[OpenSIPS-Users] Fwd: Drouting use next gateway adding too many vias SOLVED

[Richard Robson]

Hi All

I was re processing the failures in the next route block (with record 
route and topology hiding etc) as i would for a new packet instead of 
jut relaying after the next gateway is chosen.

Sorry for the noise

R

Hi,


I'm testing out new carrier which has 3 different endpoits for us to
connect to. I've setup the 3 addresses in the DR_Rules table and the
first call or calls to a different address directly work. however, when
a call fails ad I select the next gateway the originating asteris box
failed to respond to the 200 OK.

I've traced this to Asterisk compalining about too many vias in the sip
response


this is one of the packets and you can seee the two vias.

Is there anyway to stop this, is this expected behavoir or am I doing
something wrong?

its Opensips 2.2.2


Regards,


Richard

2016/12/14 17:42:16.974603 192.168.36.141:5060 -> 192.168.36.68:5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
192.168.36.68:5060;received=192.168.36.68;rport=5060;branch=z9hG4bK42faa1b4
Via: SIP/2.0/UDP
192.168.36.68:5060;received=192.168.36.68;rport=5060;branch=z9hG4bK42faa1b4
To: <sip:0800800150 at 192.168.36.141>;tag=3690726199-902585
From: <sip:+441382843843 at 192.168.36.68>;tag=as1d2fa6b6
Call-ID: 57f6d7d10a004b8943a25d41470ef82d at 192.168.36.68:5060
CSeq: 103 INVITE
Allow: UPDATE,PRACK,INFO,OPTIONS,BYE,INVITE,ACK,CANCEL
Contact: <sip:44800800150 at 141.170.9.156:5060;did=391.d98a02f3>
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 372

v=0
o=sbc-uk-mqd-k05a 280406016 1481737400 IN IP4 109.159.137.74
s=sip call
c=IN IP4 141.170.9.159
t=0 0
m=audio 40728 RTP/AVP 8 0 18 96
a=rtpmap:8 PCMA/8000
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:30
a=oldcip:109.159.137.68



[2016-12-14 17:43:19] WARNING[28124][C-00004e17] chan_sip.c: Misrouted
SIP response '183 Session Progress' with Call-ID
'57f6d7d10a004b8943a25d41470ef82d at 192.168.36.68:5060', too many vias
[2016-12-14 17:43:20] WARNING[28124][C-00004e17] chan_sip.c: Misrouted
SIP response '180 Ringing' with Call-ID
'57f6d7d10a004b8943a25d41470ef82d at 192.168.36.68:5060', too many vias
[2016-12-14 17:43:21] WARNING[28124][C-00004e17] chan_sip.c: Misrouted
SIP response '200 OK' with Call-ID
'57f6d7d10a004b8943a25d41470ef82d at 192.168.36.68:5060', too many vias
[2016-12-14 17:43:22] WARNING[28124][C-00004e17] chan_sip.c: Misrouted
SIP response '200 OK' with Call-ID
'57f6d7d10a004b8943a25d41470ef82d at 192.168.36.68:5060', too many vias
[2016-12-14 17:43:23] WARNING[28124][C-00004e17] chan_sip.c: Misrouted
SIP response '200 OK' with Call-ID
'57f6d7d10a004b8943a25d41470ef82d at 192.168.36.68:5060', too many vias
[2016-12-14 17:43:25] WARNING[28124][C-00004e17] chan_sip.c: Misrouted
SIP response '200 OK' with Call-ID
'57f6d7d10a004b8943a25d41470ef82d at 192.168.36.68:5060', too many vias


-- 
Richard Robson
Greenlight Support
01382 843843
support at greenlightcrm.com