Trunk Sip Audio problem

Here is my file.

My Asterisk server is a vps with a public ip (194.34.XXX.XXX) connected to the operator by a vpn the operator’s local network is 172.14.23.X. The operator has a Sip (172.14.23.85) and an RTP (172.14.23.90).
I created a virtual address on my VPS (10.15.21.154). It is she who communicates directly with the operator.
Outgoing calls from my PBX to the operator are OK with OK sound. But incoming calls from IMS to my PBX are established but have no audio.
I nevertheless notice that the IMS sends me in SDP the AMR codecs which are not present in Asterisk 20. is this a codec problem or another problem? should I install AMR codecs? If yes, how ? what should I do in this case

<--- Received SIP request (1448 bytes) from UDP:172.14.23.85:5060 --->
INVITE sip:+245551552665@10.15.21.154:5060;line=ekaopjw SIP/2.0
Via: SIP/2.0/UDP 172.14.23.85:5060;branch=z9hG4bKheq5tjpv5lwvcvpttfcwcuwc5;Role=3;Hpt=8e52_36
Record-Route: <sip:172.14.23.85:5060;lr;Hpt=nw_27_64f723fa_3ed_ex_8e52_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=327>
Call-ID: asbcll00kkm1hwjhkxlavzizajvuxj11nnmm@172.26.112.6
From: "698541263"<tel:98541263;noa=subscriber;srvattri=national;phone-context=+2455>;tag=njvxmuiz
To: <sip:659999877@172.14.23.85;transport=udp;user=phone>
CSeq: 1 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,UPDATE
Contact: <sip:172.14.23.85:5060;Dsp=eaea-200;Hpt=nw_27_64f723fa_3ed_ex_8e52_16;CxtId=4;TRC=ffffffff-ffffffff>
Max-Forwards: 66
Supported: timer,100rel,histinfo
Session-Expires: 1800
Min-SE: 600
P-Asserted-Identity: <tel:698541263>
P-Called-Party-ID: <tel:+245551552665>
P-Early-Media: supported,gated
Content-Length: 490
Content-Type: application/sdp

v=0
o=- 259 259 IN IP4 172.14.23.90
s=SBC call
c=IN IP4 172.14.23.90
t=0 0
m=audio 12056 RTP/AVP 101 108 8 100 96 116 97
a=rtpmap:101 AMR-WB/16000
a=fmtp:101 mode-set=0,1,2
a=rtpmap:108 AMR/8000
a=fmtp:108 mode-set=7
a=rtpmap:8 PCMA/8000
a=rtpmap:100 AMR/8000
a=fmtp:100 mode-set=0,2,4,7;mode-change-neighbor=1;mode-change-period=2
a=rtpmap:96 AMR/8000
a=fmtp:96 mode-set=0,2,4,7
a=rtpmap:116 telephone-event/16000
a=rtpmap:97 telephone-event/8000
a=ptime:20
a=3gOoBTC

<--- Transmitting SIP response (561 bytes) to UDP:172.14.23.85:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.14.23.85:5060;rport=5060;received=172.14.23.85;branch=z9hG4bKheq5tjpv5lwvcvpttfcwcuwc5;Role=3;Hpt=8e52_36
Record-Route: <sip:172.14.23.85:5060;lr;Hpt=nw_27_64f723fa_3ed_ex_8e52_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=327>
Call-ID: asbcll00kkm1hwjhkxlavzizajvuxj11nnmm@172.26.112.6
From: "698541263" <tel:98541263;phone-context=+2455;noa=subscriber;srvattri=national>;tag=njvxmuiz
To: <sip:659999877@172.14.23.85;user=phone>
CSeq: 1 INVITE
Server: ASTERISK
Content-Length:  0



<--- Transmitting SIP request (1840 bytes) to UDP:154.72.150.227:22576 --->
INVITE sip:pbxteam237698541263@154.72.150.227:22576;ob SIP/2.0
Via: SIP/2.0/UDP 194.34.XXX.XXX:5060;rport;branch=z9hG4bKPj5fd637bf-db2c-4391-bc50-0fa8ae8740a4
From: "698541263" <sip:98541263@194.34.XXX.XXX>;tag=05363a34-c03f-4731-a0b0-ad2d4c01b4d6
To: <sip:pbxteam237698541263@154.72.150.227;ob>
Contact: <sip:asterisk@194.34.XXX.XXX:5060>
Call-ID: fbd52e52-2b8d-4751-a5c7-165acb329881
CSeq: 1546 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: ASTERISK
Content-Type: application/sdp
Content-Length:  1105

v=0
o=- 1666360584 1666360584 IN IP4 194.34.XXX.XXX
s=Asterisk
c=IN IP4 194.34.XXX.XXX
t=0 0
m=audio 14016 RTP/AVP 8 98 4 0 3 111 112 5 10 122 118 123 124 125 126 127 96 7 18 110 117 119 97 9 102 115 116 107 101
a=rtpmap:8 PCMA/8000
a=rtpmap:98 CODEC2/8000
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:122 L16/12000
a=rtpmap:118 L16/16000
a=rtpmap:123 L16/24000
a=rtpmap:124 L16/32000
a=rtpmap:125 L16/44000
a=rtpmap:126 L16/48000
a=rtpmap:127 L16/96000
a=rtpmap:96 L16/192000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:119 speex/32000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

<--- Received SIP response (374 bytes) from UDP:154.72.150.227:22576 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 194.34.XXX.XXX:5060;rport=5060;received=194.34.XXX.XXX;branch=z9hG4bKPj5fd637bf-db2c-4391-bc50-0fa8ae8740a4
Call-ID: fbd52e52-2b8d-4751-a5c7-165acb329881
From: "698541263" <sip:98541263@194.34.XXX.XXX>;tag=05363a34-c03f-4731-a0b0-ad2d4c01b4d6
To: <sip:pbxteam237698541263@154.72.150.227;ob>
CSeq: 1546 INVITE
Content-Length:  0


<--- Received SIP response (583 bytes) from UDP:154.72.150.227:22576 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 194.34.XXX.XXX:5060;rport=5060;received=194.34.XXX.XXX;branch=z9hG4bKPj5fd637bf-db2c-4391-bc50-0fa8ae8740a4
Call-ID: fbd52e52-2b8d-4751-a5c7-165acb329881
From: "698541263" <sip:98541263@194.34.XXX.XXX>;tag=05363a34-c03f-4731-a0b0-ad2d4c01b4d6
To: <sip:pbxteam237698541263@154.72.150.227;ob>;tag=7157889ee5854fdc8f2a8496fa389e48
CSeq: 1546 INVITE
Contact: "pbxteams" <sip:pbxteam237698541263@192.168.100.80:54538;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length:  0


    -- PJSIP/pbxteam237698541263-0000002b is ringing
<--- Transmitting SIP response (748 bytes) to UDP:172.14.23.85:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.14.23.85:5060;rport=5060;received=172.14.23.85;branch=z9hG4bKheq5tjpv5lwvcvpttfcwcuwc5;Role=3;Hpt=8e52_36
Record-Route: <sip:172.14.23.85:5060;lr;Hpt=nw_27_64f723fa_3ed_ex_8e52_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=327>
Call-ID: asbcll00kkm1hwjhkxlavzizajvuxj11nnmm@172.26.112.6
From: "698541263" <tel:98541263;phone-context=+2455;noa=subscriber;srvattri=national>;tag=njvxmuiz
To: <sip:659999877@172.14.23.85;user=phone>;tag=4c92cb23-c53a-4358-bd3a-47ea53d54aa3
CSeq: 1 INVITE
Server: ASTERISK
Contact: <sip:10.15.21.154:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Length:  0


<--- Received SIP response (1034 bytes) from UDP:154.72.150.227:22576 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 194.34.XXX.XXX:5060;rport=5060;received=194.34.XXX.XXX;branch=z9hG4bKPj5fd637bf-db2c-4391-bc50-0fa8ae8740a4
Call-ID: fbd52e52-2b8d-4751-a5c7-165acb329881
From: "698541263" <sip:98541263@194.34.XXX.XXX>;tag=05363a34-c03f-4731-a0b0-ad2d4c01b4d6
To: <sip:pbxteam237698541263@154.72.150.227;ob>;tag=7157889ee5854fdc8f2a8496fa389e48
CSeq: 1546 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: "pbxteams" <sip:pbxteam237698541263@192.168.100.80:54538;ob>
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   321

v=0
o=- 3902907002 3902907003 IN IP4 192.168.100.80
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4010 RTP/AVP 8 101
c=IN IP4 192.168.100.80
b=TIAS:64000
a=rtcp:4011 IN IP4 192.168.100.80
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:384645182 cname:49d66e664bdc4d3a

       > 0x7f237c030530 -- Strict RTP learning after remote address set to: 192.168.100.80:4010
<--- Transmitting SIP request (483 bytes) to UDP:154.72.150.227:22576 --->
ACK sip:pbxteam237698541263@154.72.150.227:22576;ob SIP/2.0
Via: SIP/2.0/UDP 194.34.XXX.XXX:5060;rport;branch=z9hG4bKPj920a1d40-b996-4674-8883-3271441b8012
From: "698541263" <sip:98541263@194.34.XXX.XXX>;tag=05363a34-c03f-4731-a0b0-ad2d4c01b4d6
To: <sip:pbxteam237698541263@154.72.150.227;ob>;tag=7157889ee5854fdc8f2a8496fa389e48
Call-ID: fbd52e52-2b8d-4751-a5c7-165acb329881
CSeq: 1546 ACK
Max-Forwards: 70
User-Agent: ASTERISK
Content-Length:  0

<--- Transmitting SIP response (1098 bytes) to UDP:172.14.23.85:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.14.23.85:5060;rport=5060;received=172.14.23.85;branch=z9hG4bKheq5tjpv5lwvcvpttfcwcuwc5;Role=3;Hpt=8e52_36
Record-Route: <sip:172.14.23.85:5060;lr;Hpt=nw_27_64f723fa_3ed_ex_8e52_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=327>
Call-ID: asbcll00kkm1hwjhkxlavzizajvuxj11nnmm@172.26.112.6
From: "698541263" <tel:98541263;phone-context=+2455;noa=subscriber;srvattri=national>;tag=njvxmuiz
To: <sip:659999877@172.14.23.85;user=phone>;tag=4c92cb23-c53a-4358-bd3a-47ea53d54aa3
CSeq: 1 INVITE
Server: ASTERISK
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:10.15.21.154:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   220

v=0
o=- 259 261 IN IP4 10.15.21.154
s=Asterisk
c=IN IP4 10.15.21.154
t=0 0
m=audio 11102 RTP/AVP 8 97
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

Got  RTP packet from    154.72.150.227:32249 (type 08, seq 029057, ts 000800, len 000160)
Sent RTP P2P packet to 172.14.23.90:12056 (type 08, len 000160)
Got  RTP packet from    154.72.150.227:32249 (type 08, seq 029058, ts 000960, len 000160)
Sent RTP P2P packet to 172.14.23.90:12056 (type 08, len 000160)
Got  RTP packet from    154.72.150.227:32249 (type 08, seq 029059, ts 001120, len 000160)
Sent RTP P2P packet to 172.14.23.90:12056 (type 08, len 000160)
Got  RTP packet from    154.72.150.227:32249 (type 08, seq 029060, ts 001280, len 000160)
Sent RTP P2P packet to 172.14.23.90:12056 (type 08, len 000160)
Got  RTP packet from    154.72.150.227:32249 (type 08, seq 029077, ts 004000, len 000160)
Sent RTP P2P packet to 172.14.23.90:12056 (type 08, len 000160)
Got  RTP packet from    154.72.150.227:32249 (type 08, seq 029078, ts 004160, len 000160)
Sent RTP P2P packet to 172.14.23.90:12056 (type 08, len 000160)
Got  RTP packet from    154.72.150.227:32249 (type 08, seq 029079, ts 004320, len 000160)
Sent RTP P2P packet to 172.14.23.90:12056 (type 08, len 000160)
<--- Transmitting SIP response (1098 bytes) to UDP:172.14.23.85:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.14.23.85:5060;rport=5060;received=172.14.23.85;branch=z9hG4bKheq5tjpv5lwvcvpttfcwcuwc5;Role=3;Hpt=8e52_36
Record-Route: <sip:172.14.23.85:5060;lr;Hpt=nw_27_64f723fa_3ed_ex_8e52_16;CxtId=4;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=327>
Call-ID: asbcll00kkm1hwjhkxlavzizajvuxj11nnmm@172.26.112.6
From: "698541263" <tel:98541263;phone-context=+2455;noa=subscriber;srvattri=national>;tag=njvxmuiz
To: <sip:659999877@172.14.23.85;user=phone>;tag=4c92cb23-c53a-4358-bd3a-47ea53d54aa3
CSeq: 1 INVITE
Server: ASTERISK
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:10.15.21.154:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   220

v=0
o=- 259 261 IN IP4 10.15.21.154
s=Asterisk
c=IN IP4 10.15.21.154
t=0 0
m=audio 11102 RTP/AVP 8 97
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

Got  RTP packet from    154.72.150.227:32249 (type 08, seq 029080, ts 004480, len 000160)
Got  RTP packet from    154.72.150.227:32249 (type 08, seq 029088, ts 005760, len 000160)
Sent RTP P2P packet to 172.14.23.90:12056 (type 08, len 000160)
Got  RTP packet from    154.72.150.227:32249 (type 08, seq 029089, ts 005920, len 000160)
Sent RTP P2P packet to 172.14.23.90:12056 (type 08, len 000160)
<--- Received SIP request (443 bytes) from UDP:172.14.23.85:5060 --->
ACK sip:10.15.21.154:5060 SIP/2.0
Via: SIP/2.0/UDP 172.14.23.85:5060;branch=z9hG4bKhgev0ewefphppjevcg3hcth3h;Role=3;Hpt=8e52_36
Call-ID: asbcll00kkm1hwjhkxlavzizajvuxj11nnmm@172.26.112.6
From: "698541263"<tel:98541263;noa=subscriber;srvattri=national;phone-context=+2455>;tag=njvxmuiz
To: <sip:659999877@172.14.23.85;transport=udp;user=phone>;tag=4c92cb23-c53a-4358-bd3a-47ea53d54aa3
CSeq: 1 ACK
Max-Forwards: 66
Content-Length: 0


Got  RTP packet from    154.72.150.227:32249 (type 08, seq 029090, ts 006080, len 000160)
Sent RTP P2P packet to 172.14.23.90:12056 (type 08, len 000160)
Got  RTP packet from    154.72.150.227:32249 (type 08, seq 029091, ts 006240, len 000160)
Sent RTP P2P packet to 172.14.23.90:12056 (type 08, len 000160)
Got  RTP packet from    154.72.150.227:32249 (type 08, seq 029254, ts 032320, len 000160)
Sent RTP P2P packet to 172.14.23.90:12056 (type 08, len 000160)
Got  RTP packet from    154.72.150.227:32249 (type 08, seq 029255, ts 032480, len 000160)
Sent RTP P2P packet to 172.14.23.90:12056 (type 08, len 000160)
<--- Received SIP request (495 bytes) from UDP:172.14.23.85:5060 --->
BYE sip:10.15.21.154:5060 SIP/2.0
Via: SIP/2.0/UDP 172.14.23.85:5060;branch=z9hG4bK3evjptvwfpgwhfvw3p0hcc3vl;Role=3;Hpt=8e52_36
Call-ID: asbcll00kkm1hwjhkxlavzizajvuxj11nnmm@172.26.112.6
From: "698541263"<tel:98541263;noa=subscriber;srvattri=national;phone-context=+2455>;tag=njvxmuiz
To: <sip:659999877@172.14.23.85;transport=udp;user=phone>;tag=4c92cb23-c53a-4358-bd3a-47ea53d54aa3
CSeq: 2 BYE
Max-Forwards: 66
Reason: Q.850;cause=16;text="Normal call clearing"
Content-Length: 0


<--- Transmitting SIP response (468 bytes) to UDP:172.14.23.85:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.14.23.85:5060;rport=5060;received=172.14.23.85;branch=z9hG4bK3evjptvwfpgwhfvw3p0hcc3vl;Role=3;Hpt=8e52_36
Call-ID: asbcll00kkm1hwjhkxlavzizajvuxj11nnmm@172.26.112.6
From: "698541263" <tel:98541263;phone-context=+2455;noa=subscriber;srvattri=national>;tag=njvxmuiz
To: <sip:659999877@172.14.23.85;user=phone>;tag=4c92cb23-c53a-4358-bd3a-47ea53d54aa3
CSeq: 2 BYE
Server: ASTERISK
Content-Length:  0

<--- Transmitting SIP request (507 bytes) to UDP:154.72.150.227:22576 --->
BYE sip:pbxteam237698541263@154.72.150.227:22576;ob SIP/2.0
Via: SIP/2.0/UDP 194.34.XXX.XXX:5060;rport;branch=z9hG4bKPje1c02f23-7657-41fd-9a14-01fe5e80b48f
From: "698541263" <sip:98541263@194.34.XXX.XXX>;tag=05363a34-c03f-4731-a0b0-ad2d4c01b4d6
To: <sip:pbxteam237698541263@154.72.150.227;ob>;tag=7157889ee5854fdc8f2a8496fa389e48
Call-ID: fbd52e52-2b8d-4751-a5c7-165acb329881
CSeq: 1547 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: ASTERISK
Content-Length:  0


<--- Transmitting SIP request (507 bytes) to UDP:154.72.150.227:22576 --->
BYE sip:pbxteam237698541263@154.72.150.227:22576;ob SIP/2.0
Via: SIP/2.0/UDP 194.34.XXX.XXX:5060;rport;branch=z9hG4bKPje1c02f23-7657-41fd-9a14-01fe5e80b48f
From: "698541263" <sip:98541263@194.34.XXX.XXX>;tag=05363a34-c03f-4731-a0b0-ad2d4c01b4d6
To: <sip:pbxteam237698541263@154.72.150.227;ob>;tag=7157889ee5854fdc8f2a8496fa389e48
Call-ID: fbd52e52-2b8d-4751-a5c7-165acb329881
CSeq: 1547 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: ASTERISK
Content-Length:  0


<--- Received SIP response (404 bytes) from UDP:154.72.150.227:22576 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 194.34.XXX.XXX:5060;rport=5060;received=194.34.XXX.XXX;branch=z9hG4bKPje1c02f23-7657-41fd-9a14-01fe5e80b48f
Call-ID: fbd52e52-2b8d-4751-a5c7-165acb329881
From: "698541263" <sip:98541263@194.34.XXX.XXX>;tag=05363a34-c03f-4731-a0b0-ad2d4c01b4d6
To: <sip:pbxteam237698541263@154.72.150.227;ob>;tag=7157889ee5854fdc8f2a8496fa389e48
CSeq: 1547 BYE
Content-Length:  0


<--- Received SIP response (404 bytes) from UDP:154.72.150.227:22576 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 194.34.XXX.XXX:5060;rport=5060;received=194.34.XXX.XXX;branch=z9hG4bKPje1c02f23-7657-41fd-9a14-01fe5e80b48f
Call-ID: fbd52e52-2b8d-4751-a5c7-165acb329881
From: "698541263" <sip:98541263@194.34.XXX.XXX>;tag=05363a34-c03f-4731-a0b0-ad2d4c01b4d6
To: <sip:pbxteam237698541263@154.72.150.227;ob>;tag=7157889ee5854fdc8f2a8496fa389e48
CSeq: 1547 BYE
Content-Length:  0


First thing you should check is if your rtp ports are opened 10000-25000.
nat=yes in each phone’s configuration?
Firewall.
When asterisk initiates a phone call, there are always TWO ports in use if the call is a SIP phone call.
UDP Port 5060 for control and a UDP rport (random port) between 10k and 25k, chosen at the moment of the call, for the audio.
If the 5060 packet gets through, you get a ring and even a “live call” notice. But until the rport traffic succeeds, you won’t get any audio.
So: UDP5060 is succeeding, but the rport is being blocked to the agent, so no audio.

outgoing calls from the PBX to the IMS work very well there is no NAT between the pbx and the IMS, there is just a VPN which connects them. There is no firewall

What IP is that ?

my softphone is in my android phone so this ip in my android public IP

IMS == softphone ?

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