sr-users March 2018

sr-users@lists.kamailio.org

101 participants
132 discussions

Can't get route[AUTH] working as expected.
by Aqs Younas 25 Mar '18

25 Mar '18

Greetings list. I can see that I was able to bypass the default route[AUTH] if I send an invite containing from_uri which is not local but requested line containing a local user. llisten=udp:172.16.40.10:5060 route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } Below INVITE get passed above auth route. INVITE sip:60129879190@172.16.40.10 SIP/2.0 Via: SIP/2.0/UDP 139.5.177.91:5060;branch=z9hG4bK31edc7f4;rport Max-Forwards: 70 From: <sip:0128888877@139.5.177.99>;tag=as2274e806 To: <sip:60129879190@172.16.40.10> Contact: <sip:0128888877@139.5.177.91:5060> Call-ID: 7b6d32bc6c679bb23eb248b955c0ac8b@139.5.177.91:5060 CSeq: 102 INVITE User-Agent: FPBX-13.0.194.2(13.17.0) Date: Fri, 23 Mar 2018 09:33:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 321 v=0 o=root 237494576 237494576 IN IP4 139.5.177.99 s=Asterisk PBX 13.17.0 c=IN IP4 139.5.177.99 t=0 0 m=audio 15332 RTP/AVP 0 18 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv >From INVITE and route[AUTH] I can see why it is being passed. But should not it by default authenticate every request if IP address is not allowed in permission module. Br, Aqs.

2 4

Best approach for Kamailio behind nat
by Evgeny Kuchkov 25 Mar '18

25 Mar '18

Sorry for my previous mail with typos. I have the current configuration located on google cloud servers: [Alice a.a.a.a]--->(GCN NAT x.x.x.x)--->(Instance with Kam 10.0.0.1)--->(next proxy 10.0.0.2) Currently I have configured a fake alias with IP x.x.x.x in Debain and set listeners this way: listen=udp:10.0.0.1:5060 advertise x.x.x.x:5060 listen=udp:x.x.x.x:5060 200OK from Kam to Alice: Via: a.a.a.a Record-Route: <sip:alias1@10.0.0.2;lr> (used for further internal routing) Record-Route: <sip:alias2@10.0.0.2;lr> Record-Route: <sip:x.x.x.x;lr> Contact: <sip:dnis@x.x.x.x> ACK recived from Alice with reversed RR: ACK sip:dnis@x.x.x.x Via: a.a.a.a Route: <sip:x.x.x.x;lr>, <sip:alias2@10.0.0.2;lr>,<sip:alias1@10.0.0.2;lr> So what Kam does - it goes through loose_route() and dispatches two ACKs to x.x.x.x (from which I'd like to get rid of) and 10.0.0.2 First one is: ACK sip:alias1@10.0.0.2 Via: x.x.x.x Via: a.a.a.a Route: <sip:x.x.x.x;lr>, <sip:alias2@10.0.0.2;lr> And the next one: ACK sip:alias1@10.0.0.2 Via: x.x.x.x Via: x.x.x.x (Doubles Via) Via: a.a.a.a Route: <sip:alias2@10.0.0.2;lr> So my question is - how to make Kam think that x.x.x.x it's own address and stop dispatching in-dialog messages to itself? Currently I'm thinking about chooping off one Route/Record-Route in route[WITHINDLG] tight before loose_route(). But maybe there's a better way to solve my problem?

1 0

Best approach for Kamailio behind nat
by Evgeny Kuchkov 24 Mar '18

24 Mar '18

Hi all, I have the current configuration located on google clod servers: [Alice a.a.a.a]--->(GCN NAT x.x.x.x)--->(Instance with Kam 10.0.0.1)--->(next proxy 10.0.0.2) Currently I have configured a fake alias with IP x.x.x.x in Debain and set listeners this way: listen=udp:10.0.0.1:5060 advertise x.x.x.x:5060 listen=udp:x.x.x.x:5060 200OK from Kam to Alice: Via: a.a.a.a Record-Route: <sip:alias1@10.0.0.2;lr> (used for further internal routing) Record-Route: <sip:alias2@10.0.0.2;lr> Record-Route: <sip:x.x.x.x;lr> Contact: <sip:dnis@x.x.x.x> ACK recived from Alice with reversed RR: ACK sip:dnis@x.x.x.x Via: a.a.a.a Route: <sip:x.x.x.x;lr>, <sip:alias2@10.0.0.2;lr>,<sip:alias1@10.0.0.2;lr> So what Kam does - it goes through loose_route() and dispatches two ACKs to x.x.x.x (from which I'd like to get rid of) and 10.0.0.2 First one is: ACK sip:alias1@10.240.0.67 Via: x.x.x.x Via: a.a.a.a Route: <sip:x.x.x.x;lr>, <sip:alias2@10.0.0.1;lr> And the next one: ACK sip:alias1@10.240.0.67 Via: x.x.x.x Via: x.x.x.x (Doubles Via) Via: a.a.a.a Route: <sip:alias2@10.0.0.1;lr> So my question is - how to make Kam think that x.x.x.x it's own address and stop dispatching in-dialog messages to itself? Currently I'm thinking about chooping off one Route/Record-Route in route[WITHINDLG] tight before loose_route(). But maybe there's a better way to solve my problem?

1 0

dlg_manage() and dlg_flag
by Alex Balashov 24 Mar '18

24 Mar '18

Hi, One aspect of the dialog module that is very confusing to newbies is the relationship between the dlg_flag modparam and dlg_manage(). According to the documentation, dlg_manage() effectively seems to obsolete the requirement for a flag[1]: Process current SIP request with dialog module. It is an alternative to setting dialog flag for initial INVITE and Route-parameter-callback execution for within-dialog requests. Yet, if one does not set a dlg_flag modparam, the dialog module fails to load: Mar 23 15:42:29 evaristesys-legacy /usr/local/sbin/kamailio[2900]: ERROR: dialog [dialog.c:500]: mod_init(): invalid dlg flag -1!! Perhaps the flag modparam should be made optional, and any arguments for using one over the other more clearly explained in the documentation. I'd make the latter contribution, but unfortunately am not entirely sure what those are. Cheers, -- Alex [1] https://kamailio.org/docs/modules/5.1.x/modules/dialog.html#dialog.f.dlg_ma… -- 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 3

have successfully installed but ..
by abartlett＠vixtel.com.au 23 Mar '18

23 Mar '18

having trouble configuring kamalio to move calls between servers and carriers

3 3

tcpconn_main_timeout
by Kjeld Flarup 23 Mar '18

23 Mar '18

We are developing an app with a pjsip client. I see this error when trying to send an invite from kamailio. Mar 17 00:29:46 raspberrypi /usr/sbin/kamailio[18720]: ERROR: <core> [tcp_main.c:4258]: tcpconn_main_timeout(): connect 62.44.134.85:43829 failed (timeout) A few seconds before 62.44.134.85:43829 did a register. I can understand that either the TCP connection was close by the other end or no one is listening. How much can I see in a pcap from this scenario? I cannot see the invite, but somehow there must be some TCP communication to show what is failing. -- -------------------- 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

3 10

htable not visible in another thread
by Kjeld Flarup 22 Mar '18

22 Mar '18

Im trying to do a t_suspend, but cant get htable to work I save to $sht $sht(vtp=>join::$rU) = "" + $T(id_index) + ":" + $T(id_label); xlog( "L_ALERT", "Suspended transaction sht(vtp=>join::$rU) = [$sht(vtp=>join::$rU)]\n" ); Mar 19 23:36:28 raspberrypi /usr/sbin/kamailio[2167]: ALERT: <script>: Suspended transaction sht(vtp=>join::004540294149) = [23690:1238111367] But in my register thread nothing is read xlog("L_ALERT","sht(vtp=>join::$tU) = $sht(vtp=>join::$tU)\n"); Mar 19 23:36:31 raspberrypi /usr/sbin/kamailio[2114]: ALERT: <script>: sht(vtp=>join::004540294149) = <null> # -------- htable params -------------- modparam("htable", "db_url", DBURL) modparam("htable", "htable", "vtp=>size=10;autoexpire=120;dbtable=htable;dbmode=1") modparam("htable", "htable", "a=>size=6;") I never see any data in htable in the database. What am I doing wrong? -- -------------------- 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

3 3

phonenum module erratic behaviour with 5.1.2
by George Diamantopoulos 22 Mar '18

22 Mar '18

Hello all, It seems that the phonenum module in at least 5.1.2 (haven't tried previous versions) is behaving somewhat erratically. More specifically, if there are any references to any phn PVs with the key other than "number": if ( phonenum_match("1-484-555-8888", "src") ) { xlog("number norm: $phn(src=>number)\n"); xlog("number country: $phn(src=>country)\n"); dbg_pv_dump(30, "L_ERR"); } kamailio will not start: Mar 19 20:06:49 modcom-sbc-dev1 kamailio[5042]: 0(5044) DEBUG: <core> [core/pvapi.c:494]: pv_spec_lookup(): PV <$phn(src=>country)> is not in cache Mar 19 20:06:49 modcom-sbc-dev1 kamailio[5042]: 0(5044) DEBUG: phonenum [phonenum_pv.c:160]: pv_parse_phonenum_name(): phonenum [src] - key [country] Mar 19 20:06:49 modcom-sbc-dev1 kamailio[5042]: 0(5044) ERROR: phonenum [phonenum_pv.c:205]: pv_parse_phonenum_name(): error at PV phonenum name: src=>country Mar 19 20:06:49 modcom-sbc-dev1 kamailio[5042]: 0(5044) ERROR: <core> [core/pvapi.c:951]: pv_parse_spec2(): pvar "phn" has an invalid name param [src=>country] Mar 19 20:06:49 modcom-sbc-dev1 kamailio[5042]: 0(5044) ERROR: <core> [core/pvapi.c:1106]: pv_parse_spec2(): wrong char [)/41] in [$phn(src=>country)] at [17 (5)] Mar 19 20:06:49 modcom-sbc-dev1 kamailio[5042]: 0(5044) ERROR: xlog [xlog.c:513]: xdbg_fixup_helper(): wrong format[number country: $phn(src=>country) If I comment the offending line, all is well, but when dumping PVs with dbg_pv_dump(30, "L_ERR");, only $phn(src=>number) is listed, not the other keys (country, operator, region according to the docs) However, what is more disturbing is that phn(src=>number) always contains the string provided to phonenum_match() function, even if invalid characters are included, with no normalization. For example: phonenum_match("14~8%^88", "src") will result in dbg_pv_dump reporting: "$phn(src=>number)": "14~8%^88" Please note that I'm using sipwise's debian repository for stretch for kamailio, and debian's stable stretch repository for libphonenumnber (currently at 7.1.0-5+b1). I have also tried building current libphonenumber myself (version v8.9.2), but the result is the same. Is this a bug? Or am I missing something? Thanks! BR, George

3 4

Kamailio htable.dump command returns - reply too big
by José Seabra 22 Mar '18

22 Mar '18

Hello there, I have an htable with 3000 entries and when i try dump the htable through the command kamcmd htable.dump htable, this command returns "ERROR: reply too big". I have tried to increase the body size using the following modparam: modparam("ctl", "binrpc_max_body_size",1024) But the result still the same "ERROR: reply too big", do I need to configure another parameter to be able dump this htable? The kamailio version that i'm using is 4.4.2 Thank you. Regards José Seabra

2 3

Kamailio behind NAT no Audio
by Benjamin Marty 22 Mar '18

22 Mar '18

Hi, I am new to Kamailio and coming a bit from the Asterisk side. Currently I'm using the Amazon EC2 Infrastructure and set up an Kamailio with standard configuration. EC2 uses Destination NAT on their side. Thus I enabled WITH_NAT and WITH_NATSIPPING. I'm using the rtpproxy from master branch of https://github.com/miconda/rtpproxy. Besides those two parameters I also changed the listen parameter in the Kamailio config: listen=udp:10.0.0.221:5060 advertise 52.59.XXX.XXX:5060 My rtpproxy is running with following parameters: ./rtpproxy -A 52.59.XXX.XXX -F -l 10.0.0.221 -m 20000 -M 30000 -s udp:*:7722 -d INFO When I use two Softclients (Zoiper) on different Networks (with Source NAT on the Firewall) the Softclient tries to transmit the RTP Packets to the Public IP of the other Network. But the Client obviously thus doesn't get the RTP Packets. Is there an easy and documented way to solve that issue? If I understood it correctly Kamailio together with rtpproxy should be able to receive the RTP Packets and forward them to the other client. So it should signal the IP of the Kamailio to send the RTP packets to. Or am I on the complete wrong way? Regards, Benjamin

1 0

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sr-users March 2018