18 May
2017
18 May
'17
4:12 a.m.
Using set_contact_alias()/handle_ruri_alias() should get this working in
terms of routing through kamailio.
But probably is more sane to fix it in client side, other UAs may
complain on such domain part.
Cheers,
Daniel
On 17.05.17 21:23, David Villasmil wrote:
You can set the contact manually, look up
set_contact_alias() ... I'm
not sure whether this would be advisable though... you need someone
more kamailio-knowledgeable...
I think the problem is the linphone config, i sometimes use it and
have never seen that...
On Wed, May 17, 2017 at 8:45 PM Jean Cérien <cerien.jean(a)gmail.com
<mailto:cerien.jean@gmail.com>> wrote:
Thanks for the help.
I have reverted to the default config file
(https://github.com/sipwise/kamailio/blob/master/etc/kamailio.cfg),
and trying to place a call between two ua (linphone & zoiper). I
am testing totally on a LAN, clients & kamailo on the same subnet,
no nat.
Register is fine, Invite is fine, I receive the 200OK from called,
then I get the ACK from the calling, and while processing it, I
get the following errors in the log:
May 17 14:12:32 kamailio : ERROR: <core> [resolve.c:1694]:
sip_hostport2su(): could not resolve hostname: "(null)"
May 17 14:12:32 kamailio : ERROR: <core> [forward.c:495]:
forward_request(): bad host name (null), dropping packet
May 17 14:12:32 kamailio : ERROR: sl [sl_funcs.c:363]:
sl_reply_error(): ERROR: sl_reply_error used: Unresolvable
destination (478/SL)
Digging a bit more, I've noticed that the calling party, using
Linphone, has the Contact field a bit weird:
Contact: <sip:user@(null)>
Changing to another softphone that populates correctly this field
works ok. Is there a way to mitigate this external issue ? The
softphone works ok directly connected to asterisk for instance
Rgds
J.
On Tue, May 16, 2017 at 6:15 PM, David Villasmil
<david.villasmil.work(a)gmail.com
<mailto:david.villasmil.work@gmail.com>> wrote:
There isn't an ACK received, check in kamailio side to make
sure it is received. This is most probably a nat issue.
On Tue, May 16, 2017 at 11:20 PM Jean Cérien
<cerien.jean(a)gmail.com <mailto:cerien.jean@gmail.com>> wrote:
Hello
I am getting started with Kamailio (or restarted, used it
briefly years ago), with the final objective to do load
balancing.
For the time being, I am just trying to have one asterisk
and one kamailio, on the same box. I have setup a box with
an asterisk 11.3, and kamailio 4.4. I've taken the config
file
from
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb.
My idea is that asterisk runs on port 5080, while kamailio
on port 5060. Client interacts with Kamailio on port 5060.
It almost works... Registration is fine, but when I send
an invite, it is properly acknowledged (by asterisk 100
trying then 200 OK) - but the OK message gets repeated
multiple times and asterisk issues its infamous
'Retransmission timeout reached ...' - as if Kamailio
wasnt processing it. See below ngrep traces between
asterisk and kamailio
Any ideas where to look ?
Thanks
J.
#
U +18.289105 192.168.2.228:5060
<http://192.168.2.228:5060> -> 192.168.2.228:5080
<http://192.168.2.228:5080>
INVITE sip:102@192.168.2.228
<mailto:sip%3A102@192.168.2.228> SIP/2.0..Record-Route:
<sip:192.168.2.228;lr=on;ftag=1034946464>..Via:
SIP/2.0/UDP
192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b9922
4198ef593.0..Via: SIP/2.0/UDP
192.168.2.200:5085;received=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..From:
<sip:199@192.168.2.228
<mailto:sip%3A199@192.168.2.228>>;tag=1034946464..To:
<sip:102@
192.168.2.228>..Call-ID: 1571382735..CSeq: 21
INVITE..Contact: <sip:iper@(null)>..Content-Type:
application/sdp..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
REFER, NOTIFY
, MESSAGE, SUBSCRIBE, INFO..Max-Forwards:
69..User-Agent: Linphone/3.6.1 (eXosip2/4.1.0)..Subject:
Phone call..Content-Length: 437....v=0..o=199 2799 2990
IN IP4 192.
168.2.200..s=Talk..c=IN IP4 192.168.2.200..t=0
0..m=audio 7078 RTP/AVP 124 111 110 0 8 101..a=rtpmap:124
opus/48000..a=fmtp:124 useinbandfec=1;
usedtx=1..a=rtpmap:111 s
peex/16000..a=fmtp:111 vbr=on..a=rtpmap:110
speex/8000..a=fmtp:110 vbr=on..a=rtpmap:101
telephone-event/8000..a=fmtp:101 0-11..m=video 9078
RTP/AVP 103 99..a=rtpmap:103
VP8/90000..a=rtpmap:99 MP4V-ES/90000..a=fmtp:99
profile-level-id=3..
#
U +0.000683 192.168.2.228:5080 <http://192.168.2.228:5080>
-> 192.168.2.228:5060 <http://192.168.2.228:5060>
OPTIONS sip:199@192.168.2.228:5060
<http://sip:199@192.168.2.228:5060> SIP/2.0..Via:
SIP/2.0/UDP
192.168.2.228:5080;branch=z9hG4bK7c2d71fb..Max-Forwards:
70..From: "asterisk" <sip:199@192.168.2.228:5080
<http://sip:199@192.168.2.228:5080>>;
tag=as57c98c4b..To: <sip:199@192.168.2.228:5060
<http://sip:199@192.168.2.228:5060>>..Contact:
<sip:199@192.168.2.228:5080
<http://sip:199@192.168.2.228:5080>>..Call-ID:
52fa034372ce18ca2b93fc1817ad38a5@192.168.2.228:5080..CSeq:
102 OPTIONS
..User-Agent: Asterisk PBX 11.3.0..Date: Tue, 16 May
2017 21:08:29 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: re
places, timer..Content-Length: 0....
#
U +0.001035 192.168.2.228:5080 <http://192.168.2.228:5080>
-> 192.168.2.228:5060 <http://192.168.2.228:5060>
OPTIONS sip:199@192.168.2.228:5060
<http://sip:199@192.168.2.228:5060> SIP/2.0..Via:
SIP/2.0/UDP
192.168.2.228:5080;branch=z9hG4bK4248158e..Max-Forwards:
70..From: "asterisk" <sip:199@192.168.2.228:5080
<http://sip:199@192.168.2.228:5080>>;
tag=as30d9cfb4..To: <sip:199@192.168.2.228:5060
<http://sip:199@192.168.2.228:5060>>..Contact:
<sip:199@192.168.2.228:5080
<http://sip:199@192.168.2.228:5080>>..Call-ID:
4331da391bca02965b2af65254717a18@192.168.2.228:5080..CSeq:
102 OPTIONS
..User-Agent: Asterisk PBX 11.3.0..Date: Tue, 16 May
2017 21:08:29 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: re
places, timer..Content-Length: 0....
#
U +0.000407 192.168.2.228:5080 <http://192.168.2.228:5080>
-> 192.168.2.228:5060 <http://192.168.2.228:5060>
SIP/2.0 100 Trying..Via: SIP/2.0/UDP
192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via:
SIP/2.0/UDP 192.168.2.200:5085;rec
eived=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route:
<sip:192.168.2.228;lr=on;ftag=1034946464>..From:
<sip:199@192.168.2.228
<mailto:sip%3A199@192.168.2.228>>;tag=1034946464..To: <sip
:102@192.168.2.228 <mailto:102@192.168.2.228>>..Call-ID:
1571382735..CSeq: 21 INVITE..Server: Asterisk PBX
11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO, PUBLIS
H..Supported: replaces, timer..Contact:
<sip:102@192.168.2.228:5080
<http://sip:102@192.168.2.228:5080>>..Content-Length:
0....
#
U +0.003961 192.168.2.228:5080 <http://192.168.2.228:5080>
-> 192.168.2.228:5060 <http://192.168.2.228:5060>
SIP/2.0 200 OK..Via: SIP/2.0/UDP
192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via:
SIP/2.0/UDP 192.168.2.200:5085;receive
d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route:
<sip:192.168.2.228;lr=on;ftag=1034946464>..From:
<sip:199@192.168.2.228
<mailto:sip%3A199@192.168.2.228>>;tag=1034946464..To: <sip:102
@192.168.2.228
<http://192.168.2.228>>;tag=as497f35c0..Call-ID:
1571382735..CSeq: 21 INVITE..Server: Asterisk PBX
11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, I
NFO, PUBLISH..Supported: replaces, timer..Contact:
<sip:102@192.168.2.228:5080
<http://sip:102@192.168.2.228:5080>>..Content-Type:
application/sdp..Content-Length: 312....v=0..o=root
350189084 350189084 I
N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4
192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8
101..a=rtpmap:0 PCMU/8000..a=rtpmap:8
PCMA/8000..a=rtpmap:101 telep
hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - -
- -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99..
#
U +0.087859 192.168.2.228:5060 <http://192.168.2.228:5060>
-> 192.168.2.228:5080 <http://192.168.2.228:5080>
SIP/2.0 200 OK..Via: SIP/2.0/UDP
192.168.2.228:5080;branch=z9hG4bK7c2d71fb..From:
"asterisk" <sip:199@192.168.2.228:5080
<http://sip:199@192.168.2.228:5080>>;tag=as57c98c4b..To:
<sip:199@192.168.2.228:506 <http://sip:199@192.168.2.228:506>
0>;tag=524348182..Call-ID:
52fa034372ce18ca2b93fc1817ad38a5@192.168.2.228:5080..CSeq:
102 OPTIONS..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
MESSAGE, SUBSCRIBE, NOTIFY,
INFO..Accept: application/sdp..User-Agent:
Linphone/3.6.1 (eXosip2/4.1.0)..Content-Length: 0....
#
U +0.000213 192.168.2.228:5060 <http://192.168.2.228:5060>
-> 192.168.2.228:5080 <http://192.168.2.228:5080>
SIP/2.0 200 OK..Via: SIP/2.0/UDP
192.168.2.228:5080;branch=z9hG4bK4248158e..From:
"asterisk" <sip:199@192.168.2.228:5080
<http://sip:199@192.168.2.228:5080>>;tag=as30d9cfb4..To:
<sip:199@192.168.2.228:506 <http://sip:199@192.168.2.228:506>
0>;tag=939659485..Call-ID:
4331da391bca02965b2af65254717a18@192.168.2.228:5080..CSeq:
102 OPTIONS..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
MESSAGE, SUBSCRIBE, NOTIFY,
INFO..Accept: application/sdp..User-Agent:
Linphone/3.6.1 (eXosip2/4.1.0)..Content-Length: 0....
#
U +0.011138 192.168.2.228:5080 <http://192.168.2.228:5080>
-> 192.168.2.228:5060 <http://192.168.2.228:5060>
SIP/2.0 200 OK..Via: SIP/2.0/UDP
192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via:
SIP/2.0/UDP 192.168.2.200:5085;receive
d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route:
<sip:192.168.2.228;lr=on;ftag=1034946464>..From:
<sip:199@192.168.2.228
<mailto:sip%3A199@192.168.2.228>>;tag=1034946464..To: <sip:102
@192.168.2.228
<http://192.168.2.228>>;tag=as497f35c0..Call-ID:
1571382735..CSeq: 21 INVITE..Server: Asterisk PBX
11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, I
NFO, PUBLISH..Supported: replaces, timer..Contact:
<sip:102@192.168.2.228:5080
<http://sip:102@192.168.2.228:5080>>..Content-Type:
application/sdp..Content-Length: 312....v=0..o=root
350189084 350189084 I
N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4
192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8
101..a=rtpmap:0 PCMU/8000..a=rtpmap:8
PCMA/8000..a=rtpmap:101 telep
hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - -
- -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99..
#
U +0.200291 192.168.2.228:5080 <http://192.168.2.228:5080>
-> 192.168.2.228:5060 <http://192.168.2.228:5060>
SIP/2.0 200 OK..Via: SIP/2.0/UDP
192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via:
SIP/2.0/UDP 192.168.2.200:5085;receive
d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route:
<sip:192.168.2.228;lr=on;ftag=1034946464>..From:
<sip:199@192.168.2.228
<mailto:sip%3A199@192.168.2.228>>;tag=1034946464..To: <sip:102
@192.168.2.228
<http://192.168.2.228>>;tag=as497f35c0..Call-ID:
1571382735..CSeq: 21 INVITE..Server: Asterisk PBX
11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, I
NFO, PUBLISH..Supported: replaces, timer..Contact:
<sip:102@192.168.2.228:5080
<http://sip:102@192.168.2.228:5080>>..Content-Type:
application/sdp..Content-Length: 312....v=0..o=root
350189084 350189084 I
N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4
192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8
101..a=rtpmap:0 PCMU/8000..a=rtpmap:8
PCMA/8000..a=rtpmap:101 telep
hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - -
- -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99..
#
U +0.400478 192.168.2.228:5080 <http://192.168.2.228:5080>
-> 192.168.2.228:5060 <http://192.168.2.228:5060>
SIP/2.0 200 OK..Via: SIP/2.0/UDP
192.168.2.228;branch=z9hG4bK9e22.bff83e1026a667df63b99224198ef593.0;received=192.168.2.228..Via:
SIP/2.0/UDP 192.168.2.200:5085;receive
d=192.168.2.200;rport=5085;branch=z9hG4bK1247964647..Record-Route:
<sip:192.168.2.228;lr=on;ftag=1034946464>..From:
<sip:199@192.168.2.228
<mailto:sip%3A199@192.168.2.228>>;tag=1034946464..To: <sip:102
@192.168.2.228
<http://192.168.2.228>>;tag=as497f35c0..Call-ID:
1571382735..CSeq: 21 INVITE..Server: Asterisk PBX
11.3.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, I
NFO, PUBLISH..Supported: replaces, timer..Contact:
<sip:102@192.168.2.228:5080
<http://sip:102@192.168.2.228:5080>>..Content-Type:
application/sdp..Content-Length: 312....v=0..o=root
350189084 350189084 I
N IP4 192.168.2.228..s=Asterisk PBX 11.3.0..c=IN IP4
192.168.2.228..t=0 0..m=audio 17148 RTP/AVP 0 8
101..a=rtpmap:0 PCMU/8000..a=rtpmap:8
PCMA/8000..a=rtpmap:101 telep
hone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - -
- -..a=ptime:20..a=sendrecv..m=video 0 RTP/AVP 103 99..
#
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--
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