- sr-users - kamailio.org

start_recording and stop_recording inside event_route[xhttp:request]
by Володимир Іванець 05 May '21

05 May '21

Hello! I'm testing call recording with Rtpengine. It works fine when the "record-call=on" flag is added to the *rtpengine_offer* or *start_recording* is used in the *request_route*. But I was wondering if the call recording can be managed by a separate application. So I add the following lines to the *event_route[xhttp:request]* and triggered it with an HTTP request after the call was established. The call-id value was taken from the Rtpengine log and sent with the request. * if ($hu =~ "^/CALL_RECORD_START/") {* * $var(call_id) = *"call-id=" + *$(hu{s.select,2,/});* xlog("L_DBG", "$var(call_id)"); * start_recording($var(call_id));* exit; * }* In this case, Kamailio will log errors and will not send the Rtpengine command to start call recording. *Apr 27 18:25:00 test /usr/sbin/kamailio[5347]: DEBUG: <script>: call-id=3de74068278ede4417086eb82bdf5a06@127.0.0.1:5080.0* *Apr 27 18:25:00 test /usr/sbin/kamailio[5347]: DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header* *Apr 27 18:25:00 test /usr/sbin/kamailio[5347]: ERROR: rtpengine [rtpengine_funcs.c:294]: get_callid(): call-id not found* *Apr 27 18:25:00 test /usr/sbin/kamailio[5347]: ERROR: rtpengine [rtpengine.c:2383]: rtpp_function_call(): can't get Call-Id field* Have anyone tried to call functions *start_recording* and *stop_recording* inside *event_route[xhttp:request]*? Thanks a lot!

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Kamailio v5.5.0 Released - new major version is out
by Daniel-Constantin Mierla 05 May '21

05 May '21

Kamailio v5.5.0 is out – it comes with 6 new modules and a considerable set of improvements touching again more than 80 existing modules. You can read detailed release notes at: * https://www.kamailio.org/w/kamailio-v5-5-0-release-notes/ Many thanks to all developers and community members that made possible this release. v5.5.0 brings more flexibility and optimizations across many components, enhancements to distributed message queue (dmq module), load balancer, http client, STIR/SHAKEN support, IMS/VoLTE, rtpengine and tls as well as to make available more functions to KEMI interface ... just to list only a few here. Enjoy Kamailio v5.5.0! Thank you for flying Kamailio! Stay safe and healthy! Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online May 17-20, 2021 (Europe Timezone) - June 7-10, 2021 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

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dialog OPTIONS
by David Villasmil 05 May '21

05 May '21

Hello guys, I'm using the dialog module, params are: # ----- dialog params ----- modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_flag", 20) modparam("dialog", "ka_timer", 10) modparam("dialog", "ka_interval", 30) modparam("dialog", "ka_failed_limit", 2) modparam("dialog", "send_bye", 1) modparam("dialog", "timer_procs", 1) When receiving an INVITE, i'm doing dlg_set_property("ka-src"); dlg_set_property("ka-dst"); dlg_set_property("timeout-noreset"); I have a couple questions: The flow is the following: clientA>-\ kamailio<->freeswitch clientB<-/ Kamailio receives an INVITE from ClientA, which I forward to freeswitch, which in turn forwards it back to Kamailio. In reality I would like to monitor only the client legs, but I'm having problems, so I'm enabling OPTIONS on all legs. On the ClientA legs, I see OPTIONS going from kamailio to both freeswitch and the client. But on the ClientB legs i don't see any OPTIONS at all. Can someone point me in the right direction to enable it on all legs? Something else i'm seeing is even though i set "ka_failed_limit" to 2, kamailio is sending several OPTIONS before sending BYEs to both legs (clientA, since there are no OPTIONS on clientB) Thanks all, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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The plan for releasing Kamailio v5.5.0
by Daniel-Constantin Mierla 05 May '21

05 May '21

Hello, after almost 4 weeks since development was frozen, it is time to plan releasing the next major stable version, respectively 5.5.0, therefore I propose to do it on Wednesday, May 5, 2021. The branch 5.5 was already created about one week ago, several doc resources were published (core/variables/transformations/rpcs/stats cookbooks, alphabetic indexes, ...). If anyone is testing an upgrade from branch 5.4 to 5.5 and encounters changes that should be done in Kamailio config, add notes about at: * https://www.kamailio.org/wiki/install/upgrade/5.4.x-to-5.5.0 Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online May 17-20, 2021 (Europe Timezone) - June 7-10, 2021 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

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possible bug on {param.value,name}
by David Villasmil 04 May '21

04 May '21

Hello guys, I'm possible using this wrong, but here it is; I need to get a previously set parameter to rr, named "wsconnid": $var(wsconnid) = $(hdr(Route){param.value,wsconnid}) xlog("L_ERR", "Connection ID parameter: $var(wsconnid)"); And i'm getting: May 3 19:24:45 ip-10-231-32-37 /usr/sbin/kamailio[10015]: ERROR: <script>: WS Connection ID 13> May 3 19:24:49 ip-10-231-32-37 /usr/sbin/kamailio[10014]: ERROR: <script>: Got an OPTIONS We now need to get the connection id from rr This is an example: Route: <sip:10.231.32.37;r2=on;lr;wsconnid=13> The function is returning "13>" instead of "13". Thanks all, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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geting $conid on outbound INVITE
by David Villasmil 04 May '21

04 May '21

Hello guys, I don't have a lot of experience with webrtc (websocket). I'm trying to get $conid when i receive an INVITE via UDP which is GOING to a ws, how do i do that? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Re: [SR-Users] Kamailio 5.4.4 and Siptrace
by pwer spire 04 May '21

04 May '21

Hi, Any one have some idea about this? Or how it should be configured in order to still have the original response received in kamailio instead of the mapping one? Thanks in advance, Regards, Tiago

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Can't make tcp_conid_alive work with $var
by David Villasmil 03 May '21

03 May '21

Hello guys, I'm trying to do the following: On INVITE, i add a param to RR "wsconnid" which is the connection id of the WS socket. Afterwards, i get an OPTIONS including said param. Next I want to retrieve the param and check if the connection is alive. NOTE: using simply "$(hdr(Route){param.value,wsconnid})" returns an extra ">" from the end of the string, and i haven't been able to remove that any other way but with "striptail", it works fine. *** $var(conid) = $(hdr(Route){param.value,wsconnid}); $var(conid) = $(var(conid){s.striptail,1}); xlog("L_ERR", "Connection ID parameter: [$var(conid)]"); if (tcp_conid_alive("$var(conid)")) { xlog("L_ERR", "WS Connection ID $var(conid) is alive"); } else { xlog("L_ERR", "WS Connection ID $var(conid) is NOT alive"); } *** With that, "$var(conid)" contains the connection id properly: *** ERROR: <script>: Connection ID parameter: [2] *** but tcp_conid_alive complaints: *** ERROR: <core> [core/sr_module.c:1638]: get_int_fparam(): Could not convert PV to int ERROR: tcpops [tcpops_mod.c:374]: w_tcp_conid_state(): invalid parameter 'conid' (must be a number) *** Using a hardcoded number such as "1" generates no complaints. Kamailio version: version: kamailio 5.3.7 (x86_64/linux) Not sure what's going on here, any ideas? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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named account with multiple numbers
by beer Ll 03 May '21

03 May '21

Hi I have a configuration with sip accounts which authenticate with the real name of the customers ex: customer1(a)sip.myvoip.dev customer2(a)sip.myvoip.dev ... INBOUND call of multiple numbers are connected with alias_db to the accounts ex: +331234567890 | sip.myvoip.dev | customer1 | sip.myvoip.dev | +331234567891 | sip.myvoip.dev | customer1 | sip.myvoip.dev | +341234567893 | sip.myvoip.dev | customer2 | sip.myvoip.dev | +341234567894 | sip.myvoip.dev | customer2 | sip.myvoip.dev | ... and all works perfectly. OUTBOUND call after authentication with the username kamailio change the caller id and the ip of my ISP-SIP-TRUNK to relay the call with the correct CID for the ISP using the variable $fn (display name in the From header) [SIP-GW-ISP] { #out with ISP route(CID); $ru = "sip:" + $rU + "@" + "IP_ADDRESS_MY-ISP"; route(RELAY); exit; } ROUTE[CID] { $var(caller) = $(fn{re.subst,/"//g}); uac_replace_from("$var(caller)","sip:$var(caller)@$fd"); } Is possible do it with a different approach for example using Remote-Party-ID , PPI P-Preferred-Identity ? Is possible to check if the customer uses the correct outbound number present in the alias_db table ? ex for customer1 only +331234567890 or +331234567891 numbers many thanks Luca

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connect multiple asterisk to kamailio
by Vahid sadrnia 03 May '21

03 May '21

hi everybody how can i connect multiple asterisk to kamailio and user just connect to specific asterisk(not load balancing),in fact each customer have their own media gateway(asterisk) and use kamailio for topolgy hiding. by regards and thanks.

2 1

Set $uac_req(body) to raw hex value
by Nick 03 May '21

03 May '21

Hi all, It seems in the UAC module the $uac_req(body) PV is evaluated as a string, How can I set the value of the request body of a generated request, to a raw hex value without it being interpreted as a string? Cheers, Nick

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new in Kamailio 5.5.x document updated
by Henning Westerholt 03 May '21

03 May '21

Hello, the "new in Kamailio 5.5.x" document has been updated. Again, a really impressive set of changes that lead to this release: Over 1350 commits have been done in the last development period from over 70 individual contributors. You find the document at the usual place in our wiki: https://www.kamailio.org/wiki/features/new-in-5.5.x Many thanks to everybody who contributed to Kamailio. Help in updating the documentation (e.g. core cookbook, pseudo-variable and transformation section) in the wiki is really welcome, as some content is probably still missing there. Best regards, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

2 1

potential PRACK related topos bug [solved]
by Juha Heinanen 03 May '21

03 May '21

Thanks to Daniel who patched topos/topos_redis code, topos now works also in situations where requests spiral through the same proxy twice, for example: A - P1 - P2 - P1 - B. -- Juha

2 1

cannot xlog messages into /var/log/syslog
by Abdulaziz Alghosh 02 May '21

02 May '21

Hi Everyone, i am a beginner with Kamaili Server and I am trying to get along with Kamailio config file by logging some messages by xlog Module. xlog module logs two already-written messages which had been delivered with the default configuration file with <script> as a prefix. Unfortunately, I am not able to log anything further neither with xlog() nor xdbg() functions. may someone please advise? Best regards. Abdul

2 1

Dialog server behind proxy and ka_timer/ka_interval
by David Villasmil 30 Apr '21

30 Apr '21

Hello guys, So i have a setup where client | V (WS) [proxy (no dialog tracking)] | V [dialog server] | V [proxy (no dialog tracking)] (WS) | V client I'd like to know when any of the clients are not available (i.e.: crashed, shutdown unexpectedly. etc), so i enabled on the dialog server ka_timer/ka_interval. I see the OPTIONS going to the proxy when the call starts, and the proxy replying with 200 OK. If i shutdown the client, i see the OPTIONS still going, but the proxy is still responding. I assume the Proxy (kamailio) is simply responding to an OPTIONS message and not actually trying to resend it to the client. Is there anyway of doing this or something better to accomplish the same thing? Thanks! David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

1 0

Dynamically enable ka_timer, ka_interval
by David Villasmil 30 Apr '21

30 Apr '21

Hello guys, is it possible to enable the ka_timer and ka_internal dynamically on a per leg basis? i.e. Only the a-leg? Thanks guys Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

1 0

Development for v5.5.x is frozen
by Daniel-Constantin Mierla 30 Apr '21

30 Apr '21

Hello, the formal notification that the development for the next major version 5.5.0 is now frozen. The focus has to be on testing the master branch. Also, the master branch should not get commits with new features till the branch 5.5 is created, expected to happen in 2-4 weeks, a matter of how testing goes on. Meanwhile, the commits with new features in the C code can be pushed to personal branches, new pull requests can still be done, but they will be merged after branching 5.5. Can still be done commits with documentation improvements, enhancements to related tools (e.g., kamctl, kamcmd), merging exiting pull requests at this moment, exporting missing KEMI functions and completing the functionality of the new modules added in 5.5. Once the branch 5.5 is created, new features can be pushed again to master branch as usual. From that moment, the v5.5.0 should be out very soon, time used for further testing but also preparing the release of packages. If someone is not sure if a commit brings a new feature, just make a pull request and it can be discussed there on github portal or via sr-dev mailing list. A summary of what is new in upcoming 5.5 is going to be built at: * https://www.kamailio.org/wiki/features/new-in-5.5.x Upgrade guidelines will be collected at: * https://www.kamailio.org/wiki/install/upgrade/5.4.x-to-5.5.0 Everyone is more than welcome to contribute to the above wiki pages, especially to the upgrade guidelines, to help everyone else during the migration process from v5.4.x to 5.4.x. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

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Bogon IP Addresses
by HS 29 Apr '21

29 Apr '21

Hi all, We are using RTPProxy with Kamailio on Amazon EC2. Everything seems to work fine, but once in a while we would get IPs like: 10.3.14.156 and no port. Any idea what is wrong? Thanks.

2 2

Default config and options for Kamailio 5.5.0
by Daniel-Constantin Mierla 29 Apr '21

29 Apr '21

Hello, I opened a tracker issue to discuss about default kamailio.cfg to be shipped with upcoming new major version 5.5.0: * https://github.com/kamailio/kamailio/issues/2719 Add your comments on the tracker, to have everyting in a single place -- this message is to notify about on mailing list, not intended to discuss here. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online May 17-20, 2021 (Europe Timezone) - June 7-10, 2021 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

1 0

Video call
by ahmed moghazy 28 Apr '21

28 Apr '21

Hello sir, I make a video call using linphone on kamailio when i made it on local host i works fine when i give my kamailio server static public ip and working on internet and making a video call the screen is always black but the sound is available why ? Thank you sir.

2 1

To-Tag on Register Refresh method
by Duarte Rocha 28 Apr '21

28 Apr '21

Hello all, I'm using Kamailio as a registrar and i have an endpoint that sometimes sends a REGISTER method with a To-Tag when refreshing its regist. My code on this proxy shares many similarities with the default kamailio.cfg configuration file. On that occasion Kamailio answers with a 404 - Not Here since it treats it as a in-dialog request which fails the loose_route check. Why would a CPE send a REGISTER with a To-Tag? Why is a REGISTER with a To-Tag rejected by Kamailio when it could otherwise be a successful REGISTER? Is this behaviour noncompliant with the SIP protocol? I thought about making an exception in the code and having this register following the same path as a register without to-tag, but i don't know if that is unsafe and/or noncompliant with the protocol. Best Regards,

2 1

potential PRACK related topos bug
by Juha Heinanen 27 Apr '21

27 Apr '21

In case of topology: A - P1 - P2 - P1 - B is the idea to use in P1 context "in" when initial request comes to P1 from A or B and context "out" when initial request comes to P1 from P2? Does context need to be set in replies too? If so, is reply context the same as the corresponding request context? And finally, does context need to be set for in-dialog requests/replies the same way as for initial requests? -- Juha

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how to turn off memory debug messages?
by Juha Heinanen 26 Apr '21

26 Apr '21

In config file I have: debug=2 mem_summary=0 Still I get 3400+ memory related syslog message when Kamailio terminates due to sqlops not been able to connect to mysql server. According to core cookbook, memdbg and memlog should not be active when their values are greater than debug value (they default to 3) and mem_summary=0 should disable printing of memory debugging info. What is it that I'm still missing? -- Juha

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unsubscribe list
by Luis Vieira 26 Apr '21

26 Apr '21

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[ANNOUNCE]: sipnagios, a Nagios Plugin to check Call Quality in SIP VoIP (compatible with checkmk, etc)
by Giovanni Maruzzelli 23 Apr '21

23 Apr '21

Hello fellow VoIPers and RTCers, on GitHub there is an early release of sipnagios, opensource. check it out: https://github.com/gmaruzz/sipnagios sipnagios is a Nagios Plugin to check Call Quality in SIP VoIP (compatible with checkmk, etc) sipnagios implements the Nagios plugin API for monitoring and performance data.sipnagios.c is a modification of the original siprtp.c sample in pjproject distribution. Supposedly, it works on Linux, Windows, and anywhere you can compile pjproject on.It makes a call, checks all the various resulting values (mos, rtt, pdd, tta, jitter, packet loss, bytes and packets transferred, and so on). It verifies these values are included into acceptable, warning, or critical ranges.If the call has gone well, sipnagios print performance data for Nagios graphs, and returns 0.If the call fails, or if its measured values are not inside acceptable ranges, it exits with Nagios conventional WARNING or CRITICAL values. mos calculation is scraped from Julien Chavanton work (VoIP Patrol, on GitHub too) I can't even understand :) (merci Julien!) Enjoy! -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

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