Asterisk 13 does not honor invite request of a=ptime:20 a=maxptime:20

Asterisk 13 does not honor invite request of a=ptime:20 a=maxptime:20.

It sends a larger payload.

Please what to to cahnge to correst this?

Asterisk 11 responds properly.

Please see below.

Chris

CSeq: 102 INVITE
Contact: sip:8893@10.200.20.91
User-Agent: RiedelArtist/7.20.CL2#21 #21 (VoIP)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, INFO
Require: timer
Supported: timer, 100rel
Session-Expires: 3600;refresher=uac
Content-Type: application/sdp
Content-Length: 331

v=0
o=sip:8893@10.200.20.91 0 2 IN IP4 10.200.20.91
s=sip:8893@10.200.20.91
c=IN IP4 10.200.20.91
t=0 0
m=audio 5016 RTP/AVP 0 8 9 101
a=rtcp:5017 IN IP4 10.200.20.91
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=maxptime:20
a=sendrecv
<------------->
[2018-04-03 19:05:45] VERBOSE[13752] chan_sip.c: — (14 headers 15 lines) —
[2018-04-03 19:05:45] VERBOSE[13752][C-00000099] chan_sip.c: Found RTP audio format 0
[2018-04-03 19:05:45] VERBOSE[13752][C-00000099] chan_sip.c: Found RTP audio format 8
[2018-04-03 19:05:45] VERBOSE[13752][C-00000099] chan_sip.c: Found RTP audio format 9
[2018-04-03 19:05:45] VERBOSE[13752][C-00000099] chan_sip.c: Found RTP audio format 101
[2018-04-03 19:05:45] VERBOSE[13752][C-00000099] chan_sip.c: Found audio description format PCMU for ID 0
[2018-04-03 19:05:45] VERBOSE[13752][C-00000099] chan_sip.c: Found audio description format PCMA for ID 8
[2018-04-03 19:05:45] VERBOSE[13752][C-00000099] chan_sip.c: Found audio description format G722 for ID 9
[2018-04-03 19:05:45] VERBOSE[13752][C-00000099] chan_sip.c: Found audio description format telephone-event for ID 101
[2018-04-03 19:05:45] VERBOSE[13752][C-00000099] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw|alaw|g722)/video=(nothing)/text=(nothing), combined - (ulaw)
[2018-04-03 19:05:45] VERBOSE[13752][C-00000099] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2018-04-03 19:05:45] VERBOSE[13752][C-00000099] chan_sip.c: Peer audio RTP is at port 10.200.20.91:5016
[2018-04-03 19:05:45] VERBOSE[13752][C-00000099] sip/route.c: sip_route_dump: route/path hop: sip:8893@10.200.20.91
[2018-04-03 19:05:45] VERBOSE[13752][C-00000099] chan_sip.c: Transmitting (NAT) to 10.200.20.91:5060:
ACK sip:8893@10.200.20.91 SIP/2.0
Via: SIP/2.0/UDP 10.200.20.200:5060;branch=z9hG4bK76ef7110;rport
Max-Forwards: 70
From: "company HOLDINGS I " <sip:npa nxx xxxx@10.200.20.200>;tag=as6f0ebae0
To: sip:8893@10.200.20.91;tag=Xa6H857rm2XDF
Contact: <sip:npa nxx xxxx@10.200.20.200:5060>
Call-ID: 58a14462628dbe7b3712f3bc0a1dcdb6@10.200.20.200:5060
CSeq: 102 ACK
User-Agent: FPBX-13.0.194.5(13.19.1)
Content-Length: 0


[2018-04-03 19:05:45] VERBOSE[29890][C-00000099] app_dial.c: SIP/8893-00000145 answered SIP/ATT-VOIP-2-00000144
[2018-04-03 19:05:45] VERBOSE[29890][C-00000099] chan_sip.c: Audio is at 31496
[2018-04-03 19:05:45] VERBOSE[29890][C-00000099] chan_sip.c: Adding codec ulaw to SDP
[2018-04-03 19:05:45] VERBOSE[29890][C-00000099] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2018-04-03 19:05:45] VERBOSE[29890][C-00000099] chan_sip.c:
<— Reliably Transmitting (NAT) to xxx.xxx.xxx.xxx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bKktso2100787hsghmlhr0.1;received=xxx.xxx.xxx.xxx;rport=5060
From: "company HOLDINGS I " <sip:npa nxx xxxx@xxx.xxx.xxx.xxx;user=phone>;tag=1478074414-1522807545104-
To: "User npxnxx8893"sip:npxnxx8893@10.20.200.200;user=phone;tag=as6004129b
Call-ID: BW020545104040418-1166658349@F1
CSeq: 120819337 INVITE
Server: FPBX-13.0.194.5(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: sip:npxnxx8893@10.20.200.200:5060
Content-Type: application/sdp
Require: timer
Content-Length: 250

v=0
o=root 48030184 48030184 IN IP4 10.20.200.200
s=Asterisk PBX 13.19.1
c=IN IP4 10.20.200.200
t=0 0
m=audio 31496 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>
[2018-04-03 19:05:45] VERBOSE[29891][C-00000099] bridge_channel.c: Channel SIP/8893-00000145 joined ‘simple_bridge’ basic-bridge <9526ab98-34f0-4a47-ae4d-ee4d311f13fc>
[2018-04-03 19:05:45] VERBOSE[29890][C-00000099] bridge_channel.c: Channel SIP/ATT-VOIP-2-00000144 joined ‘simple_bridge’ basic-bridge <9526ab98-34f0-4a47-ae4d-ee4d311f13fc>
[2018-04-03 19:05:45] VERBOSE[13752] chan_sip.c: Really destroying SIP dialog ‘yIjP8cAJAzAntbxa4RhjHTXK1e8TEiWg’ Method: REGISTER
[2018-04-03 19:05:45] VERBOSE[13752] chan_sip.c: Really destroying SIP dialog ‘zdgmQA.2AydMUbVrKiFthZyUnECbWshP’ Method: REGISTER
[2018-04-03 19:05:45] VERBOSE[13752] chan_sip.c:
<— SIP read from UDP:xxx.xxx.xxx.xxx:5060 —>
ACK sip:npxnxx8893@10.20.200.200:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bKmdgu8s00b04igp1i1op0.1
From: "company HOLDINGS I " <sip:npa nxx xxxx@xxx.xxx.xxx.xxx;user=phone>;tag=1478074414-1522807545104-
To: "User npxnxx8893"sip:npxnxx8893@10.20.200.200;user=phone;tag=as6004129b
Call-ID: BW020545104040418-1166658349@F1
CSeq: 120819337 ACK
Contact: sip:xxx.xxx.xxx.xxx:5060;transport=udp
Max-Forwards: 69
Content-Length: 0

<------------->
[2018-04-03 19:05:45] VERBOSE[13752] chan_sip.c: — (9 headers 0 lines) —
[2018-04-03 19:05:45] VERBOSE[13752] chan_sip.c: Really destroying SIP dialog ‘.jQsv8de9MGXBsdSi5J6-JnB0slT4W…’ Method: REGISTER
[2018-04-03 19:05:45] VERBOSE[13752] chan_sip.c:
<— SIP read from UDP:10.200.20.91:5060 —>
BYE sip:npa nxx xxxx@10.200.20.200:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.20.91;rport;branch=z9hG4bKpNSFKXc4Q2XgB
Max-Forwards: 70
From: sip:8893@10.200.20.91;tag=Xa6H857rm2XDF
To: "company HOLDINGS I " <sip:npa nxx xxxx@10.200.20.200>;tag=as6f0ebae0
Call-ID: 58a14462628dbe7b3712f3bc0a1dcdb6@10.200.20.200:5060
CSeq: 30845908 BYE
Contact: sip:8893@10.200.20.91?