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@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(a)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@lists.sip-router.org] On Behalf Of Daniel Tryba
Sent: quarta-feira, 22 de Julho de 2015 16:39
To: sr-users(a)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(a)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