[OpenSIPS-Users] Replacing Adtran with OpenSIPS

Mon Sep 16 16:09:02 EDT 2019

Hi,

We are trying to replace our Adtran which is being used as an SBC for
faxing with openSIPS. But we are getting a problem with openSIPS. This
gonna be our call flow ->

Asterisk -> OpenSIPS -> SIP Trunk

Rtpproxy is running inside openSIPS. I am new to openSIPS so need help from
smart folks. When we get 200 OK (SDP) from trunk provider, our ACK is not
traversing the openSIPS. ACK is being directly sent to trunk provider.
Please view the call flow ->

2019/09/16 15:05:16.045125 10.250.110.90:5060 -> 10.10.10.172:5060
INVITE sip:6014279333 at 10.10.10.172 SIP/2.0
Via: SIP/2.0/UDP 10.250.110.90:5060;branch=z9hG4bK5b901208
Max-Forwards: 70
From: "Sunny Singh" <sip:6015863010 at 10.250.110.90>;tag=as6597b08e
To: <sip:6014279333 at 10.10.10.172>
Contact: <sip:6015863010 at 10.250.110.90:5060>
Call-ID: 3032e3ce2c4b2c9334aadd4c0d294683 at 10.250.110.90:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX certified/13.21-cert3
Date: Mon, 16 Sep 2019 20:05:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 301

v=0
o=root 736228850 736228850 IN IP4 10.250.110.90
s=Asterisk PBX certified/13.21-cert3
c=IN IP4 10.250.110.90
t=0 0
m=audio 12276 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

2019/09/16 15:05:16.045914 10.10.10.172:5060 -> 67.231.12.67:5060
INVITE sip:6014279333 at 67.231.12.67:5060 SIP/2.0
Record-Route: <sip:38.65.55.146:5060;lr;ftag=as6597b08e;did=455.89ce1c5>
Via: SIP/2.0/UDP 38.65.55.146:5060;branch=z9hG4bK424b.1d038492.0
Via: SIP/2.0/UDP 10.250.110.90:5060;branch=z9hG4bK5b901208
Max-Forwards: 69
From: "Sunny Singh" <sip:6015863010 at 10.250.110.90>;tag=as6597b08e
To: <sip:6014279333 at 10.10.10.172>
Contact: <sip:6015863010 at 10.250.110.90:5060>
Call-ID: 3032e3ce2c4b2c9334aadd4c0d294683 at 10.250.110.90:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX certified/13.21-cert3
Date: Mon, 16 Sep 2019 20:05:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 317

v=0
o=root 736228850 736228850 IN IP4 38.65.55.146
s=Asterisk PBX certified/13.21-cert3
c=IN IP4 38.65.55.146
t=0 0
m=audio 12322 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
a=nortpproxy:yes

2019/09/16 15:05:16.281867 67.231.12.67:5060 -> 10.10.10.172:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 38.65.55.146:5060;branch=z9hG4bK424b.1d038492.0
Via: SIP/2.0/UDP 10.250.110.90:5060;branch=z9hG4bK5b901208
From: "Sunny Singh" <sip:6015863010 at 10.250.110.90>;tag=as6597b08e
To: <sip:6014279333 at 10.10.10.172>;tag=gK00b184ef
Call-ID: 3032e3ce2c4b2c9334aadd4c0d294683 at 10.250.110.90:5060
CSeq: 102 INVITE
Record-Route: <sip:38.65.55.146:5060;lr;ftag=as6597b08e;did=455.89ce1c5>
Accept: application/sdp
Contact: <sip:6014279333 at 67.231.12.67:5060>
Allow: INVITE,ACK,CANCEL,BYE,PRACK,OPTIONS
Supported: replaces
Content-Length:   236
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 687409 672900 IN IP4 67.231.12.67
s=SIP Media Capabilities
c=IN IP4 67.231.12.207
t=0 0
m=audio 36368 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20

2019/09/16 15:05:16.289521 10.10.10.172:5060 -> 10.250.110.90:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.250.110.90:5060;branch=z9hG4bK5b901208
From: "Sunny Singh" <sip:6015863010 at 10.250.110.90>;tag=as6597b08e
To: <sip:6014279333 at 10.10.10.172>;tag=gK00b184ef
Call-ID: 3032e3ce2c4b2c9334aadd4c0d294683 at 10.250.110.90:5060
CSeq: 102 INVITE
Record-Route: <sip:38.65.55.146:5060;lr;ftag=as6597b08e;did=455.89ce1c5>
Accept: application/sdp
Contact: <sip:6014279333 at 67.231.12.67:5060>
Allow: INVITE,ACK,CANCEL,BYE,PRACK,OPTIONS
Supported: replaces
Content-Length: 253
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 687409 672900 IN IP4 38.65.55.146
s=SIP Media Capabilities
c=IN IP4 38.65.55.146
t=0 0
m=audio 13406 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
a=nortpproxy:yes

2019/09/16 15:05:16.290712 10.250.110.90:34022 -> 10.10.10.172:5060
ACK sip:6014279333 at 67.231.12.67:5060 SIP/2.0
Via: SIP/2.0/UDP 10.250.110.90:5060;branch=z9hG4bK5c3259ad
Route: <sip:38.65.55.146:5060;lr;ftag=as6597b08e;did=455.89ce1c5>
Max-Forwards: 70
From: "Sunny Singh" <sip:6015863010 at 10.250.110.90>;tag=as6597b08e
To: <sip:6014279333 at 10.10.10.172>;tag=gK00b184ef
Contact: <sip:6015863010 at 10.250.110.90:5060>
Call-ID: 3032e3ce2c4b2c9334aadd4c0d294683 at 10.250.110.90:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX certified/13.21-cert3
Content-Length: 0

And for this setup we have to use openSIPS residential script or trunking
script ?

Thanks
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