Multiple SDP sections?

[lrosenman]

My SIP provider (BulkVS) tells me the reason my outbound calls are being rejected is multiple SDP sections. How can I fix this?
from them:
Hi Team,

It appears your system is sending two SDP sections on your invite. Could you please just send one.

When you update it send only this one:
m=audio 9004 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=sendrecv

SIP INVITE:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.230.140:5060;branch=z9hG4bK-524287-1---9b071e5bf0b65755;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]>
From: "2146429640"<sip:[email protected]>;tag=065d111a
Call-ID Y2moXUSOWINjCQkiVX4hWQ..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE
Content-Type: application/sdp
Supported: replaces, timer
User-Agent: 3CXPhoneSystem 20.0.2.715 (715)
Remote-Party-ID: "2146429640"<sip:[email protected]>;party=calling
Content-Length: 624

v=0
o=3cxPS 6627672046174208 33954353249255425 IN IP4 70.123.41.87
s=3cxPS Audio call
c=IN IP4 70.123.41.87
t=0 0
m=audio 9004 RTP/SAVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:vr6Nxlm/cEs86l9b71411mfeaODC9M3oUA/cP8DN
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:+I/hzrPWf1YNBZrfjMtn1n13kvA0biSVC6eLWc2h
a=sendrecv
m=audio 9004 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=sendrecv

Bulk Solutions Team

 

[JohnS_3CX]

Disable SRTP mode on your trunk settings

 

[lrosenman]

Thank You, John!