Hi list, I have a sympathetic "issue" or miss-configuration with FS. SIP provider <--> FS <--> Opensips <--> Asterisk After long investigation, I saw FS merge a RE-INVITE request from Asterisk into 200OK to provider. ONly when I have 183 session progress Ex: *SDP INVITE from Provider (SIP provider -->FS)* v=0 o=Sonus_UAC 7950 990 IN IP4 ******** s=SIP Media Capabilities c=IN IP4 ********* t=0 0 m=audio 22970 RTP/AVP 8 18 2 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=ptime:20 *SDP INVITE from FS to asterisk (FS --> Opensips --> Asterisk)* v=0 o=Sonus_UAC 7950 990 IN IP4 ******** s=SIP Media Capabilities c=IN IP4 ****** t=0 0 m=audio 22970 RTP/AVP 8 18 2 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 *183 from asterisk (Asterisk --> Opensips --> FS)* v=0 o=- 7950 992 IN IP4 10.1.15.89 s=Asterisk c=IN IP4 10.1.15.89 t=0 0 m=audio 14096 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv *183 from FS to SIP provider (FS --> provider)* v=0 o=- 7950 992 IN IP4 ******* s=Asterisk c=IN IP4 ******* t=0 0 m=audio 18344 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=maxptime:150 a=rtcp:18345 a=ptime:20 *200OK from asterisk to FS (asterisk --> FS) (same code than 183)* v=0 o=- *7950 992* IN IP4 10.1.15.89 s=Asterisk c=IN IP4 10.1.15.89 t=0 0 m=audio 14096 RTP/AVP *8 18 101* a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv *RE-INVITE from asterisk (very close to 200OK)* v=0 o=- *7950 993* IN IP4 10.1.15.89 s=Asterisk c=IN IP4 10.1.15.89 t=0 0 m=audio 14096 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 a=ptime:20 a=maxptime:150 a=sendrecv *Final 200OK from FS to sip provider (FS --> sip provider)* v=0 o=- *7950 993* IN IP4 ****** s=Asterisk c=IN IP4 ****** t=0 0 m=audio 18344 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 a=maxptime:150 a=rtcp:18345 a=ptime:20 We can see FS receives REINVITE close to 200OK from asterisk, it doesn't send original 200OK to SIP provider, but "merge" 200OK with sdp from REINVITE (from asterisk) ==> same owner and codecs (7950 / 993). - owner 200OK from asterisk: 7950 992 / codecs: 8 18 101 - owner REINVITE from asterisk: 7950 993 / codecs: 8 0 18 101 - owner 200OK from FS to SIP provider: 7950 993 / codecs: 8 0 18 101 After that, FS never sends 200OK to the asterisk's REINVITE and opensips hangup call avec timeout (2 mins) (cf call flow in attachment). Do you already saw that ? How can I correct this please ? thanks in advance PS: - FreeSWITCH Version 1.10.1-release+git~20190820T165404Z~f9990221e6~64bit (git f999022 2019-08-20 16:54:04Z 64bit) - I use <param name="inbound-late-negotiation" value="true"/>
