I'm moving our Epygy M32x (running firmware version 5.3.75) to a different location on a different subnet. Outgoing calls works as they should be, however external incoming calls are disconected as soon as the user tries to pickup the call. I have not changed anything other than the Epygi's WAN IP. I'm tearing my hair out trying to fix this!
Some relevant info:
- Handsets are SNOM 320 running version 8.4.35 (latest supported version by epygi)
- Calls between handset works fine.
- calls that get routed back out (i.e to mobiles if user is not at desk) works fine
- [Internal] Calls to / from other Epygi unit works fine
ITSP provided the below info on the cause of call disconect (what they can see from their end):
Code:
From the SIP logs file, I can see that call is being disconnected because of the Invalid SDP. Please find the following details for both 200 OK and 237 BYE.
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.176.185.26;branch=z9hG4bKdfb5.1f9e13f327ce4fb974c73d43de786060.0
Via: SIP/2.0/UDP 175.45.127.201:5062;received=175.45.127.201;rport=5062;branch=z9hG4bK-571767192-3843100834-2361413535-749475168
Record-Route: <sip:#ITSP sip server#:5060;transport=udp;lr>
Record-Route: <sip:203.176.185.26;lr>
Contact: "Anonymous"<sip:#ITSP sip server#:5061>
To: <sip:#called number#@203.176.185.26;user=phone>;tag=ns6g2j4awxhtvkoc.i
From: <sip:#caller number#@175.45.127.201:5062;user=phone>;tag=1022064792-3843100834-2361413535-749475168
Call-ID: 9878eb5aa21011e59f4fc08c6015ac2c@175.45.127.201
CSeq: 1 INVITE
Content-Type: application/sdp
Server: Sippy
H323-credit-time: 21600
Content-Length: 230
v=0
o=Sippy 496830480 1 IN IP4 #ITSP sip server#
s=-
t=0 0
m=audio 0 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
BYE sip:#ITSP sip server#:5061 SIP/2.0
Via: SIP/2.0/UDP 203.176.185.26;branch=z9hG4bKafb5.037f2688149301190c0a629a7e5a4de9.0
Via: SIP/2.0/UDP 175.45.127.201:5062;received=175.45.127.201;rport=5062;branch=z9hG4bK-2524621977-3843100834-2361413535-749475168
Max-Forwards: 69
Route: <sip:#ITSP sip server#:5060;transport=udp;lr>
To: <sip:#called number#@203.176.185.26;user=phone>;tag=ns6g2j4awxhtvkoc.i
From: <sip:#caller number#@175.45.127.201:5062;user=phone>;tag=1022064792-3843100834-2361413535-749475168
Call-ID: 9878eb5aa21011e59f4fc08c6015ac2c@175.45.127.201
CSeq: 2 BYE
User-Agent: TS-v4.5.1-18fW
Reason: MVTSLocal;cause=85;text="[SIP] Invalid SDP from terminator", Q.850;cause=16;text="Normal call clearing"
Content-Length: 0"
The codec preference on the snom are: pcma, pcmu then 729.
Incoming calls from two separate ITSP are haveing the same issue (one of the is more helpful with the log).
I've searched everywhere and can't find where this SDO issue might be set / unset. I'm tearing my hair out.
How do I fix up this "[SIP] Invalid SDP from terminator"?? Please help, Thank You.