sr-users November 2014

sr-users@lists.kamailio.org

91 participants
128 discussions

by elektau＠yandex.ru

Hello All, I'm using Kamailio 4.2 running TLS only connection on non standard port 5065. I added the following line to the kamailio.cfg: listen=tls:111.222.333.444:5065 Domain SRV record as follows: _sips._tls.domain.com port 5065 When I try to connect using a softphone to domain.com login/password - "error while registering" When I connect using domain.com:5065 login/password - connection established successfully. So the question is: why I don't get redirected automatically to the specified port? If I use the default port for TLS (5061) in kamailio.cfg and in SRV, I can successfully connect using domain.com login/password Please help. Many Thanks!

10 years, 3 months

cannot register UA when domain is not valid, using usrloc, auth_db and registrar modules

by Vik Killa

Hello, I'm trying to register 1004(a)3.local (yes i know 3.local is not a valid domain) and kamailio is responding with SIP/2.0 478 Unresolvable destination (478/SL) Why does kamailio require a valid domain? Is there any way to disable kamailio from looking up the domain (i don't see a reason to look it up since it's just a REGISTER request) Here is my cfg: if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "1"); exit; } And kamilio is responding with: Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6931]: ERROR: <core> [resolve.c:1726]: sip_hostport2su(): could not resolve hostname: "3.local" Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6931]: ERROR: tm [ut.h:337]: uri2dst2(): failed to resolve "3.local" Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6931]: ERROR: tm [t_fwd.c:1778]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6931]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unresolvable destination (478/SL) Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6932]: ERROR: <core> [resolve.c:1726]: sip_hostport2su(): could not resolve hostname: "3.local" Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6932]: ERROR: tm [ut.h:337]: uri2dst2(): failed to resolve "3.local" Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6932]: ERROR: tm [t_fwd.c:1778]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6932]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unresolvable destination (478/SL) Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6933]: ERROR: <core> [resolve.c:1726]: sip_hostport2su(): could not resolve hostname: "3.local" Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6933]: ERROR: tm [ut.h:337]: uri2dst2(): failed to resolve "3.local" Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6933]: ERROR: tm [t_fwd.c:1778]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches Nov 13 16:39:41 ua01-proxy01 /usr/sbin/kamailio[6933]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unresolvable destination (478/SL) Here is SIP ----------------begin of packet ----------------------------- U 2014/11/13 16:37:36.742489 192.168.207.162:63582 -> 192.168.207.156:5060 REGISTER sip:3.local SIP/2.0 Via: SIP/2.0/UDP 192.168.16.120:63582 ;rport;branch=z9hG4bKPjbea638d5e7054e03bc4dfee6b6872275 Max-Forwards: 70 From: "1004" <sip:1003@3.local>;tag=ff09e20e549b4127a88ae6a6ec34e88b To: "1004" <sip:1003@3.local> Contact: <sip:98524760@192.168.16.120:63582 >;+sip.instance="<urn:uuid:7d4ed3bd-451a-4406-a8e2-80315408b7c5>" Call-ID: a7454fb3539740ffb385c146303a926f CSeq: 1 REGISTER Expires: 600 Supported: gruu User-Agent: Blink 0.9.1.2 (Windows) Content-Length: 0 ------------------end of packet ----------------------------- ----------------begin of packet ----------------------------- U 2014/11/13 16:37:36.743505 192.168.207.156:5060 -> 192.168.207.162:63582 SIP/2.0 478 Unresolvable destination (478/SL) Via: SIP/2.0/UDP 192.168.16.120:63582 ;rport=63582;branch=z9hG4bKPjbea638d5e7054e03bc4dfee6b6872275;received=192.168.207.162 From: "1004" <sip:1003@3.local>;tag=ff09e20e549b4127a88ae6a6ec34e88b To: "1004" <sip:1003@3.local>;tag=98beed43ea484d57160c0a87164dfce1.ff17 Call-ID: a7454fb3539740ffb385c146303a926f CSeq: 1 REGISTER Server: kamailio (4.2.0 (x86_64/linux)) Content-Length: 0 ------------------end of packet ----------------------------- ----------------begin of packet ----------------------------- U 2014/11/13 16:37:36.743625 192.168.207.156:5060 -> 192.168.207.162:63582 SIP/2.0 478 Unresolvable destination (478/TM) Via: SIP/2.0/UDP 192.168.16.120:63582 ;rport=63582;branch=z9hG4bKPjbea638d5e7054e03bc4dfee6b6872275;received=192.168.207.162 From: "1004" <sip:1003@3.local>;tag=ff09e20e549b4127a88ae6a6ec34e88b To: "1004" <sip:1003@3.local>;tag=b9d45a0d611ee9eea95e269f12755a03-ff17 Call-ID: a7454fb3539740ffb385c146303a926f CSeq: 1 REGISTER Server: kamailio (4.2.0 (x86_64/linux)) Content-Length: 0 ------------------end of packet ----------------------------- ----------------begin of packet ----------------------------- U 2014/11/13 16:37:36.750939 192.168.207.156:5060 -> 192.168.207.162:63582 SIP/2.0 478 Unresolvable destination (478/SL) Via: SIP/2.0/UDP 192.168.16.120:63582 ;rport=63582;branch=z9hG4bKPj186bbf75b3964f06893742b853533302;received=192.168.207.162 From: "1004" <sip:1003@3.local>;tag=8b337cc975ab41c3b431a335a898128a To: "1004" <sip:1003@3.local>;tag=98beed43ea484d57160c0a87164dfce1.10fc Call-ID: fb19ff0b98f542849a249cd96dcd5f8c CSeq: 1 PUBLISH Server: kamailio (4.2.0 (x86_64/linux)) Content-Length: 0

10 years, 3 months

Allow Re-invite mid call

by Eric Koome

How do I process re-invite from sip provider mid call which check connection. At the moment Kamailio replies 404 not here, and call is dropped. Eric

10 years, 3 months

avoid OPTION requests without authentication

by Julien

Hello, I received in my SIP clients an OPTION request forwarded by kamailio. The user agent of this request is 'sipvicious'. I want to avoid these kind of requests from unauthenticated users How can I do that ? -- Julien Barreau

10 years, 3 months

Re: [SR-Users] UDP-TCP bridging in-dialog SUBSCRIBE

by Dan Christian Bogos

Thanks Alex for your confirmation. In some way I was afraid about this ;). Have a good one! DanB

10 years, 3 months

Kamailio - group presence

by Marino Mileti

Hi guys, I've to servers K1 e K2. On K2 I've registered two users on K1 and 2 users on K2. I would like that a user on K1 can subscribe to status of K2.... when there's a call on K2, it has to publish "On-call" status; if there isn't any ongoin call it has to publish "Online" status. I've tried to use uac module and send "PUBLISH" from scripts and it works but there are a lot of issue with SIP-If-Match header..so on K1 presentity table on db everytime a row is added and the old will be deleted when expires. I've tried also with PUA_MI...but using fifo on K2...so user on K1 should subscribe for a "virtual" user on K2. This isn't a good idea in case of network problem. Is there anyother way or do you have any suggests? ----- Marino Mileti -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-group-presence-tp132512.ht… Sent from the Users mailing list archive at Nabble.com.

10 years, 3 months

messages delivered by MSILO are not cleared immediately

by Prasun Bheri

Hi Group, I have configured MSILO with following modparams. modparam("msilo", "db_url",DBURL) modparam("msilo", "offline_message","$rU is offline!") modparam("msilo", "extra_hdrs", "Msilo-msg: true\r\n") modparam("msilo", "outbound_proxy", MSILO_OUTBOUND_PROXY) modparam("msilo", "content_type_hdr","Content-Type: message/cpim\r\n") modparam("msilo", "extra_hdrs", "Content-Type: message/cpim\r\n") modparam("msilo", "add_date", 0) MSILO dumps all the messages to destination immediately after recipient registers and gets 200 ok for sent messages. The delivered messages are not cleared immediately and takes a minute to delete each message. Could some one explain why the delay is observed? In case you need to know I am using Kamailio 4.0.0 on CentOS 6.5 Thanks & Regards, Prasun

10 years, 3 months

Kamailio Routing Logic

by Zaka Ul Isam

Hello Friends and Gurus: I am struggling with routing and need a bit of clarification WRT following: route { if ($fU=="601"&&is_method("INVITE")) record_route(); rewritehost("10.0.10.111"); //SS IP route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } * In the above stateful forwarding what is the rationale of calling route (1) within main route block? Could the t_relay statement afyer rewriting the host part of URI not have performed the stateful forwarding? ** In route[1] there is is only an expression abt logical not condition about t_relay() (if methods in this function are not invoked??) I wonder where exactly is stateful forwarding taking place? 1- Is it the default action? 2- Or in function sl_reply_error()? Thanks a lot in advance! Zaka ________________________________ Albtelecom & Eagle Mobile invite you to protect together the Environment. Please do not print the e-mail unless really necessary. Albtelecom & Eagle Mobile iu ftojnë të mbrojmë së bashku Mjedisin. Lutemi të mos e printoni këtë komunikim elektronik nëse nuk është vërtet i nevojshëm. ________________________________ This e-mail and any files transmitted with it are confidential and intended solely for the use of the addressee/s. If you are not the intended recipient you are hereby notified that any dissemination, forwarding, copying or use of any of the information is strictly prohibited. If you receive this e-mail in error, please notify the sender immediately and delete it! Albtelecom makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. The opinions expressed in this message may belong to sender alone and may not necessarily reflect the opinions of Albtelecom. Albtelecom shall bear no liability for any loss or damage caused by software or e-mail viruses. Ky mesazh dhe çdo informacion i transmetuar në përmbajtje te këtij mesazhi është konfidencial dhe është i destinuar vetëm për marrësin e destinuar. Nëse nuk jeni marrësi i destinuar, Ju bëjmë me dije se çdo përhapje, transmetim, kopjim apo përdorim i çdo informacioni është i ndaluar. Nëse e merrni këtë mesazh gabimisht, ju lutem kontaktoni urgjentisht nisësin e tij dhe fshijeni atë. Albtelecom nuk jep asnjë garanci për saktësinë apo plotësinë e informacionit në përmbajtje të këtij mesazhi dhe nuk mban asnjë përgjegjësi për informacionin e përmbajtur, transmetimin, marrjen, ruajtjen apo përdorimin e tij në çfarëdolloj mënyre. Mendimet e shprehura në këtë mesazh mund t'i përkasin vetë nisësit dhe nuk mund të reflektojnë domosdoshmërisht qëndrimet e Albtelecom. Albtelecom nuk do të mbaje asnjë përgjegjësi për humbje ose dëme te shkaktuara nga programet apo viruset.

10 years, 3 months

result of param.value transformation when param does not exist?

by Juha Heinanen

wiki transformations page on param.value does not mention, what is the result of the transformation when named param does not exist. same with many other transformations. what is the value if referenced value does not exit? should it be mentioned on the wiki page? -- juha

10 years, 3 months

topoh and uac_replace_from()

by Alex Balashov

Hi, If I use uac_replace_from() to alter the User part of the From URI: uac_replace_from("sip:$xavp(call=>src)@$fd"); it overwrites the domain concealment that is otherwise imparted by topoh, i.e. restores the original From domain. However, uac_replace_from() does not appear to offer the option to provide a replacement user part only, so I have to specify the domain. As far as I know, there is no means of accessing the modifications topoh has made, as this is transparent from the point of view of the config. Any way around this problem? Thanks, -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

10 years, 3 months

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sr-users November 2014