sr-users July 2010

sr-users@lists.kamailio.org

64 participants
88 discussions

Fwd: Re: Fwd: Re: Kamailio and NAPTR lookup with TLS
by Matteo Campana 14 Jul '10

14 Jul '10

Hi Daniel, I have followed your instruction (I'm using kamailio 1.5) and before the t_relay("0x05") statement I've setted: $du="sip:" + $rd + ";transport=tls"; but I get the following mesasge: Jul 14 15:17:55 OVSIPP /usr/local/sbin/kamailio[7790] Request leaving server, *D_-URI='sip:garr.it;transport=tls'_* - M=INVITE RURI=sip:xxx@garr.it F=sip:yyyyy@sip.mydomqin.it:5066 T=sip:123456789@sip.mydomain.it:5066 Jul 14 15:17:55 OVSIPP /usr/local/sbin/kamailio[7790]: CRITICAL:core:mk_proxy: could not resolve hostname: "garr.it" Jul 14 15:17:55 OVSIPP /usr/local/sbin/kamailio[7790]: ERROR:tm:uri2proxy: bad host name in URI <sip:garr.it;transport=tls> Jul 14 15:17:55 OVSIPP /usr/local/sbin/kamailio[7790]: ERROR:tm:t_forward_nonack: failure to add branches Therefore I would like to use information about TLS connection (tls port?) got from NAPTR record. Doesn't it possible with kamailio 1.5 ? Many thanks, Daniel Il 08/07/2010 18.12, Matteo Campana ha scritto: > > > -------- Messaggio originale -------- > Oggetto: Re: [SR-Users] Kamailio and NAPTR lookup with TLS > Data: Thu, 08 Jul 2010 18:10:35 +0200 > Mittente: Daniel-Constantin Mierla <miconda(a)gmail.com> > A: matteo.campana(a)klarya.it > CC: sr-users(a)lists.sip-router.org > > > > Hello, > > On 7/8/10 5:59 PM, Matteo Campana wrote: > > > > Hi all, > > I'm using kamailio 1.5 with TLS module. > > I need to make ENUM query and get NAPTR record. > > > From NAPTR lookup, I'd like to relay my SIP Invite with tls protocol. > > > > How can I tell Kamailio to use TLS protocol ( instead of udp) after NAPTR lookup ? > > > > I've try to set : > > > > dns_tls_pref=1 > > dns_udp_pref=2 > > dns_tcp_pref=3 > > > > in the general section of kamailio.cfg, but I get a parse error. > > > > > these parameters were introduced in kamailio with version 3.0. > > If you need TLS then it is recommended to use 3.0 anyhow, it is a far > better implementation. That will make the life easier to migrate to > upcoming 3.1 that will bring asynchronous TLS. > > No matter what you have in R-URI, you can force TLS via setting outbound > proxy address to be a TLS uri: > > $du = "sip:__ip_or_host__;transport=tls"; > t_relay(); > > The IP or host you can take from R-URI without any problem via PV $rd. > Other option is to use function from tm - t_relay_to_tls(): > > http://kamailio.org/docs/modules/stable/modules/tm.html#t_relay_to_udp > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com/ > >

2 1

dialog module: in-early-dialog requests / race conditions
by Timo Reimann 13 Jul '10

13 Jul '10

Hello, I have two issues related to dialog module state keeping/logging that caught my attention today: 1.) The module considers requests routed in early (but not yet confirmed) dialogs to be bogus, as can be seen from a few lines of state machine code in dlg_hash.c: http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=modules_… In the logs, it shows up something like this: CRITICAL:dialog:log_next_state_dlg: bogus event 8 in state 2 for dlg 0x7f9f237ffff8 [2901:1286709577] with clid '<Call-ID>' and tags '<caller tag>' '<callee tag, always empty>' I believe that such in-early-dialog requests should not trigger bogus-event log messages as they look fine from a standard's point of view. 2.) Various race conditions may generate log messages such as following when DID mode is enabled: WARNING:dialog:dlg_onroute: unable to find dialog for <REQUEST TYPE> with route param 'faf.f69e7b4' [4015:79161711] Observed REQUEST TYPEs at our site were "BYE" when BYE requests were transmitted by both caller and callee at roughly the same time due to user behavior; and "PRACK" when a PRACK was received by the UAS after a 486 (Busy Here)/ACK pair of messages was already exchanged between the UAS and the proxy. In both cases, the associated dialog had already been destroyed when another request was received. My suggestion is to lower the log level for this kind of message because it can simply happen too often but doesn't affect proper dialog handling. Feedback? Cheers, --Timo

3 8

Several BYEs conveying Kamailio
by Maciej Bylica 11 Jul '10

11 Jul '10

Dear ALL, I have configured Kamailio 1.5 from svn with radius (radiusclient-ng) support for acc. All seem to work okay, except one thing. >From time to time I am receiving following radius error (debug from radius server on different ip) Fri Jul 9 12:22:58 2010 : Error: rlm_sql (sql): Couldn't update SQL accounting STOP record - Column 'AcctStartTime' cannot be null Fri Jul 9 12:23:08 2010 : Error: rlm_sql (sql): Couldn't update SQL accounting STOP record - Column 'AcctStartTime' cannot be null Fri Jul 9 12:23:18 2010 : Error: rlm_sql (sql): Couldn't update SQL accounting STOP record - Column 'AcctStartTime' cannot be null A few lines from radiusclient.conf # time to wait for a reply from the RADIUS server radius_timeout 10 # resend request this many times before trying the next server radius_retries 3 After closer look at this, it turned out that there is one START request and four STOPs with the same CallID (Acct-Session-Id). START and first STOP are treated as a pair, the next ones are odd. Taking about sip i do have following scenario: MGW(1)---Kamailio(2)---SIP_Provider(SER based)(3) The call flow that generate aforemtioned errors looks like below (debug on kamailio) 1 - (2) is receiving INVITE from (3) 2 - (2) is sending Giving a Try to (3) 3 - (2) is sending INVITE to (1) 4 - (2) is receiving Trying from (1) 5 - (2) is receiving Ringing from (1) 6 - (2) is receiving OK from (1) 7 - (2) is sending OK to (3) 8 - (2) is receiving ACK from (3) 9 - (2) is sending ACK to (1) ---- call---- 10 - (2) is receiving BYE from (1) (tearing down from (1) side, STOP radius request is generated) 11 - (2) is sending BYE to (3) 12 - (2) is receiving BYE from (3) (second STOP is generated, radius_retries 3, radius_timeout 10) 13 - (2) is sending BYE to (1) 14 - (2) is receiving OK from (1) 15 - (2) is sending OK to (3) 16 - (2) is receiving OK to (3) 17 - (2) is sending OK to (1) For sure there is sth wrong with SIP_Provider but how Kamilio should behave is such situation. Should second BYE (p.12) with the same callid as in p.10 be continued with p.13? Thanks in advance, Maciej.

2 3

Use AVP as regex in avp_subst
by Uriel Rozenbaum 10 Jul '10

10 Jul '10

Hey guys, I'm using Kamailio 1.5.3-notls and need to apply some re operation to an AVP. The problem I have is that my re expression is held in an AVP, so the parser is not recognizing it :( Would it be very hard to change that behavior? Any ideas/tips to implement? This code route[16] { ... avp_subst("$avp(s:ANI)/$avp(s:ANIegress)", "$avp(s:carrierAniRegex)"); ... } returns this error ERROR:avpops:fixup_subst: avpops: bad subst re $avp(s:carrierAniRegex) This code route[16] { ... avp_subst("$avp(s:ANI)/$avp(s:ANIegress)", $avp(s:carrierAniRegex)); ... } returns this error CRITICAL:core:yyerror: parse error in config file, line 751, column 51-74: syntax error CRITICAL:core:yyerror: parse error in config file, line 751, column 74-75: bad arguments Do you think some other way I can achieve this? Be aware I'm using SQLops to get realtime the value of the expression... Thanks in advance! Uriel

3 6

Kamailio, spiral with TM
by David 09 Jul '10

09 Jul '10

Hey, I have setup two Kamailio servers, my INVITEs will go through server A, which forwards to server B which forwards it back to server A. This is intential, the trouble is when the packet comes back through server B, nothing has changed so TM sees it as a retransmission. The difference between the first time through and the second time is the port 5061 instead of 5060. How can I tell tm that these are not the same transactions ? Would it be easier to have a seperate instance of Kamailio for 5061 ? Thanks, David

4 7

How to force snd_port?
by Victor Pascual Avila 09 Jul '10

09 Jul '10

Hi Folks, How could I force my sip-router to send from a given port number? Many thanks in advance, -- Victor Pascual Ávila

4 8

Kamailio and NAPTR lookup with TLS
by Matteo Campana 09 Jul '10

09 Jul '10

Hi all, I'm using kamailio 1.5 with TLS module. I need to make ENUM query and get NAPTR record. >From NAPTR lookup, I'd like to relay my SIP Invite with tls protocol. How can I tell Kamailio to use TLS protocol ( instead of udp) after NAPTR lookup ? I've try to set : dns_tls_pref=1 dns_udp_pref=2 dns_tcp_pref=3 in the general section of kamailio.cfg, but I get a parse error. Regards, Daniel

4 4

Fwd: Re: Fwd: Re: Kamailio and NAPTR lookup with TLS
by Matteo Campana 09 Jul '10

09 Jul '10

Hi klaus, Suppose I can't access to NAPTR settings. I need to manage SIP URI, so , If I right understand, the only way to use TLS protocol in kamailio 1.5 is to append ";transport=tls" in R-URI before relay. In other words I need to rewrite R-URI: $ru = $ru + ";transport=tls" ; # and the t_relay t_relay() ; something like that? Regards, Daniel Il 08/07/2010 18.45, Matteo Campana ha scritto: > > > -------- Messaggio originale -------- > Oggetto: Re: [SR-Users] Kamailio and NAPTR lookup with TLS > Data: Thu, 08 Jul 2010 18:44:27 +0200 > Mittente: Klaus Darilion <klaus.mailinglists(a)pernau.at> > A: Daniel-Constantin Mierla <miconda(a)gmail.com> > CC: matteo.campana(a)klarya.it, sr-users(a)lists.sip-router.org > > > > Am 08.07.2010 18:10, schrieb Daniel-Constantin Mierla: > > Hello, > > > > On 7/8/10 5:59 PM, Matteo Campana wrote: > >> > >> Hi all, > >> I'm using kamailio 1.5 with TLS module. > >> I need to make ENUM query and get NAPTR record. > >> > From NAPTR lookup, I'd like to relay my SIP Invite with tls protocol. > >> > >> How can I tell Kamailio to use TLS protocol ( instead of udp) after > >> NAPTR lookup ? > >> > >> I've try to set : > >> > >> dns_tls_pref=1 > >> dns_udp_pref=2 > >> dns_tcp_pref=3 > >> > >> in the general section of kamailio.cfg, but I get a parse error. > >> > > these parameters were introduced in kamailio with version 3.0. > > > > If you need TLS then it is recommended to use 3.0 anyhow, it is a far > > better implementation. That will make the life easier to migrate to > > upcoming 3.1 that will bring asynchronous TLS. > > > > No matter what you have in R-URI, you can force TLS via setting outbound > > proxy address to be a TLS uri: > > > > $du = "sip:__ip_or_host__;transport=tls"; > > t_relay(); > > IIRC we do have NAPTR support in Kamailio 1.5 - don't we? > > Then I think it should work when putting a domain into $du and makeing > sure that there is no transport parameter, no port, and NAPTR TLS record > has highest priority. > > regards > klaus > > > > > The IP or host you can take from R-URI without any problem via PV $rd. > > Other option is to use function from tm - t_relay_to_tls(): > > > > http://kamailio.org/docs/modules/stable/modules/tm.html#t_relay_to_udp > > > > Cheers, > > Daniel > > > >

2 1

Differentiation between ACK INVITE and ACK CANCEL
by inge 08 Jul '10

08 Jul '10

Hello all, I need to work on particular ACK within the configuration of SER 0.9.9. Firstly, I was thinking to make a difference using the CSeq field, but it seems that the CSeq on ACK contains the ACK himself. Probably because ACK is a request and not a response. Would it be possible to proceed in another way? Regards, Adrien .L

4 9

Kamailio integration - Registration issue
by Lucas Alvarez 08 Jul '10

08 Jul '10

Hi, I new with kamailio, I've been able to integrate kamailio 3.02 with asterisk 1.6. The only issue I'm having is if I have to restart asterisk( for some config update) I loose all the sip registration in asterisk, is there any way of fixing this?

9 13

← Newer
1
2
3
4
5
6
7
8
9
Older →

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users July 2010