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11 participants
37526 discussions

NOTIFY message without XML part
by denislav.tsonev＠musala.com 27 Mar '18

27 Mar '18

Hello, I use kamailio 5.1.1 with default configuration for PRESENCE: Route: #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif and modparams: #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif I use and GRANSTREAM phones. When I make subscribe from soft sip client (Micro sip) to the grandstreams, all NOTIFY messages are without XML. Subscriptions between two soft sip clients like Zoiper and Micro SIP works perfectly. Is this related to GRANDSTREAM phones - configuration or so... or is necessary to modify kamailio configuration and if Yes, please provide some hitns how to change kamailio config? Thanks in advance. Best regards. Denislav Raychev Tsonev | Integration and Infrastructure Engineer | Musala Soft JSC www.musala.com | t: +359 2 969 58 21 | m: +359 878 270 965 | f: +359 2 969 58 22 World Trade Center, 36 Dragan Tsankov blvd., Sofia 1057, Bulgaria

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Strange auth_challenge behaivor on 407
by Yuriy Gorlichenko 27 Mar '18

27 Mar '18

Hi I'm using version: kamailio 5.1.2 I have strange issue with auth_challenge kamialio works as NAT handler. When Register comes kamailio handles nat and save original ip:port at the received field in location table I found that when local port in contact is different than external port reply 407 on INVITE comes to externalip (from received) : internalport(from contact) And never reaches destination for exmaple Register comes from 192.168.1.10:50555 Kamailio saves at the Received 12.1.1.12:1030 but when invite comes form 12.1.1.12:1030 kamailio sends reply to 12.1.1.12:50555 i don't think that this is an issue in kamailio source code but i can find where i can configure it

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How should i configure kamailio connection freepbx?
by 星昊通技术 27 Mar '18

27 Mar '18

hello all: I install kamailio 5.1 and siremis 5.1, I want to configure kamailio connection freepbx, how should i configure kamailio vai siremis GUI? Thanks. ------------------ Best regards! alex Hiastar 深圳星昊通科技 Sangoma in APAC, 潮流网络IP话机，IPPBX Asterisk cards, SBC, NetBorder VOIP Gateway, Transcoding, 潮流网络IP话机 QQ：2933996052 website:www.hiastar.com|www.freepbx.cn, contact number: 13574109204 关注微信公众帐号：Asterisk-CN

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Help on setting UP SIP PRoxy
by Mohammed Ayub 26 Mar '18

26 Mar '18

HI, Myself Ayub from Phonology, Bangalore, India. We are trying to set up an SIP proxy server using Kamailio with Siremis as web interface. Have installed and configured kamailio and siremis but need help in config if sip domain , users and etc... Please revert. Regards Ayub

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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/

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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

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