- sr-users - kamailio.org

[IGNORE] 5.1.1 python_exec, extra chars ' 0' appended to args
by Anthony Alba 15 Feb '18

15 Feb '18

Apologies for the noise; it is working now. Not sure how I got the extra " 0" appended to the method arguments for python_exec(). AAlba

1 0

[BUG?] 5.1.1 python_exec, extra chars ' 0' appended to args
by Anthony Alba 15 Feb '18

15 Feb '18

Using 5.1, the python_exc args string is getting an extra [_0' appended When I call python_exec("method", "string"); In python: def method(self, msg, args): KSR.info("DBG: {} {} {}".format(msg, args)) This will show 'string 0'; instead of string. AAlba

1 0

RTPEngine Transcoding
by Social Boh 14 Feb '18

14 Feb '18

Hi List, I just read on the RTPEngine page the Proxy RTP now support transcoding: https://github.com/sipwise/rtpengine#transcoding is there a plan to support it? Regards -- --- I'm SoCIaL, MayBe

5 10

RTPEngine QoS statistics and leg labelling
by Alex Balashov 14 Feb '18

14 Feb '18

Hello, I am excited about the new media statistics from RTPEngine, exposed conveniently into PVs, e.g. jitter, packet loss, MOS, time offset of lowest and highest MOS scores, etc. Great work! I am, however, having a bit of trouble understanding how to properly apply this new functionality, as it is not clearly documented (IMHO): 1. What is the meaning of "global" statistics which do not take into account an "A" and a "B" side? That is to say, if I have a packet loss or a jitter or MOS score statistic using just the generic toplevel values, e.g. https://kamailio.org/docs/modules/5.1.x/modules/rtpengine.html#rtpengine.p.… What does that mean? A functional call ordinarily consists of an inbound and outbound RTP stream direction relative to the endpoints on both sides of RTPEngine. Inbound A --> server --> Outbound A Outbound B <-- server <--- Inbound B Where do the "global" stats situate themselves in relation to all this? 2. I am given to understand that the statistics for the "A" and "B" sides of a bidirectional RTP stream can be separated based on an arbitrary label, using these PVs: https://kamailio.org/docs/modules/5.1.x/modules/rtpengine.html#rtpengine.p.… https://kamailio.org/docs/modules/5.1.x/modules/rtpengine.html#rtpengine.p.… How do I use it properly? My intuition was to try something like this: rtpengine_offer("replace-origin replace-session-connection ICE=remove label=outbound"); And in the answer/manage handler on the SDP answer: rtpengine_answer("replace-origin replace-session-connection ICE=remove label=inbound"); Later, when processing a BYE and obtaining subsequent media statistics, I tried this: if(check_route_param("proxy_media=yes")) { $avp(rtpengine_a_label) = 'outbound'; $avp(rtpengine_b_label) = 'inbound'; rtpengine_delete(); route(LOG_RTPENGINE_STATS); } But this only gets me stats for the "A" leg in the JSON object I emit: { "server_addr": "udp:127.0.0.1:5050", "a_label": "outbound", "b_label": "inbound", "a_leg": { "min": { "mos": "4.4", "at": "0:04", "packetloss": "0", "jitter": "0", "roundtrip": 0 }, "max": { "mos": "4.4", "at": "0:04", "packetloss": "0", "jitter": "0", "roundtrip": 0 }, "avg": { "mos": "4.4", "packetloss": "0", "jitter": "0", "roundtrip": 0, "samples": 2 } }, "b_leg": { "min": { "mos": "<null>", "at": "<null>", "packetloss": "<null>", "jitter": "<null>" }, "max": { "mos": "<null>", "at": "<null>", "packetloss": "<null>", "jitter": "<null>" }, "avg": { "mos": "<null>", "packetloss": "<null>", "jitter": "<null>" } } } I also tried rtpengine_delete("label=inbound") (and outbound) individually. Some clarity on how this all fits together would be appreciated. :-) Many thanks to Sipwise for great work on it! -- 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 14

How to add a new module to existing Kamailio Server?
by Arun NV 14 Feb '18

14 Feb '18

I am using a *Kamailio 4.4* server. And I would like to install *http_async_client* module to this server. I have looked in the module directory(*/usr/local/lib64/kamailio/modules*) to check whether this *http_async _client* module already exists there, but it was not there. And I searched for a solution and only found <http://www.kamailio.org/dokuwiki/doku.php/tutorials:openser-modular-install> the method to include new module at the compile time. I could find the modules source code on GitHub <https://github.com/sipwise/kamailio/tree/master/modules/http_async_client>. But I don't know the steps to install a new module to the existing Kamailio Thank you, Arunbalan

2 1

Use RTPProxy in 302 scenario
by Fabian Borot 13 Feb '18

13 Feb '18

I have this scenario Customer --> Kamailio --> Provider1 The route where l route the call to Provider1 is called route["INCOMING"] and I "arm" the replies and failures to : t_on_reply("INCOMING") and t_on_failure("INCOMING"). Provider1 always replies with a 302 Redirect and in the failure_route["INCOMING'] I have this code: if (t_check_status("302")) { get_redirects("*"); t_on_branch("REDIRECT_BRANCH"); t_load_contacts(); t_next_contacts(); t_on_failure("SERIAL_FORKING"); t_relay(); exit; } In the t_on_failure("SERIAL_FORKING") block I do t_relay() as long as there are more contacts to be processed. Lets say I need to try Provider_2 and Provider_3 failure_route["SERIAL_FORKING"] { xlog("L_INFO","mylog: In Failure Route SERIAL_FORKING. Reply [$rs]\n"); if (!t_next_contacts()) { exit; } xlog("L_INFO","mylog: In Failure Route SERIAL_FORKING. Trying next Contact.\n"); t_on_failure("SERIAL_FORKING"); t_relay(); } I want to use RTPProxy in this scenario and I am using the function rtpproxy_manage() to let kamailio handle it automatically. RTPProxy ↑ | | Customer --> Kamailio --> Provider1 | | ↓ Provider N After the 302 response is processed I would like to have the signaling flow "Customer -> Kamailio --> ProviderN" and the RTP flow "Customer --> RTPProxy --> ProviderN" I have tried inserting the function in several places but can't make it work and what I mean is that sometimes the rtpproxy creates the 1st session and then deletes it as soon as I call it the 2nd time with a message similar to this one (log of rtpproxy) : "INFO:416f20b74a58fb75168175e61b2edece@192.168.1.61:5060:rtpp_cmd_delete_ematch: forcefully deleting session 1 on ports 63502/0". Sometimes the kamailio log shows this message: "ERROR: rtpproxy [rtpproxy.c:2536]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy" I think I must call rtpproxy_manage() before I send the INVITE to each of the Providers in the Contact list so they receive a SDP offer with an IP:Port combination provided by the RTPProxy (instead of the ones coming from the Customer). That would be in my route["INCOMING'] Then once Provider_N replies with a 183 with SDP or a 200 OK I would call it again to get a new session to be passed on the reply to my customer. But is the part I am struggling with. Should I add a new reply route inside the failure_route["INCOMING'] only if I am dealing with 302 and do it there ? I am out of ideas guys and any idea is welcome

2 2

Can rtpengine change network interface of estabished call?
by Yufei Tao 13 Feb '18

13 Feb '18

Hi, Is it possible to define two pairs of network interfaces for rtpengine, e.g. int/ext and up/down? If so during a call established between interfaces 'int' and 'ext', is it possible to send a re-INVITE from the calling party on interface 'ext', to update the RTP destination to an IP on a different interface 'up', for example? Or does the updated RTP destination have to be on the same interface as when the call was established? I have some success updating the RTP destination if it is on the same interface (different IP:port) however haven't managed to divert it to a different interface. I wonder if it is possible at all or me missing something? Cheers, Yufei

1 0

(no subject)
by Tihomir Culjaga 13 Feb '18

13 Feb '18

Hi, Im new to Kamailio so im sorry if some of my questions will sound a bit wierd :=) i have a setup like this: (Kamailio version 5.1) UA > Kamailio > Freeswitch Kamailio forwards registration to freeswitch without saving location. I have configured sip over websockets and im able to register to Freeswitcha via kamailio. Also, i use add_path_received() so kamailio adds a path header to register messages recv 945 bytes from udp/[192.168.50.60]:5060 at 22:38:07.283809: ------------------------------------------------------------------------ REGISTER sip:192.168.50.60 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.60;branch=z9hG4bKa929.0f66d60c4333f4a3a9d83d41a63295d9.0 Via: SIP/2.0/WSS 192.0.2.110;rport=63296;received=192.168.200.77;branch=z9hG4bK6931435 Max-Forwards: 69 To: <sip:1001@192.168.50.60> From: <sip:1001@192.168.50.60>;tag=vq30modpgd Call-ID: dlnrna9o4ngn25fb9d1vi2 CSeq: 196 REGISTER Authorization: Digest algorithm=MD5, username="1001", realm="192.168.50.60", nonce="744aaa14-4a86-4ce3-b6d6-419b0a34ee90", uri="sip:192.168.50.60", response="9b592b874474d4ffe5769f2410baceef", qop=auth, cnonce="sf7ff0fp8tm5", nc=00000001 Contact: <sip:1001@192.0.2.110 ;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:587f4694-7e55-46dc-a858-579a82556a20>";expires=600 Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: path, gruu, 100rel, outbound User-Agent: SIP.js/0.7.0 BB Content-Length: 0 Path: <sip:192.168.50.60;lr;received=192.168.200.77~63296~6> ------------------------------------------------------------------------ send 719 bytes to udp/[192.168.50.60]:5060 at 22:38:07.329444: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.50.60;branch=z9hG4bKa929.0f66d60c4333f4a3a9d83d41a63295d9.0 Via: SIP/2.0/WSS 192.0.2.110;rport=63296;received=192.168.200.77;branch=z9hG4bK6931435 From: <sip:1001@192.168.50.60>;tag=vq30modpgd To: <sip:1001@192.168.50.60>;tag=Spya88DKU9gSQ Call-ID: dlnrna9o4ngn25fb9d1vi2 CSeq: 196 REGISTER Contact: <sip:1001@192.0.2.110;transport=ws>;expires=600 Date: Fri, 09 Feb 2018 21:38:07 GMT User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Path: <sip:192.168.50.60>;lr;received=192.168.200.77~63296~6 Content-Length: 0 ------------------------------------------------------------------------ Once, registered, im trying to originate a call to the registered endpoint from freeswitch this results in an invite like this: send 1757 bytes to udp/[192.168.50.60]:5060 at 22:46:21.252549: ------------------------------------------------------------------------ INVITE sip:1001@192.0.2.110;transport=ws SIP/2.0 Via: SIP/2.0/UDP 192.168.50.65:15060;rport;branch=z9hG4bKXHcX316pDgXZB Route: <sip:192.168.50.60>;lr;received=192.168.200.77~63296~6 Max-Forwards: 70 From: "" <sip:1002@192.168.50.65>;tag=tZQ393yprj7BK To: <sip:1001@192.0.2.110;transport=ws> Call-ID: 8212b700-8885-1236-08b3-000c293ae94e CSeq: 118770326 INVITE Contact: <sip:mod_sofia@192.168.50.65:15060> User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 891 X-FS-Support: update_display,send_info Remote-Party-ID: <sip:1002@192.168.50.65 >;party=calling;screen=yes;privacy=off Freeswitch appends a route header to the invite specifying kamailio as route and a received with original uri. well, here is the issue ... im failing to hit the SIP over websocket client... Kamailio my RELAY route is like this route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) { xlog("L_ALERT", "Its a Branch, go manage that !!! <$ru>\n"); t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) { xlog("L_ALERT", "its onreply_route, go manage !!! <$ru>\n"); t_on_reply("MANAGE_REPLY"); } } if (is_method("INVITE")) { if(!t_is_set("failure_route")) { xlog("L_ALERT", "Its failure_route, go manage !!! <$ru>\n"); t_on_failure("MANAGE_FAILURE"); } } $fs = "udp:192.168.50.60:5060"; xlog("L_ALERT", "Trying to rely !!! <$ru>\n"); if (!t_relay()) { xlog("L_ALERT", "failed to rely !!! <$ru>\n"); sl_reply_error(); } exit; } here is a part of the log: Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: WARNING: <core> [core/forward.c:231]: get_send_socket2(): protocol/port mismatch (forced udp:192.168.50.60:5060, to ws:192.0.2.110:5060) Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: <core> [core/msg_translator.c:424]: clen_builder(): content-length: 916 (916) Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: WARNING: <core> [core/msg_translator.c:2765]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: <core> [core/msg_translator.c:1980]: build_req_buf_from_sip_req(): could not create Via header Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: tm [t_fwd.c:476]: prepare_new_uac(): could not build request Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: tm [t_fwd.c:1735]: t_forward_nonack(): failure to add branches Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: tm [t_funcs.c:334]: t_relay_to(): t_forward_nonack returned error -2 (-2) Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: tm [t_funcs.c:352]: t_relay_to(): -2 error reply generation delayed Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: *** cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio.cfg] l=744 a=26 n=xlog Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ALERT: <script>: failed to rely !!! <sip:1001@192.0.2.110;transport=ws> is that because i set a $fu before t_relay() ? any idea how to approach the issue ?

2 4

onend_route() ?
by Wilkins, Steve 13 Feb '18

13 Feb '18

I meant to reply to Kamailio mailing list. From: Wilkins, Steve Sent: Friday, February 9, 2018 12:40 PM To: 'miconda(a)gmail.com' <miconda(a)gmail.com> Subject: RE: [SR-Users] t_relay()? Hi Daniel, I am not sure what I am doing wrong but I cannot get the forwarded message to print. I am using onend_route() and trying to print the results using xdbg. I have searched for examples and have had no luck. Thank you From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: Friday, February 9, 2018 7:48 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>; Wilkins, Steve <swwilkins(a)mitre.org<mailto:swwilkins@mitre.org>> Subject: Re: [SR-Users] t_relay()? Hello, before t_relay() you can't see what is going to the network, because some headers are added when knowing target address and local socket (e.g, Via, Record-Route). You can use onsend_route for seeing the request just before writing to the network. https://www.kamailio.org/wiki/cookbooks/devel/core#onsend_route $snd(buf) will give outgoing buffer -- for more see $sndfrom(...) and $sndto(...) vars. Cheers, Daniel On 09.02.18 11:24, Wilkins, Steve wrote: Hi All, Could someone please tell me, what variable to I use to see what t_relay() fowarding in its message? Thank you, -Steve _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla www.twitter.com/miconda<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda> Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com<http://www.asipto.com> Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com<http://www.kamailioworld.com>

2 1

Request Timeouts
by Wilkins, Steve 13 Feb '18

13 Feb '18

Hello, This is actually a follow up to my last question about some providers calls being dropped. I noticed that when calls are dropped, I see a Kamailio log message "Request Timeout" about 5-10 seconds before the call is dropped. I am wondering what could cause this while a call is in progress. Does Kamailio need to respond to Asterisk with a '200 OK' for an INFO request sent to Kamailio from Asterisk? I am not sure what else could cause this call dropping behavior. Note: It does not appear to be caused from a missing 'ACK'. Thank you, -Steve

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What is the mechanism used for sending push notification from Kamailio, For both IOS and Android?
by Arun NV 13 Feb '18

13 Feb '18

I am creating custom phone client, which uses *Kamailio *as a proxy and it's required to send *push notification* from *Kamailio *to the clients when a call towards the client reached on *Kamailio*. I found a module for APNS here* https://github.com/tvntsr/push <https://github.com/tvntsr/push>*. But I am developing both Android and IOS clients. So What mechanism I can use for the Android side? Is there any module available for *GCM*(Google Cloud Messaging) in Kamailio? Thank you, Arunbalan

2 1

is_audio_on_hold() doesn't support rfc3264
by Julia Boudniatsky 13 Feb '18

13 Feb '18

FYI, After some tests found, that function "*is_audio_on_hold()*" (textops) *supports only rfc 254*3, where "hold" is indicated by setting the "c" destination addresses for the media streams to zero (0.0.0.0). The newest *rfc3264*, where "hold" is indicated by a=sendonly/inactive, *is not supported*. Kamailio version 4.4.6. Best regards, Julia

4 8

Load testing: Taking CPU & MEM stats
by jeevan Chaitanya 12 Feb '18

12 Feb '18

Hi All, Found python library to take system stats during the load. >>> from getstats import GetStats >>> stat_obj = GetStats.getstats(dur=300,rate=10) >>> stat_obj.run_stats() Running.......... ***** Completed capturing the STATS ***** ######################################## # stats_2018-02-12_16-47-47_.csv # ######################################## Doc URL: http://getstats.readthedocs.io/en/latest/ If you require use and let me know. Thanks, K.Jeevan.

1 0

Very inconsistent calls
by Wilkins, Steve 12 Feb '18

12 Feb '18

Hello, I have found that calls are very inconsistent. I use Kamailio 5, Asterisk 14. When certain Providers like (Sorenson, ZVRS) make calls into a WebRTC client (tryit-jssip), sometime the calls stay up until I close them (10-15 minutes), others times those calls drop in 30 seconds. This is extremely confusing...does anyone else experience this type of behavior? -Steve

4 7

t_suspend usage in branch_route
by Aymeric Moizard 09 Feb '18

09 Feb '18

Hi, I'm trying to use t_suspend in my kamailio.cfg from a branch_route. I'm using kamailio 5.1.1 on debian jessie from http://deb.kamailio.org/kamailio51 >From the documentation: http://www.kamailio.org/docs/modules/5.1.x/modules/tmx.html#tmx.f.t_suspend i can see that "This function can be used from ANY_ROUTE ." However, the INVITE is still forwarded for this branch. So obviously the "outgoing transaction" is not suspended. May be "t_suspend" is only suspending "incoming transactions"? I tried to call "drop()". This prevent the INVITE from being sent. But in such case, i have an answer is sent immediatly to the caller (SIP/2.0 500 I'm terribly sorry, server error occurred (6/SL)) I want the script to wait as expected after my "t_suspend". And in the block indicated by t_continue, i want to append a new branch. Any way to do this? Otherwise, do you advise me to control the list of branch before t_relay? I guess I can access the list. Not sure how.. so if you have any tips... Tks Aymeric -- Antisip - http://www.antisip.com

3 9

I: - w_ro_ccr(): Unable to find dialog and cannot do Ro charging without it
by Angelo Colucci 09 Feb '18

09 Feb '18

Hello Everyone, I've been resolved the problem. It was the setflag(FLT_ACC) at the beginning of the block if (is_method("INVITE")) { setflag(FLT_ACC); <----- using instead following function now the dialog is correctly created. setflag(FLT_DIALOG); Best Regards Angelo Da: Angelo Colucci Inviato: giovedì 8 febbraio 2018 18:33 A: 'sr-users(a)lists.kamailio.org' <sr-users(a)lists.kamailio.org> Oggetto: [SR-Users] - w_ro_ccr(): Unable to find dialog and cannot do Ro charging without it Hello Everyone, I'm new in Kamailio environment. We trying to setup a IMS Credit Control Client as per RFC 4006. We've trying to implement the suggesting configuration in ims_charging module doc. The Kamailio version is 5.1.1 Stable. The Server is configured with two local users (A and B) registered on local mysql database. The scenario is the following: 1. In the request_route object we trying to intercept the first INVITE and trigger a diameter credit control request using Ro_CCR function. 2. When A calls B without triggering mechanism the call is fine. 3. When A calls B with credit control triggering, the calls failed and the diamater request is not generated. In the debug we find the following error: exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio.cfg] l=540 a=25 n=xlog ERROR: <script>: Individuato primo INVITE: generata nuova transazione exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio.cfg] l=541 a=63 n=assign exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio.cfg] l=541 a=29 n=Ro_CCR DEBUG: ims_charging [ims_charging_mod.c:494]: w_ro_ccr(): Ro CCR initiated: direction:orig, reservation_units:30, route_name:CHARGING_CCR_REPLY, incoming_trunk_id:1 outgoing_trunk_id:1 DEBUG: ims_dialog [dlg_hash.c:926]: internal_get_dlg(): no dialog callid='yH5kdBGSrM6CowdkkRHTUw..' found DEBUG: ims_dialog [dlg_hash.c:956]: get_dlg(): no dialog callid='yH5kdBGSrM6CowdkkRHTUw..' found DEBUG: ims_dialog [dlg_handlers.c:1950]: dlg_get_msg_dialog(): dlg with callid 'yH5kdBGSrM6CowdkkRHTUw..' not found ERROR: ims_charging [ims_charging_mod.c:511]: w_ro_ccr(): Unable to find dialog and cannot do Ro charging without it The error indicates that w_ro_ccr function doesn't find any dialog. Following you can find the relative configuration of kamailio.cfg. The request_route block is equal to default configuration except the following command: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_DEBUG loadmodule "cdp" loadmodule "cdp_avp" loadmodule "ims_charging" loadmodule "ims_dialog" modparam("ims_charging", "origin_host", "kcc-t-lu-03.mnc049.mcc222.3gppnetwork.org") modparam("ims_charging", "origin_realm", "mnc049.mcc222.3gppnetwork.org") modparam("ims_charging", "destination_realm", "mnc049.mcc222.3gppnetwork.org") modparam("ims_charging","interim_update_credits",30) modparam("ims_charging","timer_buffer",5) # -- ims_dialog params -- modparam("ims_dialog", "dlg_flag", FLT_DIALOG) request_route { ... if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting t_newtran(); $var(cc_ret) = Ro_CCR("CHARGING_CCR_REPLY", "orig", 30, "1", "1"); if ($var(cc_ret) < 0) { xlog("Chiamata a Ro_CCR() fallita\n"); sl_send_reply("402","Payment required"); exit; } } ... } route[CHARGING_CCR_REPLY]{ xlog("cca_return code is $avp(s:cca_return_code): $avp(s:cca_result_code)\n"); switch ($avp(s:cca_return_code)){ case 1: #success break; case -1: #failure switch ($avp(s:cca_result_code)){ case 4010: send_reply("402", "Payment required - Unsufficient funds"); break; case 5006: send_reply("486", "Line limit exceeded"); break; default: send_reply("402","Payment required ($avp(s:cca_result_code))"); break; } default: xlog("L_ERR", "Unknown return code from CCR: [$avp(s:cca_return_code)] \n"); send_reply("500", "Charging Error"); break; } } What's goes wrong? Thanks in advance Angelo Colucci

1 0

Regular exepression issue in permissions module
by Paulo Ferreira 09 Feb '18

09 Feb '18

Hi list, I'm testing permissions module with kamailio 5.1.1 and I was struggling with regular expressions in trusted requests part of module. I noticed that 9[0-9]{4} doesn't work as expected so I had to change it for 9[0-9][0-9][0-9][0-9] to work. Is it supposed to be like that? Is there other way in this module to simplify the expression since it is not accepting the POSIX way? Thanks, Paulo -- ------------------------------------------- Paulo Ferreira VoIP@RCTS - Área de Infraestruturas Aplicacionaisz FCCN http://www.fccn.pt/ Av. do Brasil, n.º 101 1700-066 Lisboa - Portugal Telefone|Phone: +351 218440100; Fax: +351 218472167 Aviso de Confidencialidade/Disclaimer Esta mensagem é exclusivamente destinada ao seu destinatário, podendo conter informação CONFIDENCIAL, cuja divulgação está expressamente vedada nos termos da lei. Caso tenha rececionado indevidamente esta mensagem,solicitamos-lhe que nos comunique esse mesmo facto por esta via ou para o telefone +351 218440100 devendo apagar o seu conteúdo de imediato. This message is intended exclusively for its addressee. It may contain CONFIDENTIAL information protected by law. If this message has been received by error, please notify us via e-mail or by telephone +351 218440100 and delete it immediately.

2 4

t_relay()?
by Wilkins, Steve 09 Feb '18

09 Feb '18

Hi All, Could someone please tell me, what variable to I use to see what t_relay() fowarding in its message? Thank you, -Steve

3 2

- w_ro_ccr(): Unable to find dialog and cannot do Ro charging without it
by Angelo Colucci 08 Feb '18

08 Feb '18

Hello Everyone, I’m new in Kamailio environment. We trying to setup a IMS Credit Control Client as per RFC 4006. We’ve trying to implement the suggesting configuration in ims_charging module doc. The Kamailio version is 5.1.1 Stable. The Server is configured with two local users (A and B) registered on local mysql database. The scenario is the following: 1. In the request_route object we trying to intercept the first INVITE and trigger a diameter credit control request using Ro_CCR function. 2. When A calls B without triggering mechanism the call is fine. 3. When A calls B with credit control triggering, the calls failed and the diamater request is not generated. In the debug we find the following error: exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio.cfg] l=540 a=25 n=xlog ERROR: <script>: Individuato primo INVITE: generata nuova transazione exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio.cfg] l=541 a=63 n=assign exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio.cfg] l=541 a=29 n=Ro_CCR DEBUG: ims_charging [ims_charging_mod.c:494]: w_ro_ccr(): Ro CCR initiated: direction:orig, reservation_units:30, route_name:CHARGING_CCR_REPLY, incoming_trunk_id:1 outgoing_trunk_id:1 DEBUG: ims_dialog [dlg_hash.c:926]: internal_get_dlg(): no dialog callid='yH5kdBGSrM6CowdkkRHTUw..' found DEBUG: ims_dialog [dlg_hash.c:956]: get_dlg(): no dialog callid='yH5kdBGSrM6CowdkkRHTUw..' found DEBUG: ims_dialog [dlg_handlers.c:1950]: dlg_get_msg_dialog(): dlg with callid 'yH5kdBGSrM6CowdkkRHTUw..' not found ERROR: ims_charging [ims_charging_mod.c:511]: w_ro_ccr(): Unable to find dialog and cannot do Ro charging without it The error indicates that w_ro_ccr function doesn’t find any dialog. Following you can find the relative configuration of kamailio.cfg. The request_route block is equal to default configuration except the following command: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_DEBUG loadmodule "cdp" loadmodule "cdp_avp" loadmodule "ims_charging" loadmodule "ims_dialog" modparam("ims_charging", "origin_host", "kcc-t-lu-03.mnc049.mcc222.3gppnetwork.org") modparam("ims_charging", "origin_realm", "mnc049.mcc222.3gppnetwork.org") modparam("ims_charging", "destination_realm", "mnc049.mcc222.3gppnetwork.org") modparam("ims_charging","interim_update_credits",30) modparam("ims_charging","timer_buffer",5) # -- ims_dialog params -- modparam("ims_dialog", "dlg_flag", FLT_DIALOG) request_route { … if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting t_newtran(); $var(cc_ret) = Ro_CCR("CHARGING_CCR_REPLY", "orig", 30, "1", "1"); if ($var(cc_ret) < 0) { xlog("Chiamata a Ro_CCR() fallita\n"); sl_send_reply("402","Payment required"); exit; } } … } route[CHARGING_CCR_REPLY]{ xlog("cca_return code is $avp(s:cca_return_code): $avp(s:cca_result_code)\n"); switch ($avp(s:cca_return_code)){ case 1: #success break; case -1: #failure switch ($avp(s:cca_result_code)){ case 4010: send_reply("402", "Payment required - Unsufficient funds"); break; case 5006: send_reply("486", "Line limit exceeded"); break; default: send_reply("402","Payment required ($avp(s:cca_result_code))"); break; } default: xlog("L_ERR", "Unknown return code from CCR: [$avp(s:cca_return_code)] \n"); send_reply("500", "Charging Error"); break; } } What's goes wrong? Thanks in advance Angelo Colucci

1 0

Feature/help android fcm and ios related push notification module
by Dipen Patel 08 Feb '18

08 Feb '18

hello.. Now most sip soft client in required push wakeup phone. So for that required android, ios push notifications module. I have seen some one have developed 'Tsilo' module based on GCM but it is old. So if possible so please upgrade for Android FCM so it iwll supported for Android and ios both. Thanks

3 2

Using t_relay_to_tcp()
by Wilkins, Steve 08 Feb '18

08 Feb '18

Hi All, Some providers only allow tcp so I switched from t_relay to t_relay_to_tcp(); however, nothing now gets relayed. Has anyone else had luck switching? Thank you, -Steve

3 2

Random 'BYE" sent?
by Wilkins, Steve 08 Feb '18

08 Feb '18

Hello All, I am running Kamailio 5 and using Asterisk 14 as PBX. I have a strange issue. When I receive calls from some providers to a WebRTC client, calls hang up after about 25 seconds. This does not happen with all providers and does not happen with WebRTC client to WebRTC client calls. It appears the "BYE" is coming from Kamailio. Any ideas? Thank you, -Steve

3 2

Re: [SR-Users] How to avoid automatic string to int conversion?
by Yuriy Gorlichenko 08 Feb '18

08 Feb '18

Hi. Yep! I found issue was in if. Fixed it. thx for response! On Feb 8, 2018 18:44, "Daniel-Constantin Mierla" <miconda(a)gmail.com> wrote: It looks like you are using it in some conditional expression (like if(...)) or some assignment with integer expression. Can you paste here the part of the config where you use it? Cheers, Daniel On 08.02.18 04:08, Yuriy Gorlichenko wrote: Hi im using kamailio 5.1 with native cfg file and using json_get_value( method) form jsonso module When i get json string that contains strings and i need to have these strings kamailio tries to convert string i find into int and affcource it can't then i see warning and can't get vaules of the string i have rval_expr_eval_int(): rval expression conversion to int failed (24,6-24,14) How to say kamailio to not convers strng to int? _______________________________________________ Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 0

Info: Kamailio World 2018: First Group Of Speakers Announced
by Daniel-Constantin Mierla 08 Feb '18

08 Feb '18

Hello, the details for the first group of speakers at Kamailio World 2018 have been published: - https://www.kamailioworld.com/k06/speakers/ So far they come from three continents: Europe, North America and Asia, many presenting for the first time at our event. The two sessions present at all editions so far will be there also in 2018, at our 6th edition, respectively Dangerous Demos with James Body and VUC Visions with Randy Resnick. Besides covering various use cases for Kamailio, Asterisk or FreeSwitch, the sessions go into WebRTC, VoLTE/IMS, IoT, blockchains for telecommunications or scalability using NoSQL data storage systems. Definitely another edition with very interesting content – soon we will publish more details about the sessions as well. You can register now to benefit of the early registration price: - https://www.kamailioworld.com/k06/registration/ Looking forward to meeting many of you at Kamailio World Conference, May 14-16, 2018, in Berlin, Germany! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 0

How to avoid automatic string to int conversion?
by Yuriy Gorlichenko 08 Feb '18

08 Feb '18

Hi im using kamailio 5.1 with native cfg file and using json_get_value( method) form jsonso module When i get json string that contains strings and i need to have these strings kamailio tries to convert string i find into int and affcource it can't then i see warning and can't get vaules of the string i have rval_expr_eval_int(): rval expression conversion to int failed (24,6-24,14) How to say kamailio to not convers strng to int?

2 1

dmq_usrloc - which server?
by Alex Balashov 07 Feb '18

07 Feb '18

Hi, The Socket parameter of location entries is not replicated under dmq_usrloc. To make matters worse, the technical characteristics of the registrar module are such that I cannot straightforwardly modify the incoming Contact to add a rider indicating the server on which the register was received. The AoR domain is the same for all registrations received on any server in the DMQ mesh. To reach the registrants, I need to forward the registration to the correct server that corresponds to the true provenance of the registration. What is a reasonable way of finding out what that is? I don't see any method of injecting such values into the replicated contacts, as dmq_usrloc does not provide hooks for this. Thanks, -- 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 19

any interest in segmentation fault at startup ?
by Jean Cérien 07 Feb '18

07 Feb '18

Hi I guess I'm a bit too creative when it comes to kamailio syntax.... just edited the cfg file and got the following error upon startup: kamctl start INFO: Starting Kamailio : /usr/sbin/kamctl: line 1915: 5294 Segmentation fault $OSERBIN -P $PID_FILE -f $ETCDIR/kamailio.cfg $STARTOPTIONS > /dev/null 2> /dev/null ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed I assume it is my syntax in cfg that creates this, I just need to correct it, but if there is interest, I can provide the faulty cfg file J.

4 4

Re: [SR-Users] Blind Transfer WS Error
by Matteo Campana 07 Feb '18

07 Feb '18

Dear Kamailians, does anyone have any idea on the issue below? Alessio told me that is always reproducible, and he can also provide captures and more logs. Thanks in advance! Matteo *From: *sr-users <sr-users-bounces(a)lists.kamailio.org> on behalf of Alessio Miranda <smirolandia(a)gmail.com> *Reply-To: *"Kamailio (SER) - Users Mailing List" < sr-users(a)lists.kamailio.org> *Date: *Tuesday, 16 January 2018 at 14:18 *To: *"sr-users(a)lists.kamailio.org" <sr-users(a)lists.kamailio.org> *Subject: *[SR-Users] Blind Transfer WS Error Hi, I have a problem with the blind transfer via WebSocket, debugging from the browser we see that in some cases when the REFER is done and the sip message "202 Accepted" arrives, the message "180 Ringing" arrives first and then the "100 Trying" and the transfer goes in error with "500 JsSIP Internal Error" I have an architecture with Kamailio in frontend and asterisk in backend on the same network, the sip dialogue between FE and BE is in SIP / UDP. The WebRTC client connects to WebSocket at Kamailio's public address. We have verified that the sip messages exchanged between kamailio and asterisk are always correct. Kamailio Ver. 4.4.4 Asterisk Ver. 13.18.3 PJSIP Ver. 2.6 JSSIP Ver. 3.2.2 Thanks Alessio

2 2

Contact and Record-Route are using Private IP Address- is this correct?
by Wilkins, Steve 07 Feb '18

07 Feb '18

Hello All, Contact and Record-Route are using Private IP Address - is this correct? I though the Public IP should be used. I do not know what setting is causing this to happen. Does anyone have any ideas? Thank you, -Steve

3 4

Info: Markus Monka replacing Marcus Hunger in management group
by Daniel-Constantin Mierla 07 Feb '18

07 Feb '18

Hello, Markus Monka has just replaced Marcus Hunger in Kamailio project management group. Marcus (still at sipgate) has moved to work more on frontend applications than backend, no longer interacting with Kamailio project. Markus Monka has managed the VoIP operations at sipgate for more than 15 years, helping the project over the years with various resources, mainly in respect of organizing events and testing infrastructure, sipgate being one of the oldest VoIP services using Kamailio (since first releases of SER). The change is reflected now on the website: * https://www.kamailio.org/w/management/ Warm welcome to Markus and many thanks to both of them for what they did so far for Kamailio project! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 0

Info: Kamailio project administration updates
by Daniel-Constantin Mierla 07 Feb '18

07 Feb '18

Hello, after several discussions at some of the past IRC devel meetings, finally we started to build a team to be involved more actively in the administration of Kamailio. The project has grown steadily, not only in terms of code, but also packaging, continuous integration, social networking interactions as well as participation to events world wide. For a better coordination and ability to handle related tasks, I invited the most active developers and community members to join so called Kamailio Administration Team, the initial details about it are published as part of management page on kamailio.org: * https://www.kamailio.org/w/management/ Some more details about its rules and purpose: * https://www.kamailio.org/w/kadmin/ It will still take some time to get it properly rolling, more or less now looking to see if the community has suggestions/improvements on what can be done in these aspects. I wrote to all mailing lists for the initial announcement, but you can just reply to sr-users, if you are subscribed there, otherwise it's fine to write back to any of the lists. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 0

Re: [SR-Users] Contact and Record-Route are using Private IP Address- is this correct?
by Ammar Tinawi 07 Feb '18

07 Feb '18

Is it mandatory to have an interface with public ip in order to overcome NAT problems ?

1 0

Issue with Socket selection on Forwarding ACK message
by Thomas Carvello 06 Feb '18

06 Feb '18

Hello, i have an issue with my Kamailio 4.1.9 configuration. This configuration is multi-homed, we have*two network* interfaces, one on a private network and on the public Internet. Kamailio is configured to listen on port 5060 and 5066 on both interfaces. We register two users Alice and Bob on the public Internet using port 5066. Both users are behind a NAT and we capture the SIP exchange on the proxy server. We have set the parameter mhomed=1 When Alice calls Bob, we have Alice Proxy Bob src=5063 dst=5066 INVITE ------------------> src=5066 ------ INVITE ---------------> dst=5060 dst=5066 <------- 200 OK -------------- src=5060 dst=5063 <------- 200 OK --------- src=5066 src=5063 dst=5066 -------- ACK -----------> *src=5060 (blocked by NAT)* ------ ACK-----x dst=5060 The ACK packet gets relayed with the wrong source port. Then the NAT rejects the packet and the call cannot be established. For some reason, when Bob calls Alice, the call is correctly established. Could it be because Bob happend to use 5060 as local port? Also, if we set nhomed=0 it works BUT we are not sure that multi homed is handled correctly. I was wondering if you have encounter this issue before? I have investigated the code for selection socket and what is the logic of this selection ? /*How does kamailo knows that it should choose 5066 as src port if the user is registered using port 5066 instead of 5066?* / Thank you for your time. Thomas

4 8

rtpproxy - no media connection
by Steve 05 Feb '18

05 Feb '18

Dear sr-users, I am trying to set up a simple Kamailio server with rtpproxy on the same host computer, behind a NAT router. “Run your own Skype-like service” is my set-up guide, but I’m using Kamailio version 5.1 so I modified the config files that came with the download. My problem is that SIP clients can register over the internet via TLS encryption with the server and establish a call session connection, but the ZRTP media connection is not being made. Once the call session connection is made, both clients send out UDP packets to each other, but neither receives them—the media traffic is one way and being dropped somewhere. I’ve tried different rtpproxy start settings. If I start the rtpproxy service with “sudo /etc/init.d/rtpproxy start” then it uses the default “unix:/var/run/rtpproxy/rtpproxy.sock”. I’ve also tried starting with “sudo rtpproxy –l /my_public_ip/ –s udp:localhost:7722” which then asks for either –F (superuser) or –u (user). I’ve tried it both with options “–F” and “-u kamailio kamailio”. All the variations tried so far have the same result as noted above. The server behind the router is in a DMZ zone that bypasses the router firewall. Is there anything in the user-list archives that addresses this issue? Or some likely place to start troubleshooting the problem? Any suggestions would be greatly appreciated. Thanks in advance. Steve --- This email has been checked for viruses by Avast antivirus software. https://www.avast.com/antivirus

1 0

IMS (ng-voice vs github)
by Amar Tinawi 04 Feb '18

04 Feb '18

Hello Dears, can you please tell me what are the differences between the github version of ims modules and the ng-voice modules introduced by Mr Carsten repo ? Thanks in advance

2 1

Fosdem dinner
by Daniel-Constantin Mierla 03 Feb '18

03 Feb '18

Hello, in the past 10 years or so, many of us met at Fosdem conference in Brussels (the edition this year happens during February 3-4) and had a dinner on Saturday evening. I know already couple of people from community that will go there, so the question is if there is any interest to organize again such event. If yes, it will be again on Saturday, on the evening of the 3rd of February. At the past editions, it turned into an all-RTC dinner, but we can consider to have it alone again (eventually with few other invited friends), if there are too many or people want a smaller meeting. Reply this week if you want to join, so we can estimate how many seats we need and have enough time to find a place that can accommodate us. More about Fosdem can be found at: - https://fosdem.org There is a devroom on Sunday with topics covering real time communications. I will have a presentation there. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

10 13

SDP Rewrite issues
by Steven Meyer 02 Feb '18

02 Feb '18

Our topology is: Client <> Client Router <> Internet <> Firewall (NAT) <> Kamailio <> FS Unfortunately, when enabling SDP Rewrite: fix_nated_sdp, 0x08, it rewrites packets to the firewall external IP, not the client router external IP. Right now we have to do a local NATed IP on each phone.

1 0

Routing based on A and B numbers
by Grant Bagdasarian 02 Feb '18

02 Feb '18

Hi, Which module would be best suited for routing calls based on A and B numbers? We're currently using the LCR module, with some additional htables to add this functionality, but I was wondering if there is a module which does exactly this? Or perhaps a combination of modules? Regards, Grant Bagdasarian Senior Developer +31765727054<tel:+31765727054> cm.com<https://cm.com> [cid:image001.png@01D39C36.945A76C0]

2 2

Distributed Registrations.
by Володимир Іванець 02 Feb '18

02 Feb '18

Hello! Yesterday I read email conversation "Distributed Presence between Multiple Kamailios" where it was suggested to use Presence module with new dmq parameter. So was wondering if someone was thinking about approach to share registrations between multiple Kamailio servers without using database? Thanks!

2 1

Contact Header need update?
by Duarte Rocha 02 Feb '18

02 Feb '18

Greetings, My proxy does some number normalization in order to guarantee E164 numeration. It also verifies for valid Calling Numbers and replaces them if needed. In order to this, i'm updating the $fU pseudo-variable. Should i update the "Contact" Header too? I've made some testing and i can get calls to work with keeping the Contact intact and changing it to the new number. What is the most correct approach? Cheers, Duarte Rocha

1 0

Distributed Presence between Multiple Kamailios
by SamyGo 01 Feb '18

01 Feb '18

Hi, I'm working on multiple Kamailios that have users distributed evenly among them, I now need to enable presence in those Kamailio boxes. The scenario is a mesh where any user may have subscribed to BLF of other user registered & calling on any other proxy. What modules can help me achieve this? can this be done without creating a new Presence-Node for the whole environment? Hope to get some suggestions on how experts here have done this. Looking forward to positive response. Regards, Sammy

3 5

Case insensitive regexp in textops::search() function
by Andrey Deykunov 01 Feb '18

01 Feb '18

Hi, I have an htable containing some black list words (i.e. user IDs or phone models). To find matches of the words in every message I use the following script: sht_iterator_start("i1", "black_list"); while(sht_iterator_next("i1")) { if ( search($shtitval(i1)) ) { // do something } } sht_iterator_end("i1"); How can I modify search($shtitval(i1)) to get case insensitive matching? The example below does not work: sht_iterator_start("i1", "black_list"); while(sht_iterator_next("i1")) { $var(re) = $shtitval(i1) + "/i"; if ( search($var(re)) ) { // do something } } sht_iterator_end("i1"); Other possible modifications of $var(re) aren't working too. Thanks, Andrey

2 2

Partial/regex match on hash table key?
by Anthony Joseph Messina 31 Jan '18

31 Jan '18

Is is possible (without iterating through all entries in a hash table) to return the value based on a partial or regex match to the key? Something like... $var(one)=$sht(a=>*$var(two)) -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery F9B6 560E 68EA 037D 8C3D D1C9 FF31 3BDB D9D8 99B6

2 2

question about force_send_socket
by Fabian Borot 31 Jan '18

31 Jan '18

our kamailo has 2 IPs, a public (87.x.x.x) and a private (192.168.1.1). It is listening on all of them. For all calls it listens on the public IP but for some calls I need to forward the incoming INVITE to a server in the private network so I do a " force_send_socket(privateip:port)" before the route that forwards the INVITE to the server in the private network and that works. But then when the server in the private network replies with 302 I have to contact the servers inside the Contact header using the Public IP now so I do this hoping that it will switch back to the public IP: if (t_check_status("302")) { force_send_socket(PUBLIC_IP:5060); get_redirects("*"); t_relay(); } But it stays in the private, so the INVITEs to the servers inside the Contact header are sent from the Private IP and they are blocked. This would be the flow: customer INVITE -> kamailio(publicIP) kamailio (publicIP) 100 trying -> customer kamailio(privateIP) INVITE -> Server(privateIP) Server(privateIP) 302 redirect -> kamailio(privateIP) kamailio(privateIP) ACK -> Server(privateIP) after I would like this to happen kamailio(publicIP) INVITE -> Server1(publicIP) but instead this happens: kamailio(privateIP) INVITE -> Server1(publicIP) how can I switch back to the public IP to continue with this call flow? thank you

2 2

[4.2] how to send out BYE successfully to close a TCP/TLS connection which misses the connection ID?
by Jack Wang 31 Jan '18

31 Jan '18

For some reasons, My settings will cause kamailio to be restarted even if there were connections connected to it. And when one SIP client in a connection wanted to hangup the call with a WebRTC Client (SIP over websocket) after the proxy-restarting, local client was closed, however , the other side was not. >From the log below it seems that the connection missed. Jan 30 18:19:31 ./kamailio[27857]: WARNING: <core> [msg_translator.c:2777]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jan 30 18:19:31 ./kamailio[27857]: ERROR: tm [t_msgbuilder.c:1365]: assemble_via(): assemble_via: via building failed Jan 30 18:19:31 ./kamailio[27857]: ERROR: tm [t_msgbuilder.c:1545]: build_uac_req(): build_uac_req(): Error while assembling Via Jan 30 18:19:31 ./kamailio[27857]: ERROR: tm [uac.c:379]: t_uac_prepare(): t_uac: Error while building message So, Is it equivalent to connection ID missing? And is there any way to make Websocket found the connection? Btw, after proxy-restarting for seconds , I typed `sh kamctl stats` : websocket:ws_current_connections = 1 It showed that the WebRTC Client will re-connect automatically, So I think this connection could be found by proxy, Any suggestion? Thanks :)

1 2

Normal behaviour with loose_routing
by Loic Chabert 30 Jan '18

30 Jan '18

Hi list, I have a setup configurered has diagram below: UAC -> Proxy LB (proxy lb) -> Proxy Router/registrar (proxy B) -> Asterisk SBC My UAC is SIPP and sending a classic call scenario: INVITE - ACK - BYE. But i have some trouble to understand which queries should be loose routed and which queries should be dispatched by dispatcher module. My first invite request has been dispatched to proxy B by proxy LB. After that, ACK and BYE request should be loose_routed or should be dispatched as other request ? My proxy B is a statefull proxy, logging cdrs and other stuff ... >From my point of view, subsequent request should be loose_routed to keep dialog informations consistent. My subsequent requests have record-route headers. Should i use theses headers to route my calls ? Loose routing features only work with "Route" header ? What is the difference beetwen Route headers and Record-route headers ? Thanks in advance, Regards.

3 5

rtjson + uac + event - OK, ACK and BYE display name not updated
by Diego Nadares 30 Jan '18

30 Jan '18

Hi Guys, We are modifying "from" and "to" headers with rtjson module via a json struct. It works fine in INVITE, TRYING, RINGING and OK but not in the following messages. Kamailio It's changing ONLY "from" and "to" URI BUT NOT the DISPLAY NAME. I think it's the tm module but I can't figure out how to confirm this. Maybe I need to configure something else in my cfg or could it be a bug? My json struct that is handled by rtjson module response.routing = "serial"; //response.routing = "parallel"; response.routes = []; response.routes[0] = {}; response.routes[0].uri = "sip:1111111111@172.16.213.31:5060"; response.routes[0].headers = { // Headers to be modified "from": { "display": "1111111111", "uri": "sip:111111111@172.16.213.38" }, "to": { "display": "2222222222", "uri": "sip:2222222222@172.16.213.38" }, }; response.routes[0].headers.extra = {};//"X-Hdr-A: abc\r\nX-Hdr-B: bcd\r\n"; This is part of my cfg .... modparam("dialog","dlg_flag", FLD_START) modparam("uac","restore_mode","auto") modparam("uac", "restore_dlg", 1) .... # account only INVITEs if (is_method("INVITE") && !has_totag()) { setflag(FLT_ACC); # do accounting * setflag(FLD_START); # do accounting* setflag(FLT_ACCMISSED); setflag(FLT_ACCFAILED); } ..... >From source/Aleg: *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>.* To dest/Bleg: *From: 1111111111 <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: 2222222222 <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>.* Incorrect display name in BYE and ACK: *From: sipp <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: sut <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* interface: any filter: ( port 5060 ) and (ip or ip6) # U 2016/12/06 15:52:13.638731 172.16.213.21:5060 -> 172.16.213.38:5060 INVITE sip:5555555555@172.16.213.38:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-0. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Contact: "SIPP" <sip:sipp@172.16.213.21:5060>. Max-Forwards: 70. Subject: Performance Test. Content-Type: multipart/mixed;boundary=uniqueBoundary. Remote-Party-ID: <sip:sipp@172.16.213.21 >:5060;party=calling;id-type=subscriber;privacy=off;screen=no>. Content-Length: 549. . --uniqueBoundary. Content-Type: application/sdp. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.21. s=-. c=IN IP4 172.16.213.21. t=0 0. m=audio 6000 RTP/AVP 8. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11,16. . --uniqueBoundary. Content-Type: application/gtd. Content-Disposition: signal;handling=optional. . IAM,. CGN,04,y,1,y,4,1143617400. CIC,0000000573. CPC,09. CPN,02,y,1,52381660. FCI,n,n,n,n,y,n,n,u. GCI,f7140cc78a611601838a002128d7e512. NOC,0,n,1,n. PRN,q761*,AR*,oper2,1993. TMR,02. . --uniqueBoundary--. ................ # U 2016/12/06 15:52:13.734159 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-0. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Server: kamailio (4.4.4 (x86_64/linux)). Content-Length: 0. . ................ # U 2016/12/06 15:52:13.878600 172.16.213.38:5060 -> 172.16.213.31:5060 INVITE sip:1111111111@172.16.213.31:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bKabaa.840cc83114975a34a37b7af020d37b65.1. *From: 1111111111 <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: 2222222222 <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Max-Forwards: 69. Subject: Performance Test. Content-Type: multipart/mixed;boundary=uniqueBoundary. Content-Length: 788. Contact: <sip:btpsh-58470879-6c61-1@172.16.213.38>. . --uniqueBoundary. Content-Type: application/sdp. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.38. s=-. c=IN IP4 172.16.213.38. t=0 0. m=audio 40236 RTP/AVP 8. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11,16. a=sendrecv. a=rtcp:40237. a=ice-ufrag:QNCzXhUr. a=ice-pwd:Ci90BRpdZH51Hx5BaPgGMrGCeh. a=candidate:P1djl8LFNb5kexbG 1 UDP 2130706431 172.16.213.38 40236 typ host. a=candidate:P1djl8LFNb5kexbG 2 UDP 2130706430 172.16.213.38 40237 typ host. . --uniqueBoundary. Content-Type: application/gtd. Content-Disposition: signal;handling=optional. . IAM,. CGN,04,y,1,y,4,1143617400. CIC,0000000573. CPC,09. CPN,02,y,1,52381660. FCI,n,n,n,n,y,n,n,u. GCI,f7140cc78a611601838a002128d7e512. NOC,0,n,1,n. PRN,q761*,AR*,oper2,1993. TMR,02. . --uniqueBoundary--. ................ # U 2016/12/06 15:52:13.879441 172.16.213.31:5060 -> 172.16.213.38:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bKabaa.840cc83114975a34a37b7af020d37b65.1. *From: 1111111111 <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: 2222222222 <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Contact: <sip:172.16.213.31:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 137. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.31. s=-. c=IN IP4 172.16.213.31. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. ................ # U 2016/12/06 15:52:13.936565 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 180 Ringing. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Content-Type: application/sdp. Content-Length: 316. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-0. Contact: <sip:atpsh-58470879-6c61-2@172.16.213.38>. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.38. s=-. c=IN IP4 172.16.213.38. t=0 0. m=audio 40248 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=sendrecv. a=rtcp:40249. a=candidate:P1djl8LFNb5kexbG 1 UDP 2130706431 172.16.213.38 40248 typ host. a=candidate:P1djl8LFNb5kexbG 2 UDP 2130706430 172.16.213.38 40249 typ host. ................ # U 2016/12/06 15:52:20.882165 172.16.213.31:5060 -> 172.16.213.38:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bKabaa.840cc83114975a34a37b7af020d37b65.1. *From: 1111111111 <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: 2222222222 <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Contact: <sip:172.16.213.31:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 213. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.31. s=-. c=IN IP4 172.16.213.31. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. ................ # U 2016/12/06 15:52:21.173798 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 200 OK. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Content-Type: application/sdp. Content-Length: 380. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-0. Contact: <sip:atpsh-58470879-6c61-2@172.16.213.38>. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.38. s=-. c=IN IP4 172.16.213.38. t=0 0. m=audio 40248 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. a=rtcp:40249. a=candidate:P1djl8LFNb5kexbG 1 UDP 2130706431 172.16.213.38 40248 typ host. a=candidate:P1djl8LFNb5kexbG 2 UDP 2130706430 172.16.213.38 40249 typ host. ................ # U 2016/12/06 15:52:21.174390 172.16.213.21:5060 -> 172.16.213.38:5060 ACK sip:atpsh-58470879-6c61-2@172.16.213.38 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-4. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 ACK. Contact: sip:sipp@172.16.213.21:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . ................ # U 2016/12/06 15:52:21.248172 172.16.213.38:5060 -> 172.16.213.31:5060 ACK sip:172.16.213.31:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bKabaa.98116b5795d4c30f52156187a72059c8.0. *From: sipp <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: sut <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 ACK. Max-Forwards: 69. Subject: Performance Test. Content-Length: 0. Contact: <sip:btpsh-58470879-6c61-1@172.16.213.38>. . ................ # U 2016/12/06 15:52:51.178826 172.16.213.21:5060 -> 172.16.213.38:5060 BYE sip:atpsh-58470879-6c61-2@172.16.213.38 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-7. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 2 BYE. Contact: sip:sipp@172.16.213.21:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . ................ # U 2016/12/06 15:52:51.468639 172.16.213.38:5060 -> 172.16.213.31:5060 BYE sip:172.16.213.31:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bK7baa.3cb44935b506e759e3418501b0688b08.0. *From: sipp <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: sut <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 2 BYE. Max-Forwards: 69. Subject: Performance Test. Content-Length: 0. Contact: <sip:btpsh-58470879-6c61-1@172.16.213.38>. . ................ # U 2016/12/06 15:52:51.469153 172.16.213.31:5060 -> 172.16.213.38:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bK7baa.3cb44935b506e759e3418501b0688b08.0. *From: sipp <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: sut <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 2 BYE. Contact: <sip:172.16.213.31:5060;transport=UDP>. Content-Length: 0. . ................ # U 2016/12/06 15:52:51.519423 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 200 OK. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 2 BYE. Content-Length: 0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-7. Contact: <sip:atpsh-58470879-6c61-2@172.16.213.38>. . ................ exit 13 received, 0 dropped Sorry for my english. Thanks in advance! Diego.

2 7

Session Timers
by Duarte Rocha 30 Jan '18

30 Jan '18

Hey guys, I have two issues about Session Timers that need some clarification. 1) I have my Kamailio between my clients CPE's and my GW to the PSTN. I want to configure it in a way that i always try to make the CPE do the refreshing so I can save resources on my kamailio and in the GW. If the CPE doesn't support it, the refresh should be done by the GW. Is it possible to do this test and configuration in every call ? 2) SIP has the MIN-SE mechanic and is well documented, but is it possible to implement a MAX-SE in any way ? Thanks for the help. If you need more details about the issue, please let me know. Cheers

2 1

Re: [SR-Users] Problem to install Kameilio from Ubuntu packages
by Victor Seva 30 Jan '18

30 Jan '18

2018-01-26 17:46 GMT+01:00 Alain Bonnefoy - XOOL <a.bonnefoy(a)xool.fr>: > > > I tried to install Kameilio on Ubuntu 16.04 from what explained in the > wiki <https://www.kamailio.org/wiki/packages/debs>. > Wiki updated. Thanks

1 0

Re: [SR-Users] Problem to install Kameilio from Ubuntu packages
by Bastian Triller 30 Jan '18

30 Jan '18

There another command on http://deb.kamailio.org for people behind firewalls which block port 11371: % wget -O- http://deb.kamailio.org/kamailiodebkey.gpg | sudo apt-key add - On Jan 26, 2018 5:46 PM, "Alain Bonnefoy - XOOL" <a.bonnefoy(a)xool.fr> wrote: > Hello, > > > > I tried to install Kameilio on Ubuntu 16.04 from what explained in the > wiki <https://www.kamailio.org/wiki/packages/debs>. > > > > But when I execute the command to add your GPG key, I just get replied > that the timeout occured. > > > > What to do ? > > > > Alain > > Provenance : Courrier <https://go.microsoft.com/fwlink/?LinkId=550986> > pour Windows 10 > > > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > >

1 0

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