msg_parser.c:604]: parse_msg(): SIP Request: 3(8350) DEBUG: <core> [core/parser/msg_parser.c:606]: parse_msg(): method: <BYE> 3(8350) DEBUG: <core> [core/parser/msg_parser.c:608]: parse_msg(): uri: <sip:34606147628@10.100.10.67:5060> 3(8350) DEBUG: <core> [core/parser/msg_parser.c:610]: parse_msg(): version: <SIP/2.0> 3(8350) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-9680-1-8>; state=16 3(8350) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 3(8350) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=2 3(8350) DEBUG: <core> [core/parser/msg_parser.c:494]: parse_headers(): this is the first via 3(8350) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=5 3(8350) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 3(8350) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [50]; uri=[sip:34606147628@10.100.10.67:5069] 3(8350) DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): to body ["Test" <sip:34606147628@10.100.10.67:5069>] 3(8350) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <2> <BYE> 3(8350) DEBUG: <core> [core/receive.c:205]: receive_msg(): --- received sip message - request - call-id: [1-9680(a)10.100.10.67] - cseq: [2 BYE] 3(8350) DEBUG: <core> [core/receive.c:248]: receive_msg(): preparing to run routing scripts... 3(8350) INFO: <script>: hereeeeeeeeeee 3(8350) DEBUG: tm [t_lookup.c:1320]: t_newtran(): msg id=2 , global msg id=1 , T on entrance=0xffffffffffffffff 3(8350) DEBUG: <core> [core/parser/msg_parser.c:183]: get_hdr_field(): content_length=0 3(8350) DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header 3(8350) DEBUG: tm [t_lookup.c:477]: t_lookup_request(): start searching: hash=13771, isACK=0 3(8350) DEBUG: tm [t_lookup.c:435]: matching_3261(): RFC3261 transaction matching failed 3(8350) DEBUG: tm [t_lookup.c:659]: t_lookup_request(): no transaction found 3(8350) DEBUG: <core> [core/md5utils.c:67]: MD5StringArray(): MD5 calculated: 5b69d64cadac24fd6fc477c074a9b4c9 3(8350) INFO: <script>: LEL 3(8350) DEBUG: <core> [core/msg_translator.c:159]: check_via_address(): (10.100.10.67, 10.100.10.67, 0) 3(8350) INFO: <script>: hereeeeeeeeeee 3(8350) DEBUG: tm [t_funcs.c:372]: t_relay_to(): new transaction forwarded 3(8350) DEBUG: <core> [core/receive.c:289]: receive_msg(): request-route executed in: 265 usec 3(8350) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 3(8350) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 3(8350) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 3(8350) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 3(8350) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 3(8350) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 3(8350) DEBUG: <core> [core/xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) 3(8350) DEBUG: <core> [core/receive.c:378]: receive_msg(): cleaning up 7(8354) DEBUG: <core> [core/udp_server.c:492]: udp_rcv_loop(): received on udp socket: (106/100/396) [[SIP/2.0 200 OK 0D 0A Via: SIP/2.0/UDP 52.88.180.226:5069;branch=z9hG4bKbc53.5b69d64cadac24fd6fc477c074a9b]] 7(8354) DEBUG: <core> [core/parser/msg_parser.c:614]: parse_msg(): SIP Reply (status): 7(8354) DEBUG: <core> [core/parser/msg_parser.c:616]: parse_msg(): version: <SIP/2.0> 7(8354) DEBUG: <core> [core/parser/msg_parser.c:618]: parse_msg(): status: <200> 7(8354) DEBUG: <core> [core/parser/msg_parser.c:620]: parse_msg(): reason: <OK> 7(8354) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKbc53.5b69d64cadac24fd6fc477c074a9b4c9.0>; state=9 7(8354) DEBUG: <core> [core/parser/parse_via.c:2681]: parse_via(): parsing via: next via 7(8354) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-9680-1-8>; state=16 7(8354) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 7(8354) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=2 7(8354) DEBUG: <core> [core/parser/msg_parser.c:494]: parse_headers(): this is the first via 7(8354) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=5 7(8354) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 7(8354) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [50]; uri=[sip:34606147628@10.100.10.67:5069] 7(8354) DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): to body ["Test" <sip:34606147628@10.100.10.67:5069>] 7(8354) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <2> <BYE> 7(8354) DEBUG: <core> [core/receive.c:205]: receive_msg(): --- received sip message - reply - call-id: [1-9680(a)10.100.10.67] - cseq: [2 BYE] 7(8354) DEBUG: tm [t_lookup.c:1019]: t_check_msg(): msg (0x7f334b5f3768) id=2 global id=1 T start=0xffffffffffffffff 7(8354) DEBUG: tm [t_lookup.c:897]: t_reply_matching(): t_reply_matching: hash 13771 label 0 branch 0 7(8354) DEBUG: tm [t_lookup.c:951]: t_reply_matching(): reply (0x7f334b5f3768) matched an active transaction (T=0x7f33466359d8)! 7(8354) DEBUG: tm [t_lookup.c:1088]: t_check_msg(): msg (0x7f334b5f3768) id=2 global id=2 T end=0x7f33466359d8 7(8354) DEBUG: tm [t_reply.c:2209]: reply_received(): org. status uas=0, uac[0]=0 local=0 is_invite=0) 7(8354) DEBUG: tm [t_reply.c:1265]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=200 7(8354) DEBUG: tm [t_reply.c:1788]: relay_reply(): branch=0, save=0, relay=0 icode=0 7(8354) DEBUG: <core> [core/msg_translator.c:2270]: generate_res_buf_from_sip_res(): old size: 396, new size: 310 7(8354) DEBUG: <core> [core/msg_translator.c:2288]: generate_res_buf_from_sip_res(): copied size: orig:107, new: 21, rest: 289 msg= SIP/2.0 200 OK Via: SIP/2.0/UDP 10.100.10.67:5061;branch=z9hG4bK-9680-1-8 From: "BTS" <sip:34917019888@10.100.10.67:5061>;tag=9680SIPpTag001 To: "Test" <sip:34606147628@10.100.10.67:5069>;tag=5 Call-ID: 1-9680(a)10.100.10.67 CSeq: 2 BYE Contact: <sip:10.100.10.67:5060;transport=UDP> Content-Length: 0 7(8354) DEBUG: tm [t_reply.c:1631]: cleanup_uac_timers(): RETR/FR timers reset 7(8354) DEBUG: <core> [core/receive.c:362]: receive_msg(): reply-route executed in: 137 usec 7(8354) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 7(8354) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 7(8354) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 7(8354) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 7(8354) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 7(8354) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 7(8354) DEBUG: <core> [core/xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) 7(8354) DEBUG: <core> [core/receive.c:378]: receive_msg(): cleaning up 18(8365) DEBUG: tm [h_table.c:129]: free_cell_helper(): freeing transaction 0x7f334663b040 from timer.c:654 18(8365) DEBUG: tm [h_table.c:129]: free_cell_helper(): freeing transaction 0x7f33466359d8 from timer.c:654 1_children.log 1(7698) DEBUG: <core> [core/udp_server.c:492]: udp_rcv_loop(): received on udp socket: (106/100/634) [[INVITE sip:34606147628@10.100.10.67:5069 SIP/2.0 0D 0A Via: SIP/2.0/UDP 10.100.10.67:5061;branch=z9hG4bK-]] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:604]: parse_msg(): SIP Request: 1(7698) DEBUG: <core> [core/parser/msg_parser.c:606]: parse_msg(): method: <INVITE> 1(7698) DEBUG: <core> [core/parser/msg_parser.c:608]: parse_msg(): uri: <sip:34606147628@10.100.10.67:5069> 1(7698) DEBUG: <core> [core/parser/msg_parser.c:610]: parse_msg(): version: <SIP/2.0> 1(7698) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-8173-1-0>; state=16 1(7698) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 1(7698) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=2 1(7698) DEBUG: <core> [core/parser/msg_parser.c:494]: parse_headers(): this is the first via 1(7698) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=10 1(7698) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [44]; uri=[sip:34606147628@10.100.10.67:5069] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): to body ["Javi" <sip:34606147628@10.100.10.67:5069> ] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <1> <INVITE> 1(7698) DEBUG: <core> [core/receive.c:205]: receive_msg(): --- received sip message - request - call-id: [1-8173(a)10.100.10.67] - cseq: [1 INVITE] 1(7698) DEBUG: <core> [core/receive.c:248]: receive_msg(): preparing to run routing scripts... 1(7698) INFO: <script>: hereeeeeeeeeee 1(7698) DEBUG: tm [t_lookup.c:1320]: t_newtran(): msg id=1 , global msg id=0 , T on entrance=0xffffffffffffffff 1(7698) DEBUG: <core> [core/parser/msg_parser.c:183]: get_hdr_field(): content_length=217 1(7698) DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header 1(7698) DEBUG: tm [t_lookup.c:477]: t_lookup_request(): start searching: hash=65453, isACK=0 1(7698) DEBUG: tm [t_lookup.c:435]: matching_3261(): RFC3261 transaction matching failed 1(7698) DEBUG: tm [t_lookup.c:659]: t_lookup_request(): no transaction found 1(7698) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=8173SIPpTag001 1(7698) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 1(7698) DEBUG: <core> [core/md5utils.c:67]: MD5StringArray(): MD5 calculated: 8f25665e8c867e73b07da3cb1f702171 1(7698) DEBUG: tm [t_funcs.c:324]: t_relay_to(): new INVITE 1(7698) DEBUG: <core> [core/msg_translator.c:159]: check_via_address(): (10.100.10.67, 10.100.10.67, 0) 1(7698) DEBUG: tm [t_reply.c:596]: _reply_light(): reply sent out. buf=0x7fe2c208bea0: SIP/2.0 100 trying -..., shmem=0x7fe2bd0c9f40: SIP/2.0 100 trying - 1(7698) DEBUG: tm [t_reply.c:606]: _reply_light(): finished 1(7698) INFO: <script>: LEL 1(7698) DEBUG: <core> [core/msg_translator.c:159]: check_via_address(): (10.100.10.67, 10.100.10.67, 0) 1(7698) INFO: <script>: hereeeeeeeeeee 1(7698) DEBUG: tm [t_funcs.c:372]: t_relay_to(): new transaction forwarded 1(7698) DEBUG: <core> [core/receive.c:289]: receive_msg(): request-route executed in: 607 usec 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) 1(7698) DEBUG: <core> [core/receive.c:378]: receive_msg(): cleaning up 1(7698) DEBUG: <core> [core/udp_server.c:492]: udp_rcv_loop(): received on udp socket: (106/100/397) [[SIP/2.0 100 Trying 0D 0A Via: SIP/2.0/UDP 52.88.180.226:5069;branch=z9hG4bKdaff.8f25665e8c867e73b07da3cb1]] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:614]: parse_msg(): SIP Reply (status): 1(7698) DEBUG: <core> [core/parser/msg_parser.c:616]: parse_msg(): version: <SIP/2.0> 1(7698) DEBUG: <core> [core/parser/msg_parser.c:618]: parse_msg(): status: <100> 1(7698) DEBUG: <core> [core/parser/msg_parser.c:620]: parse_msg(): reason: <Trying> 1(7698) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKdaff.8f25665e8c867e73b07da3cb1f702171.0>; state=9 1(7698) DEBUG: <core> [core/parser/parse_via.c:2681]: parse_via(): parsing via: next via 1(7698) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-8173-1-0>; state=16 1(7698) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 1(7698) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=2 1(7698) DEBUG: <core> [core/parser/msg_parser.c:494]: parse_headers(): this is the first via 1(7698) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=10 1(7698) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [44]; uri=[sip:34606147628@10.100.10.67:5069] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): to body ["Javi" <sip:34606147628@10.100.10.67:5069> ] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <1> <INVITE> 1(7698) DEBUG: <core> [core/receive.c:205]: receive_msg(): --- received sip message - reply - call-id: [1-8173(a)10.100.10.67] - cseq: [1 INVITE] 1(7698) DEBUG: tm [t_lookup.c:1019]: t_check_msg(): msg (0x7fe2c208a768) id=2 global id=1 T start=0xffffffffffffffff 1(7698) DEBUG: tm [t_lookup.c:897]: t_reply_matching(): t_reply_matching: hash 65453 label 0 branch 0 1(7698) DEBUG: tm [t_lookup.c:951]: t_reply_matching(): reply (0x7fe2c208a768) matched an active transaction (T=0x7fe2bd0c76d8)! 1(7698) DEBUG: tm [t_lookup.c:1088]: t_check_msg(): msg (0x7fe2c208a768) id=2 global id=2 T end=0x7fe2bd0c76d8 1(7698) DEBUG: tm [t_reply.c:2209]: reply_received(): org. status uas=100, uac[0]=0 local=0 is_invite=1) 1(7698) DEBUG: <core> [core/parser/msg_parser.c:183]: get_hdr_field(): content_length=0 1(7698) DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header 1(7698) INFO: <script>: LAL 1(7698) DEBUG: tm [t_reply.c:1265]: t_should_relay_response(): ->>>>>>>>> T_code=100, new_code=100 1(7698) DEBUG: tm [t_reply.c:1788]: relay_reply(): branch=0, save=0, relay=-1 icode=0 1(7698) DEBUG: <core> [core/receive.c:362]: receive_msg(): reply-route executed in: 118 usec 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) 1(7698) DEBUG: <core> [core/receive.c:378]: receive_msg(): cleaning up 1(7698) DEBUG: <core> [core/udp_server.c:492]: udp_rcv_loop(): received on udp socket: (106/100/479) [[SIP/2.0 302 Moved Temporarily 0D 0A Via: SIP/2.0/UDP 52.88.180.226:5069;branch=z9hG4bKdaff.8f25665e8c867e]] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:614]: parse_msg(): SIP Reply (status): 1(7698) DEBUG: <core> [core/parser/msg_parser.c:616]: parse_msg(): version: <SIP/2.0> 1(7698) DEBUG: <core> [core/parser/msg_parser.c:618]: parse_msg(): status: <302> 1(7698) DEBUG: <core> [core/parser/msg_parser.c:620]: parse_msg(): reason: <Moved Temporarily> 1(7698) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKdaff.8f25665e8c867e73b07da3cb1f702171.0>; state=9 1(7698) DEBUG: <core> [core/parser/parse_via.c:2681]: parse_via(): parsing via: next via 1(7698) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-8173-1-0>; state=16 1(7698) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 1(7698) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=2 1(7698) DEBUG: <core> [core/parser/msg_parser.c:494]: parse_headers(): this is the first via 1(7698) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=3 1(7698) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 1(7698) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [50]; uri=[sip:34606147628@10.100.10.67:5069] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): to body ["Javi" <sip:34606147628@10.100.10.67:5069>] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <1> <INVITE> 1(7698) DEBUG: <core> [core/receive.c:205]: receive_msg(): --- received sip message - reply - call-id: [1-8173(a)10.100.10.67] - cseq: [1 INVITE] 1(7698) DEBUG: tm [t_lookup.c:1019]: t_check_msg(): msg (0x7fe2c208a768) id=3 global id=2 T start=0xffffffffffffffff 1(7698) DEBUG: tm [t_lookup.c:897]: t_reply_matching(): t_reply_matching: hash 65453 label 0 branch 0 1(7698) DEBUG: tm [t_lookup.c:951]: t_reply_matching(): reply (0x7fe2c208a768) matched an active transaction (T=0x7fe2bd0c76d8)! 1(7698) DEBUG: tm [t_lookup.c:1088]: t_check_msg(): msg (0x7fe2c208a768) id=3 global id=3 T end=0x7fe2bd0c76d8 1(7698) DEBUG: tm [t_reply.c:2209]: reply_received(): org. status uas=100, uac[0]=100 local=0 is_invite=1) 1(7698) DEBUG: <core> [core/parser/msg_parser.c:183]: get_hdr_field(): content_length=63 1(7698) DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header 1(7698) INFO: <script>: LAL {"sdp_data":"3466666666","sdp_data_2":null,"nrn":null} 1(7698) DEBUG: tm [t_reply.c:1265]: t_should_relay_response(): ->>>>>>>>> T_code=100, new_code=302 1(7698) DEBUG: <core> [core/parser/msg_parser.c:614]: parse_msg(): SIP Reply (status): 1(7698) DEBUG: <core> [core/parser/msg_parser.c:616]: parse_msg(): version: <SIP/2.0> 1(7698) DEBUG: <core> [core/parser/msg_parser.c:618]: parse_msg(): status: <302> 1(7698) DEBUG: <core> [core/parser/msg_parser.c:620]: parse_msg(): reason: <Moved Temporarily> 1(7698) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKdaff.8f25665e8c867e73b07da3cb1f702171.0>; state=9 1(7698) DEBUG: <core> [core/parser/parse_via.c:2681]: parse_via(): parsing via: next via 1(7698) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-8173-1-0>; state=16 1(7698) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 1(7698) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=2 1(7698) DEBUG: <core> [core/parser/msg_parser.c:494]: parse_headers(): this is the first via 1(7698) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=3 1(7698) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 1(7698) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [50]; uri=[sip:34606147628@10.100.10.67:5069] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): to body ["Javi" <sip:34606147628@10.100.10.67:5069>] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <1> <INVITE> 1(7698) DEBUG: <core> [core/parser/msg_parser.c:183]: get_hdr_field(): content_length=63 1(7698) DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header 1(7698) INFO: <script>: LOOOOOOOOL {"sdp_data":"3466666666","sdp_data_2":null,"nrn":null} 1(7698) DEBUG: tm [t_lookup.c:1019]: t_check_msg(): msg (0x7fe2bd0ca850) id=0 global id=0 T start=0x7fe2bd0c76d8 1(7698) DEBUG: tm [t_lookup.c:1091]: t_check_msg(): T (0x7fe2bd0c76d8) already found for msg (0x7fe2bd0ca850)! 1(7698) DEBUG: tmx [t_var.c:534]: pv_get_tm_reply_code(): reply code is <302> 1(7698) INFO: <script>: Call: 302 received, trying to load contacts 1(7698) DEBUG: uac_redirect [rd_funcs.c:72]: get_redirect(): resume branch=0 1(7698) DEBUG: uac_redirect [rd_funcs.c:79]: get_redirect(): checking branch=0 (added=0) 1(7698) DEBUG: uac_redirect [rd_funcs.c:83]: get_redirect(): branch=0 is a redirect (added=0) 1(7698) DEBUG: uac_redirect [rd_funcs.c:159]: sort_contacts(): sort_contacts: <sip:34666988297@10.100.10.67:5060> q=10 1(7698) DEBUG: uac_redirect [rd_funcs.c:295]: shmcontact2dset(): adding contact <sip:34666988297@10.100.10.67:5060> 1(7698) DEBUG: <core> [sruid.c:183]: sruid_next(): new sruid is [rdir-5be42212-1e12-1] (1 / 20) 1(7698) DEBUG: tm [t_serial.c:191]: ki_t_load_contacts(): nr_branches is 1 1(7698) DEBUG: tm [t_serial.c:194]: ki_t_load_contacts(): nothing to do - only one contact! 1(7698) INFO: <script>: LEL 1(7698) DEBUG: <core> [core/msg_translator.c:159]: check_via_address(): (10.100.10.67, 10.100.10.67, 0) 1(7698) INFO: <script>: hereeeeeeeeeee 1(7698) DEBUG: tm [t_reply.c:1788]: relay_reply(): branch=0, save=0, relay=-1 icode=0 1(7698) DEBUG: <core> [core/receive.c:362]: receive_msg(): reply-route executed in: 581 usec 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) 1(7698) DEBUG: <core> [core/receive.c:378]: receive_msg(): cleaning up 1(7698) DEBUG: <core> [core/udp_server.c:492]: udp_rcv_loop(): received on udp socket: (106/100/397) [[SIP/2.0 100 Trying 0D 0A Via: SIP/2.0/UDP 52.88.180.226:5069;branch=z9hG4bKdaff.8f25665e8c867e73b07da3cb1]] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:614]: parse_msg(): SIP Reply (status): 1(7698) DEBUG: <core> [core/parser/msg_parser.c:616]: parse_msg(): version: <SIP/2.0> 1(7698) DEBUG: <core> [core/parser/msg_parser.c:618]: parse_msg(): status: <100> 1(7698) DEBUG: <core> [core/parser/msg_parser.c:620]: parse_msg(): reason: <Trying> 1(7698) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKdaff.8f25665e8c867e73b07da3cb1f702171.1>; state=9 1(7698) DEBUG: <core> [core/parser/parse_via.c:2681]: parse_via(): parsing via: next via 1(7698) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-8173-1-0>; state=16 1(7698) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 1(7698) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=2 1(7698) DEBUG: <core> [core/parser/msg_parser.c:494]: parse_headers(): this is the first via 1(7698) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=10 1(7698) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [44]; uri=[sip:34606147628@10.100.10.67:5069] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): to body ["Javi" <sip:34606147628@10.100.10.67:5069> ] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <1> <INVITE> 1(7698) DEBUG: <core> [core/receive.c:205]: receive_msg(): --- received sip message - reply - call-id: [1-8173(a)10.100.10.67] - cseq: [1 INVITE] 1(7698) DEBUG: tm [t_lookup.c:1019]: t_check_msg(): msg (0x7fe2c208a768) id=4 global id=3 T start=0xffffffffffffffff 1(7698) DEBUG: tm [t_lookup.c:897]: t_reply_matching(): t_reply_matching: hash 65453 label 0 branch 1 1(7698) DEBUG: tm [t_lookup.c:951]: t_reply_matching(): reply (0x7fe2c208a768) matched an active transaction (T=0x7fe2bd0c76d8)! 1(7698) DEBUG: tm [t_lookup.c:1088]: t_check_msg(): msg (0x7fe2c208a768) id=4 global id=4 T end=0x7fe2bd0c76d8 1(7698) DEBUG: tm [t_reply.c:2209]: reply_received(): org. status uas=100, uac[1]=0 local=0 is_invite=1) 1(7698) DEBUG: <core> [core/parser/msg_parser.c:183]: get_hdr_field(): content_length=0 1(7698) DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header 1(7698) INFO: <script>: LAL 1(7698) DEBUG: tm [t_reply.c:1265]: t_should_relay_response(): ->>>>>>>>> T_code=100, new_code=100 1(7698) DEBUG: tm [t_reply.c:1788]: relay_reply(): branch=1, save=0, relay=-1 icode=0 1(7698) DEBUG: <core> [core/receive.c:362]: receive_msg(): reply-route executed in: 38 usec 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) 1(7698) DEBUG: <core> [core/receive.c:378]: receive_msg(): cleaning up 1(7698) DEBUG: <core> [core/udp_server.c:492]: udp_rcv_loop(): received on udp socket: (106/100/482) [[SIP/2.0 503 Service Unavailable 0D 0A Via: SIP/2.0/UDP 52.88.180.226:5069;branch=z9hG4bKdaff.8f25665e8c86]] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:614]: parse_msg(): SIP Reply (status): 1(7698) DEBUG: <core> [core/parser/msg_parser.c:616]: parse_msg(): version: <SIP/2.0> 1(7698) DEBUG: <core> [core/parser/msg_parser.c:618]: parse_msg(): status: <503> 1(7698) DEBUG: <core> [core/parser/msg_parser.c:620]: parse_msg(): reason: <Service Unavailable> 1(7698) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKdaff.8f25665e8c867e73b07da3cb1f702171.1>; state=9 1(7698) DEBUG: <core> [core/parser/parse_via.c:2681]: parse_via(): parsing via: next via 1(7698) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-8173-1-0>; state=16 1(7698) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 1(7698) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=2 1(7698) DEBUG: <core> [core/parser/msg_parser.c:494]: parse_headers(): this is the first via 1(7698) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=3 1(7698) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 1(7698) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [50]; uri=[sip:34606147628@10.100.10.67:5069] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): to body ["Javi" <sip:34606147628@10.100.10.67:5069>] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <1> <INVITE> 1(7698) DEBUG: <core> [core/receive.c:205]: receive_msg(): --- received sip message - reply - call-id: [1-8173(a)10.100.10.67] - cseq: [1 INVITE] 1(7698) DEBUG: tm [t_lookup.c:1019]: t_check_msg(): msg (0x7fe2c208a768) id=5 global id=4 T start=0xffffffffffffffff 1(7698) DEBUG: tm [t_lookup.c:897]: t_reply_matching(): t_reply_matching: hash 65453 label 0 branch 1 1(7698) DEBUG: tm [t_lookup.c:951]: t_reply_matching(): reply (0x7fe2c208a768) matched an active transaction (T=0x7fe2bd0c76d8)! 1(7698) DEBUG: tm [t_lookup.c:1088]: t_check_msg(): msg (0x7fe2c208a768) id=5 global id=5 T end=0x7fe2bd0c76d8 1(7698) DEBUG: tm [t_reply.c:2209]: reply_received(): org. status uas=100, uac[1]=100 local=0 is_invite=1) 1(7698) DEBUG: <core> [core/parser/msg_parser.c:183]: get_hdr_field(): content_length=64 1(7698) DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header 1(7698) INFO: <script>: LAL {"sdp_data":"34619796636","sdp_data_2":null,"nrn":null} 1(7698) DEBUG: tm [t_reply.c:1265]: t_should_relay_response(): ->>>>>>>>> T_code=100, new_code=503 1(7698) INFO: <script>: LOOOOOOOOL {"sdp_data":"3466666666","sdp_data_2":null,"nrn":null} 1(7698) DEBUG: tm [t_lookup.c:1019]: t_check_msg(): msg (0x7fe2bd0cbb88) id=0 global id=0 T start=0x7fe2bd0c76d8 1(7698) DEBUG: tm [t_lookup.c:1091]: t_check_msg(): T (0x7fe2bd0c76d8) already found for msg (0x7fe2bd0cbb88)! 1(7698) DEBUG: tmx [t_var.c:534]: pv_get_tm_reply_code(): reply code is <503> 1(7698) DEBUG: tm [t_lookup.c:1019]: t_check_msg(): msg (0x7fe2bd0cbb88) id=0 global id=0 T start=0x7fe2bd0c76d8 1(7698) DEBUG: tm [t_lookup.c:1091]: t_check_msg(): T (0x7fe2bd0c76d8) already found for msg (0x7fe2bd0cbb88)! 1(7698) DEBUG: tmx [t_var.c:534]: pv_get_tm_reply_code(): reply code is <503> 1(7698) INFO: <script>: LEL 1(7698) DEBUG: <core> [core/msg_translator.c:159]: check_via_address(): (10.100.10.67, 10.100.10.67, 0) 1(7698) INFO: <script>: hereeeeeeeeeee 1(7698) DEBUG: tm [t_reply.c:1788]: relay_reply(): branch=1, save=0, relay=-1 icode=0 1(7698) DEBUG: <core> [core/receive.c:362]: receive_msg(): reply-route executed in: 188 usec 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) 1(7698) DEBUG: <core> [core/receive.c:378]: receive_msg(): cleaning up 1(7698) DEBUG: <core> [core/udp_server.c:492]: udp_rcv_loop(): received on udp socket: (106/100/404) [[SIP/2.0 180 Ringing 0D 0A Via: SIP/2.0/UDP 52.88.180.226:5069;branch=z9hG4bKdaff.8f25665e8c867e73b07da3cb]] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:614]: parse_msg(): SIP Reply (status): 1(7698) DEBUG: <core> [core/parser/msg_parser.c:616]: parse_msg(): version: <SIP/2.0> 1(7698) DEBUG: <core> [core/parser/msg_parser.c:618]: parse_msg(): status: <180> 1(7698) DEBUG: <core> [core/parser/msg_parser.c:620]: parse_msg(): reason: <Ringing> 1(7698) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKdaff.8f25665e8c867e73b07da3cb1f702171.2>; state=9 1(7698) DEBUG: <core> [core/parser/parse_via.c:2681]: parse_via(): parsing via: next via 1(7698) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-8173-1-0>; state=16 1(7698) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 1(7698) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=2 1(7698) DEBUG: <core> [core/parser/msg_parser.c:494]: parse_headers(): this is the first via 1(7698) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=3 1(7698) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 1(7698) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [50]; uri=[sip:34606147628@10.100.10.67:5069] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): to body ["Javi" <sip:34606147628@10.100.10.67:5069>] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <1> <INVITE> 1(7698) DEBUG: <core> [core/receive.c:205]: receive_msg(): --- received sip message - reply - call-id: [1-8173(a)10.100.10.67] - cseq: [1 INVITE] 1(7698) DEBUG: tm [t_lookup.c:1019]: t_check_msg(): msg (0x7fe2c208a768) id=6 global id=5 T start=0xffffffffffffffff 1(7698) DEBUG: tm [t_lookup.c:897]: t_reply_matching(): t_reply_matching: hash 65453 label 0 branch 2 1(7698) DEBUG: tm [t_lookup.c:951]: t_reply_matching(): reply (0x7fe2c208a768) matched an active transaction (T=0x7fe2bd0c76d8)! 1(7698) DEBUG: tm [t_lookup.c:1088]: t_check_msg(): msg (0x7fe2c208a768) id=6 global id=6 T end=0x7fe2bd0c76d8 1(7698) DEBUG: tm [t_reply.c:2209]: reply_received(): org. status uas=100, uac[2]=0 local=0 is_invite=1) 1(7698) DEBUG: <core> [core/parser/msg_parser.c:183]: get_hdr_field(): content_length=0 1(7698) DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header 1(7698) INFO: <script>: LAL 1(7698) DEBUG: tm [t_reply.c:1265]: t_should_relay_response(): ->>>>>>>>> T_code=100, new_code=180 1(7698) DEBUG: tm [t_reply.c:1788]: relay_reply(): branch=2, save=0, relay=2 icode=0 1(7698) DEBUG: <core> [core/msg_translator.c:2270]: generate_res_buf_from_sip_res(): old size: 404, new size: 318 1(7698) DEBUG: <core> [core/msg_translator.c:2288]: generate_res_buf_from_sip_res(): copied size: orig:112, new: 26, rest: 292 msg= SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.100.10.67:5061;branch=z9hG4bK-8173-1-0 From: "BTS" <sip:34917019888@10.100.10.67:5061>;tag=8173SIPpTag001 To: "Javi" <sip:34606147628@10.100.10.67:5069>;tag=3 Call-ID: 1-8173(a)10.100.10.67 CSeq: 1 INVITE Contact: <sip:10.100.10.67:5060;transport=UDP> Content-Length: 0 1(7698) DEBUG: <core> [core/receive.c:362]: receive_msg(): reply-route executed in: 93 usec 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) 1(7698) DEBUG: <core> [core/receive.c:378]: receive_msg(): cleaning up 1(7698) DEBUG: <core> [core/udp_server.c:492]: udp_rcv_loop(): received on udp socket: (106/100/595) [[SIP/2.0 200 OK 0D 0A Via: SIP/2.0/UDP 52.88.180.226:5069;branch=z9hG4bKdaff.8f25665e8c867e73b07da3cb1f702]] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:614]: parse_msg(): SIP Reply (status): 1(7698) DEBUG: <core> [core/parser/msg_parser.c:616]: parse_msg(): version: <SIP/2.0> 1(7698) DEBUG: <core> [core/parser/msg_parser.c:618]: parse_msg(): status: <200> 1(7698) DEBUG: <core> [core/parser/msg_parser.c:620]: parse_msg(): reason: <OK> 1(7698) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKdaff.8f25665e8c867e73b07da3cb1f702171.2>; state=9 1(7698) DEBUG: <core> [core/parser/parse_via.c:2681]: parse_via(): parsing via: next via 1(7698) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-8173-1-0>; state=16 1(7698) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 1(7698) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=2 1(7698) DEBUG: <core> [core/parser/msg_parser.c:494]: parse_headers(): this is the first via 1(7698) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=3 1(7698) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 1(7698) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [50]; uri=[sip:34606147628@10.100.10.67:5069] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): to body ["Javi" <sip:34606147628@10.100.10.67:5069>] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <1> <INVITE> 1(7698) DEBUG: <core> [core/receive.c:205]: receive_msg(): --- received sip message - reply - call-id: [1-8173(a)10.100.10.67] - cseq: [1 INVITE] 1(7698) DEBUG: tm [t_lookup.c:1019]: t_check_msg(): msg (0x7fe2c208a768) id=7 global id=6 T start=0xffffffffffffffff 1(7698) DEBUG: tm [t_lookup.c:897]: t_reply_matching(): t_reply_matching: hash 65453 label 0 branch 2 1(7698) DEBUG: tm [t_lookup.c:951]: t_reply_matching(): reply (0x7fe2c208a768) matched an active transaction (T=0x7fe2bd0c76d8)! 1(7698) DEBUG: tm [t_lookup.c:1088]: t_check_msg(): msg (0x7fe2c208a768) id=7 global id=7 T end=0x7fe2bd0c76d8 1(7698) DEBUG: tm [t_reply.c:2209]: reply_received(): org. status uas=180, uac[2]=180 local=0 is_invite=1) 1(7698) DEBUG: <core> [core/parser/msg_parser.c:183]: get_hdr_field(): content_length=161 1(7698) DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header 1(7698) INFO: <script>: LAL v=0 o=user1 53655765 2353687637 IN IP4 10.100.10.67 s=- c=IN IP4 10.100.10.67 t=0 0 m=audio 6000 RTP/AVP 0 18 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 1(7698) DEBUG: tm [t_reply.c:1265]: t_should_relay_response(): ->>>>>>>>> T_code=180, new_code=200 1(7698) DEBUG: tm [t_reply.c:1788]: relay_reply(): branch=2, save=0, relay=2 icode=0 1(7698) DEBUG: <core> [core/msg_translator.c:2270]: generate_res_buf_from_sip_res(): old size: 595, new size: 509 1(7698) DEBUG: <core> [core/msg_translator.c:2288]: generate_res_buf_from_sip_res(): copied size: orig:107, new: 21, rest: 488 msg= SIP/2.0 200 OK Via: SIP/2.0/UDP 10.100.10.67:5061;branch=z9hG4bK-8173-1-0 From: "BTS" <sip:34917019888@10.100.10.67:5061>;tag=8173SIPpTag001 To: "Javi" <sip:34606147628@10.100.10.67:5069>;tag=3 Call-ID: 1-8173(a)10.100.10.67 CSeq: 1 INVITE Contact: <sip:10.100.10.67:5060;transport=UDP> Content-Type: application/sdp Content-Length: 161 v=0 o=user1 53655765 2353687637 IN IP4 10.100.10.67 s=- c=IN IP4 10.100.10.67 t=0 0 m=audio 6000 RTP/AVP 0 18 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 1(7698) DEBUG: tm [t_reply.c:1631]: cleanup_uac_timers(): RETR/FR timers reset 1(7698) DEBUG: <core> [core/receive.c:362]: receive_msg(): reply-route executed in: 185 usec 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) 1(7698) DEBUG: <core> [core/receive.c:378]: receive_msg(): cleaning up 1(7698) DEBUG: <core> [core/udp_server.c:492]: udp_rcv_loop(): received on udp socket: (106/100/323) [[ACK sip:34606147628@10.100.10.67:5060 SIP/2.0 0D 0A Via: SIP/2.0/UDP 10.100.10.67:5061;branch=z9hG4bK-817]] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:604]: parse_msg(): SIP Request: 1(7698) DEBUG: <core> [core/parser/msg_parser.c:606]: parse_msg(): method: <ACK> 1(7698) DEBUG: <core> [core/parser/msg_parser.c:608]: parse_msg(): uri: <sip:34606147628@10.100.10.67:5060> 1(7698) DEBUG: <core> [core/parser/msg_parser.c:610]: parse_msg(): version: <SIP/2.0> 1(7698) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-8173-1-0>; state=16 1(7698) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 1(7698) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=2 1(7698) DEBUG: <core> [core/parser/msg_parser.c:494]: parse_headers(): this is the first via 1(7698) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=3 1(7698) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 1(7698) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [50]; uri=[sip:34606147628@10.100.10.67:5069] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): to body ["Test" <sip:34606147628@10.100.10.67:5069>] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <1> <ACK> 1(7698) DEBUG: <core> [core/receive.c:205]: receive_msg(): --- received sip message - request - call-id: [1-8173(a)10.100.10.67] - cseq: [1 ACK] 1(7698) DEBUG: <core> [core/receive.c:248]: receive_msg(): preparing to run routing scripts... 1(7698) DEBUG: sl [sl_funcs.c:387]: sl_filter_ACK(): too late to be a local ACK! 1(7698) INFO: <script>: hereeeeeeeeeee 1(7698) DEBUG: tm [t_lookup.c:1320]: t_newtran(): msg id=8 , global msg id=7 , T on entrance=0xffffffffffffffff 1(7698) DEBUG: <core> [core/parser/msg_parser.c:183]: get_hdr_field(): content_length=0 1(7698) DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header 1(7698) DEBUG: tm [t_lookup.c:477]: t_lookup_request(): start searching: hash=65453, isACK=1 1(7698) DEBUG: tm [t_lookup.c:435]: matching_3261(): RFC3261 transaction matching failed 1(7698) DEBUG: tm [t_lookup.c:659]: t_lookup_request(): no transaction found 1(7698) DEBUG: tm [t_funcs.c:284]: t_relay_to(): forwarding ACK statelessly 1(7698) DEBUG: <core> [core/md5utils.c:67]: MD5StringArray(): MD5 calculated: 8f25665e8c867e73b07da3cb1f702171 1(7698) DEBUG: <core> [core/msg_translator.c:159]: check_via_address(): (10.100.10.67, 10.100.10.67, 0) 1(7698) DEBUG: <core> [core/forward.c:557]: forward_request(): Sending: ACK sip:34606147628@10.100.10.67:5060 SIP/2.0 Via: SIP/2.0/UDP 52.88.180.226:5069;branch=z9hG4bKdaff.8f25665e8c867e73b07da3cb1f702171.0 Via: SIP/2.0/UDP 10.100.10.67:5061;branch=z9hG4bK-8173-1-0 From: "BTS" <sip:34917019888@10.100.10.67:5061>;tag=8173SIPpTag001 To: "Test" <sip:34606147628@10.100.10.67:5069>;tag=3 Call-ID: 1-8173(a)10.100.10.67 CSeq: 1 ACK Max-Forwards: 70 Subject: 1 Content-Length: 0 . 1(7698) DEBUG: <core> [core/forward.c:559]: forward_request(): orig. len=323, new_len=414, proto=1 1(7698) INFO: <script>: hereeeeeeeeeee 1(7698) DEBUG: <core> [core/receive.c:289]: receive_msg(): request-route executed in: 109 usec 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) 1(7698) DEBUG: <core> [core/receive.c:378]: receive_msg(): cleaning up 1(7698) DEBUG: <core> [core/udp_server.c:492]: udp_rcv_loop(): received on udp socket: (106/100/338) [[BYE sip:34606147628@10.100.10.67:5060 SIP/2.0 0D 0A Via: SIP/2.0/UDP 10.100.10.67:5061;branch=z9hG4bK-817]] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:604]: parse_msg(): SIP Request: 1(7698) DEBUG: <core> [core/parser/msg_parser.c:606]: parse_msg(): method: <BYE> 1(7698) DEBUG: <core> [core/parser/msg_parser.c:608]: parse_msg(): uri: <sip:34606147628@10.100.10.67:5060> 1(7698) DEBUG: <core> [core/parser/msg_parser.c:610]: parse_msg(): version: <SIP/2.0> 1(7698) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-8173-1-8>; state=16 1(7698) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 1(7698) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=2 1(7698) DEBUG: <core> [core/parser/msg_parser.c:494]: parse_headers(): this is the first via 1(7698) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=3 1(7698) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 1(7698) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [50]; uri=[sip:34606147628@10.100.10.67:5069] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): to body ["Test" <sip:34606147628@10.100.10.67:5069>] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <2> <BYE> 1(7698) DEBUG: <core> [core/receive.c:205]: receive_msg(): --- received sip message - request - call-id: [1-8173(a)10.100.10.67] - cseq: [2 BYE] 1(7698) DEBUG: <core> [core/receive.c:248]: receive_msg(): preparing to run routing scripts... 1(7698) INFO: <script>: hereeeeeeeeeee 1(7698) DEBUG: tm [t_lookup.c:1320]: t_newtran(): msg id=9 , global msg id=8 , T on entrance=(nil) 1(7698) DEBUG: <core> [core/parser/msg_parser.c:183]: get_hdr_field(): content_length=0 1(7698) DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header 1(7698) DEBUG: tm [t_lookup.c:477]: t_lookup_request(): start searching: hash=65450, isACK=0 1(7698) DEBUG: tm [t_lookup.c:435]: matching_3261(): RFC3261 transaction matching failed 1(7698) DEBUG: tm [t_lookup.c:659]: t_lookup_request(): no transaction found 1(7698) DEBUG: <core> [core/md5utils.c:67]: MD5StringArray(): MD5 calculated: 72541dabc2c0991bc1a987a58afe468a 1(7698) INFO: <script>: LEL 1(7698) DEBUG: <core> [core/msg_translator.c:159]: check_via_address(): (10.100.10.67, 10.100.10.67, 0) 1(7698) INFO: <script>: hereeeeeeeeeee 1(7698) DEBUG: tm [t_funcs.c:372]: t_relay_to(): new transaction forwarded 1(7698) DEBUG: <core> [core/receive.c:289]: receive_msg(): request-route executed in: 223 usec 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) 1(7698) DEBUG: <core> [core/receive.c:378]: receive_msg(): cleaning up 1(7698) DEBUG: <core> [core/udp_server.c:492]: udp_rcv_loop(): received on udp socket: (106/100/396) [[SIP/2.0 200 OK 0D 0A Via: SIP/2.0/UDP 52.88.180.226:5069;branch=z9hG4bKaaff.72541dabc2c0991bc1a987a58afe4]] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:614]: parse_msg(): SIP Reply (status): 1(7698) DEBUG: <core> [core/parser/msg_parser.c:616]: parse_msg(): version: <SIP/2.0> 1(7698) DEBUG: <core> [core/parser/msg_parser.c:618]: parse_msg(): status: <200> 1(7698) DEBUG: <core> [core/parser/msg_parser.c:620]: parse_msg(): reason: <OK> 1(7698) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKaaff.72541dabc2c0991bc1a987a58afe468a.0>; state=9 1(7698) DEBUG: <core> [core/parser/parse_via.c:2681]: parse_via(): parsing via: next via 1(7698) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-8173-1-8>; state=16 1(7698) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 1(7698) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=2 1(7698) DEBUG: <core> [core/parser/msg_parser.c:494]: parse_headers(): this is the first via 1(7698) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=3 1(7698) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 1(7698) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [50]; uri=[sip:34606147628@10.100.10.67:5069] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): to body ["Test" <sip:34606147628@10.100.10.67:5069>] 1(7698) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <2> <BYE> 1(7698) DEBUG: <core> [core/receive.c:205]: receive_msg(): --- received sip message - reply - call-id: [1-8173(a)10.100.10.67] - cseq: [2 BYE] 1(7698) DEBUG: tm [t_lookup.c:1019]: t_check_msg(): msg (0x7fe2c208a768) id=10 global id=9 T start=0xffffffffffffffff 1(7698) DEBUG: tm [t_lookup.c:897]: t_reply_matching(): t_reply_matching: hash 65450 label 0 branch 0 1(7698) DEBUG: tm [t_lookup.c:951]: t_reply_matching(): reply (0x7fe2c208a768) matched an active transaction (T=0x7fe2bd0ccd40)! 1(7698) DEBUG: tm [t_lookup.c:1088]: t_check_msg(): msg (0x7fe2c208a768) id=10 global id=10 T end=0x7fe2bd0ccd40 1(7698) DEBUG: tm [t_reply.c:2209]: reply_received(): org. status uas=0, uac[0]=0 local=0 is_invite=0) 1(7698) DEBUG: tm [t_reply.c:1265]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=200 1(7698) DEBUG: tm [t_reply.c:1788]: relay_reply(): branch=0, save=0, relay=0 icode=0 1(7698) DEBUG: <core> [core/msg_translator.c:2270]: generate_res_buf_from_sip_res(): old size: 396, new size: 310 1(7698) DEBUG: <core> [core/msg_translator.c:2288]: generate_res_buf_from_sip_res(): copied size: orig:107, new: 21, rest: 289 msg= SIP/2.0 200 OK Via: SIP/2.0/UDP 10.100.10.67:5061;branch=z9hG4bK-8173-1-8 From: "BTS" <sip:34917019888@10.100.10.67:5061>;tag=8173SIPpTag001 To: "Test" <sip:34606147628@10.100.10.67:5069>;tag=3 Call-ID: 1-8173(a)10.100.10.67 CSeq: 2 BYE Contact: <sip:10.100.10.67:5060;transport=UDP> Content-Length: 0 1(7698) DEBUG: tm [t_reply.c:1631]: cleanup_uac_timers(): RETR/FR timers reset 1(7698) DEBUG: <core> [core/receive.c:362]: receive_msg(): reply-route executed in: 74 usec 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(7698) DEBUG: <core> [core/xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) 1(7698) DEBUG: <core> [core/receive.c:378]: receive_msg(): cleaning up 3(7700) DEBUG: tm [h_table.c:129]: free_cell_helper(): freeing transaction 0x7fe2bd0c76d8 from timer.c:654 3(7700) DEBUG: tm [h_table.c:129]: free_cell_helper(): freeing transaction 0x7fe2bd0ccd40 from timer.c:654

3 5

dlflags interface deprecated
by AL S 13 Nov '18

13 Nov '18

Hi, We are running the old release of Kamailio 3.3.7 and getting the following notice log: kamailio: NOTICE: <core> [sr_module.c:627]: /usr/local/lib64/kamailio/modules_k/perl.so: exports dlflags interface is deprecated and it will not be supported in newer versions; consider using mod_register() instead Is there a way to fix this issue on the same Kamailio3.3.7 code? Thanks, AL

3 2

Execution Sequence of the Request_Route
by Abdulaziz Alghosh 13 Nov '18

13 Nov '18

Hello, I migrated from Kamailio 3.0.3 to Kamailio 5.1.4. and have two questions about the execution sequence of the Request_Route. The following condition trys to authenticate the first INVITE or REGISTER request which normally does not have the Authorization Header: route (auth) .. .. .. route[auth]{ if ( !proxy_authorize("myrealm", "subscriber") ) { proxy_challenge("myrealm", "0"); sl_reply_error(); exit; }; return; } For sure the Return Code of "proxy_authorize" will not be one (1) at the beginneing, so the proxy_challenge() will generate the "407 Proxy Authentication Required" request and the agent will send the INVITE or the REGISTER again with the Authorization-Header and the call or the registration attempt succeeds. During my observation to the new returned codes (values) of both proxy_authorize() and proxy_challenge() in Kamailio 5.1.4, I noticed the following differnet execution sequence of this "Request_Route". at Kamailio 3.0.3, and exactly after the proxy_challenge("myrealm", "0") command was executed, no more commands were excuted as if it jumped out of the if statement, and the proxy was wating for the new INVITE or REGISTER. Hence the sl_reply_error() did not send any error reply. I wrote several logging commad before and after the proxy_challenge("myrealm", "0"), so I am pretty sure that the proxy_challenge("myrealm", "0") is last excuted command to be excuted here. at Kamailio 5.1.4, the behaviour was more logical and the proxy executes commands which come under the proxy_challenge("myrealm", "0") like logging commands or sl_reply_error() which generates usually the error reply " 500 I'm terribly sorry, server error occurred". My first question is: May somebody explain this different behavior? My second Question: assume that the agent will send a wrong credentials and response, does the proxy_challenge("myrealm", "0") function terminate this dialouge with 401 Unauthorized Error or I must add the following "if condition" afterwards: if ( !proxy_authorize("myrealm", "subscriber") ) { proxy_challenge("myrealm", "0"); if (!$rc){ sl_reply_error(); '#### or I will be more specific sl_send_reply (code, reason) } exit; }; Thanks and best regards Abdulaziz Alghosh

2 1

Redirects - are q values required to implement serial forking?
by Mike Montgomery 12 Nov '18

12 Nov '18

We have a situation where we get a redirect from a carrier with a list of carriers to send the transaction to in the contact header. None of the carriers in the contact header have q values assigned. We have implemented our redirect handling logic using the UAC_REDIRECT module (get_redirects()) with help from TM module (t-load_contacts(), t_next_contact()). (failure_route code is at bottom of this note) We find that when we receive a contact header with q values, we can launch requests serially using t_next_contact() and individually work through the contact list until we get a positive response. When we get a redirect with no q values in the contact header we find that all contacts get loaded and launched in parallel. Although not stated in the documentation, it appears that in order to implement serial contact redirect handling, q values must be present in the contact header. Based on this assumption, and knowing that the redirecting carrier cannot include q values, we have attempted to modify the contact header and find that we are unable to add q values to the contact header. We have attempted without success to use remove_hf("Contact”) followed by append_hf("Contact: $(avp(contact)) \r\n”); where $(avp(contact)) is the avp where we have rewritten the contact header to include q values. Any guidance on how to handle this issue is appreciated. _________________________________________________________- Our failure route code: if (t_check_status("3[0-9][0-9]")) { xlog("L_INFO", "----> REDIRECT Contact Header --> $T_rpl($ct) --- Method $rm --- From $T_req($fu) --- To $T_req($tu) --- Destination $T_req($du) \n"); get_redirects("*:*"); if (!t_load_contacts()) { xlog("L_INFO", "-t_load_contacts() FAILED /n"); }; xlog("L_INFO", "####->LOAD CONTACT LIST $xavp(tm_contacts[0]=>uri) \n"); if (t_next_contacts()) { xlog("L_INFO", "-t_next_contact() loaded successfully /n"); xlog("L_INFO", "####-> NEXT CONTACT LIST $xavp(tm_contacts[0]=>uri) \n"); }; t_on_failure("serial"); if(t_relay()){ xlog("L_INFO","----->Relayed in MANAGE_FAILURE --- Method $rm --- Contact Header $ct \n"); }; exit; } else { xlog("L_INFO","-----> NO REDIRECT --- Method $rm --- Contact Header $ct \n"); exit; } failure_route["serial"] { xlog ("L_INFO", "Entered failure_route(serial) /n"); if (t_check_status("3[0-9][0-9]")) { xlog("L_INFO", "----> DOUBLE REDIRECT Contact Header --> $T_rpl($ct) --- Method $rm --- From $T_req($fu) --- To $T_req($tu) --- Destination $T_req($du) \n"); $xavp(tm_contacts[*]) = $null; xlog("L_INFO", "XXX-> tm_contacts reset to Null $xavp(tm_contacts[0]=>uri)\n"); get_redirects("*:*"); if (!t_load_contacts()) { xlog("L_INFO", "-t_load_contacts() FAILED /n"); }; xlog("L_INFO", "####-> LOAD 2 CONTACT LIST $xavp(tm_contacts[0]=>uri) \n"); } if (!t_next_contacts()) { exit; } xlog("L_INFO", "-t_next_contact() loaded successfully /n"); xlog("L_INFO", "####-> NEXT CONTACT LIST $xavp(tm_contacts[0]=>uri) \n"); t_on_failure("serial"); if(t_relay()){ xlog("L_INFO","----->Relayed in failure_route(serial) --- Method $rm --- Contact Header $ct \n"); }; } Best Regards, Mike Montgomery Principal, Trusted Technical Services, Inc. o - 919 924 4703 x3000 m - 919 539 6133

2 1

Adding Perl Module to kamailio-5.1.6
by AL S 12 Nov '18

12 Nov '18

I am trying to install Perl module on Ubuntu 18.04 / Kamailio kamailio-5.1.6 and getting the following response: $ sudo apt-get install kamailio-perl-modules Reading package lists... Done Building dependency tree Reading state information... Done kamailio-perl-modules is already the newest version (5.1.2-1ubuntu2). You might want to run 'apt --fix-broken install' to correct these. The following packages have unmet dependencies: kamailio-perl-modules : Depends: kamailio (= 5.1.2-1ubuntu2) but it is not installable E: Unmet dependencies. Try 'apt --fix-broken install' with no packages (or specify a solution). Is there a better way to add perl module? PS: libperl-dev already installed. Thanks, AL

2 1

siptrace module doesn't close db connections
by Roman Dissauer 12 Nov '18

12 Nov '18

Hi, it seems that siptrace module doesn’t close db connections. I’m on Kamailio 5.1.4 from official debian stretch repository with postgresql db module enabled. siptrace config: modparam("siptrace", "db_url", DBURL) modparam("siptrace", "duplicate_uri", "sip:193.32.51.189:9060") modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "hep_version", 3) modparam("siptrace", "hep_capture_id", 1001) modparam("siptrace", "trace_mode", 1) Here you can see the open connections from the kamailio host: # select client_addr, state, query from pg_stat_activity where client_addr = '10.23.101.61'; client_addr | state | query --------------+-------+---------------------------------------------------------------- 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' 10.23.101.61 | idle | select table_version from version where table_name='sip_trace' (31 Zeilen) another example, other installation with same kamailio version 5.1.4: # select client_addr, state, query from pg_stat_activity where client_addr = '10.23.102.1'; client_addr | state | query -------------+-------+---------------------------------------------------------------- 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' 10.23.102.1 | idle | select table_version from version where table_name='sip_trace' (39 Zeilen) Thanks Roman

2 2

Single session start
by José Antonio Gutiérrez Delgado 12 Nov '18

12 Nov '18

Hi, I would like to know if it is possible to make the last register of a user the active register and if possible the only one. I'm with an Android application and sometimes I can not deregister properly before the new registration, for example when the network changes, so there are times when the hanging signals do not work correctly when there is more than one register for a user. Is it possible to avoid that there is more than one register for a user? Is it possible to configure the last register to delete the previous ones or to be a single register? Thanks in advance and I await your answers. -- José Antonio Gutiérrez Delgado Responsable técnico Oficina: (+34) 923040031 <http://www.arsoft-company.com/> *Este mensaje se dirige exclusivamente a su destinatario/s y puede contener información privilegiada o confidencial. Si no es vd. el destinatario indicado, queda notificado de que la utilización, divulgación y/o copia sin autorización está prohibida en virtud de la legislación vigente. Si ha recibido este mensaje por error, le rogamos que nos lo comunique inmediatamente por esta misma vía y proceda a su destrucción.* *Antes de imprimir, piensa si es necesario. Cuidar el medioambiente es tarea de todos.*

3 2

Custom authentication
by João Sávio 12 Nov '18

12 Nov '18

Hello! I was reading the documentation of Kamailio, And, although there are some authentication modules, I couldn't find a way to implement a custom authentication for SIP Register. Something that allows me, for example, to call a web server and returns if the user is authenticated, or the user data. Am I missing anything? Do you know if it's possible to do this using another way? Thanks João

2 2

high availability and load balancing
by eyas barhouk 10 Nov '18

10 Nov '18

hello dears, i'm using kamailio 5.1 as IMS , and i have the following servers: * p-cscf * i-cscf-1 ,icscf-2 * s-cscf-1,scscf-2 i need to do high availability and load balancing between my servers , but i have some questions like do i have to use keepalived or dispatcher module and what the benefits of each one of them ,and on load balancing method should i do it for the register requests or only for invite requests or just for every thing ?? if any one has an idea or any thing to guide me i will be thankful thanks in advance and regards

3 3

Client Registration and Call over Kamailio
by serkan celik 09 Nov '18

09 Nov '18

Hi all, I am using Kamailio as a proxy in front of a 3rd part Open IMS (Not IMS modules in kamailio). I have defined Kamailio as an outbound proxy in Soft clients(I have configured IMS users info in Soft Clients). When I send register from client to IMS, Register message is sent to IMS over Kamailio. Kamailio forwards this received Register to the 3rd party IMS. Finally the client is registered into IMS over Kamailio. However when I try to make a call from registered users (both registered over Kamailio), IMS sends 403 Forbidden - "Not Registered! You must register first with a S-CSCF" to Kamailio. When I debugged the issue I found the reason why IMS sends 403. When IMS receives an Invite message from Kamailio it checks whether source IP which Invite is sent registered or not. As I make registration over Kamailio without changing contact header, IMS registers the user with the contact IP (which is the client IP). So to make a call from Soft client (IMS users) over Kamailio I also need to register Kamailio source IP as well.To deal with the issue I have changed the contact address of Register message coming from Soft client with Kamailio IP on Kamailio. Then IMS registered Kamailio IP as a contact when registering IMS user. But this time when I make a call from Soft client over Kamailio, IMS sends Termination Invite to Kamailio. So to finalize the call Kamailio has to route this termination Invite to the terminating IMS user . Is there any better approach to overcome this issue. Which is the better approach to use Kamailio in front of IMS. I want to both register and make a call from Soft clients over Kamailio to the IMS. Regards, Serkan CELIK

3 5

systemd kamailio.service - RFC Wants=network-online.target
by Kevin Olbrich 08 Nov '18

08 Nov '18

Hi! The systemd kamailio.service file for debian (in my case, it's stretch) has the following unit options set: [Unit] > Description=Kamailio (OpenSER) - the Open Source SIP Server > After=network.target It would be better to change this to: [Unit] > Description=Kamailio (OpenSER) - the Open Source SIP Server > Wants=network-online.target > After=network-online.target Why? With network.target the service starts as soon as the network setup is started, not explicitly after configuring (=IP) it [1]. This creates a race condition on very fast hardware like mine (booting incl. all services takes 10s). Even with static IPs, kamailio often fails directly at boot because eth1 (connection to DB cluster) is not up. [I have never met this problem on systems which ran on HDDs.] Setting "network-online.target" as Wants + After makes sure, Kamailio is started after all interfaces are ready to serve. Units that strictly require a configured network connection should pull in > network-online.target (via a Wants= type dependency) and order themselves > after it. This target unit is intended to pull in a service that delays > further execution until the network is sufficiently set up. What precisely > this requires is left to the implementation of the network managing service. > Perfect example for this is the Shorewall package (firewall solution): [Unit] > Description=Shorewall IPv4 firewall > Wants=network-online.target > After=network-online.target > Conflicts=iptables.service firewalld.service RFC Kind regards Kevin [1] https://www.freedesktop.org/software/systemd/man/systemd.special.html

3 6

cfg_lock_helper(): lock set not initialized
by Kjeld Flarup 08 Nov '18

08 Nov '18

Hi What does this error actually mean? I get it with 5.1.4 Nov 6 13:27:02 scantronpbx /usr/sbin/kamailio[2149]: ERROR: cfgutils [cfgutils.c:708]: cfg_lock_helper(): lock set not initialized (attempt to do op: 1 on: u1) -- -------------------- 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

2 1

Phone does not set "Expire-header" but "Contact expire", immediately expires
by Kevin Olbrich 07 Nov '18

07 Nov '18

Hi! I have a phone (Mitel 6930) that does not send an Expire-header but "expire" in Contact during REGISTER: Session Initiation Protocol (REGISTER) > Request-Line: REGISTER sip:pbx.example.com SIP/2.0 > Message Header > Via: SIP/2.0/TLS 192.168.6.16:5061;branch=xxxxxxxxxxxxxxxx;rport > Route: <sip:pbx.example.com;lr=lr;transport=tls> > Max-Forwards: 70 > From: "Example" <sip:121212_6666@pbx.example.com>;tag=b0cec92c3e > To: "Example" <sip:121212_6666@pbx.example.com> > Call-ID: aaaaaaaaaaaaaaaaa > CSeq: 1190442896 REGISTER > Accept-Language: de > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, > PRACK, SUBSCRIBE, INFO, PUBLISH > Allow-Events: talk, hold, conference, LocalModeStatus > Contact: "Example" <sip:121212_6666@192.0.2.1:62336 > >;+sip.instance="<urn:uuid:00000000-0000-1000-8000-08000FB888F5>";expires=300 > Supported: path, gruu > User-Agent: Mitel 6930/5.0.0.2041 > Aastra-Line: 1 > Aastra-Mac: AAAAAAAAAAAA > Content-Length: 0 This results in successful register, kamailio forwards REGISTER to asterisk (endpoint registered) and immediatly kamailio sends a new REGISTER with Expire set to 0 to asterisk: Session Initiation Protocol (REGISTER) > Request-Line: REGISTER sip:192.168.99.112:5060 SIP/2.0 > Message Header > Via: SIP/2.0/UDP 192.168.99.111:5060 > ;branch=z9hG4bK9f9e.599cf243000000000000000000000000.0 > To: <sip:121212_6666@192.168.99.112> > From: <sip:121212_6666@192.168.99.112 > >;tag=7e4f6993bfbf831f6b016fea6683cdca-b98a > CSeq: 10 REGISTER > Call-ID: 2bc5fd1061b31723-1350(a)192.168.99.111 > Max-Forwards: 70 > Content-Length: 0 > User-Agent: kamailio (5.1.6 (x86_64/linux)) > Contact: <sip:121212_6666@192.168.99.111:5060> > Expires: 0 The phone is then unregistered - the phone also shows "no service". It looks like Kamailio generates this message on it's own (and I don't know why). I am still learning to implement kamailio as proxy. Reading the docs, I would expect Kamailio to automatically eval the correct parameter: > If the processed message contains neither Expires HFs nor expires contact > parameters, this value will be used for newly created usrloc records. [1] What did I miss? Phones which use the Expire-header are working fine. Kind regards Kevin [1] default_expires - http://www.kamailio.org/docs/modules/5.2.x/modules/registrar.html

3 10

busy lamp field
by Kevin Olbrich 06 Nov '18

06 Nov '18

Hi! Can someone share his kamailio.cfg with working busy lamp field in a kamailio-as-proxy scenario? My config is based on the example shipped with 5.1.x and asterisk as the backend. Calling works fine but PUBLISH / SUBSCRIBE is handled by kamailio itself. I am not sure but this might be the part where I should relay the request to asterisk(?): #!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 Currently the lights are "broken", some phones show them on, some off. Kind regards Kevin

2 1

Kamailio SQL error
by Marcos Pytel 06 Nov '18

06 Nov '18

Hello team! Im using kamailio in sipwise! I get this errors: Nov 5 11:51:34 sipwise proxy[1943]: ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query Nov 5 11:51:34 sipwise proxy[1943]: ERROR: presence [subscribe.c:2514]: update_db_subs_timer(): unsuccessful sql insert Nov 5 11:51:34 sipwise proxy[1943]: ERROR: db_mysql [km_dbase.c:127]: db_mysql_submit_query(): driver error on query: Duplicate entry '6c23f96-0001-0003-0030(a)10.7.255.250-f3067022b00c564156251ba2f28f' for key 'active_watchers_idx' (1062) How can I fix it? Thank you! Marcos.

2 2

db_mongodb_get_columns(): field [username] not found in result iterator
by Denys Pozniak 06 Nov '18

06 Nov '18

Hello! I want to keep user location in mongodb, but Kamailio does not start well. Configuration: loadmodule "usrloc.so" loadmodule "db_mongodb.so" modparam("usrloc", "db_insert_null", 1) modparam("usrloc", "db_url", "mongodb://mongodb.test/kamailio") modparam("usrloc", "db_mode", 2) modparam("usrloc", "timer_interval", 30) modparam("usrloc", "desc_time_order", 1) Mongodb: rs0:PRIMARY> use kamailio switched to db kamailio rs0:PRIMARY> db.location.find() { "_id" : ObjectId("5be024d6b9d8febf4ae19681"), "name" : "location", "version" : 9, "columns" : { "id" : { "type" : "int", "default" : null, "null" : false }, "ruid" : { "type" : "string", "default" : "", "null" : false }, "username" : { "type" : "string", "default" : "", "null" : false }, "domain" : { "type" : "string", "default" : null, "null" : true }, "contact" : { "type" : "string", "default" : "", "null" : false }, "received" : { "type" : "string", "default" : null, "null" : true }, "path" : { "type" : "string", "default" : null, "null" : true }, "expires" : { "type" : "datetime", "default" : "2030-05-28 21:32:15", "null" : false }, "q" : { "type" : "double", "default" : 1, "null" : false }, "callid" : { "type" : "string", "default" : "Default-Call-ID", "null" : false }, "cseq" : { "type" : "int", "default" : 1, "null" : false }, "last_modified" : { "type" : "datetime", "default" : "2000-01-01 00:00:01", "null" : false }, "flags" : { "type" : "int", "default" : 0, "null" : false }, "cflags" : { "type" : "int", "default" : 0, "null" : false }, "user_agent" : { "type" : "string", "default" : "", "null" : false }, "socket" : { "type" : "string", "default" : null, "null" : true }, "methods" : { "type" : "int", "default" : null, "null" : true }, "instance" : { "type" : "string", "default" : null, "null" : true }, "reg_id" : { "type" : "int", "default" : 0, "null" : false }, "server_id" : { "type" : "int", "default" : 0, "null" : false }, "connection_id" : { "type" : "int", "default" : 0, "null" : false }, "keepalive" : { "type" : "int", "default" : 0, "null" : false }, "partition" : { "type" : "int", "default" : 0, "null" : false } } } { "_id" : ObjectId("5be02a77175610be86a1a589"), "username" : "dummy_user" } Syslog: Nov 5 12:51:01 dev-rtpengine /usr/sbin/kamailio[21543]: ERROR: db_mongodb [mongodb_dbase.c:416]: db_mongodb_get_columns(): field [username] not found in result iterator Nov 5 12:51:01 dev-rtpengine /usr/sbin/kamailio[21543]: ERROR: db_mongodb [mongodb_dbase.c:815]: db_mongodb_store_result(): failed to set the columns Nov 5 12:51:01 dev-rtpengine /usr/sbin/kamailio[21543]: ERROR: db_mongodb [mongodb_dbase.c:1018]: db_mongodb_query(): failed to store result Nov 5 12:51:01 dev-rtpengine /usr/sbin/kamailio[21543]: ERROR: db_mongodb [mongodb_dbase.c:1027]: db_mongodb_query(): failed to do the query Nov 5 12:51:01 dev-rtpengine /usr/sbin/kamailio[21543]: ERROR: usrloc [udomain.c:457]: preload_udomain(): db_query failed Nov 5 12:51:01 dev-rtpengine /usr/sbin/kamailio[21543]: ERROR: usrloc [usrloc_mod.c:430]: child_init(): child(1): failed to preload domain 'location' Nov 5 12:51:01 dev-rtpengine /usr/sbin/kamailio[21543]: ERROR: <core> [core/sr_module.c:946]: init_mod_child(): error while initializing module usrloc (/usr/lib64/kamailio/modules/usrloc.so) (idx: 1 rank: 1 desc: [udp receiver child=0 sock=10.10.10.10:5060]) Nov 5 12:51:01 dev-rtpengine /usr/sbin/kamailio[21543]: ERROR: <core> [core/pt.c:340]: fork_process(): init_child failed for process 1, pid 21543, "udp receiver child=0 sock=10.10.10.10:5060" Nov 5 12:51:01 dev-rtpengine /usr/sbin/kamailio[21543]: CRITICAL: <core> [main.c:1607]: main_loop(): Cannot fork 2018-11-05T13:07:19.117+0100 D COMMAND [conn367] run command kamailio.$cmd { find: "version", filter: { table_name: "location" }, projection: { table_version: 1 } } 2018-11-05T13:07:19.118+0100 I COMMAND [conn367] command kamailio.version command: find { find: "version", filter: { table_name: "location" }, projection: { table_version: 1 } } planSummary: COLLSCAN keysExamined:0 docsExamined:2 cursorExhausted:1 numYields:0 nreturned:1 reslen:148 locks:{ Global: { acquireCount: { r: 2 } }, Database: { acquireCount: { r: 1 } }, Collection: { acquireCount: { r: 1 } } } protocol:op_query 0ms 2018-11-05T13:07:19.118+0100 D COMMAND [conn367] run command kamailio.$cmd { find: "location", filter: { username: "dummy_user" }, projection: { username: 1 } } 2018-11-05T13:07:19.119+0100 I COMMAND [conn367] command kamailio.location command: find { find: "location", filter: { username: "dummy_user" }, projection: { username: 1 } } planSummary: COLLSCAN keysExamined:0 docsExamined:2 cursorExhausted:1 numYields:0 nreturned:1 reslen:155 locks:{ Global: { acquireCount: { r: 2 } }, Database: { acquireCount: { r: 1 } }, Collection: { acquireCount: { r: 1 } } } protocol:op_query 0ms 2018-11-05T13:07:19.178+0100 D COMMAND [conn368] run command kamailio.$cmd { find: "location", filter: {}, projection: { username: 1, contact: 1, expires: 1, q: 1, callid: 1, cseq: 1, flags: 1, cflags: 1, user_agent: 1, received: 1, path: 1, socket: 1, methods: 1, last_modified: 1, ruid: 1, instance: 1, reg_id: 1, server_id: 1, connection_id: 1, keepalive: 1 } } 2018-11-05T13:07:19.179+0100 I COMMAND [conn368] command kamailio.location command: find { find: "location", filter: {}, projection: { username: 1, contact: 1, expires: 1, q: 1, callid: 1, cseq: 1, flags: 1, cflags: 1, user_agent: 1, received: 1, path: 1, socket: 1, methods: 1, last_modified: 1, ruid: 1, instance: 1, reg_id: 1, server_id: 1, connection_id: 1, keepalive: 1 } } planSummary: COLLSCAN keysExamined:0 docsExamined:2 cursorExhausted:1 numYields:0 nreturned:2 reslen:180 locks:{ Global: { acquireCount: { r: 2 } }, Database: { acquireCount: { r: 1 } }, Collection: { acquireCount: { r: 1 } } } protocol:op_query 0ms [root@dev-rtpengine kamailio]# kamailio -v version: kamailio 5.1.6 (x86_64/linux) 7d1964 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 ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 7d1964 compiled on 11:10:15 Nov 4 2018 with gcc 4.8.5 -- BR, Denys Pozniak

2 5

Fwd: Fw: kamailio5.0.0 support presence
by chaos wu 06 Nov '18

06 Nov '18

Hi Kamailio expert, i want to use kamailio to simulate presence server, after receive subscribe, i need kamailio reply 200 OK, and then send out Notify, but now, it just forward the subscribe, don't reply, could you help check my cfg? thanks in advance. BRs

2 1

Development is open for future release v5.3.x series
by Daniel-Constantin Mierla 06 Nov '18

06 Nov '18

Hello, with the creation of branch 5.2 done this morning, the master branch is from now on open for adding new features, to be part of future release series v5.3.x. Any bug fix committed to master that applies to 5.2.x or older stable branches should be backported as usual with git cherry-pick to appropriate branches like 5.2 or 5.1. Expect that v5.2.0 will be released in about 2-3 weeks from now. Based on the workflow used during the past years, the next future release v5.3.0 should be out after another 8-10 months of development, plus 1-2 months of testing, so sometime in the summer or autumn of 2019. 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 0

Branch 5.2 has been created
by Daniel-Constantin Mierla 06 Nov '18

06 Nov '18

Hello, the branch 5.2 has been created, to be used for releasing v5.2.x series. To check out this branch, the following commands can be used: git clone https://github.com/kamailio/kamailio kamailio-5.2 cd kamailio-5.2 git checkout -b 5.2 origin/5.2 Pushing commits in this branch: git push origin 5.2:5.2 Note that 5.2 is an official stable branch, so only bug fixes, missing kemi exports (discuss on sr-dev if not sure) or improvements to documentation or helper tools can be pushed to this branch. As usual, if there is a bug fixed, commit and push first to master branch and then cherry pick to 5.2 branch. In few weeks, the first release from branch 5.2 will be out, respectively Kamailio v5.2.0. 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 0

Use of SCTP on public SIP server
by Kevin Olbrich 06 Nov '18

06 Nov '18

Hi! I read about the SCTP module for Kamailio and I am not sure if I should load it (currently I am not). Does this provide any benefit when Kamailio is used as proxy in front of Asterisk? At least from Wikipedia and the kamailio docs, it sounds like I can load it and it does it's magic. The multihomed part caught my attention. Kinds regards Kevin

3 2

How register with STORE_PLAINTEXT_PW=0
by José Antonio Gutiérrez Delgado 05 Nov '18

05 Nov '18

Hello, I recently discovered that the Kamailio database stores the user's password directly. I have found that by changing STORE_PLAINTEXT_PW = 0 in the configuration file, it stops storing the password in the database, but if I make this change I can no longer register the users in Kamailio. How should I register if I do not store the password? Have I missed a step? I have read that it is related to the HA1 field but it has not been clear how to do it. I hope your help. Thank you. -- José Antonio Gutiérrez Delgado Responsable técnico Oficina: (+34) 923040031 <http://www.arsoft-company.com/> *Este mensaje se dirige exclusivamente a su destinatario/s y puede contener información privilegiada o confidencial. Si no es vd. el destinatario indicado, queda notificado de que la utilización, divulgación y/o copia sin autorización está prohibida en virtud de la legislación vigente. Si ha recibido este mensaje por error, le rogamos que nos lo comunique inmediatamente por esta misma vía y proceda a su destrucción.* *Antes de imprimir, piensa si es necesario. Cuidar el medioambiente es tarea de todos.*

2 2

Storing in-progress dialogs in database
by jenus＠cyberchaos.nl 05 Nov '18

05 Nov '18

Hi all, I'm testing with the new dlg_db_load_callid function to replicate dialogs to another (standby) kamailio instance on failover. This works fine for calls that are connected. But i'm missing the dialogs in "ringing" state. Looks like dialogs are written to database by the dialog module when we receive the 200OK/ACK for the invite transaction. The result is that the call gets connected on the other kamailio instance but without a dialog. Is it possible to store the early_dialogs that are in rining state in the database? If that is not the case can i still create a dialog when i only receive the 200OK/ACK to the INVITE transaction (INVITE was processed by another instance) since TM is not aware of the INVITE transaction? I also tried to reject the 200OK to the INVITE in the on reply route, but running into the same issue that TM does not know the INVITE transaction. Thanks, Jan

2 1

string in ims_diameter_server
by Iman Mohammadi 04 Nov '18

04 Nov '18

We have a problem putting data to an AVP object. The data type is byte array (:byte[]). If we convert that array to an integer and put it in 'int32' field then the module just gets the first 4 bytes. If we convert it to base64 or hexString and put it in 'string' field then it treats it as a regular string. And any other field beside 'int32' and 'string' will be ignored. So the question is, how we put a byte array inside an AVP and send it to the module so the diameter gets the data as a byte array? Best Regards Iman Mohammadi

1 0

Branching GIT repository for 5.2.x release series
by Daniel-Constantin Mierla 03 Nov '18

03 Nov '18

Hello, if there is no other opinion, I am considering to create git branch 5.2 at the beginning of next week (likely on Monday evening), to be used for 5.2.x release series. >From that moment, the master branch can get new features and the appropriate bug fixes have to be backported to branch 5.2 as well. With this step, the first release in 5.2.x series, respectively v5.2.0, should be out in about 2-3 weeks. 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

2 1

Crash ims_diameter_server
by Iman Mohammadi 03 Nov '18

03 Nov '18

When the below format is sent from rest , kamailio crashes List[{ List[{ List[{ ]} ]} ]} (Json with 3 nested lists), With 2 lists it works properly, When diameter is translated to json with 3 lists by this module it also works properly, Json format is correct too. gdb: gdb) bt full #0 0x00007f5e23c4a11e in diameterserver_add_avp (m=0x0, d=0x7f5e1b5c2240 "", len=12, avp_code=431, flags=64, vendorid=0, data_do=2, func=0x7f5e23c51d6c <__FUNCTION__.20148> "parselist") at avp_helper.c:208 avp = 0x7f5e1b5c3bd0 __FUNCTION__ = "diameterserver_add_avp" #1 0x00007f5e23c4d0c8 in parselist (response=0x0, list=0x7fff9f21a8d0, item=0x11b67d0, level=2) at avp_helper.c:309 i = 1 flags = 64 x = "p\250!\237" avp_list = {head = 0x0, tail = 0x0} avp_list_s = {s = 0x7f5e1b5c2240 "", len = 12} __FUNCTION__ = "parselist" #2 0x00007f5e23c4cffc in parselist (response=0x7f5e1b5c3ab8, list=0x0, item=0x11b6550, level=1) at avp_helper.c:304 subitem = 0x11b67d0 i = 0 flags = 64 x = "\000\000\000" avp_list = {head = 0x0, tail = 0x0} avp_list_s = {s = 0x7fff9f21a8f0 "@\251!\237\377\177", len = 600070897} __FUNCTION__ = "parselist" #3 0x00007f5e23c4db3a in addAVPsfromJSON (response=0x7f5e1b5c3ab8, json=0x7f5e23e53950 <responsejson>) at avp_helper.c:349 subitem = 0x11b6550 i = 4 __FUNCTION__ = "addAVPsfromJSON" root = 0x11b4210 #4 0x00007f5e23c3fbdb in callback_cdp_request (request_in=0x7f5e1b5c19b0, param=0x0) at ims_diameter_server.c:193 fmsg = 0xab7840 <_faked_msg> backup_rt = 1 ctx = {rec_lev = 0, run_flags = 0, last_retcode = 1, jmp_env = {{__jmpbuf = {1, -5713032302318786866, 7971288, 7971288, 140042162528420, 0, -5713032302339758386, 5712822384154044110}, __mask_was_saved = 0, __saved_mask = {__val = { 140735863172072, 127, 0, 140042162532340, 140042389578761, 4222451713, 140042162532340, 140042162532340, 140042162532340, 140042162532340, 140042162532358, 140042162532467, 140042162532340, 140042162532467, 0, 0}}}}} response = 0x7f5e1b5c3ab8 __FUNCTION__ = "callback_cdp_request" #5 0x00007f5e24a705c0 in api_callback (p=0x7f5e1b598f50, msg=0x7f5e1b5c19b0, ptr=0x0) at api_process.c:83 t = 0x7f5e1b598f50 auto_drop = 32767 h = 0x7f5e1b5b3358 ---Type <return> to continue, or q <return> to quit--- x = {type = (unknown: 2669784000), handler = {requestHandler = 0x7f5e23c3eed1 <callback_cdp_request>, responseHandler = 0x7f5e23c3eed1 <callback_cdp_request>}, param = 0x3ee9f21ac10, next = 0x7f5e1b5988b8, prev = 0x19f210069} type = REQUEST_HANDLER rsp = 0x7f5e1b5c19b0 __FUNCTION__ = "api_callback" #6 0x00007f5e24a857d7 in worker_process (id=5) at worker.c:346 t = {p = 0x7f5e1b598f50, msg = 0x7f5e1b5c19b0} cb = 0x7f5e1b59df30 r = 128 __FUNCTION__ = "worker_process" #7 0x00007f5e24a62a8e in diameter_peer_start (blocking=0) at diameter_peer.c:242 pid = 0 k = 5 p = 0x36 __FUNCTION__ = "diameter_peer_start" #8 0x00007f5e24a559bc in cdp_child_init (rank=0) at cdp_mod.c:243 __FUNCTION__ = "cdp_child_init" #9 0x0000000000547e54 in init_mod_child (m=0x7f5e28656fb8, rank=0) at core/sr_module.c:943 __FUNCTION__ = "init_mod_child" #10 0x0000000000547b16 in init_mod_child (m=0x7f5e28657c38, rank=0) at core/sr_module.c:939 __FUNCTION__ = "init_mod_child" #11 0x0000000000547b16 in init_mod_child (m=0x7f5e28658010, rank=0) at core/sr_module.c:939 __FUNCTION__ = "init_mod_child" #12 0x0000000000547b16 in init_mod_child (m=0x7f5e28658790, rank=0) at core/sr_module.c:939 __FUNCTION__ = "init_mod_child" #13 0x0000000000547b16 in init_mod_child (m=0x7f5e28658d68, rank=0) at core/sr_module.c:939 __FUNCTION__ = "init_mod_child" #14 0x00000000005481e6 in init_child (rank=0) at core/sr_module.c:970 No locals. #15 0x00000000004250f5 in main_loop () at main.c:1701 i = 8 pid = 1964 si = 0x0 si_desc = "udp receiver child=7 sock=10.25.5.23:5060\000y\000\000\000\000\000x\333i(^\177\000\000@\264!\237\377\177", '\000' <repeats 18 times>, "`\263!\237\377\177\000\000\347L\340(^\177\000\000\360\260!\237\377\177\000\000\060\000\000\000\060\000\000\000\260\261!\237\377\177\000\000\350\226i(^\177\000" nrprocs = 8 ---Type <return> to continue, or q <return> to quit--- woneinit = 1 __FUNCTION__ = "main_loop" #16 0x000000000042b99b in main (argc=13, argv=0x7fff9f21b448) at main.c:2638 cfg_stream = 0x111c020 c = -1 r = 0 tmp = 0x7fff9f21bf69 "" tmp_len = 32606 port = 693647725 proto = 0 options = 0x74cd90 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:Y:" ret = -1 seed = 898203166 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x7fff9f21b300 p = 0x7fff9f21b4b8 "\207\277!\237\377\177" st = {st_dev = 15, st_ino = 13240, st_nlink = 2, st_mode = 16877, st_uid = 110, st_gid = 116, __pad0 = 0, st_rdev = 0, st_size = 60, st_blksize = 4096, st_blocks = 0, st_atim = {tv_sec = 1540797448, tv_nsec = 544589798}, st_mtim = {tv_sec = 1540797905, tv_nsec = 662909589}, st_ctim = {tv_sec = 1540798045, tv_nsec = 563947001}, __glibc_reserved = {0, 0, 0}} __FUNCTION__ = "main" (gdb) (gdb) Where is problem? Thanks

3 4

Store TLS handshake date from Kamailio to an external DB
by Toffi Bossol 02 Nov '18

02 Nov '18

Dear community, I am new in Kamailio hence the following question: Does any one know how to tell Kamailio to store the TLS session data in an external DB? Many Thanks in advance! BR; Toffi

3 7

Installing rtpengine from RPM on CentOS
by Pan Christensen 02 Nov '18

02 Nov '18

Hello! According to https://github.com/sipwise/rtpengine/blob/master/el/README.el.md , we should be able to install rtpengine from a CentOS repository using yum. We're unable to find the packages. Any suggestions? Pan B. Christensen Developer Phonect AS Mail: pan.christensen(a)phonect.no<mailto:pan.christensen@phonect.no> + 47 41 88 88 00 [mail_footer]

5 5

Parse Header error when send Register to P-CSCF using ims-pcscf
by Hưng Nguyễn Tiến 02 Nov '18

02 Nov '18

Hi, i'm newbie in kamailio, I tried installed ims kamailio from ng-repository packages, Then when I tried to send register to SIP server using SIPp, I received this log [Proxy-CSCF][28505]: ERROR: <core> [core/parser/msg_parser.c:331]: parse_headers(): bad header field [(null)] [Proxy-CSCF][28500]: ERROR: <core> [core/parser/msg_parser.c:331]: parse_headers(): bad header field [(null)] [Proxy-CSCF][28496]: WARNING: <core> [core/receive.c:192]: receive_msg(): parsing relevant headers failed [Proxy-CSCF][28505]: message repeated 2 times: [ WARNING: <core> [core/receive.c:192]: receive_msg(): parsing relevant headers failed] [Proxy-CSCF][28492]: ERROR: <core> [core/parser/msg_parser.c:675]: parse_msg(): ERROR: parse_msg: message=<#002#020#002#021#023�#023��z'��z'#006��[%�#007> Thank you -- Nguyễn Tiến Hưng

2 1

Diversion headers access and message too long error
by Joan Salvatella 01 Nov '18

01 Nov '18

Hello, We are working on a project that's using a Twilio SIP trunk that diverts calls from the PSTN to our cluster that is hosting two active-active Kamailio 5.1 instances that redirect traffic to several Asterisk 15.4 instances. On this setup we are facing 2 issues: - *Diversion headers access: *Currently, Kamailio only supports access to the last diversion header but since we are receiving traffic from Twilio (it sets the last Diversion header) we are unable to see if the caller was diverted to our SIP trunk. Is there a workaround for this? Any external module that would support this use case? - *Message Too Long Error:* Since Twilio uses long URIs to define its resources, the SIP messages being handled by Kamailio are sometimes too big and generate a "Message Too Long error". I have been able to temporarily patch this removing unused headers using remove_hf_re and remove_hf but it still fails from time to time. Is there a way to split the UDP packet to mitigate this issue? or what other options could be considered? Thanks and regards,

3 5

[CentOS 7] Kamailio 5.1.x mongodb package missing in OpenSUSE repo.
by Denys Pozniak 01 Nov '18

01 Nov '18

Hello! How can I add module DB_MONGODB to existing installation of Kamailio 5.1.6 , if there is no package in OpenSUSE repo (home:kamailio:v5.1.x-rpms.repo) ? -- BR, Denys Pozniak

2 1

Topos on high load
by Igor Olhovskiy 01 Nov '18

01 Nov '18

Hi! We’ve run into really interesting issue while testing Kamailio with sipp (15 CPS, 300 calls - 1 failed call, more CPS - more failed calls) with topos involved. And at some point Kamailio starts to send 2 packets on received INVITE. 1 - with topos applied, 1 - with not. Main problem is to collect logs at this point, cause load is really high. Is it possible some internal race condition while topos involved? PCAP is attached. I’ve made a picture to illustrate a bit an issue Regards, Igor

2 2

Determine destination address family
by Isaac 31 Oct '18

31 Oct '18

Hello, How can I determine the address family (IPv6 or IPv4) of a destination? The destination in this case is a hostname. I'd like to set a flag if the destination is IPv6 but not sure how to go about it. Thanks, Isaac

2 2

Can't add permanent location
by David Villasmil 31 Oct '18

31 Oct '18

Hello all, I'm trying to add a permanent location. i’m using db_mode 1, use_domain = 1, i tried that command kamctl rpc ul.add location 361119997(a)m.domain.com sip:361119997@4.3.2.1 3600 0 sip:4.3.2.1:5060 Response is: { "jsonrpc": "2.0", "error": { "code": -32000, "message": "Execution Error" }, "id": 25950 } Anyone knows what this might be? David

2 5

USRLOC db_mode=4 and server_id
by Darren Williams 31 Oct '18

31 Oct '18

I am currently using multiple instances of Kamailio with a common database and using the usrloc module with db_mode=4 and server_id to filter the contacts that get loaded at startup on each server. The problem I seem to have is that using the RPC call ul.add, I cannot see any parameter to send to specify the server_id. The server_id that the contact is given is 0. If I manually go into the db and modify the server_id everything works as expected. I think I must be missing something somewhere as I'm sure you shouldn't need to do this. Can anyone help please. Thanks Daz

3 2

help with regex extracting from $rb
by Sergiu Pojoga 31 Oct '18

31 Oct '18

Hi there, May be some regex gurus can help me out. Being aware that regex's are tricky but not seeing another alternative in this particular case. Trying to extract MOSLQ field from $rb. $var(moslq) = $(rb{re.subst,/.*MOSLQ=([0-9].[0-9])\s+MOSCQ=([0-9].[0-9]).*/\1/s}); Above regex works fine IF body contains a single matching line. However, if there's 2 lines as in the below example - it return $var(moslq)=0.0, the last found value. I need only *LocalMetrics *values, so I tried this regex which works fine in regex101.com with flag 'multi-line' and *without flag 'global'*, but doesn't work in Kamailio. $var(moslq) = $(rb{re.subst,/.*MOSLQ=([0-9].[0-9])\s+MOSCQ=([0-9].[0-9])[\nRemoteMetrics]?.*/\1/s}); Assuming $rb is as follows: VQSessionReport CallID:1ca9258d285a539e3c1048205bf38d41@mypb.net LocalMetrics: Timestamps:START=2018-10-29T15:04:38Z STOP=2018-10-29T15:06:13Z QualityEst:EXTRI=127 MOSLQ=4.2 MOSCQ=4.4 RemoteMetrics: Timestamps:START=2018-10-29T15:04:38Z STOP=2018-10-29T15:06:13Z QualityEst:RCQ=0 EXTRI=0 MOSLQ=0.0 MOSCQ=0.0 Thanks in advance.

3 4

generate and forward a NOTIFY message
by AL S 31 Oct '18

31 Oct '18

Hi, I was wondering if Kamailio can generate and forward a NOTIFY message upon receiving an INVITE. Thanks, AL

3 2

Kamailio 5.1.x support under ubuntu 18.04 (beaver)
by Andrew Chen 30 Oct '18

30 Oct '18

Hi, We are planning to upgrade our Kamailio infrastructure to Kam 5.1.6 using Ubuntu 18.04. Is this a supported setup given that under the Debia package list support, I don't see beaver in there at all? Stable releases Debian/Ubuntu packages with latest release of 5.1 (choose the distribution you're interested in): deb http://deb.kamailio.org/kamailio51 wheezy main deb-src http://deb.kamailio.org/kamailio51 wheezy main deb http://deb.kamailio.org/kamailio51 jessie main deb-src http://deb.kamailio.org/kamailio51 jessie main deb http://deb.kamailio.org/kamailio51 stretch main deb-src http://deb.kamailio.org/kamailio51 stretch main deb http://deb.kamailio.org/kamailio51 buster main deb-src http://deb.kamailio.org/kamailio51 buster main deb http://deb.kamailio.org/kamailio51 precise main deb-src http://deb.kamailio.org/kamailio51 precise main deb http://deb.kamailio.org/kamailio51 trusty main deb-src http://deb.kamailio.org/kamailio51 trusty main deb http://deb.kamailio.org/kamailio51 xenial main deb-src http://deb.kamailio.org/kamailio51 xenial main Thanks. -- Andy Chen -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

1 0

How is ruid calculated
by Kjeld Flarup 28 Oct '18

28 Oct '18

I got two tablets registering with the same username, this used to work but now I got a bug report that only one of them gets a call. I found out that the problem is in the ruid column as they both end up with this ruid value uloc-1-5bd0abe9-83c-e4 How is that ruid value calculated, and how can I change this? -- -------------------- 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

2 1

kamailio as proxy for sipclients behind NAT without rtpproxy
by Karthik Kondapaneni 28 Oct '18

28 Oct '18

I want to use kamailio as a forwarding proxy infront of SIP registrar like asterisk/freeswitch without any rtpproxy to reduce the load on server and to be able to handle more requests . I am using default kamailio config file registration is working fine but call is not connecting . So sip clients should register on asterisk(or freeswitch) and able to receive inbound calls too . I am interested in only REGISTER & INVITE messages to restrict the messages based on sipclient's source ip . Please can someone help me with a working configuration for this setup . This is the topology .. (Sipclients with or without NAT ) ---------> (Kamailio without rtpproxy)-------------->(Asterisk or freeswitch or anyother sipserver )

1 0

add_authinfo_hdr not Added in 200 OK
by Seshu Kumar 26 Oct '18

26 Oct '18

Hello Guys, I am using Kamailio version 5.0.4 in IMS mode. I have added ims_auth in kamailio.cfg in S-Server. And added line modparam("ims_auth", "add_authinfo_hdr", 1) I am not finding “Authentication-Info” header being added in 200 OK responses. Anything more i need to enable. Kindly someone give a clue. Or this is known problem? Regards Seshu Kumar

9 21

Postgres: server version < 9.5 does not support insert_update
by Minh Phan 25 Oct '18

25 Oct '18

Hello List, I have just installed the version 5.1.6 and I get this warning message when Kamailio starts: > db_postgres_new_connection(): server version < 9.5 does not support insert_update Does it mean Kamailio will not be working properly with my current Postgres Server (9.4)? What are the consequence of not having support for insert_update command? Best regards, Minh Phan

4 3

Kamailio Slack channel
by Alex Balashov 25 Oct '18

25 Oct '18

Hi, I'd like to submit for the community's consideration the idea of having a Kamailio Slack channel. Emmanuel Schmidbauer took the initiative to create one: https://kamailio.slack.com/ The sign-in link is: https://join.slack.com/t/kamailio/shared_invite/enQtNDY1NDc5MzMwNDIzLWJlNTF… I for one think this is a good idea and, even if it doesn't ultimately get traction, it should be seriously considered by all. Yes, I am aware of the open-source-purist objection to Slack; it is proprietary and perceived to be a "walled garden" of sorts. Furthermore, for those of us who have been on the Internet and involved in technology for a significant amount of time, the irony is not lost that it is little more than a packaging of the "IRC" experience in the bloated form of a modern JavaScript application. I am also very reluctant to contribute in any way to the Balkanisation or fracturing of project-related communication channels. That thought gives me no comfort. These are all concerns of which I'm mindful. Nevertheless: - A great deal of professional real-time messaging and communication in many organisations has moved to Slack. This is also true of our company, and most companies we work with. Thus, a Slack channel for Kamailio would be highly complementary to a platform many other people are already using. - Slack provides a polished and cohesive user experience, which is why it has become so popular. There are clients for every platform, and a seamless user experience that is also highly compatible with mobile and tablet. One can shoehorn IRC into these media, but it clearly is not designed for them. - There are already lots of open-source discussion channels and developer forums on Slack. - Slack's inline Markdown and other conveniences make it _much_ easier to have discussions about code, configurations, etc., since it is easy to insert inline monospace and multiline blobs, attachments, etc. This is what inspired my switch to it. - A lot of people positioned within the current wave of thinking about IT and technology, and thus a big part of our candidate user base, have never used IRC. Anyway, for those interested, I propose that you join Emmanuel and I. Let's see how it works out? -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 4

Why set this flag on registration?
by Isaac 25 Oct '18

25 Oct '18

Hello, In the example for a dual-stack IPv6/IPv4 Kamailio setup here: https://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6 the "FLB_IPV6" flag is being sent on IPv6 registrations. Why set this flag on IPv6 registrations? If I understand correctly, registration requests never make it beyond !save("location so it's not like the flag can be used further in the dial-plan. The code in question: route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if(af==INET6) setbflag(FLB_IPV6); if (!save("location")) sl_reply_error(); exit; } } Any help is appreciated. Thanks

2 2

Kamailio Question
by Brian Longwell 25 Oct '18

25 Oct '18

I'm trying to setup a simple proxy with NAT/RTPproxy. I have started with the basic cfg file and added nat and forced rtpproxy. The proxy is to reside in between endpoints and multiple SIP servers. Currently calls from endpoints are working great but my issue is calls from the server to the endpoint. The SIP messages are good until the 200 OK sent from the phone back, Kamailio is not fixing the contact so it's sending the private IP of the phone back to the server so all subsequent traffic is attempting to route to the private IP. I have tried using the fix_nated_contact everywhere I could think but still no difference. Attached is sip capture and the kamailio.cfg i'm currently using. Thank you, Brian

1 0

RtpEngine and iLBC support
by Lars Olsson 25 Oct '18

25 Oct '18

Hi, Can anyone give a quick answer regarding RtpEngine and support for iLBC codec? I have followed the instructions how to build and enable codecs according to the GitHub manual. But iLBC codec is missing when performing "rtpengine --codecs". I have tried to google around for iLBC references, but the only thing that I have found so far is libilbc, which is a lib for ffmpeg, and it looks like ffmpeg needs to be recompiled. What is the correct/fastest way forward for getting iLBC support on RtpEngine on debian 9? The purpose it to test a-law to iLBC transcoding. Best Regards, Lars

2 1

Odd behavior comparing strings
by Duarte Rocha 25 Oct '18

25 Oct '18

Greetings, I found an odd behaviour in Kamailio and i would like to know if it's intentional. I've made this snippet for showing : $var(teste) = $null; if($var(teste) == "Test") { xerr("TEST : If returned true"); } The "if" returns true if the $var is $null or 0. I'm using the "cmp_str()" now in order to avoid this issue. Best Regards, Duarte Rocha

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Re: [SR-Users] kamailio + asterisk + webrtc
by arish haque 25 Oct '18

25 Oct '18

Hi, I am using this command for starting RTPengine - /usr/sbin/rtpengine --table=0 --interface=192.168.1.x\!182.75.xx.yy --listen-ng=127.0.0.1:22222 --pidfile=/var/run/rtpengine.pid --log-level=6 --log-facility=local1 in kamailio.cfg --> listen=MY_SIP_ADDR advertise 182.75.xx.yy:5060 But still there is no RTP from outside network. Still not able to sort out what is the issue.If pfsense is blocking rtp packets or any other. but i have very config on pfsense. Thanks & Regards, Arish Haque

2 1

Re: [SR-Users] : hight shared memory usage
by Laurent Schweizer 25 Oct '18

25 Oct '18

Hello, No, both server have the same script, the only difference is that they are connected to different DB. We increase the max TLS connections because we plan to move some customer on this proxy and we want to be ready to accept more than 2K TLS connection . Laurent -----Message d'origine----- De : Henning Westerholt <hw(a)kamailio.org> Envoyé : mercredi, 24 octobre 2018 22:46 À : sr-users(a)lists.kamailio.org Cc : Laurent Schweizer <laurent.schweizer(a)peoplefone.com> Objet : [JUNK-SPAM] Re: [SR-Users] : hight shared memory usage Am Mittwoch, 24. Oktober 2018, 15:24:05 CEST schrieb Laurent Schweizer: > I have 2 proxy running and we have a very big difference of memory > usage between them, moreover we have more connected users on the proxy > with less memory usage. > > We have this issue on the first proxy, today we have to increase the > shared memory as it was no more possible to setup new TCP connection. > > We use the same version of the proxy, script is quite the same, one > difference, on the proxy with the issue we increase the number of max > TLS connection to 20K > > Any idea ? > is TLS consuming more memory than TCP or UDP ? Hello Laurent, TLS is using more memory than TCP and than UDP, due to the higher workload and overhead for crypto/connection tracking. But looking to your registered user count this should not make a big difference in the end. Do you have maybe a different usage pattern on the first proxy comparing to the second one? So that different parts of the cfg or the code are executed? Why do you increased the number of max TLS connections on the first proxy, you had also some issues here? > First proxy (with high memory usage): > > > [cycle #: 3; if constant make sure server lives] Kamailio Runtime > Details: > kamailio 5.0.6 (x86_64/linux) > now: Wed Oct 24 15:18:55 2018 > up_since: Wed Oct 24 13:21:12 2018 > uptime: 7063 > > Transaction Statistics: > tmx:UAS_transactions = 56240 tmx:active_transactions = 4 > tmx:UAC_transactions = 0 tmx:inuse_transactions = 18 > > Stateless Server Statistics: > sl:sent_replies = 134317 sl:sent_err_replies = 0 > > UsrLoc Statistics: > usrloc:location_ch-contacts = 1463 usrloc:location_ch-users = 1181 > usrloc:location_ch-expires = 1076 usrloc:registered_users = 1181 > > Core Statistics: > core:rcv_requests = 214220 core:rcv_replies = 179145 > core:fwd_requests = 5495 core:fwd_replies = 73 > > Shared Memory Statistics: > shmem:fragments = 529 shmem:real_used_size = > 15862690496 shmem:free_size = 36566109504 shmem:total_size = > 52428800000 shmem:max_used_size = 16198927224 shmem:used_size = > 15854961104 > > > > cat /proc/net/netstat | cut -d" " -f12-17 ; cat /proc/net/sockstat TW > TWRecycled TWKilled PAWSPassive PAWSActive PAWSEstab > 339118 0 684 3195 0 4648 > InBcastOctets OutBcastOctets InCsumErrors InNoECTPkts InECT1Pkts > InECT0Pkts > 14620680 0 0 19142166496 9683 133135 > sockets: used 1951 > TCP: inuse 1029 orphan 0 tw 2 alloc 1036 mem 15 > UDP: inuse 540 mem 446 > UDPLITE: inuse 0 > RAW: inuse 0 > FRAG: inuse 0 memory 0 > > > > ------------------------------------------- > > > Second proxy (memory is OK) : > > [cycle #: 2; if constant make sure server lives] Kamailio Runtime > Details: > kamailio 5.0.6 (x86_64/linux) > now: Wed Oct 24 15:19:39 2018 > up_since: Mon Jul 2 00:31:48 2018 > uptime: 9902871 > > Transaction Statistics: > tmx:UAS_transactions = 86712048 tmx:active_transactions = 92 > tmx:UAC_transactions = 0 tmx:inuse_transactions = 178 > > Stateless Server Statistics: > sl:sent_replies = 641635175 sl:sent_err_replies = 0 > > UsrLoc Statistics: > usrloc:location-contacts = 3971 usrloc:location-users = 3845 > usrloc:location-expires = 286685 usrloc:registered_users = 3845 > > Core Statistics: > core:rcv_requests = 752644955 core:rcv_replies = 1181197213 > core:fwd_requests = 14482059 core:fwd_replies = 206909 > > Shared Memory Statistics: > shmem:fragments = 28437 shmem:real_used_size = 56447456 > shmem:free_size = 2091036192 shmem:total_size = 2147483648 > shmem:max_used_size = 82099768 shmem:used_size = 43354160 > > > cat /proc/net/netstat | cut -d" " -f12-17 ; cat /proc/net/sockstat TW > TWRecycled TWKilled PAWSPassive PAWSActive PAWSEstab > 273625 0 0 0 0 3524 > InBcastOctets OutBcastOctets InCsumErrors InNoECTPkts InECT1Pkts > InECT0Pkts > 2010 0 1 179031265416 81564 21432766 > sockets: used 4964 > TCP: inuse 576 orphan 0 tw 3 alloc 681 mem 105 > UDP: inuse 3908 mem 3117 > UDPLITE: inuse 0 > RAW: inuse 0 > FRAG: inuse 0 memory 0 Best regards, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio security assessment - https://skalatan.de/de/assessment

2 1

DMQ and DIALOG
by Phillman25 Kyriacou 24 Oct '18

24 Oct '18

Hi List! I currently have 3 Kamailio's (Version 5.1) in an ACTIVE-ACTIVE-ACTIVE scenario behind a load balancer all 3 nodes are listening and advertising the same virtual ip of the load balancer. Im using dmq module together with dialogue module to sync the state of the dialog to all 3 nodes. Is there a way or module that i can use to also sync every transaction? Problem that im facing right now is that an INVITE will come to Kamailio1 and the 100 TRYING or 183 RINGING or 200OK will go to Kamailio2 and since Kamailio1 hasn't received a response will retransmit the INVITE. Or is there a way with dmq to also update the other nodes for responses like 100 ,183 or 200 OK so that Kamailio1 knows not to retransmit? Maybe im going down the wrong path? :) Any assistance would be greatly appreciated My config for dmq and dialog: #=====================DIALOG==================================== modparam("dialog", "enable_dmq", 1) #=====================DMQ======================================= modparam("dmq", "server_address", "sip:x.x.x.x:5080") modparam("dmq", "notification_address", "sip:y.y.y.y:5080") modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 4) modparam("dmq", "ping_interval", 30) request_route { # MANAGE ALL DIALOGS dlg_manage(); if(is_method("KDMQ")) { dmq_handle_message(); } Thanks! Phillip

4 5

: hight shared memory usage
by Laurent Schweizer 24 Oct '18

24 Oct '18

Dear all, I have 2 proxy running and we have a very big difference of memory usage between them, moreover we have more connected users on the proxy with less memory usage. We have this issue on the first proxy, today we have to increase the shared memory as it was no more possible to setup new TCP connection. We use the same version of the proxy, script is quite the same, one difference, on the proxy with the issue we increase the number of max TLS connection to 20K Any idea ? is TLS consuming more memory than TCP or UDP ? BR Laurent First proxy (with high memory usage): [cycle #: 3; if constant make sure server lives] Kamailio Runtime Details: kamailio 5.0.6 (x86_64/linux) now: Wed Oct 24 15:18:55 2018 up_since: Wed Oct 24 13:21:12 2018 uptime: 7063 Transaction Statistics: tmx:UAS_transactions = 56240 tmx:active_transactions = 4 tmx:UAC_transactions = 0 tmx:inuse_transactions = 18 Stateless Server Statistics: sl:sent_replies = 134317 sl:sent_err_replies = 0 UsrLoc Statistics: usrloc:location_ch-contacts = 1463 usrloc:location_ch-users = 1181 usrloc:location_ch-expires = 1076 usrloc:registered_users = 1181 Core Statistics: core:rcv_requests = 214220 core:rcv_replies = 179145 core:fwd_requests = 5495 core:fwd_replies = 73 Shared Memory Statistics: shmem:fragments = 529 shmem:real_used_size = 15862690496 shmem:free_size = 36566109504 shmem:total_size = 52428800000 shmem:max_used_size = 16198927224 shmem:used_size = 15854961104 cat /proc/net/netstat | cut -d" " -f12-17 ; cat /proc/net/sockstat TW TWRecycled TWKilled PAWSPassive PAWSActive PAWSEstab 339118 0 684 3195 0 4648 InBcastOctets OutBcastOctets InCsumErrors InNoECTPkts InECT1Pkts InECT0Pkts 14620680 0 0 19142166496 9683 133135 sockets: used 1951 TCP: inuse 1029 orphan 0 tw 2 alloc 1036 mem 15 UDP: inuse 540 mem 446 UDPLITE: inuse 0 RAW: inuse 0 FRAG: inuse 0 memory 0 ------------------------------------------- Second proxy (memory is OK) : [cycle #: 2; if constant make sure server lives] Kamailio Runtime Details: kamailio 5.0.6 (x86_64/linux) now: Wed Oct 24 15:19:39 2018 up_since: Mon Jul 2 00:31:48 2018 uptime: 9902871 Transaction Statistics: tmx:UAS_transactions = 86712048 tmx:active_transactions = 92 tmx:UAC_transactions = 0 tmx:inuse_transactions = 178 Stateless Server Statistics: sl:sent_replies = 641635175 sl:sent_err_replies = 0 UsrLoc Statistics: usrloc:location-contacts = 3971 usrloc:location-users = 3845 usrloc:location-expires = 286685 usrloc:registered_users = 3845 Core Statistics: core:rcv_requests = 752644955 core:rcv_replies = 1181197213 core:fwd_requests = 14482059 core:fwd_replies = 206909 Shared Memory Statistics: shmem:fragments = 28437 shmem:real_used_size = 56447456 shmem:free_size = 2091036192 shmem:total_size = 2147483648 shmem:max_used_size = 82099768 shmem:used_size = 43354160 cat /proc/net/netstat | cut -d" " -f12-17 ; cat /proc/net/sockstat TW TWRecycled TWKilled PAWSPassive PAWSActive PAWSEstab 273625 0 0 0 0 3524 InBcastOctets OutBcastOctets InCsumErrors InNoECTPkts InECT1Pkts InECT0Pkts 2010 0 1 179031265416 81564 21432766 sockets: used 4964 TCP: inuse 576 orphan 0 tw 3 alloc 681 mem 105 UDP: inuse 3908 mem 3117 UDPLITE: inuse 0 RAW: inuse 0 FRAG: inuse 0 memory 0

2 2

How is ruid calculated
by Kjeld Flarup 24 Oct '18

24 Oct '18

I got two tablets registering with the same username, this used to work but now I got a bug report that only one of them gets a call. I found out that the problem is in the ruid column as they both end up with this ruid value uloc-1-5bd0abe9-83c-e4 How is that ruid value calculated, and how can I change this? -- -------------------- 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

1 0

Fwd: tcp Free fragment not found
by Diego Nadares 24 Oct '18

24 Oct '18

---------- Mensaje reenviado --------- De: Diego Nadares <dnadares(a)gmail.com> Fecha: El vie, 5 oct. 2018 a las 15:46 Asunto: Re: [SR-Users] tcp Free fragment not found Para: Daniel-Constantin Mierla <miconda(a)gmail.com> Hi Daniel, Sorry for the delay. -M 32 -m 128 Should I increase any/both of these params? Thanks again! Diego El jue., 4 oct. 2018 a las 3:31, Daniel-Constantin Mierla (< miconda(a)gmail.com>) escribió: > Hello, > > what are the values for -m and -M command line parameters? > > Cheers, > Daniel > > On 03.10.18 16:37, Diego Nadares wrote: > > Hi guys! I hope you are doing well. > > Any hint on this? I changed to json but sometimes i get the same error. > > 2018-10-03T10:12:11.066539-03:00 drwsip11 /usr/local/sbin/kamailio[28173]: > ERROR: <core> [mem/f_malloc.c:415]: fm_search_defrag(): fm_search_defrag(0 > x7f575c4ad000, 5352056); Free fragment not found! > 2018-10-03T10:12:11.067552-03:00 drwsip11 /usr/local/sbin/kamailio[28173]: > ERROR: <core> [mem/f_malloc.c:498]: fm_malloc(): fm_malloc(0x7f575c4ad000, > 5352056) called from core: tcp_main.c: _wbufq_add(636), module: core; > Free fragment not found! > 2018-10-03T10:12:11.068128-03:00 drwsip11 /usr/local/sbin/kamailio[28182]: > ERROR: <core> [tcp_main.c:3496]: handle_ser_child(): received CON_ERROR for > 0x7f575f6a6918 (id 12638), refcnt 3, flags 0x4018 > 2018-10-03T10:12:11.068397-03:00 drwsip11 /usr/local/sbin/kamailio[28173]: > ERROR: sl [../../forward.h:220]: msg_send_buffer(): tcp_send failed > 2018-10-03T10:12:11.068929-03:00 drwsip11 /usr/local/sbin/kamailio[28173]: > ERROR: xmlrpc [xmlrpc.c:795]: send_reply(): Error while sending reply > 2018-10-03T10:12:11.092080-03:00 drwsip11 /usr/local/sbin/kamailio[28173]: > WARNING: <core> [tcp_read.c:1666]: handle_io(): F_TCPCONN connection marked > as bad: 0x7f575f6a6918 id 12638 refcnt 1 > > Is there any stat to see? Or any additional configuration to make? > > Here is my tcp related config > > tcp_connection_lifetime=3605 > tcp_conn_wq_max=30000000 > tcp_rd_buf_size=5273226 > > We are running kamilio 4.4.5 > > Thanks in advance! > > Diego > > > > > El vie., 26 may. 2017 a las 16:12, Diego Nadares (<dnadares(a)gmail.com>) > escribió: > >> Hi guys! >> >> We have four a scripts that sends dlg.list command to kamailio through >> xmlrpc. They are not executed simultaneously. Sometimes, we are getting >> the following errors: >> >> fm_search_defrag(): fm_search_defrag(0x7f212c25a000, 5273984); Free >> fragment not found!2017-05-26T13:54:20.438736-03:00 drwsip02 >> /usr/local/sbin/kamailio[19833]: ERROR: <core> [mem/f_malloc.c:498]: >> fm_malloc(): fm_malloc(0x7f212c25a000, 5273984) called from core: >> tcp_main.c: tcpconn_new(957), module: core; Free fragment not >> found!2017-05-26T13:54:20.439101-03:00 drwsip02 >> /usr/local/sbin/kamailio[19833]: ERROR: <core> [tcp_main.c:959]: >> tcpconn_new(): mem. allocation failure2017-05-26T13:54:20.439344-03:00 >> drwsip02 /usr/local/sbin/kamailio[19833]: ERROR: <core> [tcp_main.c:3985]: >> handle_new_connect(): tcpconn_new failed, closing socket >> >> I all ready increased this two variables: >> >> tcp_conn_wq_max=30000000 >> tcp_rd_buf_size=5273226 >> >> How can I see how much memory xmlrpc is using? >> >> Any advice will be appreciated. >> >> Thanks in advance! >> >> Diego >> >> > > _______________________________________________ > Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierla -- www.asipto.comwww.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 > >

2 3

Double Session Description Protocol Version (v) 0 data when using rtpengine
by Wilkins, Steve 24 Oct '18

24 Oct '18

Hello all, I noticed double Session Description Protocol Version (v) 0 data in the SDP section when using rtpengine with Kamailio. Has any else noticed this? Is there a way for Kamailio to remove one of them? Thank you, -Steve

3 12

Double Session Description Protocol Version (v) 0 data when using rtpengine
by Wilkins, Steve 24 Oct '18

24 Oct '18

I was looking at the sdpops_mod.c code hoping that there was an easy way to remove a sess_version (Session Description Protocol Version (v): 0) structure, but there does not appear to be that functionality in that module. It might be a bit of work to remove a duplicate sess_version line. I am using Kamailio 5.1, does anyone know if this is an issue in later versions? Thank you, -Steve

2 4

Re: [SR-Users] htable 1 Millon registers
by Henning Westerholt 24 Oct '18

24 Oct '18

Am Dienstag, 23. Oktober 2018, 23:45:49 CEST schrieb Mauricio Tejeda: > Hello Henning. > Thanks for answered. > Yes, the failure is during the kamailio load. > > I'm increased the shared memory but there is no change, the failure > persists. > > I will try with redis Hello Mauricico, ok - if you want to look people on this list into the error you need to give more details, like some actual error messages from the Kamailio log. But as already pointed out, going with redis is indeed a valid option. Best regards, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio security assessment - https://skalatan.de/de/assessment

1 0

Info: Pre-announcement about Kamailio World 2019
by Daniel-Constantin Mierla 24 Oct '18

24 Oct '18

Hello, being asked couple of times recently, for now I am sending a short note to say that the next edition of Kamailio World Conference is likely to take place at the same location in the city center of Berlin, Germany, during May 6-8, 2019. There are still some bits to sort out before making the dates final, so just have the dates in mind at this moment. Once all is set from organising point of view, we will make the official announcement. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

kamailio+asterisk+webrtc
by arish haque 23 Oct '18

23 Oct '18

Hi, Thank you for the reply! I tried advertising udp port like this listen=udp:192.168.1.x:5060 advertise 182.70.xx.yy:5060. How to advertise RTP ports? Kamailio+RTPengine on a Centos s/m with a priv ip - 192.168.1.x Asterisk is on another CentOs s/m with a priv ip - 192.168.1.y pfsense on pub ip - 182.75.xx.yy PORT FORWARDING on pfsense- sip and rtp ports to kamailio+rtpengine server 192.168.1.x. How to make RTPengine listen on public interface-- 182.75.181.70? sip.conf ======== [general] context=external realm=192.168.1.y bindaddr=0.0.0.0:5080 tcpenable=no allowoverlap=no transport=udp,wss,ws srvlookup=yes qualify=yes directmedia=no canreinvite=no ;---------NAT SUPPORT--------- externip=182.75.xx.yy localnet=192.168.1.0/255.255.255.0 nat=yes ;nat=no //tried but no RTP ;nat=force_rport,comedia //tried but no RTP ;-----------RTP SUPPORT---------------- rtpkeepalive=30 disallow=all allow=gsm allow=ulaw allow=opus allow=alaw allow=g723 ================== and i am using asterisk realtime support for mysql to store peer details. With Regards, Arish Haque

1 0

kamailio + asterisk + webrtc
by arish haque 23 Oct '18

23 Oct '18

Hi, Thank you for the reply! I tried advertising udp port like this listen=udp:192.168.1.x:5060 advertise 182.70.xx.yy:5060. How to advertise RTP ports? Kamailio+RTPengine on a Centos s/m with a priv ip - 192.168.1.x Asterisk is on another CentOs s/m with a priv ip - 192.168.1.y pfsense on pub ip - 182.75.xx.yy PORT FORWARDING on pfsense- sip and rtp ports to kamailio+rtpengine server 192.168.1.x. sip.conf ======== [general] context=external realm=192.168.1.y bindaddr=0.0.0.0:5080 tcpenable=no allowoverlap=no transport=udp,wss,ws srvlookup=yes qualify=yes directmedia=no canreinvite=no ;---------NAT SUPPORT--------- externip=182.75.xx.yy localnet=192.168.1.0/255.255.255.0 nat=yes ;nat=no //tried but no RTP ;nat=force_rport,comedia //tried but no RTP ;-----------RTP SUPPORT---------------- rtpkeepalive=30 disallow=all allow=gsm allow=ulaw allow=opus allow=alaw allow=g723 ================== How to make RTPengine listen to public interface-- 182.75.181.70? With Regards, Arish Haque

1 0

kamailio + asterisk + webrtc
by arish haque 23 Oct '18

23 Oct '18

Hi, I tried advertising udp port like this listen=udp:192.168.1.x:5060 advertise 182.70.xx.yy:5060. How to advertise RTP ports? Kamailio+RTPengine on a Centos s/m with a priv ip - 192.168.1.x Asterisk is on another CentOs s/m with a priv ip - 192.168.1.y pfsense on pub ip - 182.75.xx.yy PORT FORWARDING on pfsense- sip and rtp ports to kamailio+rtpengine server 192.168.1.x. Config and logs are attached. Thank you for the reply! With Regards, Arish Haque

1 0

Re: [SR-Users] kamailio + asterisk + webrtc
by arish haque 23 Oct '18

23 Oct '18

Hi Fred, I have advertised the udp port in kamailio.cfg like below - listen=udp:192.168.1.x:5060 advertise 182.70.xx.yy:5060 But how to advertise rtp Ports?? I have attached sip.conf *Asterisk 1----\ * * \ ------ **Kamailio+RTPPROXY ----** FW (pfsense **182.75.xx.yy public-IP)**--- INTERNET** \ **(private ip - 192.168.1.x)* *Asterisk n------\ * *Th**anks & Regards,* *Arish Haque*

3 2

htable 1 Millon registers
by Mauricio Tejeda 23 Oct '18

23 Oct '18

Good afternoon. Trying to use the htables module to load an extensive table in memory, I can not start kamailio. If the table contains few records (5 and 10 in my examples), this works without problems, for everything else I lose the connection to the database "Lost connection to MySQL server during query" I modified the timeout of mariadb but I can not work. Someone with a similar problem or ideas on how to solve it? Thank you

3 3

RTPEngine set IDs
by Alex Balashov 22 Oct '18

22 Oct '18

Hi, I have a question about RTPEngine set IDs: As of Kamailio 5.x, it appears the RTPEngine module has got quite smart. Even if I don't set `setid_avp`, it still knows which RTPEngine instance and set ID was engaged by an initial rtpengine_offer() / rtpengine_manage() command. This knowledge persists even to delete calls in CANCEL / BYE handling, or in subsequent offer/answer calls in reinvites or UPDATEs. That's great, and very convenient. It saves me the trouble of having to store the set ID in an RR param or as a dialog variable, because everything Just Works. The problem, of course, is that if I restart Kamailio, all this state is forgotten. So, for example, if I set up a call, answer it, restart Kamailio, then hang it up, the delete command will go to the default set and not necessarily the non-zero set chosen. If I want proper RTPEngine call teardown to persist across restarts, as far as I can tell I will need to store the set ID in an RR param or some other reasonably stateless way. Am I missing anything here? Does the `rtpengine` module offer some means of persisting this data across restarts? As far as I can tell, its database backing is for the set definitions only and not for call state. Cheers, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

1 0

kamailio + asterisk + webrtc
by arish haque 22 Oct '18

22 Oct '18

Hi all, I am using kamailio as sip server and asterisk as media server behind pf-sense firewall (NAT) with a public ip. Sip signalling as well as rtp packets are flowing correctly when the endpoints are within a network. But when trying from outside network sip signalling is working perfectly but there is no RTP. Public ip behind pfsense -> 182.70.xx.yy kamailio+rtpengine and asterisk servers are on private ips --> 192.168.1.x kamailio ip - 192.168.1.x asterisk ip - 192.168.1.y Please, feel free to ask for more information. Thanks & Regards, Arish Haque

4 3

Kamailio forwarding call via Public IP Address vs Private IP Address
by Wilkins, Steve 21 Oct '18

21 Oct '18

Good morning All, I have two almost Identical (Kamailio=>Asterisk) systems set up, but one of them is receiving calls and forwarding them to the Asterisk Public IP, and the other is Forwarding to the Asterisk Private IP. The call is coming in to the number and domain in both scenarios. Any ideas on what would cause this? Thank you, -Steve

1 2

Set a system variable in the request route
by Nicolas Breuer 20 Oct '18

20 Oct '18

Hello, How can I change a system variable within the route ? I can change a variable witk kamctl but any ideas in the code ? Thanks

2 2

Variable Kamailio version
by Nicolas Breuer 20 Oct '18

20 Oct '18

Hello, Can we get the Kamailio version into a variable ? Thanks

3 3

pause/waiting Timer kamailio
by eyas barhouk 19 Oct '18

19 Oct '18

Hello Dears if i need to make kamailio able to pause request processing for a Period of time is that possible ? for example when kamailio receive an INVITE request then it have to wait until 10 seconds before relay it to the second server. thanks in advance .

6 7

Acoounting with acc_radius when dialog terminates by default_timeout
by Denys Pozniak 19 Oct '18

19 Oct '18

Hello! How can I send RADIUS Accounting-Request in case of terminating dialog via default_timeout timer? I tried to catch Kamailio's own generated BYE in event_route AND tried to use cdr_expired_dlg_enable , but it looks like acc module is not triggered. event_route[dialog:end] { if ( dlg_isflagset(FLT_ACC_START) && is_method("BYE") ) { setflag(FLT_ACC); } dlg_resetflag(FLT_ACC_START); } modparam("acc", "log_flag", FLT_ACC) modparam("acc", "cdr_expired_dlg_enable", 1) modparam("acc_radius", "radius_config", "/etc/radiusclient/radiusclient.conf") modparam("acc_radius", "radius_flag", FLT_ACC) modparam("acc_radius", "radius_missed_flag", FLT_ACCMISSED) modparam("dialog", "default_timeout", 3600) modparam("dialog", "send_bye", 1) version: kamailio 4.4.2 (x86_64/linux) 892ad6 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 ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 892ad6 compiled on 12:58:38 Aug 28 2017 with gcc 4.8.5 -- BR, Denys Pozniak

2 2

Help
by Olivier KOFFI 19 Oct '18

19 Oct '18

Good morning I want to use kamailio to send calls to an asterisk. My problem is that I want to define call prefixes. example If the subscriber calls the 6000, please call us 7000 on transfer to asterisk. Subscribers are at 6000.6001, etc. I want to know exactly which line to modify in kamailio.cfg thank you Sincerely KOFFI Koffi Olivier +225 79830005 +225 59098608 Directeur Technique LIFOR Sarl

2 1

Event route websocket:closed and function unregister
by Володимир Іванець 19 Oct '18

19 Oct '18

Hello all! I have *event_route[websocket:closed]* configured. By using *$si*, *$sp* variables and data from special hash table I can figure out information about account that was disconnected. I'm trying to use *unregister* function against but with no luck. 1. I'm creating variable which contains (as an example): " sip:100-test@127.0.0.1" string, where "100-test" is username. 2. lookup("location", "$var(user_uri)") - runs fine. I'm getting a match and lots of useful variables including *$ruid* are being set: */usr/sbin/kamailio[27169]: DEBUG: registrar [lookup.c:256]: lookup_helper(): contact for [100-test] found by address* 3. But unfortenately *unregister* with same variable does not work. unregister("location", "$var(user_uri)"): *DEBUG: <core> [core/parser/parse_uri.c:1282]: parse_uri(): bad uri, state 0 parsed: <you@> (4) / <you(a)kamailio.org <you(a)kamailio.org>> (16)* *ERROR: <core> [core/parser/parse_to.c:81]: parse_to_uri(): failed to parse To uri* *ERROR: registrar [save.c:1040]: unregister(): failed to extract Address Of Record* I was trying to manipulate *$var(user_uri)* variable's content but could not make it work. It's interesting that variant with *$ruid* - *unregister("location", "", "$ruid")* will work but unfortenately it works only with *modparam("usrloc", "db_mode", 1)*: *ERROR: usrloc [urecord.c:655]: delete_urecord_by_ruid(): delete_urecord_by_ruid currently available only in db_mode=3* I was wonderring if someone could tell me if I'm just using *unregister* function with uri parameter in wrong format or there might be some other problem. Thanks a lot!

2 4

Dispatcher Return
by Duarte Rocha 19 Oct '18

19 Oct '18

Greetings, I'm using Kamailio with Dispatcher and now i have the need to have more than one destination with the same IP:Port belonging to the same Destination Group. Those destinations will have different configurations unrelated to the Dispatcher. However, after ds_select_dst I only have access to IP and Port. Is it possible to Dispatcher to return another fields from the same table row? Or return any other way to identify the row. Thank you for your help. Best Regards, Duarte Rocha

1 0

Questions
by Olivier KOFFI 19 Oct '18

19 Oct '18

Bonjour Je veux utiliser kamailio pour transmettre des appels vers asterisk. Mon problème est que je veux définir les prefixes d'appels. exemple Si l’abonné appel le 6000 kamailio gère l'appel.mais s'il compose le 7000 on transfert à asterisk. Les abonné sont au 6000,6001,etc je veux savoir exactement quelle ligne modifier dans kamailio.cfg merci Cordialement KOFFI Koffi Olivier +225 79830005 +225 59098608 Directeur Technique LIFOR Sarl

2 1

HTTP Post with JSON
by eyas barhouk 18 Oct '18

18 Oct '18

Hello Dears i'm trying to send POST HTTP request with JSON content on Kamailio V 5.2.0. i have two main headers "content type" & "Authentication " in addition i have the json content . when i'm using curl request it look like : Curl -H "Authentication ket = xxxxx" -H "content type : JSON" -d " {the JSON content}" https://xxxxx.com/send So how can i use the Http_client module to send such requests ?? thanks in advance .

3 10

Http_async_client stops script execution
by Denys Pozniak 18 Oct '18

18 Oct '18

Hello! Please help me to understand how http_async_client module behaves. I am going to use it for call routing according to the http response. if (is_method("INVITE")) { t_newtran(); xlog("L_ERR", "ASYNC0-$fU-$rU-$rm---- \n"); http_set_timeout(1000); http_set_method("POST"); http_append_header("Content-Type: application/json"); http_async_query("http://10.10.10.10:8080/RPC", "{\"jsonrpc\": \"2.0\", \"method\": \"ul.lookup\", \"params\": [\"location\",\"user\"], \"id\": 1}", "HTTP_REPLY"); xlog("L_ERR", "ASYNC1-$fU-$rU-$rm---- \n"); } route[HTTP_REPLY] { xlog("L_ERR", "HTTP_REPLY: ok $http_ok \n"); xlog("L_ERR", "HTTP_REPLY: status $http_rs\n"); xlog("L_ERR", "HTTP_REPLY: body $http_rb\n"); xlog("L_ERR", "HTTP_REPLY: error $http_err)\n"); } In syslog I see HTTP_REPLY log strings with correct response, but do not see ASYNC1 string and script execution stops. What's wrong with this? # kamailio -v version: kamailio 4.4.2 (x86_64/linux) 892ad6 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 ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 892ad6 compiled on 12:58:38 Aug 28 2017 with gcc 4.8.5 -- BR, Denys Pozniak

2 8

No way to send SIP reply from app_perl or jsonrpc modules
by M S 18 Oct '18

18 Oct '18

Hi, I have a use case where I need to send a statefull (or stateless) reply to non-INVITE SIP transaction from PERL script using app_perl module's "moduleFunction" method, https://kamailio.org/docs/modules/5.0.x/modules/app_perl.html#ID-7b0efd72c8… It fails with error "function not found, missing load module?", even though the documentation says that at least "sl_send_reply" method should work. As an alternative, I tried to use JSONRPCS module to call RPC methods "tm.reply" and "tm.reply_callid" from PERL script (using libcurl), https://kamailio.org/docs/modules/5.0.x/modules/jsonrpcs.html https://kamailio.org/docs/modules/5.0.x/modules/tm.html#tm.rpc.reply https://kamailio.org/docs/modules/5.0.x/modules/tm.html#tm.rpc.reply_callid However, they only seems to work for INVITE transactions. If I try to use them for e.g. MESSAGE request I get "Request Timeout" on libcurl and "500 I am terribly sorry" on SIP client. Are these bugs or suppose to work this way? If so, then, is there any way i can send a SIP reply to non-INVITE SIP transaction from outside of the kamailio.cfg script? Please help. P.S. I am using Kamailio 5.0.x on debian wheezy 32bit virtual machine. Thank you.

1 0

Script @Startup
by Nicolas Breuer 17 Oct '18

17 Oct '18

Hello All, Is there a way to execute some codes/scripts at the launch of the Kamailio Server ? So only once when we launch the server. Thanks

2 2

Re: [SR-Users] Missing documentation ims_usrloc_scscf module
by Pan Christensen 17 Oct '18

17 Oct '18

Thanks Henning and Carsten. It should also be noted that the module is required by both IMS_ISC and IMS_REGISTRAR_SCSCF. The lacking documentation of IMS_USRLOC_SCSCF also make those modules difficult to implement. Pan B. Christensen Developer Phonect AS > -----Original Message----- > From: Henning Westerholt <hw(a)kamailio.org> > Sent: tirsdag 16. oktober 2018 20:27 > To: sr-users(a)lists.kamailio.org > Cc: Carsten Bock <carsten(a)ng-voice.com>; Pan Christensen > <pan.christensen(a)phonect.no> > Subject: *** SPAM *** Re: [SR-Users] Missing documentation > ims_usrloc_scscf module > > Am Dienstag, 16. Oktober 2018, 13:31:08 CEST schrieb Carsten Bock: > > can you open a ticket on the GitHub Tracker for this? Otherwise this > > might get lost at some point... I don't have time to take care of it > > immediately as I am travelling this and next week, more likely I would > > take care of it later in November. > > Hello Carsten, > > i have opened already a ticket about that one month ago. It is already > assigned to ngvoice. ;-) > > https://github.com/kamailio/kamailio/issues/1644 > > Best regards, > > Henning > > -- > Henning Westerholt - https://skalatan.de/blog/ Kamailio security assessment > - https://skalatan.de/de/assessment

1 0

Sharing xavp’s between nodes.
by Igor Olhovskiy 17 Oct '18

17 Oct '18

Hi! Is there any possibility to share xavp’s between Kamailio nodes? Idea is to share result of ds_select_dst between several nodes and as 5.2 Kamailio stores result in xavp - need to share this structure as well Right now sharing between nodes is done on sht + dmq and I’m not sure I can save xavp in sht. Regards, Igor

3 3

kamcli v1.1.0 released
by Daniel-Constantin Mierla 17 Oct '18

17 Oct '18

Hello, kamcli tool just got its first official release, a step forward in the process of packaging it. It is a command line tool that aims to be a modern alternative to the venerable kamctl, being written in Python and having already a large set of implemented sub-commands, such as: * subscriber – manage SIP subscribers * ul – manage user location records * address – manage permissions address records * aliasdb – manage database aliases * db – manage kamailio database content * dialog – manage active calls (dialog) * dialplan – manage dialplan records * dispatcher – manage load balancer (dispatcher) * group – manage group membership records (acl) * moni – continuous refresh of the values for a list of statistics * mtree – manage memory trees (mtree) * ps – print the details for kamailio running processes * rpc – interact with kamailio via jsonrpc control commands (alias of jsonrpc) * rpcmethods – return the list of available RPC methods (commands) * speeddial – manage speed dial records * srv – server management commands (sockets, aliases, …) * stats – get kamailio internal statistics * tls – management commands for TLS profiles and connections * uptime – print the uptime for kamailio instance The news article for this release is available at: * https://www.kamailio.org/w/2018/10/kamcli-v1-1-0-released/ The corresponding tag in the github repository being: * https://github.com/kamailio/kamcli/releases/tag/v1.1.0 How to install kamcli and examples of usage can be found at: * https://github.com/kamailio/kamcli/blob/v1.1/README.md 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

2 6

Kamailio topoh and t_check_trans not working.
by José Seabra 17 Oct '18

17 Oct '18

Hi all, I noticed that using the module TOPOH with the parameter mask call id enabled and the t_check_trans together isn't working correctly. When i receive the BYE message my script call the function t_check_trans() in order to check if a message belongs or is related to a transaction. This function always returns false when used with the topoh mask call id enabled, once that the BYE belongs to an existing dialog it must return true. In the logs it says something like transaction doesn't match dialog using the RFC3261. If the TOPOH mask call id option is disabled everything works perfectly. Anyone here already faced this behavior? Thanks Regards -- José Seabra

2 3

Missing documentation
by Pan Christensen 16 Oct '18

16 Oct '18

Hello. The documentation for the ims_usrloc_scscf module seems to be missing: Not Found The requested URL /docs/modules/stable/modules/ims_usrloc_scscf.html was not found on this server. ________________________________ Apache/2.4.10 (Debian) Server at www.kamailio.org Port 443 Pan B. Christensen Utvikler Phonect AS Mail: pan.christensen(a)phonect.no<mailto:pan.christensen@phonect.no> + 47 41 88 88 00 [mail_footer]

4 5

Re: [SR-Users] Problem with callid as htable key
by Daniel-Constantin Mierla 16 Oct '18

16 Oct '18

If no operator is matched in the first two characters of expression part for $shtcn(...), then regexp is used. That's because initially the $shtcn() was doing only regexp matching, later it was added support for additional operators. Yes, == is the operator for equal. Cheers, Daniel On 16.10.18 11:53, jenus(a)cyberchaos.nl wrote: > Daniel, > > Ok so if i'm correct i can prefix the callid with for example 1234 to > avoid matching the eq. In other words eq will only match the regex > code if the callid string starts with it? > > And by mathing on equal you mean: $shtcn(ht_test=>==$var(htid)) ? > > Thanks, > > Jan > > Daniel-Constantin Mierla schreef op 2018-10-16 10:16: >> Just to add that regexp matching when using keys from the sip message >> such as call-id is exposed further to some issues -- for example, it is >> common to have . (dot) in call-id, which for a regexp is matching any >> char, so a call id having 123.456 is matching 123x456. So depending on >> what you want to count, might be better to match by equal or prefix >> rules. >> >> Cheers, >> Daniel >> >> On 16.10.18 10:03, jenus(a)cyberchaos.nl wrote: >>> Daniel, >>> >>> Thanks for the quick response, that indeed solves my problem. >>> >>> >>> Jan >>> >>> Daniel-Constantin Mierla schreef op 2018-10-16 09:55: >>>> Hello, >>>> >>>> the expression in the $shtcn(table=>exp) can have some operators at >>>> the >>>> beginning, see: >>>> >>>> - >>>> https://www.kamailio.org/wiki/cookbooks/devel/pseudovariables#shtcn_htable_… >>>> >>>> >>>> >>>> So, eq is one of them, the safest is to explicitly set the operator if >>>> you cannot predict what is the key, like in this case. If the operator >>>> is missing, then the exp is matched as regular expression, so you can >>>> do: >>>> >>>> $shtcn(ht_test=>~~$var(htid)) >>>> >>>> Cheers, >>>> Daniel >>>> >>>> >>>> On 16.10.18 09:47, jenus(a)cyberchaos.nl wrote: >>>>> Hello, >>>>> >>>>> I'm running kamailio 5.1.4 and running into a issue with htable. I >>>>> use >>>>> it to store some dialog data so i can use it in a http route. When i >>>>> use the callid as a key it creates a issue for callid's starting with >>>>> eq. Here some example code that i use to reproduce the issue: >>>>> >>>>> >>>>> modparam("htable", "htable", "ht_test=>size=16;autoexpire=10") >>>>> >>>>> $var(htable-val) = $_s($dlg(h_entry)) + "::" + $_s($dlg(h_id)) + "::" >>>>> + $Ts; >>>>> xlog("L_WARN", "HT_TEST: Adding $var(htable-val) to ht_test with key >>>>> eq1234"); >>>>> $var(htid) = "eq1234"; >>>>> $sht(ht_test=>$var(htid)) = $var(htable-val); >>>>> $var(htable-val) = $null; >>>>> >>>>> if ($shtcn(ht_test=>$var(htid)) > 0) >>>>> { >>>>> xlog("L_WARN", "HT_TEST: Key eq1234 found!"); >>>>> } else { >>>>> xlog("L_WARN", "HT_TEST: Key eq1234 NOT found!"); >>>>> } >>>>> $var(htid) = $null; >>>>> >>>>> Results in : >>>>> >>>>> Oct 16 09:36:31 sip03 /usr/sbin/kamailio[22344]: WARNING: <script>: >>>>> HT_TEST: Adding 531::926::1539675391 to ht_test with key eq1234 >>>>> Oct 16 09:36:31 sip03 /usr/sbin/kamailio[22344]: WARNING: <script>: >>>>> HT_TEST: Key eq1234 NOT found! >>>>> >>>>> >>>>> When i prefix eq with a 1 it works fine: >>>>> >>>>> modparam("htable", "htable", "ht_test=>size=16;autoexpire=10") >>>>> >>>>> $var(htable-val) = $_s($dlg(h_entry)) + "::" + $_s($dlg(h_id)) + "::" >>>>> + $Ts; >>>>> xlog("L_WARN", "HT_TEST: Adding $var(htable-val) to ht_test with key >>>>> eq1234"); >>>>> $var(htid) = "1eq1234"; >>>>> $sht(ht_test=>$var(htid)) = $var(htable-val); >>>>> $var(htable-val) = $null; >>>>> >>>>> if ($shtcn(ht_test=>$var(htid)) > 0) >>>>> { >>>>> xlog("L_WARN", "HT_TEST: Key eq1234 found!"); >>>>> } else { >>>>> xlog("L_WARN", "HT_TEST: Key eq1234 NOT found!"); >>>>> } >>>>> $var(htid) = $null; >>>>> >>>>> Results in: >>>>> >>>>> Oct 16 09:37:25 sip03 /usr/sbin/kamailio[22555]: WARNING: <script>: >>>>> HT_TEST: Adding 3050::1098::1539675445 to ht_test with key eq1234 >>>>> Oct 16 09:37:25 sip03 /usr/sbin/kamailio[22555]: WARNING: <script>: >>>>> HT_TEST: Key eq1234 found! >>>>> >>>>> >>>>> Am i running into a bug? Or are there some restrictions on the >>>>> key's i >>>>> can use? I'm still going trough the logs to check if there are more >>>>> callid's that have a issue with matching. >>>>> >>>>> >>>>> Thanks, >>>>> >>>>> Jan Hazenberg >>>>> >>>>> _______________________________________________ >>>>> Kamailio (SER) - Users Mailing List >>>>> sr-users(a)lists.kamailio.org >>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> > -- 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 0

Re: [SR-Users] Problem with callid as htable key
by Daniel-Constantin Mierla 16 Oct '18

16 Oct '18

Just to add that regexp matching when using keys from the sip message such as call-id is exposed further to some issues -- for example, it is common to have . (dot) in call-id, which for a regexp is matching any char, so a call id having 123.456 is matching 123x456. So depending on what you want to count, might be better to match by equal or prefix rules. Cheers, Daniel On 16.10.18 10:03, jenus(a)cyberchaos.nl wrote: > Daniel, > > Thanks for the quick response, that indeed solves my problem. > > > Jan > > Daniel-Constantin Mierla schreef op 2018-10-16 09:55: >> Hello, >> >> the expression in the $shtcn(table=>exp) can have some operators at the >> beginning, see: >> >> - >> https://www.kamailio.org/wiki/cookbooks/devel/pseudovariables#shtcn_htable_… >> >> >> So, eq is one of them, the safest is to explicitly set the operator if >> you cannot predict what is the key, like in this case. If the operator >> is missing, then the exp is matched as regular expression, so you can >> do: >> >> $shtcn(ht_test=>~~$var(htid)) >> >> Cheers, >> Daniel >> >> >> On 16.10.18 09:47, jenus(a)cyberchaos.nl wrote: >>> Hello, >>> >>> I'm running kamailio 5.1.4 and running into a issue with htable. I use >>> it to store some dialog data so i can use it in a http route. When i >>> use the callid as a key it creates a issue for callid's starting with >>> eq. Here some example code that i use to reproduce the issue: >>> >>> >>> modparam("htable", "htable", "ht_test=>size=16;autoexpire=10") >>> >>> $var(htable-val) = $_s($dlg(h_entry)) + "::" + $_s($dlg(h_id)) + "::" >>> + $Ts; >>> xlog("L_WARN", "HT_TEST: Adding $var(htable-val) to ht_test with key >>> eq1234"); >>> $var(htid) = "eq1234"; >>> $sht(ht_test=>$var(htid)) = $var(htable-val); >>> $var(htable-val) = $null; >>> >>> if ($shtcn(ht_test=>$var(htid)) > 0) >>> { >>> xlog("L_WARN", "HT_TEST: Key eq1234 found!"); >>> } else { >>> xlog("L_WARN", "HT_TEST: Key eq1234 NOT found!"); >>> } >>> $var(htid) = $null; >>> >>> Results in : >>> >>> Oct 16 09:36:31 sip03 /usr/sbin/kamailio[22344]: WARNING: <script>: >>> HT_TEST: Adding 531::926::1539675391 to ht_test with key eq1234 >>> Oct 16 09:36:31 sip03 /usr/sbin/kamailio[22344]: WARNING: <script>: >>> HT_TEST: Key eq1234 NOT found! >>> >>> >>> When i prefix eq with a 1 it works fine: >>> >>> modparam("htable", "htable", "ht_test=>size=16;autoexpire=10") >>> >>> $var(htable-val) = $_s($dlg(h_entry)) + "::" + $_s($dlg(h_id)) + "::" >>> + $Ts; >>> xlog("L_WARN", "HT_TEST: Adding $var(htable-val) to ht_test with key >>> eq1234"); >>> $var(htid) = "1eq1234"; >>> $sht(ht_test=>$var(htid)) = $var(htable-val); >>> $var(htable-val) = $null; >>> >>> if ($shtcn(ht_test=>$var(htid)) > 0) >>> { >>> xlog("L_WARN", "HT_TEST: Key eq1234 found!"); >>> } else { >>> xlog("L_WARN", "HT_TEST: Key eq1234 NOT found!"); >>> } >>> $var(htid) = $null; >>> >>> Results in: >>> >>> Oct 16 09:37:25 sip03 /usr/sbin/kamailio[22555]: WARNING: <script>: >>> HT_TEST: Adding 3050::1098::1539675445 to ht_test with key eq1234 >>> Oct 16 09:37:25 sip03 /usr/sbin/kamailio[22555]: WARNING: <script>: >>> HT_TEST: Key eq1234 found! >>> >>> >>> Am i running into a bug? Or are there some restrictions on the key's i >>> can use? I'm still going trough the logs to check if there are more >>> callid's that have a issue with matching. >>> >>> >>> Thanks, >>> >>> Jan Hazenberg >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> sr-users(a)lists.kamailio.org >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > -- 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

2 1

Problem with callid as htable key
by jenus＠cyberchaos.nl 16 Oct '18

16 Oct '18

Hello, I'm running kamailio 5.1.4 and running into a issue with htable. I use it to store some dialog data so i can use it in a http route. When i use the callid as a key it creates a issue for callid's starting with eq. Here some example code that i use to reproduce the issue: modparam("htable", "htable", "ht_test=>size=16;autoexpire=10") $var(htable-val) = $_s($dlg(h_entry)) + "::" + $_s($dlg(h_id)) + "::" + $Ts; xlog("L_WARN", "HT_TEST: Adding $var(htable-val) to ht_test with key eq1234"); $var(htid) = "eq1234"; $sht(ht_test=>$var(htid)) = $var(htable-val); $var(htable-val) = $null; if ($shtcn(ht_test=>$var(htid)) > 0) { xlog("L_WARN", "HT_TEST: Key eq1234 found!"); } else { xlog("L_WARN", "HT_TEST: Key eq1234 NOT found!"); } $var(htid) = $null; Results in : Oct 16 09:36:31 sip03 /usr/sbin/kamailio[22344]: WARNING: <script>: HT_TEST: Adding 531::926::1539675391 to ht_test with key eq1234 Oct 16 09:36:31 sip03 /usr/sbin/kamailio[22344]: WARNING: <script>: HT_TEST: Key eq1234 NOT found! When i prefix eq with a 1 it works fine: modparam("htable", "htable", "ht_test=>size=16;autoexpire=10") $var(htable-val) = $_s($dlg(h_entry)) + "::" + $_s($dlg(h_id)) + "::" + $Ts; xlog("L_WARN", "HT_TEST: Adding $var(htable-val) to ht_test with key eq1234"); $var(htid) = "1eq1234"; $sht(ht_test=>$var(htid)) = $var(htable-val); $var(htable-val) = $null; if ($shtcn(ht_test=>$var(htid)) > 0) { xlog("L_WARN", "HT_TEST: Key eq1234 found!"); } else { xlog("L_WARN", "HT_TEST: Key eq1234 NOT found!"); } $var(htid) = $null; Results in: Oct 16 09:37:25 sip03 /usr/sbin/kamailio[22555]: WARNING: <script>: HT_TEST: Adding 3050::1098::1539675445 to ht_test with key eq1234 Oct 16 09:37:25 sip03 /usr/sbin/kamailio[22555]: WARNING: <script>: HT_TEST: Key eq1234 found! Am i running into a bug? Or are there some restrictions on the key's i can use? I'm still going trough the logs to check if there are more callid's that have a issue with matching. Thanks, Jan Hazenberg

2 2

Problem with Kamailio 4.2.0
by José Antonio Gutiérrez Delgado 16 Oct '18

16 Oct '18

Hi, I'm using Kamailio 4.2.0 and I'm developing a SIP-based application. Today I made a test connecting two devices directly to an S7 sharing the Internet, I usually use a VPN even if they are both connected to the same network because in the future they will be in different networks.Without the VPN calls were not received. This has never happened to me, I have repeated tests and in an S6 it has also happened to me. So I have checked my Kamailio server and I have detected that when executing: kamctl ul show, the received field did not appear: With VPN: Contact:: sip:pruebaT@47.60.42.40:10924;transport=udp Q= Expires:: 1992 Callid:: iwNoYb~QR- Cseq:: 21 User-agent:: Unknown (belle-sip/1.6.1) Received:: sip:47.60.42.40:10924 State:: CS_NEW Flags:: 0 Cflag:: 64 Socket:: udp:10.0.0.4:5060 Methods:: 4294967295 Ruid:: uloc-5a9d9097-9fa-431c Instance:: <urn:uuid:8b50dde3-d07f-4752-9a24-7b8ddf219a99> Reg-Id:: 0 Last-Keepalive:: 1539614732 Last-Modified:: 1539614732 Without VPN: Contact:: sip:pruebaT@37.10.156.189:51163;transport=udp Q= Expires:: 1996 Callid:: webjvHycBs Cseq:: 21 User-agent:: Unknown (belle-sip/1.6.1) State:: CS_NEW Flags:: 0 Cflag:: 0 Socket:: udp:10.0.0.4:5060 Methods:: 4294967295 Ruid:: uloc-5a9d9097-9fc-1feb Instance:: <urn:uuid:89306a81-e2d1-4b7a-8206-107e3cbc0688> Reg-Id:: 0 Last-Keepalive:: 1539614531 Last-Modified:: 1539614531 Would someone know why it happened? Thanks -- José Antonio Gutiérrez Delgado Responsable técnico Oficina: (+34) 923040031 <http://www.arsoft-company.com/> *Este mensaje se dirige exclusivamente a su destinatario/s y puede contener información privilegiada o confidencial. Si no es vd. el destinatario indicado, queda notificado de que la utilización, divulgación y/o copia sin autorización está prohibida en virtud de la legislación vigente. Si ha recibido este mensaje por error, le rogamos que nos lo comunique inmediatamente por esta misma vía y proceda a su destrucción.* *Antes de imprimir, piensa si es necesario. Cuidar el medioambiente es tarea de todos.*

3 2

Hi guys,
by shailendra kumar 15 Oct '18

15 Oct '18

I am fresher and want to learn about Kamailio Soft switch, Anyone can provide me link or document from where i can learn. I will be thankful for such help, Thanks & Regards, Shailendra Kumar

2 1

Subscription from asterisk
by willy 15 Oct '18

15 Oct '18

Hello all. I'm trying to share states and presence from kamailio between asterisks. Is there a way to do this in kamailio? Thank you, willy d

2 1

SDP in fifth 200 OK is not the same as in first four
by Roman Romanchenko 15 Oct '18

15 Oct '18

Hi all, I have an issue with SDP from Kamailio side in the following scenario: simple call from A to B. When B answers a call and sends '200 OK' to A, A delays ACK for a while, so '200 OK' re-transmission happens. After the fourth try to re-send '200 OK' towards A, Kamailio changes SDP (specifically connection information 'c', IP address become address of A instead of proxy IP as it was in the previous attempts). At some point A sends ACK and call is established, but last received '200 OK' has invalid SDP and it prevents successful media establishment. I wonder if there any re-transmission counter or whatever what can affect this scenario and change SDP? Roman Romanchenko VoIP devel Sipwise GmbH

2 1

Dialog variables and persistence
by Alex Balashov 15 Oct '18

15 Oct '18

Hi, I forget, can $dlg_var(...) variables nowadays be reconstituted, along with dialog state in general, using database backing? What about profile membership? Last time I brushed with this topic was probably in 4.x days, and as I remember the general idea was that dialog state could be recovered on restart, but without dialog variables or profile counts. Is this accurate? Is it still true? Or has the amount of state serialised into database persistence increased since then? Thanks much! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 1

UPDATE relayed from wrong socket
by Sergiu Pojoga 15 Oct '18

15 Oct '18

Hi ppl, I have this problem with call transfer, may be someone can help. The phone to the far right is registered with the Registrar to the far left using two PATH headers (trespassing two proxy ports, 5070 then 5060). As you can see in the graph below, after receiving the UPDATE request, Kamailio relays it further from port 5060, I expect it to be from 5070 just like the dialog forming INVITE and the CANCEL afterwards. [image: image.png] The UPDATE has a to-tag, but unlike the original INVITE - it has no Route header!??? route[*WITHINDLG*] { if (!has_totag()) return; if (loose_route()) { route(DLGURI); if (is_method("BYE")) { ... } else if ( is_method("ACK") ) { route(NATMANAGE); } else if ( is_method("NOTIFY") ) { record_route(); } route(RELAY); exit; } if ( is_method("ACK") ) { ... } # handle UPDATE method for in-dialog requests if (is_method("*UPDATE*")) { route(DLGURI); record_route(); route(RELAY); } } Thanks in advance.

2 7

Distorted uri / Route header replaces the uri
by Abdulaziz Alghosh 15 Oct '18

15 Oct '18

Greetings, in order to upgrade kamailio server from 3.0.3 to 5.1.4 version, I am conducting test calls and it is not completelly successful. The main components I have are: 1- Media Gateway (Asterisk) : 192.168.226.133 2- Signalling Gateway (Kamailio 5.1.4) 192.168.226.132 3- Global Dispatcher (Kamailio 3.0.3 in front of the Signalling Gateway.) : 192.168.226.130 Both caller and callee have the dispatcher's IP as a proxy and their registration requests is passed to be registered at the Signalling Gateway (Kamailio 5.1.4) and saved on the MySQL server. The caller reaches the callee properly, who in its turn answeres the call and sends the OK msg to the caller who gets it. Unfortunately, when the caller sends the ACK for this INVITE to the callee, the ACK does not reach the callee because it is trapped by the dispatcher at the last stage because the uri was dropped and the Route header was used instead. This behaviour is abnormal since as I know the Route header should be converted into a Via header and the uri stays untouchable. The ACK was looped into the dispatcher itself .i.e. the dispatcher sent it to itself adding always the via header ""Via: SIP/2.0/UDP 192.168.226.130;branch=0"" endlessly. Once again, here is an example of the ACK before the dispatcher and how the dispatcher forwards the ACK afterwards. the supposed path is as following: caller --> GDP--> SGW --> MGW--> SGW --> GDP --> callee Although the GDP (before the calle after the SGW) received the uri well from SGW inside the ACK as it appears below: ACK sip:yyyyyyyyyyyy@192.168.226.130:5060;rinstance=1608f7c153f40093;transport=UDP SIP/2.0 Record-Route: <sip:192.168.226.132;lr=on;ftag=as3b989c5f> Via: SIP/2.0/UDP 192.168.226.132;branch=z9hG4bK136f.1e8349c235fff01ee1849f3456f8868a.0 Via: SIP/2.0/UDP 192.168.226.133:5060;rport=5060;branch=z9hG4bK34871fe9 Route: <sip:192.168.226.130;lr;ftag=as3b989c5f;did=d5b.87890dc6> Max-Forwards: 69 From: "xxxxxxxxxxxxxx" <sip:xxxxxxxxxxxx@domain.com>;tag=as3b989c5f To: <sip:yyyyyyyyyyyyy@192.168.226.132:5060>;tag=1af2c424 Contact: <sip:xxxxxxxxx@192.168.226.133:5060> Call-ID: 7ed2ab533d4c43086746d0e620bdff06(a)domain.com CSeq: 102 ACK User-Agent: MGW Content-Length: 0 The global dipstacher (GDP) has used the Route header (which has no R-URI yyyyyyyyy) instead of the uri: ACK sip:192.168.226.130;lr;ftag=as3b989c5f;did=d5b.87890dc6 SIP/2.0 Record-Route: <sip:192.168.226.132;lr=on;ftag=as3b989c5f> Via: SIP/2.0/UDP 192.168.226.130;branch=0 Via: SIP/2.0/UDP 192.168.226.132;rport=5060;branch=z9hG4bK136f.1e8349c235fff01ee1849f3456f8868a.0 Via: SIP/2.0/UDP 192.168.226.133:5060;rport=5060;branch=z9hG4bK34871fe9 Max-Forwards: 68 From: "xxxxxxxxxx" <sip:xxxxxxxxxxx@domain.com>;tag=as3b989c5f To: <sip:yyyyyyyyy@192.168.226.132:5060>;tag=1af2c424 Contact: <sip:xxxxxxx@192.168.226.132:5060> Call-ID: 7ed2ab533d4c43086746d0e620bdff06(a)domain.com CSeq: 102 ACK User-Agent: MGW Content-Length: 0 Thanks in advance for your contribution Abdulaziz

2 1

DIALPLAN Module: doubts about regular expression usage
by Stefano Bertuola 15 Oct '18

15 Oct '18

Hi Experts. I did some tests with DIALPLAN module and I have some doubts about how to define the regular expressions in the database. For example, using the start character '^' (beginning of the string), it looks not working: kamcmd> dialplan.dump 1 { DPID: 1 ENTRIES: { ENTRY: { PRIO: 1 MATCHOP: 2 MATCHEXP: ^(\+39)(.*)$ MATCHLEN: 0 SUBSTEXP: ^(\+39)(.*)$ REPLEXP: \2 ATTRS: 1 } } } kamcmd> dialplan.translate 1 "+39123456" error: 500 - No translation Removing it, it works... but not as desired: kamcmd> dialplan.dump 1 { DPID: 1 ENTRIES: { ENTRY: { PRIO: 1 MATCHOP: 1 MATCHEXP: (\+39)(.*)$ MATCHLEN: 0 SUBSTEXP: (\+39)(.*)$ REPLEXP: \2 ATTRS: 1 } } } kamcmd> dialplan.translate 1 "+39123456" { Output: 123456" Attributes: 1 } kamcmd> dialplan.translate 1 "xxxxxxx+39123456" { Output: 123456" Attributes: 1 } Can someone help me understand how the regular expression should be used in DIALPLAN, please? Br. Stefano

4 8

DIALPLAN Module: how to get/use the Attributes value
by Stefano Bertuola 14 Oct '18

14 Oct '18

Hi experts. I intended ATTRIBUTES is a value that is returned by dialplan's dp_translate as the result as shown by kamcmd command: kamcmd> dialplan.translate 2 s:+3912345 { Output: 3912345 Attributes: test } Is my understanding correct? I tried many ways, including using AVP and XAVP for the result, but no success: dp_translate("2", "$rU/$xavp(dp_result)"); pv_xavp_print(); DEBUG: dialplan [dialplan.c:222]: dp_get_ivalue(): dpid is 2 from constant argument DEBUG: dialplan [dialplan.c:243]: dp_get_svalue(): searching 7 DEBUG: dialplan [dialplan.c:331]: dp_translate_f(): input is +3912345 DEBUG: dialplan [dp_repl.c:592]: translate(): regex operator testing over [+3912345] DEBUG: dialplan [dp_repl.c:662]: translate(): found a matching rule 0x7f56aef690d8: pr 1, match_exp ^\+39..... DEBUG: dialplan [dp_repl.c:481]: rule_translate(): copying match <3912345> token size 2 DEBUG: dialplan [dialplan.c:340]: dp_translate_f(): input +3912345 with dpid 2 => output 3912345 INFO: <core> [xavp.c:494]: xavp_print_list_content(): +++++ start XAVP list: 0x7f56aeffaaa8 (level=0) INFO: <core> [xavp.c:498]: xavp_print_list_content(): *** XAVP name: dp_result INFO: <core> [xavp.c:499]: xavp_print_list_content(): XAVP id: 3539756229 INFO: <core> [xavp.c:500]: xavp_print_list_content(): XAVP value type: 2 INFO: <core> [xavp.c:509]: xavp_print_list_content(): XAVP value: 3912345 INFO: <core> [xavp.c:531]: xavp_print_list_content(): ----- end XAVP list: 0x7f56aeffaaa8 (level=0) If possible, how can I get both OUTPUT and ATTRIBUTES as result? Stefano

2 1

db_mysql when compiling Kamailio
by Wilkins, Steve 12 Oct '18

12 Oct '18

Hello All, When db_mysql is selected in the make, the make looks for -lmariadb. However, I want MySQL, not mariadb. Is there a way to let the make know that MySQL is preferred ov mariadb? Thank you

2 1

registrar timer and database modes
by eyas barhouk 11 Oct '18

11 Oct '18

hello dears , i'm using kamailio 5.0.1 as IMS platform and have the following issues : * for Module "ims_usrloc_scscf" there is no documentation and i can't find a way to set database mode =2 and define the tables . * for Module "ims_registrar_scscf"when the user Unregistered it takes about 1~2 minutes to remove it from the memory on database mode 0 . so is there a way to minimize the timer when the user Unregistered and remove it immediately from the memory . thanks in advance & best regards.

2 1

Kamailio with Asterisk as Media Relay
by Benjamin Marty 11 Oct '18

11 Oct '18

Hello Ok, that sounds perfect. Is there a sample kamailio configuration file of how to achieve such a thing?

3 2

Addition of a code of conduct to the Kamailio repository
by Henning Westerholt 11 Oct '18

11 Oct '18

Hello, with commit a213d9d1 I introduced a Code of Conduct to the Kamailio repository [1]. This topic was initially discussed during the Kamailio development workshop, started from Daniel. During the development workshop we came to the conclusion that this is a policy decision and needs broader discussion in the Kamailio Management board and also the Kamailio Admin-Team. This discussion started last Friday, and after several days of discussion we decided to choose an existing Code of Conduct. We adapted the CoC of the Debian Project [2] and did a few changes for readability and also to adapt it to the (smaller) scope of our project. This CoC is in my opinion a good, generic and pragmatic document. Many people in our community are already in one way or the other involved in the Debian project. I am aware that this is (especially right now) a hot topic for some people. My personal opinion is that we managed to get a great fit to the existing Kamailio community culture. The history of this project shows that we are an open project and welcome contributions from anybody. So I don't expect any impact because of this change. We also ensured that the CoC only applies inside the Kamailio project, and that the Kamailio Management board is the authoritative instance in any conflicts about this topic. I don't want to start a classical "bike-shedding" discussion on this list about this topic.. But if you have any questions about this, lets us know. You can also contact the Kamailio Management Board, if you want to keep it private. Best regards, Henning Westerholt [1] https://github.com/kamailio/kamailio/blob/master/CODE_OF_CONDUCT.md [2] https://www.debian.org/code_of_conduct

1 0

Websocket configuration for kamailio 5.1.6
by Priyaranjan Nayak 09 Oct '18

09 Oct '18

Hi All, I have compiled the kamailio v5.1.6 source code with tls and websocket module enabled in ubuntu 18.04 LTS. Now I am able to start the kamailio in UDP and TCP port 5060. I wanted to run the kamailio in WS, WSS and TLS port as well. I have generated the private_key, certificate and ca_list and provided the absolute file path in tls.cfg file. I observed tls.so and websocket.so is present in the module path after compile and installation of code. I have used below command to compile and install. 1. make proper 2. make PREFIX="/usr/local/kamailio-5.1" include_modules="db_mysql tls websocket" cfg 3. make all 4. make install I have attached the kamailio.cfg and tls.cfg. Could you please suggest the modification in configuration file to run in WS, WSS and TLS port ? Thanks Priyaranjan

2 1

Kamailio 5.1.6 Debian Package -> SEGFAULT
by Floimair Florian 09 Oct '18

09 Oct '18

Hi! I just upgrade from 5.1.5 to 5.1.6 on my machine and I see Kamailio segfaulting (SIGSEGV). Unfortunately the systemd service is also not restarted in this case. I will generate coredumps and look into the backtrace and provide you with any insights I can gather.

1 0

force_send_socket
by David Villasmil 08 Oct '18

08 Oct '18

Hello all, I'm very rusty when it comes to variables :( There's a part of my config where i load a parameter from usr_preferences to know whether i should append a branch and call him to a fixed IP as well as to their current location. I tried doing this on the REQUEST_ROUTE and append a branch like: # This will check if the TO user is an office user if (avp_db_load("$to", "$avp(office)")) { if ( $avp(gsm)==1 ) { xlog("L_ERR","[REQUEST_ROUTE]: [$rm] callid: $ci from=$fU to=$tU should be ($avp(gsm)) branched - Appending branch\n"); append_branch("sip:$tU@1.2.3.4:5060"); } } And later when the script hits the RELAY, the branch is actually sent, BUT from the *PRIVATE IP.* I tried adding: force_send_socket(67.215.186.219:5060); Right after the "append_branch" to set the socket from where to send, but kamailio is still sending from the private ip. Any ideas? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

5 11

Kamailio with Asterisk as Media Relay
by Benjamin Marty 08 Oct '18

08 Oct '18

Hello I would like to ask if there is any available informations on how to use Kamailio as a SIP Proxy but letting (multiple) Asterisk Instances do the Media Relaying. Is this scenario supported and if yes are there any tutorials or documentation available for it? I know the Kamailio Dispatcher Module, but this does as far as I know a dispatching of also the SIP Traffic. Not just the Media Traffic.

4 4

set_advertised_address with variable
by Patrick Wakano 08 Oct '18

08 Oct '18

Hello list, Hope you all doing fine! I am using version 5.0.7 and I was trying to use the set_advertised_address with an avp variable but it does not replace with the avp value.... I then find out this thread https://lists.kamailio.org/pipermail/sr-users/2016-March/092327.html, in which is mentioned: *"I expect those functions not to support pvs, but I think this issomething to address in 5.0 and get all core functions working with pvs."* Looks like this was actually not done.... is that correct? Is there any plan to do it? In my case I want to use the set_advertised_address after reading the value from DB, so I can't use the advertised_address or listen options.... Any idea on how I could achieve that? Thank you, Kind regards, Patrick Wakano

3 6

Sending INVITE from loopback subnet
by Joonas Kamailio 08 Oct '18

08 Oct '18

Dear List, I am trying to achieve a scenario where I can recognize the original peer and mask it as peer from subnet 10.0.7.0/24 with kamailio. Basic IP connectivity is established with the following network configuration (Debian9): auto lo:1 iface lo:1 inet static address 10.0.0.0/24 And of course I have routed 10.0.0.0/24 to servers public IP address (hereby known as x.x.x.1) in my router. Please consider the following scenario: - Peer1 192.168.0.1 is sending INVITE to kamailio at IP x.x.x.1 - Kamailio sends INVITE to peer2 at 172.10.0.1 from socket 10.0.0.1 - Peer2 receives the INVITE and knows that since it came from IP 10.0.0.1 it means that INVITE is originally from peer1 - IF peer2 sends INVITE to kamailio, it should be sent to peerX from socket 10.0.0.2 This works if I list both 10.0.0.1 and 10.0.0.2 with 'listen=' parameter but things get out of hand if I have multiple peers. In my tests I have noticed that 60 peers equals 60 listen parameters which equals roughly 500 kamailio processes. At some point there are just too many processes and everything melts. If I omit the listen parameter altogether, kamailio has only one socket from /24 subnet and it is 10.0.0.0. Error message when listen parameter is omitted: [pv_core.c:2612]: pv_set_force_sock(): no socket found to match [10.0.0.1] My kamailio has the following configuration in request_route: $avp(e_private) = $(sht(get_private=>$si)); $fs = $avp(e_private); I know that the case is a bit confusing and totally something normal people won't do. If you have any questions I am happy to answer. Best Regards, Joonas Keskitalo

2 1

Server_id expects number only
by Denys Pozniak 08 Oct '18

08 Oct '18

Hello! I need to separate Kamailio's dmq_usrloc by server_id and will be much easily if server_id can accept string values also, like: server_id="192.168.100.100" But currently it accepts only integer: https://www.kamailio.org/wiki/cookbooks/5.1.x/core#server_id Syslog: 0(11488) CRITICAL: <core> [core/cfg.y:3447]: yyerror_at(): parse error in config file /etc/kamailio/global.cfg, line 126, column 11-24: syntax error 0(11488) CRITICAL: <core> [core/cfg.y:3447]: yyerror_at(): parse error in config file /etc/kamailio/global.cfg, line 126, column 11-24: number expected -- BR, Denys Pozniak

2 4

High number of aborted mysql connections
by Sergiu Pojoga 08 Oct '18

08 Oct '18

Hi all, I'm trying to understand why some Kamailio processes become Connor MacLeod and go to live forever causing high # of aborted connections. While watching in time over *SHOW PROCESSLIST*, some 12 kamailio connections in *SLEEP* state, most of them are reused over time (time value gets reset). However, a select few go on to SLEEP forever for as much as the mysql's *wait_timeout* allows them to. Default was 28800 sec, that's how long they lived. If I lower this to say 600 sec, even higher abort rate is observed. I suppose my questions are: 1. Is this normal to have so many SLEEPing connections? 2. What is the recommended MySQL *wait_timeout* value under Kamailio? 3. Any other suggestions as to why such a high rate of aborted connections? Merci!

2 2

Intercepting local reply attributes
by Alex Balashov 06 Oct '18

06 Oct '18

Hi, I have Kamailio in a weird interop/adaptation layer role with the following call flow: SUBSCRIBE to Kamailio with Expires: 0 ---> <-- immediate 200 OK from Kamailio [some time passes] <-- NOTIFY with additional info from Kamailio 200 OK to Kamailio ---> The NOTIFY follows a series of lengthy async HTTP interrogations, so it's constructed at a later point in time with $uac_req(...). The problem is that the standard which regulates this is extremely eccentric in the following way: despite the immediate expiration of the subscription, the NOTIFY is expected to conform to the format of an in-dialog request as if a dialog existed. Don't ask. It's as idiotic as it sounds. But it's a Broadsoft "standard". I can match the To URI / To-tag by capturing it from the incoming From header on the SUBSCRIBE and reproducing it into the outgoing To header on the NOTIFY. The issue is that my From-tag on the NOTIFY does not match the To-tag on the 200 OK for the SUBSCRIBE. So, from the point of view of the subscriber, the NOTIFY is "From" a different party to the one that "initiated" a "dialog" in response to its subscription. I'm just sending the "200 OK" for the SUBSCRIBE like this: --- append_to_reply("Event: $hdr(Event)\r\n"); append_to_reply("Expires: $hdr(Expires)\r\n"); append_to_reply("Contact: <sip:$sel(cfg_get.domain.sip_domain):$Rp>\r\n"); send_reply("200", "OK"); --- I am not aware of anything like an onsend_route that I can use to gain access to the To header of this reply. Nor am I aware of any means of removing headers from a reply and injecting my own To header with a self-generated To-tag I can later mimic in the From tag of the NOTIFY. Any creative workarounds would be appreciated! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

1 0

kamailio + rtpproxy behind NAT
by Bonjour Madame 05 Oct '18

05 Oct '18

Hello, I have kamailio and rtpproxy running on the same server located behind a router with NAT just like below. UA(internet)---public ip---router(NAT)--Kamailio+RTPproxy(private ip)--UA(private network) There are several issues like one way audio. I need users on the internet to communicate with each other, users on the private network to communicate with each other, and users on the internet and the private network to communicate with each other. What is the best approach? Best regards, Clarence

3 5

SIP communication does not reach phone when STUN is used
by Kevin Olbrich 05 Oct '18

05 Oct '18

Hi! I noticed a minor problem when I set up the STUN config of my phone. When the server is set, I am unable to reach the endpoint from kamailio (Phone1 -> Kamailio -> Asterisk -> same Kamilio -> no outgoing SIP flow to Phone2). If I do not set it, it seems to fallback to it's RFC1918 address which is then detected by nat_uac_test() and I can reach the phone. During my tests with Asterisk directly, this did not make any difference. Seems like Asterisk checks more than FLAG 19/8 in nat_uac_test(). If I am right, I strictly used the SIP source port for answers instead of any self-detection using STUN or similar. My kamailio node is connected to the internet directly and 95 percent of (planned) phones are behind random NATs. In example kamailio.cfg there are these settings: # Caller NAT detection > route[NATDETECT] { > #!ifdef WITH_NAT > force_rport(); > if (nat_uac_test("19")) { > if (is_method("REGISTER")) { > fix_nated_register(); > } else { > if(is_first_hop()) { > set_contact_alias(); > } > } > setflag(FLT_NATS); > } > #!endif > return; > } AND (modified for direct RTP flow): # RTPProxy control and signaling updates for NAT traversal > route[NATMANAGE] { > #!ifdef WITH_NAT > if (is_request()) { > if(has_totag()) { > if(check_route_param("nat=yes")) { > setbflag(FLB_NATB); > } > } > } > if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; > if(nat_uac_test("8")) { > fix_nated_sdp("15"); > if (is_request()) { > if (!has_totag()) { > if(t_is_branch_route()) { > add_rr_param(";nat=yes"); > } > } > } > if (is_reply()) { > if(isbflagset(FLB_NATB)) { > if(is_first_hop()) > set_contact_alias(); > } > } > #!endif > return; > } I checked the docs but I am not sure which FLAGs are safe to use (or break things). Any thoughts or hints are welcome. Thank you very much! Kind regards Kevin

2 1

Re: [SR-Users] Kamailio as TCP UDP proxy
by M K 05 Oct '18

05 Oct '18

Thanks, that was very helpful. But now I get an "ERROR: <core> [tcp_main.c:4070]: handle_tcpconn_ev(): connect <Client IP>:<Port> failed" displayed when trying to register. The SIP client establishes the connection to the proxy via TCP, but does not receive an answer. Could this indicate a faulty NAT-ALG?

2 1

clang compiler warnings
by Juha Heinanen 05 Oct '18

05 Oct '18

I just made a test and, for sure, clang warns about unused variables: $ clang -Wunused-variable -I/usr/include/re -I/usr/include/baresip -lbaresip -lrem -lre baresip.c baresip.c:48:6: warning: unused variable 'unused' [-Wunused-variable] int unused; ^ 1 warning generated. -- Juha

3 3

Kamailio memory manager
by Igor Olhovskiy 05 Oct '18

05 Oct '18

Hi! What is the best way to debug messages like... 29(42) ERROR: [core/mem/q_malloc.c:290]: qm_find_free(): qm_find_free(0x7fad301cf010, 56); Free fragment not found! 29(42) ERROR: [core/mem/q_malloc.c:423]: qm_malloc(): qm_malloc(0x7fad301cf010, 56) called from core: core/data_lump.c: dup_lump_list_r(537), module: core; Free fragment not found! 29(42) ERROR: [core/data_lump.c:573]: dup_lump_list_r(): out of mem 29(42) ERROR: [core/data_lump.c:573]: dup_lump_list_r(): out of mem 29(42) ERROR: [core/data_lump.c:573]: dup_lump_list_r(): out of mem 29(42) ERROR: [core/data_lump.c:573]: dup_lump_list_r(): out of mem 29(42) ERROR: [core/data_lump.c:573]: dup_lump_list_r(): out of mem 29(42) ERROR: [core/data_lump.c:573]: dup_lump_list_r(): out of mem 29(42) ERROR: [core/data_lump.c:573]: dup_lump_list_r(): out of mem 29(42) ERROR: [core/data_lump.c:573]: dup_lump_list_r(): out of mem 29(42) ERROR: tm [t_fwd.c:1735]: t_forward_nonack(): failure to add branches Cause core.shmmem shows total: 536870912 free: 522859104 used: 12224144 real_used: 14011808 max_used: 14479048 fragments: 10560 Also mod.stats all pkg Module: core { init_io_wait(475): 6960 init_io_wait(524): 6312 new_db_id(272): 512 dupl_string_name(72): 32 db_do_init2(298): 488 mk_switch_cond_table(43): 144 sr_wtimer_add(330): 48 as_asciiz(161): 24 init_modules(1026): 16 rpc_hash_add(146): 2048 db_allocate_columns(150): 32 db_new_result(114): 56 get_hdr_field(116): 1160 db_allocate_columns(160): 16 cnt_hash_add(383): 1024 register_select_table(458): 48 add_callback(59): 320 sr_wtimer_init(308): 136 cfg_new_group(79): 704 cfg_declare(50): 3360 fix_sock_str(420): 480 init_dst_set(83): 32208 grp_hash_add(234): 808 fix_hostname(1301): 152 add_alias(91): 464 add_alias(93): 256 dupl_string(48): 176 parse_headers(325): 432 subst_parser(291): 144 subst_parser(274): 64 exp_optimize_right(411): 24 fix_expr(541): 600 mk_case_stm(3775): 400 fixup_regexp_null(213): 72 mk_elem(90): 560 route_new_list(200): 1344 fix_param(1196): 1336 parse_select(204): 1488 fix_param(1257): 536 fix_match_rve(3028): 320 mk_rval_expr1(2616): 142656 tr_new(1646): 1464 fixup_pvar_all(279): 1624 fix_param(1162): 16144 mk_action(118): 350168 mk_rval_expr2(2674): 489552 mk_rval_expr_v(2537): 1475600 yyparse(2757): 8792 pv_parse_format(1150): 89064 pv_cache_add(347): 31456 parse_params2(585): 1400 set_mod_param_regex(110): 376 set_mod_param_regex(121): 832 tr_table_add(1875): 400 sr_cmd_exports_convert(235): 36288 register_module(280): 6624 load_module(496): 2640 new_sock_info(235): 224 fix_socket_list(1509): 48 new_sock_info(230): 3920 add_alias(93): 56 add_alias(91): 96 yyparse(1774): 64 yyparse(667): 152 yyparse(2306): 128 yyparse(648): 144 sr_push_yy_state(1666): 32 sr_push_yy_state(1656): 104 sr_push_yy_state(1596): 184 addstr(1413): 415752 rpc_hash_add(100): 25656 str_hash_alloc(59): 512 pv_table_add(236): 17864 pv_init_buffer(2055): 327680 pv_init_buffer(2045): 320 init_nonsip_hooks(43): 24 init_rlist(146): 8 mk_rval_expr_v(2548): 1264 route_add(124): 5872 str_hash_alloc(59): 768 rval_get_str(1256): 56632 pp_define(1788): 2248 init_counters(122): 128 cnt_hash_add(332): 20608 str_hash_alloc(59): 1280 Total: 3602152 } Module: rtpengine { child_init(1812): 256 Total: 256 } Module: db_mysql { db_mysql_new_connection(65): 96 db_mysql_new_connection(75): 3040 Total: 3136 } Module: siptrace { mod_init(344): 384 Total: 384 } Module: sl { sl_register_callback(460): 88 Total: 88 } Module: acc { parse_acc_extra(116): 1512 Total: 1512 } Module: dialog { dlg_bridge_init_hdrs(66): 80 Total: 80 } Module: rr { register_rrcb(61): 80 Total: 80 } Module: db_text { dbt_get_columns(67): 64 Total: 64 } Module: permissions { get_pathname(242): 64 Total: 64 } Module: auth { generate_random_secret(239): 32 generate_random_secret(238): 32 Total: 64 } Module: xlog { mod_init(214): 4104 xdbg_fixup_helper(500): 6032 xlog_fixup_helper(535): 26992 Total: 37128 } Module: dmq { make_socket_str_from_uri(149): 24 Total: 24 } Module: sqlops { sqlops_tr_buffer_init(46): 2048 sql_init_con(83): 240 Total: 2288 } Module: textops { tr_txt_parse_re(212): 32 hname_fixup(3217): 768 fixup_method(3287): 1384 Total: 2184 } Module: pv { tr_parse_string(2343): 120 tr_parse_string(2332): 32 tr_parse_string(2378): 96 tr_parse_string(2368): 184 tr_parse_string(2544): 576 add_var(58): 3616 add_var(65): 1112 Total: 5736 } Module: textopsx { fixup_hname_param(598): 352 Total: 352 } Module: avpops { avpops_parse_pvar(57): 416 Total: 416 } Module: userblacklist { check_blacklist_fixup(522): 16 Total: 16 } Module: htable { fixup_ht_key(348): 176 pv_parse_ht_name(158): 520 Total: 696 } Module: dialplan { dp_trans_fixup(396): 296 Total: 296 } Module: dispatcher { Total: 0 } Module: tmx { Total: 0 } Module: tm { Total: 0 } Module: kex { Total: 0 } Module: usrloc { Total: 0 } Module: topos { Total: 0 } Module: utils { Total: 0 } Module: app_python { Total: 0 } Module: cfgutils { Total: 0 } Module: nathelper { Total: 0 } Where is the place to look at? Means what module can cause this? Regards, Igor

2 3

Astricon: OpenSource Soiree
by Giovanni Maruzzelli 05 Oct '18

05 Oct '18

Hello friends and collegues, A nice get together has been organized for evening Wednesday 10th at 7pm The idea is to have drinks and chitchat about opensource telephony and webrtc, and all opensource project founders, members, and practitioners are invited. Please join me, as distinguished participants too! Check it out at http://party.officering.com (no animals has been harmed in this mail) -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

2 2

Kamailio as TCP UDP proxy
by M K 05 Oct '18

05 Oct '18

Hi, I'm struggling with the configuration to use Kamailio as TCP UDP proxy: [SIP Client] <-(TCP)-> [Proxy] <-(UDP)-> [SIP Server] should be the setup. I only have one client and one SIP server. How do I get Kamailio configured as a proxy so that translation between TCP and UDP works (and vice versa)? I have not yet found any documentation and examples that have caused messages to arrive at my SIP server. I think I've been on the wrong track so far and I'm looking forward to any help. Thank you in advance, Laqoore

3 2

Development for Kamailio v5.2 series is frozen
by Daniel-Constantin Mierla 05 Oct '18

05 Oct '18

Hello, short note to mark the freezing of development for Kamailio v5.2 series. For few weeks, please do not push new features in the master branch. Once the branch 5.2 is created (expected to happen in 3-4 weeks from now), the master branch becomes again open for new feature. Meanwhile the focus has to be on testing current code. Work on related tools (e.g., kamctl) or documentation can still be done as well as getting the new modules in 5.2 in good shape, plus adding exports to kemi interface (which should not interfere with old code). The entire testing phase is expected to be 4 to 6 weeks, then the release of v5.2.0 will be out. 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

PATH when used with multiple listening ports
by Sergiu Pojoga 04 Oct '18

04 Oct '18

Hi again, When acting as an intermediary proxy for a non-colocated Registrar, the REGISTER request coming in on one port and going out on another, shouldn't the *add_path[received]()* generate *two* nodes in the Path header? Otherwise, how are the subsequent requests supposed to be routed back to the UAC? Scenario with Kamailio proxy on IP 1.1.1.1 with dual port listeners UAC => proxy:5070 <=> proxy:5060 => Registrar:5060 Path: <sip:1.1.1.1;lr;received=sip:123.123.123.123:59593> I would expect it to be something like: Path: <sip:1.1.1.1;lr>,<sip:1.1.1.1:5070;lr;received=sip: 123.123.123.123:59593> modparam("rr", "enable_double_rr", 1) modparam("path", "use_received", 1) Merci!

2 2

Kamailio in a mobile world - ts_append issue
by Dan Quinney 04 Oct '18

04 Oct '18

Using "Kamailio in a mobile world" as a base, I have the following routes to store information about the current transaction, however, if the transaction is already cancelled registrations for the user will still trigger `ts_append`, raising an error as the htable hasn't expired yet. Is there anyway to remove the htable entry when the original transaction cancels? Or something else that I'm missing?

2 1

Kamailio v5.1.6 Released
by Daniel-Constantin Mierla 04 Oct '18

04 Oct '18

Hello, Kamailio SIP Server v5.1.6 stable release is out. This is a maintenance release of the latest stable branch, 5.1, that includes fixes since the release of v5.1.5. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.1.x. Deployments running previous v5.1.x versions are strongly recommended to be upgraded to v5.1.6. For more details about version 5.1.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2018/10/kamailio-v5-1-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! 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

Roadmap to next major release - v5.2.0
by Daniel-Constantin Mierla 04 Oct '18

04 Oct '18

Hello, as it was discussed during our last IRC devel meeting before the summer, the time to release a new major version should be this autumn. With the Kamailio Developers Meeting planned for September 27-28, 2018, I think it is better to freeze the code couple of days after that event. So I propose the following milestones: * on Thursday, October 4, 2018 - last development day, code for 5.2.0 will be frozen at the end of the day * 4 to 6 weeks of testing * release sometime in November, to be decided more precisely after the first weeks of testing, some rough candidates being within the weeks of Nov 5 or Nov 19 If anyone has something to add or adjust, let's discuss on sr-users mailing list. 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

2 3

Planing the release of Kamailio v5.1.6
by Daniel-Constantin Mierla 04 Oct '18

04 Oct '18

Hello, a matter of the schedule to be agreed in the morning and the availability during the Kamailio Developers Meeting, we may do a new stable release from branch 5.1.6 on Thursday, Sep 27, 2018, in the afternoon. Given that several of us will be in the same room, the special goal would be to try to automate the process as much as possible. If the time doesn't allow it, then the release will be done next week, a specific day to be decided by the end of this week. Anyhow, this is the informal notification to report as soon as possible any issues you are aware of and not yet in bug tracker, so they get a chance to be fixed. 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

2 4

tcp Free fragment not found
by Diego Nadares 04 Oct '18

04 Oct '18

Hi guys! We have four a scripts that sends dlg.list command to kamailio through xmlrpc. They are not executed simultaneously. Sometimes, we are getting the following errors: fm_search_defrag(): fm_search_defrag(0x7f212c25a000, 5273984); Free fragment not found!2017-05-26T13:54:20.438736-03:00 drwsip02 /usr/local/sbin/kamailio[19833]: ERROR: <core> [mem/f_malloc.c:498]: fm_malloc(): fm_malloc(0x7f212c25a000, 5273984) called from core: tcp_main.c: tcpconn_new(957), module: core; Free fragment not found!2017-05-26T13:54:20.439101-03:00 drwsip02 /usr/local/sbin/kamailio[19833]: ERROR: <core> [tcp_main.c:959]: tcpconn_new(): mem. allocation failure2017-05-26T13:54:20.439344-03:00 drwsip02 /usr/local/sbin/kamailio[19833]: ERROR: <core> [tcp_main.c:3985]: handle_new_connect(): tcpconn_new failed, closing socket I all ready increased this two variables: tcp_conn_wq_max=30000000 tcp_rd_buf_size=5273226 How can I see how much memory xmlrpc is using? Any advice will be appreciated. Thanks in advance! Diego

2 2

Freeing already Freed pointer
by Hamid Hashmi 04 Oct '18

04 Oct '18

I have been receiving CRITICAL core logs since I have updated my service to the following version. Print out of kamailio internals Version: kamailio 5.1.5 (x86_64/linux) d06080 Default config: /usr/local/SIPSer/etc/kamailio/kamailio.cfg Default paths to modules: /usr/local/SIPSer/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: d06080 Compiled with: gcc 4.8.5 Compiled on: 14:17:56 Sep 13 2018 Thank you for flying kamailio! I have been starting kamailio with the following command $ /usr/local/sbin/kamailio -P /run/kamailio/kamailio.pid -f /usr/local/etc/kamailio/kamailio.cfg -m 1024 -M 512 Sample Critical Logs are below Sep 27 14:40:19 kamailio_eu SIPSer[11289]: 923226363266 923000000001 27872 INVITE haYKxpYM1QR8XTU37WdVbxf4VkEITe7A CRITICAL: <core> [core/mem/q_malloc.c:504]: qm_free(): BUG: freeing already freed pointer (0x7f30c8a4df30), called from tm: t_reply.c: relay_reply(1920), first free tm: h_table.c: free_cell_helper(155) - aborting Sep 27 14:40:19 kamailio_eu SIPSer[11289]: 923226363266 923000000001 27872 INVITE haYKxpYM1QR8XTU37WdVbxf4VkEITe7A CRITICAL: <core> [core/mem/q_malloc.c:504]: qm_free(): BUG: freeing already freed pointer (0x7f30c8a4cc40), called from tm: h_table.c: free_cell_helper(153), first free tm: h_table.c: free_cell_helper(153) - aborting Its also causing service crash. Complete logs are here<https://paste.ee/p/q65DE>. Regards Hamid R. Hashmi

2 2

is_first_hop() fails to detect 'myself' when using advertise
by Sergiu Pojoga 04 Oct '18

04 Oct '18

Hi ppl, If I add a FQDN advertise address to the listen statement, *is_first_hop() <https://kamailio.org/docs/modules/4.3.x/modules/siputils.html#siputils.f.is…>* seems to return FALSE no matter how I play with the listen and alias settings. listen=udp:1.1.1.1:5060 advertise sip.mydomain.net:5060 alias="sip.mydomain.net:5060" This statement doesn't get executed in the *onreply_route* *onreply_route[MANAGE_REPLY] {* *if(is_first_hop())* *set_contact_alias(); # contact alias doesn't get set?* *}* *is_first_hop()* *"The function returns true if the proxy is first hop after the original sender. For incoming SIP replies, it means that top Record-Route URI is 'myself' and source address is not matching it"* *This is the reply.* 2018/10/03 17:03:38.276846 2.2.2.2:62033 -> 1.1.1.1:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP sip.mydomain.net:5060 ;branch=z9hG4bK49b3.bfbe15.0;received=1.1.1.1 Via: SIP/2.0/UDP 10.10.10.10:5060;rport=5060;branch=z9hG4bK1af96136 *Record-Route*: <sip:sip.mydomain.net;lr=on;did=4cc.734> Contact: <sip:888@10.17.0.35:62033> Suggestions? Thanks.

3 2

What is the typical network setup for kamailio?
by Kevin Olbrich 03 Oct '18

03 Oct '18

Hi! I am working successfully with Kamailio in my lab setup where Kamailio is the SBC for Asterisk. The network layout is looking like this: SIP-Phone <== PUBLIC NET ==> Kamailio (SBC) <== PRIVATE NET ==> Asterisk <== PUBLIC NET ==> Carrier Each public network is reachable from the internet and has a local firewall with IP whitelists. The internal SIP transactions are UDP-only but for external phones I would like to also listen for TCP/TLS. For this layout to work with rtpproxy (before we move on to RTPengine), we have to enable mhomed in Kamailio. We also have some routing issues with packets leaving with the wrong IP via rtpproxy (when call between carrier and external phone needs to be bridged). Most examples show that Asterisk is deployed on the same network as the external interface of Kamailio (-> Asterisk exposed to the public network). In our tests, this works much better but I have great security concerns because this Asterisk instance itself does not need to be reachable from external. How do other users deploy Kamailio in front of Asterisk or similar as SBC to secure internals? There is lot of docs for Kamailio's config but IMHO less for the setup as DMZ (SBC) proxy. Thank you very much. Kind regards Kevin

7 15

Using TLS on a load-balancing cluster behind DNS-SRV
by Kevin Olbrich 03 Oct '18

03 Oct '18

Hi! Which hostname do I need to request for the certificate when the servers are load-balanced using DNS-SRV? Do I need to get the cert for the DNS-SRV subdomain (without _sip._tls) or for the servers, eg. server0{1,2,3}.pbx.example.com ? Thank you! Kevin

3 2

rr module not detecting nat=ws route param
by Jon Bonilla (Manwe) 02 Oct '18

02 Oct '18

Hi all. I'm having problems with an indialog ACK request which is not being routed correctly. I call loose_route and print: $du=sip:MYIP:2443;transport=ws;r2=on;lr=on;did=1d1.7702;nat=ws $route_uri=sip:MYIP;r2=on;lr=on;did=1d1.7702;nat=ws But when I call check_route_param("nat=ws") I get false result. version: kamailio 5.1.4 any hints? cheers, Jon -- https://pekepbx.com https://www.issabel.com/multitenant

3 6

Dialog lifetime for dialogs synced with DMQ
by jenus＠cyberchaos.nl 02 Oct '18

02 Oct '18

Hello, I'm using DMQ to sync dialogs between a active and standby kamailio node. I noticed that when the standby takes over dialog's they do not expire after the lifetime i set on the other node. Is there a way to make sure that calls are terminated after the dialog lifetime expires? I see that the timer is replicated to the other node, but nothing happens after the dialog expires. The event routes aren't executed as well. Thanks, Jan

2 1

Fwd: kamailio with evapi crashing on high volume
by Jayesh Nambiar 02 Oct '18

02 Oct '18

Hello Daniel, Just a quick update on this. After running from the 5.1 branch twice it happened that Server became unresponsive. It never crashed and on forceful shutdown it only generated a core with shutdown routine. Here is what the syslog contained off while the calls were being sent to kamailio but kamailio was not processing any calls: Sep 22 16:09:04 triviamedia /usr/local/kamailio/sbin/kamailio[16006]: CRITICAL: <core> [core/mem/q_malloc.c:502]: qm_free(): BUG: freeing already freed pointer (0x7fcd23ce5648), called from tm: h_table.c: free_cell_helper(235), first free tm: h_table.c: free_cell_helper(235) - aborting Sep 22 16:09:04 triviamedia /usr/local/kamailio/sbin/kamailio[16006]: CRITICAL: <core> [core/mem/q_malloc.c:502]: qm_free(): BUG: freeing already freed pointer (0x7fcd24a38ea8), called from tm: h_table.c: free_cell_helper(234), first free tm: h_table.c: free_cell_helper(234) - aborting Sep 22 16:09:04 triviamedia /usr/local/kamailio/sbin/kamailio[16006]: CRITICAL: <core> [core/mem/q_malloc.c:502]: qm_free(): BUG: freeing already freed pointer (0x7fcd23ce5648), called from tm: h_table.c: free_cell_helper(235), first free tm: h_table.c: free_cell_helper(235) - aborting Sep 22 16:09:04 triviamedia /usr/local/kamailio/sbin/kamailio[16006]: CRITICAL: <core> [core/mem/q_malloc.c:502]: qm_free(): BUG: freeing already freed pointer (0x7fcd24a38ea8), called from tm: h_table.c: free_cell_helper(234), first free tm: h_table.c: free_cell_helper(234) - aborting Sep 22 16:09:04 triviamedia /usr/local/kamailio/sbin/kamailio[16006]: CRITICAL: <core> [core/mem/q_malloc.c:502]: qm_free(): BUG: freeing already freed pointer (0x7fcd23ce5648), called from tm: h_table.c: free_cell_helper(235), first free tm: h_table.c: free_cell_helper(235) - aborting Sep 22 16:09:04 triviamedia /usr/local/kamailio/sbin/kamailio[16006]: CRITICAL: <core> [core/mem/q_malloc.c:502]: qm_free(): BUG: freeing already freed pointer (0x7fcd24a38ea8), called from tm: h_table.c: free_cell_helper(234), first free tm: h_table.c: free_cell_helper(234) - aborting Sep 22 16:09:04 triviamedia /usr/local/kamailio/sbin/kamailio[16006]: CRITICAL: <core> [core/mem/q_malloc.c:502]: qm_free(): BUG: freeing already freed pointer (0x7fcd23ce5648), called from tm: h_table.c: free_cell_helper(235), first fre And here's the "bt full" of the core file: [New LWP 16003] Cannot access memory at address 0x7fcd64e1a148 Cannot access memory at address 0x7fcd64e1a140 Failed to read a valid object file image from memory. Core was generated by `/usr/local/kamailio/sbin/kamailio -P /var/run/siptrunk.pid -f /usr/local/carrie'. Program terminated with signal SIGABRT, Aborted. #0 0x00007fcd6443f428 in ?? () (gdb) bt full #0 0x00007fcd6443f428 in ?? () No symbol table info available. Backtrace stopped: Cannot access memory at address 0x7ffed40e9e68 Thanks, - Jayesh On Sat, Sep 15, 2018 at 7:09 PM Jayesh Nambiar <jayesh1017(a)gmail.com> wrote: > Hello Daniel, > I have now git cloned from 5.1 branch and restarted kamailio today. Also > an observation; that when one particular customer with around 40-50cps > capacity starts his dialing is when the kamailio kept crashing. I will > observe in the coming week and report of any crashes accordingly. > > Thanks, > > - Jayesh > > On Wed, Sep 12, 2018 at 3:11 AM Jayesh Nambiar <jayesh1017(a)gmail.com> > wrote: > >> Thank you Daniel. I will deploy from 5.1 branch. I will be able to update >> it in next two days. Since this is a production system, I will have to have >> find a proper time slot to get this updated. Will let you know once that is >> done. >> Thanks again. >> >> - Jayesh >> >> On Tue, Sep 11, 2018 at 2:16 PM Daniel-Constantin Mierla < >> miconda(a)gmail.com> wrote: >> >>> Hi Jayesh, >>> >>> I looked over the weekend and seems like a race or a double free. I >>> backported some patches to tm module from master to 5.1 branch (a counter >>> to see if there is more than one free). Would you be able to deploy from >>> branch 5.1 or nightly builds for 5.1? With the next crash I can evaluate >>> (rule out or go for it) if it is double free or not. >>> >>> I haven't written you yet, because I planned to dig more into libev >>> (which is used by evapi), to see if it creates some threads that can mess >>> up with some kamailio globals, although I expect it is event loop in a >>> single thread. Other work related project took a lot of time these days, so >>> not there yet ... >>> Cheers, >>> Daniel >>> >>> On 11.09.18 09:56, Jayesh Nambiar wrote: >>> >>> Hi Daniel, >>> Just a gentle follow up on this. Could you find anything useful? >>> >>> Thanks, >>> >>> - Jayesh >>> >>> >>>>>> On 07.09.18 09:56, Jayesh Nambiar wrote: >>>>>> >>>>>> Hi Daniel, >>>>>> This happens randomly. It is not a testbed and never reproducible on >>>>>> the test server. I can still give you access if you wish to analyse the >>>>>> core files to check whats happening here exactly. >>>>>> It simply feels like on heavy load the evapi socket gives up. Earlier >>>>>> I was running 5.0.2 and in that kamailio would get stuck by not sending any >>>>>> async events on evapi socket. Not even the evapi:connection-closed event >>>>>> gets triggered. After I upgraded to latest stable version (5.1.5), it at >>>>>> least started crashing. Here are two core dumps put onto pastebin. >>>>>> https://pastebin.com/nn6gJapm >>>>>> https://pastebin.com/ph7b8vFH >>>>>> >>>>>> Thanks for all the support, >>>>>> >>>>>> - Jayesh >>>>>> >>>>>> On Fri, Sep 7, 2018 at 1:55 AM Daniel-Constantin Mierla < >>>>>> miconda(a)gmail.com> wrote: >>>>>> >>>>>>> Hello, >>>>>>> >>>>>>> are you able to reproduce it somehow? Or just happens randomly? >>>>>>> >>>>>>> Is it on a testbed where I could get access to investigate the >>>>>>> corefiles? If not, then I will ask for more details from cores over email, >>>>>>> first would be 'bt full' for both cores and 'p *dead_cell' for the second >>>>>>> one. >>>>>>> Cheers, >>>>>>> Daniel >>>>>>> >>>>>>> On 05.09.18 17:47, Jayesh Nambiar wrote: >>>>>>> >>>>>>> Hi Daniel, >>>>>>> Have got these core dumps. Let me know if I should be doing a bt >>>>>>> full. I'll pastebin and send. Thanks, >>>>>>> >>>>>>> Core was generated by `/usr/local/kamailio/sbin/kamailio -P >>>>>>> /var/run/siptrunk.pid -f /usr/local/carrie'. >>>>>>> Program terminated with signal SIGSEGV, Segmentation fault. >>>>>>> #0 0x0000000000505d30 in sip_msg_shm_clone (org_msg=0x7f6ab0d9f618, >>>>>>> sip_msg_len=0x7ffdddb2e8bc, clone_lumps=1) at core/sip_msg_clone.c:491 >>>>>>> 491 LUMP_LIST_LEN(len, org_msg->add_rm); >>>>>>> (gdb) bt >>>>>>> #0 0x0000000000505d30 in sip_msg_shm_clone (org_msg=0x7f6ab0d9f618, >>>>>>> sip_msg_len=0x7ffdddb2e8bc, clone_lumps=1) at core/sip_msg_clone.c:491 >>>>>>> #1 0x00007f6af0bdf68d in fake_req (shmem_msg=0x7f6ab0d9f618, >>>>>>> extra_flags=0, uac=0x7f6ab1738980, len=0x7ffdddb2e8bc) at t_reply.c:854 >>>>>>> #2 0x00007f6af0c3aa27 in t_continue_helper (hash_index=58039, >>>>>>> label=413633661, rtact=0x7f6af10500f0, cbname=0x0, cbparam=0x0) at >>>>>>> t_suspend.c:293 >>>>>>> #3 0x00007f6af0c3eed4 in t_continue (hash_index=58039, >>>>>>> label=413633661, route=0x7f6af10500f0) at t_suspend.c:583 >>>>>>> #4 0x00007f6aae4dd010 in w_t_continue (msg=0x7ffdddb2fa60, >>>>>>> idx=0x7f6af1098e90 "8\306\t\361j\177", lbl=0x7f6af1098ff0 >>>>>>> "\240\264\t\361j\177", rtn=0x7f6af1099150 "0\275\t\361j\177") at >>>>>>> tmx_mod.c:760 >>>>>>> #5 0x000000000045b477 in do_action (h=0x7ffdddb2f850, >>>>>>> a=0x7f6af109ab38, msg=0x7ffdddb2fa60) at core/action.c:1085 >>>>>>> #6 0x0000000000467fd5 in run_actions (h=0x7ffdddb2f850, >>>>>>> a=0x7f6af1096630, msg=0x7ffdddb2fa60) at core/action.c:1565 >>>>>>> #7 0x000000000045b234 in do_action (h=0x7ffdddb2f850, >>>>>>> a=0x7f6af10a0f80, msg=0x7ffdddb2fa60) at core/action.c:1058 >>>>>>> #8 0x0000000000467fd5 in run_actions (h=0x7ffdddb2f850, >>>>>>> a=0x7f6af10a0f80, msg=0x7ffdddb2fa60) at core/action.c:1565 >>>>>>> #9 0x0000000000468797 in run_top_route (a=0x7f6af10a0f80, >>>>>>> msg=0x7ffdddb2fa60, c=0x0) at core/action.c:1654 >>>>>>> #10 0x00007f6aabe79370 in evapi_run_cfg_route (evenv=0x7ffdddb30250, >>>>>>> rt=3, rtname=0x7f6aac08cb18 <_evapi_rts+56>) at evapi_dispatch.c:161 >>>>>>> #11 0x00007f6aabe7f271 in evapi_recv_client (loop=0x7f6aabe698e0, >>>>>>> watcher=0x27af5e0, revents=1) at evapi_dispatch.c:467 >>>>>>> #12 0x00007f6aabc5fd73 in ev_invoke_pending () from >>>>>>> /usr/lib/x86_64-linux-gnu/libev.so.4 >>>>>>> #13 0x00007f6aabc633de in ev_run () from >>>>>>> /usr/lib/x86_64-linux-gnu/libev.so.4 >>>>>>> #14 0x00007f6aabe7867c in ev_loop (loop=0x7f6aabe698e0, flags=0) at >>>>>>> /usr/include/ev.h:835 >>>>>>> #15 0x00007f6aabe83fc6 in evapi_run_dispatcher (laddr=0x7f6af0f72300 >>>>>>> "127.0.0.1", lport=8060) at evapi_dispatch.c:705 >>>>>>> #16 0x00007f6aabe6e262 in child_init (rank=0) at evapi_mod.c:213 >>>>>>> #17 0x0000000000542cad in init_mod_child (m=0x7f6af0f71b70, rank=0) >>>>>>> at core/sr_module.c:943 >>>>>>> #18 0x0000000000542971 in init_mod_child (m=0x7f6af0f72968, rank=0) >>>>>>> at core/sr_module.c:939 >>>>>>> #19 0x0000000000542971 in init_mod_child (m=0x7f6af0f73d38, rank=0) >>>>>>> at core/sr_module.c:939 >>>>>>> #20 0x0000000000542971 in init_mod_child (m=0x7f6af0f74670, rank=0) >>>>>>> at core/sr_module.c:939 >>>>>>> #21 0x0000000000542971 in init_mod_child (m=0x7f6af0f76708, rank=0) >>>>>>> at core/sr_module.c:939 >>>>>>> #22 0x0000000000542971 in init_mod_child (m=0x7f6af0f76c08, rank=0) >>>>>>> at core/sr_module.c:939 >>>>>>> #23 0x0000000000542971 in init_mod_child (m=0x7f6af0f770d0, rank=0) >>>>>>> at core/sr_module.c:939 >>>>>>> #24 0x0000000000542971 in init_mod_child (m=0x7f6af0f77cf0, rank=0) >>>>>>> at core/sr_module.c:939 >>>>>>> #25 0x0000000000542971 in init_mod_child (m=0x7f6af0f78808, rank=0) >>>>>>> at core/sr_module.c:939 >>>>>>> #26 0x0000000000542971 in init_mod_child (m=0x7f6af0f78bd8, rank=0) >>>>>>> at core/sr_module.c:939 >>>>>>> #27 0x0000000000542971 in init_mod_child (m=0x7f6af0f794c8, rank=0) >>>>>>> at core/sr_module.c:939 >>>>>>> #28 0x0000000000542971 in init_mod_child (m=0x7f6af0f79920, rank=0) >>>>>>> at core/sr_module.c:939 >>>>>>> #29 0x0000000000542971 in init_mod_child (m=0x7f6af0f7a330, rank=0) >>>>>>> at core/sr_module.c:939 >>>>>>> #30 0x0000000000542971 in init_mod_child (m=0x7f6af0f7afd0, rank=0) >>>>>>> at core/sr_module.c:939 >>>>>>> #31 0x0000000000542971 in init_mod_child (m=0x7f6af0f7bc80, rank=0) >>>>>>> at core/sr_module.c:939 >>>>>>> #32 0x000000000054303d in init_child (rank=0) at core/sr_module.c:970 >>>>>>> #33 0x0000000000425399 in main_loop () at main.c:1701 >>>>>>> #34 0x000000000042bd5c in main (argc=13, argv=0x7ffdddb31088) at >>>>>>> main.c:2638 >>>>>>> >>>>>>> And this: >>>>>>> [New LWP 15804] >>>>>>> [Thread debugging using libthread_db enabled] >>>>>>> Using host libthread_db library >>>>>>> "/lib/x86_64-linux-gnu/libthread_db.so.1". >>>>>>> Core was generated by `/usr/local/kamailio/sbin/kamailio -P >>>>>>> /var/run/siptrunk.pid -f /usr/local/carrie'. >>>>>>> Program terminated with signal SIGSEGV, Segmentation fault. >>>>>>> #0 0x00007f6af0b90b70 in free_cell_helper >>>>>>> (dead_cell=0x7f6ab0a6baa8, silent=0, fname=0x7f6af0c8f630 "timer.c", >>>>>>> fline=654) at h_table.c:230 >>>>>>> 230 foo = tt->next; >>>>>>> (gdb) bt >>>>>>> #0 0x00007f6af0b90b70 in free_cell_helper >>>>>>> (dead_cell=0x7f6ab0a6baa8, silent=0, fname=0x7f6af0c8f630 "timer.c", >>>>>>> fline=654) at h_table.c:230 >>>>>>> #1 0x00007f6af0c24409 in wait_handler (ti=932640643, >>>>>>> wait_tl=0x7f6ab0a6bb28, data=0x7f6ab0a6baa8) at timer.c:654 >>>>>>> #2 0x00000000004bb445 in timer_list_expire (t=932640643, >>>>>>> h=0x7f6ab03ad158, slow_l=0x7f6ab03ae480, slow_mark=271) at core/timer.c:874 >>>>>>> #3 0x00000000004bb8ab in timer_handler () at core/timer.c:939 >>>>>>> #4 0x00000000004bbd30 in timer_main () at core/timer.c:978 >>>>>>> #5 0x00000000004250f9 in main_loop () at main.c:1691 >>>>>>> #6 0x000000000042bd5c in main (argc=13, argv=0x7ffdddb31088) at >>>>>>> main.c:2638 >>>>>>> >>>>>>> On Wed, Sep 5, 2018 at 3:13 PM Daniel-Constantin Mierla < >>>>>>> miconda(a)gmail.com> wrote: >>>>>>> >>>>>>>> Hello, >>>>>>>> >>>>>>>> the backtrace doesn't show any hint about kamailio, only from >>>>>>>> closelog() up. >>>>>>>> >>>>>>>> It may be the core generated by shutdown procedure, have you >>>>>>>> enabled one core file per pid/process? If not, do it and reproduce the >>>>>>>> issue again, you may get two core files, one being the runtime issue and >>>>>>>> the other one from shutdown procedure, which likely is an effect of the >>>>>>>> other one. The one from the runtime is more relevant. >>>>>>>> >>>>>>>> Cheers, >>>>>>>> Daniel >>>>>>>> >>>>>>>> On 05.09.18 10:09, Jayesh Nambiar wrote: >>>>>>>> >>>>>>>> Hello, >>>>>>>> I'm using kamailio 5.1.5 with evapi. I have a node.js connecting >>>>>>>> with kamailio evapi to which I send events and also consume events based on >>>>>>>> which I do the routing. I have 8 evapi workers defined in the config. >>>>>>>> The problem is that kamailio randomly crashes on high load. I'm >>>>>>>> assuming that it is related to the evapi module as rest of the config is >>>>>>>> pretty straight forward. I could get a core file and here's the core dump: >>>>>>>> [New LWP 14042] >>>>>>>> [Thread debugging using libthread_db enabled] >>>>>>>> Using host libthread_db library >>>>>>>> “/lib/x86_64-linux-gnu/libthread_db.so.1”. >>>>>>>> Core was generated by `/usr/local/kamailio/sbin/kamailio -P >>>>>>>> /var/run/siptrunk.pid -f /usr/local/carrie’. >>>>>>>> Program terminated with signal SIGABRT, Aborted. >>>>>>>> #0 0x00007f9995283428 in __GI_raise (sig=sig@entry=6) at >>>>>>>> ../sysdeps/unix/sysv/linux/raise.c:54 >>>>>>>> 54 ../sysdeps/unix/sysv/linux/raise.c: No such file or directory. >>>>>>>> (gdb) bt >>>>>>>> #0 0x00007f9995283428 in __GI_raise (sig=sig@entry=6) at >>>>>>>> ../sysdeps/unix/sysv/linux/raise.c:54 >>>>>>>> #1 0x00007f999528502a in __GI_abort () at abort.c:89 >>>>>>>> #2 0x000000000041a029 in sig_alarm_abort (signo=14) at main.c:646 >>>>>>>> #3 <signal handler called> >>>>>>>> #4 0x00007f999534f497 in __libc_cleanup_routine (f=<optimized >>>>>>>> out>) at ../sysdeps/nptl/libc-lockP.h:291 >>>>>>>> #5 closelog () at ../misc/syslog.c:415 >>>>>>>> #6 0x0000000000000000 in ?? () >>>>>>>> >>>>>>>> Any help in this regards is would allow me to identify the reason >>>>>>>> of the crash. Thanks for the support. >>>>>>>> >>>>>>>> - Jayesh >>>>>>>> >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> Daniel-Constantin Mierla -- www.asipto.comwww.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 >>>>>>>> >>>>>>>> >>>>>>> -- >>>>>>> Daniel-Constantin Mierla -- www.asipto.comwww.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 >>>>>>> >>>>>>> >>>>>> -- >>>>>> Daniel-Constantin Mierla -- www.asipto.comwww.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 >>>>>> >>>>>> >>>>> -- >>>>> Daniel-Constantin Mierla -- www.asipto.comwww.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 >>>>> >>>>> >>> -- >>> Daniel-Constantin Mierla -- www.asipto.comwww.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 >>> >>>

2 2

[DMQ] dmq_send_message function usage.
by Denys Pozniak 02 Oct '18

02 Oct '18

Hello! I have set of Kamailios which distributed location via dmq_usrloc. Could I construct synthetic DMQ message from separate Kamailio node to this cluster to get location info for the specific user? If yes, please explain how to do it. -- BR, Denys Pozniak

3 3

Forcing outbound transport to TCP
by Ivan Ribakov 02 Oct '18

02 Oct '18

I’m new to Kamailio and currently evaluating it’s capabilities. One of the things I would like to be able to do is to change outbound message transport based on some dynamic logic. Kamailio setup: I’m running on a setup from this guide - http://kamailio.org/docs/tutorials/5.1.x/kamailio-install-guide-git/ <http://kamailio.org/docs/tutorials/5.1.x/kamailio-install-guide-git/> and using default configuration file. In a basic scenario of one2one call, I was able to make Kamailio to forward INVITE to callee over TCP by issuing a REGISTER request with “transport=tcp” parameter. Although it worked, as far as I understand that means all communication with the callee will be happening over TCP. I would like to be able to change transport dynamically so looking for a way to do that from config file. I tried modifying $ru/$du in the following manner, but it didn’t seem to work. # default config content if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting # added to try to force TCP transport $du = $du + ";transport=tcp"; } Any pointers as to how can I modify the outbound message transport? Do I need to use some extra modules or can it be achieved with a basic setup? Thanks in advance, Ivan

4 5

Registrar Module
by Dan Quinney 01 Oct '18

01 Oct '18

Hi, So, route[REGISTRAR_1] will, when the max_contacts is reached, reply to the endpoint with a 503 error containing a "P-Registrar-Error: Too many registered contacts" header - this seems wrong to me so I've tried to work around it with route[REGISTRAR_2] which returns the right response, but, also writes "INFO: registrar [save.c:615]: test_max_contacts(): too many contacts for AOR" to the log file, which is messy. Does anyone have any pointers for doing this better? Thanks, Dan route[REGISTRAR_1] { if ( is_method("REGISTER") ) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); setbflag(FLB_NATSIPPING); } if ($avp(maximum_contacts) != $null) { $xavp(reg=>max_contacts) = $avp(maximum_contacts); } save("location"); } return; } route[REGISTRAR_2] { if ( is_method("REGISTER") ) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); setbflag(FLB_NATSIPPING); } if ($avp(maximum_contacts) != $null) { $xavp(reg=>max_contacts) = $avp(maximum_contacts); } save("location", "0x02"); switch ($retcode) { case -1: sl_send_reply("500", "Server Internal Error"); exit; case -2: sl_send_reply("403", "Forbidden"); exit; }; sl_send_reply("200", "OK"); } return; }

2 1

RTPEngine Question
by Wilkins, Steve 01 Oct '18

01 Oct '18

Hello All, I have an issue where Kamailio-RTPEngine-Asterisk calls work good when a softphone UAC on an IOS phone makes an Inbound call to a WebRTC client. However, if the softphone UAC is on Windows, it does not work (No Audio/Video). I noticed in the Wireshark trace that when using IOS I see STUN packets, but when using Windows I see no STUN packets. I believe this may be my issue, but I am not sure why the IOS would create STUN packets but when using Windows I get no STUN packets. Does anyone have any thoughts on what could be causing this? Thank you,

1 0

Re: [SR-Users] Problem with replicating dialogs using DMQ
by Charles Chance 01 Oct '18

01 Oct '18

Hi Jan, Setting multi_notify parameter should fix. Cheers, Charles On Mon, 1 Oct 2018 at 08:26, <jenus(a)cyberchaos.nl> wrote: > Charles, > > I have the follwing parameters configured: > > # ----- dmq params ----- > modparam("dmq", "server_address", "sip:10.10.10.1:5060") > modparam("dmq", "num_workers", 4) > modparam("dmq", "ping_interval", 90) > modparam("dmq", "notification_address", > "sip:sip04.voip.domain.test:5060") > > And the output of dmq.list_nodes is: > > { > host: 10.10.10.2 > port: 5060 > resolved_ip: 10.10.10.2 > status: active > last_notification: 0 > local: 0 > } > { > host: sip03.voip.domain.test > port: 5060 > resolved_ip: 10.10.10.1 > status: active > last_notification: 0 > local: 0 > } > { > host: sip04.voip.domain.test > port: 5060 > resolved_ip: 10.10.10.2 > status: active > last_notification: 0 > local: 0 > } > { > host: 10.10.10.1 > port: 5060 > resolved_ip: 10.10.10.1 > status: active > last_notification: 0 > local: 1 > } > > > As you can see the nodes are listed twice, one time with the ip and one > time with the hostname. If i change the notification_address to a ip > address the duplicate entry's are gone. > > Thanks, > > Jan > > Charles Chance schreef op 2018-09-28 20:54: > > Hello, > > > > Can you share your dmq config (modparams) and the output of > > dmq.list_nodes? > > > > Cheers, > > > > Charles > > > > On Thu, 27 Sep 2018 at 12:51, Jan Hazenberg <jenus321(a)gmail.com> > > wrote: > > > >> In the output of "kamcmd dmq.list_nodes" i see nodes twice. One with > >> ip address of the host and one with the hostname of the host. > >> > >> Jan > >> > >> On Thu, Sep 27, 2018 at 12:47 PM Jan Hazenberg <jenus321(a)gmail.com> > >> wrote: > >> > >> Daniel, > >> > >> It indeed sends duplicate DMQ messages, one to the ip of the other > >> node and one to the hostname of the other node. > >> > >> Unknown request: KDMQ sip:dialog@sip04.voip.domain.acc:5060 | > >> (application/json) > >> Unknown request: KDMQ sip:dialog@10.10.10.12:5060 [5] | > >> (application/json) > >> > >> Is there any way to disable this? > >> > >> Thanks, > >> > >> Jan > >> > >> On Thu, Sep 27, 2018 at 11:50 AM Daniel-Constantin Mierla > >> <miconda(a)gmail.com> wrote: > >> > >> Hello, > >> > >> the log message says that the dialog was already added to the timer. > >> Do you see retransmissions of the DMQ requests on the network with > >> ngrep/wireshark/...? > >> Cheers, > >> Daniel > >> > >> On 27.09.18 10:40, Jan Hazenberg wrote: > >> > >> Hello, > >> > >> I'm trying to replicate active dialogs using the DMQ module. I see > >> the dialogs on the other nodes but it generates a error message in > >> the logs on all the nodes: > >> > >> Sep 27 10:17:31 sip04 /usr/sbin/kamailio[27367]: CRITICAL: dialog > >> [dlg_timer.c:130]: insert_dlg_timer(): Trying to insert a bogus dlg > >> tl=0x7efdca1bebf8 tl->next=0x7efdca14cb80 tl->prev=0x7efdca14cb80 > >> Sep 27 10:17:31 sip04 /usr/sbin/kamailio[27367]: CRITICAL: dialog > >> [dlg_dmq.c:309]: dlg_dmq_handle_msg(): Unable to insert dlg timer > >> 0x7efdca1beb98 [2643:3725] > >> > >> Even with the timeout_avp option disabled i still see the error > >> messages. I tested this on kamailio 5.0.3 and 5.1.3 with the same > >> results. > >> > >> With debug enabled i see the correct data is replicated to the other > >> nodes: > >> > >> dlg_dmq_replicate_action(): sending serialized data > >> > > > {"action":2,"h_entry":1328,"h_id":409,"state":4,"start_ts":1538036606,"lifetime":7200,"tag1":"1635359980","tag2":"as2d80799c","cseq1":"21","cseq2":"0","route_set1":"","route_set2":"","contact1":" > sip:sipuser@10.10.10.10:1057;alias=10.10.10.10~1055~1","contact2":" > sip:+31101234567@10.10.10.10:5060 > >> [6]"} > >> > >> "lifetime":7200 is in the serialized data, and when i check on the > >> other node with "kamcmd dlg.list" i see that the lifetime data is > >> not updated. > >> > >> Thanks, > >> > >> Jan > >> > >> _______________________________________________ > >> Kamailio (SER) - Users Mailing List > >> sr-users(a)lists.kamailio.org > >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > >> > >> -- > >> Daniel-Constantin Mierla -- www.asipto.com [1] > >> www.twitter.com/miconda [2] -- www.linkedin.com/in/miconda [3] > >> Kamailio World Conference -- www.kamailioworld.com [4] > >> Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- > >> www.asipto.com [1] > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > -- > > > > CHARLES CHANCE > > Managing Director > > > > t. 0330 120 1200 m. 07932 063 891 > > Sipcentric Ltd. Company registered in England & Wales no. 7365592. > > Registered office: Faraday Wharf, Innovation Birmingham Campus, Holt > > Street, Birmingham Science Park, Birmingham B7 4BB. > > > > Links: > > ------ > > [1] http://www.asipto.com > > [2] http://www.twitter.com/miconda > > [3] http://www.linkedin.com/in/miconda > > [4] http://www.kamailioworld.com > > [5] http://sip:dialog@10.10.10.12:5060 > > [6] http://sip:+31101234567@10.10.10.10:5060 > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > -- *Charles Chance* Managing Director t. 0330 120 1200 m. 07932 063 891 -- 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

Problem with replicating dialogs using DMQ
by Jan Hazenberg 01 Oct '18

01 Oct '18

Hello, I'm trying to replicate active dialogs using the DMQ module. I see the dialogs on the other nodes but it generates a error message in the logs on all the nodes: Sep 27 10:17:31 sip04 /usr/sbin/kamailio[27367]: CRITICAL: dialog [dlg_timer.c:130]: insert_dlg_timer(): Trying to insert a bogus dlg tl=0x7efdca1bebf8 tl->next=0x7efdca14cb80 tl->prev=0x7efdca14cb80 Sep 27 10:17:31 sip04 /usr/sbin/kamailio[27367]: CRITICAL: dialog [dlg_dmq.c:309]: dlg_dmq_handle_msg(): Unable to insert dlg timer 0x7efdca1beb98 [2643:3725] Even with the timeout_avp option disabled i still see the error messages. I tested this on kamailio 5.0.3 and 5.1.3 with the same results. With debug enabled i see the correct data is replicated to the other nodes: dlg_dmq_replicate_action(): sending serialized data {"action":2,"h_entry":1328,"h_id":409,"state":4,"start_ts":1538036606,"lifetime":7200,"tag1":"1635359980","tag2":"as2d80799c","cseq1":"21","cseq2":"0","route_set1":"","route_set2":"","contact1":"sip:sipuser@10.10.10.10:1057 ;alias=10.10.10.10~1055~1","contact2":"sip:+31101234567@10.10.10.10:5060"} "lifetime":7200 is in the serialized data, and when i check on the other node with "kamcmd dlg.list" i see that the lifetime data is not updated. Thanks, Jan

4 5

registrar lookup with different URI
by Maarten Ureel 30 Sep '18

30 Sep '18

Hello I want the registar module to lookup a user with a different 'name' as the RURI. We have the extension & domain in variables, but the RURI is ext@ip-of-server, e.g. 200(a)1.1.1.1<mailto:200@1.1.1.1>, while we want to perform the lookup for 200(a)somedomain.tld<mailto:200@somedomain.tld>. We have somedomain.tld in a header, so we have it in a variable. However, if we perform the lookup like this (using Python module): lookup_result = KSR.registrar.lookup('location') We get a negative result, and the debug logs show that it tries to locate 200(a)1.1.1.1<mailto:200@1.1.1.1> - which makes sense. So according to docs you can specify an URI, however when I do this: lookup_result = KSR.registrar.lookup('location' '200(a)somedomain.tld') lookup_result is "False" and the function is not executed, according to docs because it cannot obtain the parameters for this function. How can we force the registrar module to lookup a different user?

1 0

Dialog timout reset when receiving a re-INVITE
by jenus＠cyberchaos.nl 30 Sep '18

30 Sep '18

Hello, I have a issue with setting the dialog timeout for a call. It works fine until i receive a re-INVITE. The re-INVITE seems to reset the dialog timeout back to the default value configured in the module parameters (7200 in my case). I use "dlg_set_timeout("900");" to set the timeout to 900 seconds. When i receive a re-INVITE i see the following in the debug logs: DEBUG: dialog [dlg_handlers.c:647]: get_dlg_timeout(): invalid AVP value, using default timeout This happens even with the "timeout_avp" module parameter disabled. I'm running kamailio 5.1.3. Thanks, Jan

3 2

Restoring htable shared variables via dmq after Kamailio restart
by Denys Pozniak 28 Sep '18

28 Sep '18

Hello! I have 2 nodes that replicates htable variables via DMQ. How can I recover them after one node restarting? -- BR, Denys Pozniak

3 2

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sr-users