[SR-Users] New SIP INVITE from UAS (new dialogue)

[Joao Alves]

Hi,

Doing further step into this, I have realized that despite the SIP INVITE being sent, I cannot create/associate a new dialogue or transaction that I'll need for managing the SIP session.

So following the uac_req_send() I tried to do:
 
t_newtran();
record_route();

dlg_manage();
if(is_known_dlg()) {
	xlog("Request $rm from $ci is in-dialog\n");
  }

  
9(44916) exec: *** cfgtrace:request_route=[SIP_INVITE] c=[/etc/kamailio/kamailio.cfg] l=972 a=24 n=uac_req_send
 9(44916) DEBUG: tm [uac.c:249]: t_uac_prepare(): DEBUG:tm:t_uac: next_hop=<sip:1000 at 188.165.231.30:12060;rtcweb-breaker=no;transport=udp;ws-src-ip=81.84.0.236;ws-src-port=1025;ws-src-proto=ws>
 9(44916) DEBUG: tm [uac.c:150]: dlg2hash(): DEBUG: dlg2hash: 39161
 9(44916) exec: *** cfgtrace:request_route=[SIP_INVITE] c=[/etc/kamailio/kamailio.cfg] l=973 a=24 n=t_newtran
 9(44916) DEBUG: tm [t_lookup.c:1312]: t_newtran(): DEBUG: t_newtran: msg id=1 , global msg id=0 , T on entrance=0xffffffffffffffff
 9(44916) ERROR: tm [t_lookup.c:453]: t_lookup_request(): ERROR: TM module: t_lookup_request: too few headers
 9(44916) exec: *** cfgtrace:request_route=[SIP_INVITE] c=[/etc/kamailio/kamailio.cfg] l=974 a=24 n=record_route
 9(44916) ERROR: <core> [parser/parse_from.c:53]: parse_from_header(): ERROR:parse_from_header: bad msg or missing FROM header
 9(44916) ERROR: rr [record.c:407]: record_route(): From parsing failed
 9(44916) exec: *** cfgtrace:request_route=[SIP_INVITE] c=[/etc/kamailio/kamailio.cfg] l=976 a=24 n=dlg_manage
 9(44916) ERROR: dialog [dlg_handlers.c:1561]: dlg_manage(): bad TO header
 9(44916) exec: *** cfgtrace:request_route=[SIP_INVITE] c=[/etc/kamailio/kamailio.cfg] l=985 a=16 n=if
 9(44916) exec: *** cfgtrace:request_route=[SIP_INVITE] c=[/etc/kamailio/kamailio.cfg] l=977 a=24 n=is_known_dlg
 9(44916) ERROR: dialog [dlg_handlers.c:652]: pre_match_parse(): bad request or missing CALLID/TO hdr :-/


Am I doing something wrong here, as it seems the headers fields are just being "hardcoded" on the SIP INVITE are not updated on the context for that new outbound session (and thus causing the typical dialogue/transaction functions to fail...).
 
I realized already from one early comment I got from Alex Balashov that this is not a typical usecase for kamailio (being a sip proxy), but is this a show stopper?

Thanks,
Joao
-----Original Message-----
From: Joao Alves 
Sent: quarta-feira, 22 de Julho de 2015 17:02
To: sr-users at lists.sip-router.org
Subject: RE: [SR-Users] New SIP INVITE from UAS (new dialogue)

Hi Daniel,

Yes, you're right. It was fragmented at UDP level. I've repeated with TCP as transport and the SDP is complete. 

Thanks again,
Joao 

-----Original Message-----
From: sr-users [mailto:sr-users-bounces at lists.sip-router.org] On Behalf Of Daniel Tryba
Sent: quarta-feira, 22 de Julho de 2015 16:39
To: sr-users at lists.sip-router.org
Subject: Re: [SR-Users] New SIP INVITE from UAS (new dialogue)

On Wednesday 22 July 2015 14:01:02 Joao Alves wrote:
> In relation with the SDP size, I originally just compared with the 
> source one (see attached).  What I just did was to double check using 
> an online tool and confirmed that the original has 1266 bytes (as also 
> indicated by the Content's length) while the one sent has only 1077 bytes.

I saw the capture after my message. The message is fragmented and not shown correctly in the capture. There shouldn't be any problem (unless there are network issues between you and destination).

_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users at lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

This message and the information contained herein is proprietary and confidential and subject to the Amdocs policy statement,
you may review at http://www.amdocs.com/email_disclaimer.asp