Hello Andrei,
OpenSIPS preserves the interface when doing a relay (uses as outbound
same interface as inbound) if not otherwise instructed by routing info
(Route hdrs) or scripting.
As ACK is a sequential request which is routed based on Route hdr, to
see where the problem is, it is a must to see the full call capture from
OpenSIPS - from first INVITE to ACK, both messages coming and leaving
OpenSIPS - please post it on pastebin or so.
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 04/12/2013 11:10 PM, Andrei Grav wrote:
Hi,
I am facing some strange situation.
Opensips is listening on multiple ports on a single public IP
193.xx.xx.20 on ports: 5060, 26999, 36999
Asterisk is on 193.xx.xx.24:5060
Sometimes Opensips respond from 5060 to a 200OK instead received port.
U 188.xx.xx.173:53929 -> 193.xx.xx.20:26999
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
193.xx.xx.20:26999;received=193.xx.xx.20;branch=z9hG4bK46cd.df220482.1.
Via: SIP/2.0/UDP
193.xx.xx.24:5060;rport=5060;received=193.xx.xx.24;branch=z9hG4bK218bf88e.
Record-Route:
<sip:193.xx.xx.20:26999;lr;r2=on;ftag=as61ef6194;did=49d.46da2342>.
Record-Route:
<sip:193.xx.xx.20;lr;r2=on;ftag=as61ef6194;did=49d.46da2342>.
Call-ID: 5ebb1c01580da34c694b320e0627d...@sip.mydomain.com
<mailto:5ebb1c01580da34c694b320e0627d...@sip.mydomain.com>.
From: "User" <sip:850...@sip.mydomain.com
<mailto:sip%3a850...@sip.mydomain.com>>;tag=as61ef6194.
To: <sip:850...@sip.mydomain.com
<mailto:sip%3a850...@sip.mydomain.com>>;tag=Yab-MAmXkF5lAQq2E6QpifQ8xa9xDoU7.
CSeq: 102 INVITE.
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS.
Contact: <sip:850...@188.xx.xx.173:53929;ob>.
Supported: replaces, 100rel, timer, norefersub.
Content-Type: application/sdp.
Content-Length: 294.
.
v=0.
o=- 3574781465 3574781466 IN IP4 188.xx.xx.173.
s=pjmedia.
c=IN IP4 188.xx.xx.173.
t=0 0.
m=audio 4006 RTP/AVP 18 101.
c=IN IP4 188.xx.xx.173.
a=rtcp:4007 IN IP4 188.xx.xx.173.
a=sendrecv.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
U 193.xx.xx.20:5060 -> 188.xx.xx.173:53929
ACK sip:850...@188.xx.xx.173:53929;ob SIP/2.0.
Via: SIP/2.0/UDP 193.xx.xx.20:26999;branch=z9hG4bK46cd.df220482.3.
Via: SIP/2.0/UDP
193.xx.xx.24:5060;rport=5060;received=193.xx.xx.24;branch=z9hG4bK63fbcba6.
Max-Forwards: 69.
From: "User" <sip:850...@sip.mydomain.com
<mailto:sip%3a850...@sip.mydomain.com>>;tag=as61ef6194.
To: <sip:850...@sip.mydomain.com
<mailto:sip%3a850...@sip.mydomain.com>>;tag=Yab-MAmXkF5lAQq2E6QpifQ8xa9xDoU7.
Contact: <sip:850...@193.xx.xx.24:5060>.
Call-ID: 5ebb1c01580da34c694b320e0627d...@sip.mydomain.com
<mailto:5ebb1c01580da34c694b320e0627d...@sip.mydomain.com>.
CSeq: 102 ACK.
User-Agent: PBX.
Content-Length: 0.
.
the last response should send the response from port 26999 to be ok
... or the call is hanged up after 32 seconds
U 193.xx.xx.20:26999 -> 188.xx.xx.173:53929
any advice ?
Thank you,
Andrei
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