No voice is heard on an incoming/outgoing call

FreePBX Endpoints
1

On FreePBX I have two eth… LAN 192.168.1.a/24 (phones are also in this network) and for VoIP trunk 10.224.33.xx to connect for Operator host 10.40.0.x
Peer settings:

username=7232aaabbb
fromuser=72324aaabbb
type=peer
qualify=yes
port=5060
insecure=invite,port
host=10.40.0.x
context=from-trunk
allow=alaw,ulaw

Codecs are well, NAT is not used
In Logs I see

<— Reliably *Transmitting (NAT) *to 10.40.0.x:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.40.0.x:5060;branch=z9hG4bK-y2qlr-dmkyt;received=10.40.0.x;rport=5060
From: “87773aaabbb” <sip:87773aaabbb @ 10.40.0.1:5060;user=phone>;tag=fe4jw4rly5
To: “7232aaabbb” <sip:7232aaabbb @ 10.40.0.x:5060;user=phone>;tag=as7c7dce47
Call-ID: [email protected]
CSeq: 268 INVITE
Server: FPBX-16.0.33(18.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:7232aaabbb @ 10.224.33.xx:5060>
Content-Type: application/sdp
Content-Length: 248

This is rtp set debug on

Sent RTP packet to 10.40.0.xx:19442 (type 00, seq 021178, ts 007040, len 000160)
Sent RTP packet to 10.40.0.xx:19442 (type 00, seq 021179, ts 007200, len 000160)
Sent RTP packet to 10.40.0.xx:19442 (type 00, seq 021180, ts 007360, len 000160)
Sent RTP packet to 10.40.0.xx:19442 (type 00, seq 021181, ts 007520, len 000160)
Sent RTP packet to 10.40.0.xx:19442 (type 00, seq 021182, ts 007680, len 000160)
Got RTP packet from 192.168.1.yy:5048 (type 00, seq 018013, ts 2205280, len 000160)
Sent RTP packet to 10.40.0.xx:19442 (type 08, seq 021183, ts 2205280, len 000160)
Got RTP packet from 192.168.1.yy:5048 (type 00, seq 018014, ts 2205440, len 000160)
Sent RTP packet to 10.40.0.xx:19442 (type 08, seq 021184, ts 2205440, len 000160)
Got RTP packet from 192.168.1.yy:5048 (type 00, seq 018015, ts 2205600, len 000160)
Sent RTP packet to 10.40.0.xx:19442 (type 08, seq 021185, ts 2205600, len 000160)
Got RTP packet from 192.168.1.yy:5048 (type 00, seq 018016, ts 2205760, len 000160)
Sent RTP packet to 10.40.0.xx:19442 (type 08, seq 021186, ts 2205760, len 000160)
Got RTP packet from 192.168.1.yy:5048 (type 00, seq 018017, ts 2205920, len 000160)
Sent RTP packet to 10.40.0.xx:19442 (type 08, seq 021187, ts 2205920, len 000160)
Got RTP packet from 192.168.1.yy:5048 (type 00, seq 018018, ts 2206080, len 000160)
Sent RTP packet to 10.40.0.xx:19442 (type 08, seq 021188, ts 2206080, len 000160)
Got RTP packet from 192.168.1.yy:5048 (type 00, seq 018019, ts 2206240, len 000160)
Sent RTP packet to 10.40.0.xx:19442 (type 08, seq 021189, ts 2206240, len 000160)
Got RTP packet from 192.168.1.yy:5048 (type 00, seq 018020, ts 2206400, len 000160)
Sent RTP packet to 10.40.0.xx:19442 (type 08, seq 021190, ts 2206400, len 000160)
Got RTP packet from 192.168.1.yy:5048 (type 00, seq 018021, ts 2206560, len 000160)
Sent RTP packet to 10.40.0.xx:19442 (type 08, seq 021191, ts 2206560, len 000160)
Got RTP packet from 192.168.1.yy:5048 (type 00, seq 018022, ts 2206720, len 000160)
Sent RTP packet to 10.40.0.xx:19442 (type 08, seq 021192, ts 2206720, len 000160)
Got RTP packet from 192.168.1.yy:5048 (type 00, seq 018023, ts 2206880, len 000160)
Sent RTP packet to 10.40.0.xx:19442 (type 08, seq 021193, ts 2206880, len 000160)
Got RTP packet from 192.168.1.yy:5048 (type 00, seq 018024, ts 2207040, len 000160)

What else can I check?

2

You’ve deleted the part of the 200 OK that actually matters, for media, and you haven’t provided any of the INVITE

3

SIP Debugging enabled

<— SIP read from UDP:10.40.0.1:5060 —>
INVITE sip:7232492078 @ 10.224.33.10:5060;user=phone SIP/2.0
Accept: application/sdp
Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS,REGISTER,REFER,SUBSCRIBE,MESSAGE,PUBLISH
Call-ID: 4662966fipd390bpl @ 10.40.0.1
Contact: “87773300792” <sip:87773300792 @ 10.40.0.1:5060;user=phone>
CSeq: 137 INVITE
Expires: 3600
From: “87773300792” <sip:87773300792 @ 10.40.0.1:5060;user=phone>;tag=gfhospxx0t
To: “7232492078” <sip:7232492078 @ 10.40.0.1:5060;user=phone>
Organization: IskraTel
Supported: 100rel
User-Agent: SI3000
Via: SIP/2.0/UDP 10.40.0.1:5060;branch=z9hG4bK-p36ko-c0sfk
Resource-Priority: q735.4
Max-Forwards: 70
Subject: Call from CS6111
P-Asserted-Identity: “87773300792” <sip:87773300792 @ 10.40.0.1:5060;user=phone>
Content-Length: 282
Content-Type: application/sdp
Content-Disposition: session;handling=required

v=0
o=- 2127603 7513105 IN IP4 10.40.0.8
s=-
c=IN IP4 10.40.0.8
b=AS:64
t=0 0
m=audio 17954 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 0-15
a=ptime:20
a=sendrecv
<------------->
— (20 headers 15 lines) —
Sending to 10.40.0.1:5060 (NAT)
Sending to 10.40.0.1:5060 (NAT)
Using INVITE request as basis request - 4662966fipd390bpl @ 10.40.0.1
Found peer ‘7232492078’ for ‘87773300792’ from 10.40.0.1:5060
Got SDP version 7513105 and unique parts [- 2127603 IN IP4 10.40.0.8]
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.40.0.8:17954
Looking for 7232492078 in from-trunk (domain 10.224.33.10)
sip_route_dump: route/path hop: <sip:87773300792 @ 10.40.0.1:5060;user=phone>

<— Transmitting (NAT) to 10.40.0.1:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.40.0.1:5060;branch=z9hG4bK-p36ko-c0sfk;received=10.40.0.1;rport=5060
From: “87773300792” <sip:87773300792 @ 10.40.0.1:5060;user=phone>;tag=gfhospxx0t
To: “7232492078” <sip:7232492078 @ 10.40.0.1:5060;user=phone>
Call-ID: 4662966fipd390bpl @ 10.40.0.1
CSeq: 137 INVITE
Server: FPBX-16.0.33(18.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:7232492078 @ 10.224.33.10:5060>
Content-Length: 0

<------------>
Audio is at 18300
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (NAT) to 10.40.0.1:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.40.0.1:5060;branch=z9hG4bK-p36ko-c0sfk;received=10.40.0.1;rport=5060
From: “87773300792” <sip:87773300792 @ 10.40.0.1:5060;user=phone>;tag=gfhospxx0t
To: “7232492078” <sip:7232492078 @ 10.40.0.1:5060;user=phone>;tag=as1c8a3329
Call-ID: 4662966fipd390bpl @ 10.40.0.1
CSeq: 137 INVITE
Server: FPBX-16.0.33(18.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:7232492078 @ 10.224.33.10:5060>
Content-Type: application/sdp
Content-Length: 276

v=0
o=root 1696347143 1696347143 IN IP4 10.224.33.10
s=Asterisk PBX 18.16.0
c=IN IP4 10.224.33.10
t=0 0
m=audio 18300 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
Audio is at 18872
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.1.141:5060:
INVITE sip:707 @ 192.168.1.141:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.120:5060;branch=z9hG4bK34ff4a2a;rport
Max-Forwards: 70
From: “87773300792” <sip:87773300792 @ 192.168.1.120>;tag=as5def761e
To: <sip:707 @ 192.168.1.141:5060>
Contact: <sip:87773300792 @ 192.168.1.120:5060>
Call-ID: 08944d22387d40a3499258a478837dfb @ 192.168.1.120:5060
CSeq: 102 INVITE
User-Agent: FPBX-16.0.33(18.16.0)
Date: Thu, 25 Apr 2024 16:14:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: “87773300792” <sip:87773300792 @ 192.168.1.120>
Content-Type: application/sdp
Content-Length: 278

v=0
o=root 1572244175 1572244175 IN IP4 192.168.1.120
s=Asterisk PBX 18.16.0
c=IN IP4 192.168.1.120
t=0 0
m=audio 18872 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— SIP read from UDP:192.168.1.141:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.120:5060;branch=z9hG4bK34ff4a2a;rport=5060
From: “87773300792” <sip:87773300792 @ 192.168.1.120>;tag=as5def761e
To: <sip:707 @ 192.168.1.141:5060>
Call-ID: 08944d22387d40a3499258a478837dfb @ 192.168.1.120:5060
CSeq: 102 INVITE
Supported: replaces, path, timer
User-Agent: Grandstream GXP1625 1.0.7.13
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (10 headers 0 lines) —

<— SIP read from UDP:192.168.1.141:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.120:5060;branch=z9hG4bK34ff4a2a;rport=5060
From: “87773300792” <sip:87773300792 @ 192.168.1.120>;tag=as5def761e
To: <sip:707 @ 192.168.1.141:5060>;tag=616664531
Call-ID: 08944d22387d40a3499258a478837dfb @ 192.168.1.120:5060
CSeq: 102 INVITE
Contact: <sip:707 @ 192.168.1.141:5060>
Supported: replaces, path, timer
User-Agent: Grandstream GXP1625 1.0.7.13
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (12 headers 0 lines) —
sip_route_dump: route/path hop: <sip:707 @ 192.168.1.141:5060>
Reliably Transmitting (NAT) to 192.168.1.124:5060:
OPTIONS sip:214 @ 192.168.1.124:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.120:5060;branch=z9hG4bK143a407f;rport
Max-Forwards: 70
From: “Unknown” <sip:Unknown @ 192.168.1.120>;tag=as7a53d8b6
To: <sip:214 @ 192.168.1.124:5060>
Contact: <sip:Unknown @ 192.168.1.120:5060>
Call-ID: 4100287f2e4bf40e7c9f38e3783efffc @ 192.168.1.120:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-16.0.33(18.16.0)
Date: Thu, 25 Apr 2024 16:14:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.1.124:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.120:5060;branch=z9hG4bK143a407f;rport=5060
From: “Unknown” <sip:Unknown @ 192.168.1.120>;tag=as7a53d8b6
To: <sip:214 @ 192.168.1.124:5060>;tag=1036342107
Call-ID: 4100287f2e4bf40e7c9f38e3783efffc @ 192.168.1.120:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1625 1.0.7.13
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘4100287f2e4bf40e7c9f38e3783efffc @ 192.168.1.120:5060’ Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.1.131:5060:
OPTIONS sip:305 @ 192.168.1.131:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.120:5060;branch=z9hG4bK6cea0fa3;rport
Max-Forwards: 70
From: “Unknown” <sip:Unknown @ 192.168.1.120>;tag=as11acc4db
To: <sip:305 @ 192.168.1.131:5060>
Contact: <sip:Unknown @ 192.168.1.120:5060>
Call-ID: 16974ec2613428d2074a55e6498385be @ 192.168.1.120:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-16.0.33(18.16.0)
Date: Thu, 25 Apr 2024 16:14:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.1.131:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.120:5060;branch=z9hG4bK6cea0fa3;rport=5060
From: “Unknown” <sip:Unknown @ 192.168.1.120>;tag=as11acc4db
To: <sip:305 @ 192.168.1.131:5060>;tag=1979425374
Call-ID: 16974ec2613428d2074a55e6498385be @ 192.168.1.120:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1625 1.0.7.13
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘16974ec2613428d2074a55e6498385be @ 192.168.1.120:5060’ Method: OPTIONS

<— SIP read from UDP:192.168.1.141:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.120:5060;branch=z9hG4bK34ff4a2a;rport=5060
From: “87773300792” <sip:87773300792 @ 192.168.1.120>;tag=as5def761e
To: <sip:707 @ 192.168.1.141:5060>;tag=616664531
Call-ID: 08944d22387d40a3499258a478837dfb @ 192.168.1.120:5060
CSeq: 102 INVITE
Contact: <sip:707 @ 192.168.1.141:5060>
Supported: replaces, path, timer
User-Agent: Grandstream GXP1625 1.0.7.13
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 236

v=0
o=707 8000 8000 IN IP4 192.168.1.141
s=SIP Call
c=IN IP4 192.168.1.141
t=0 0
m=audio 5056 RTP/AVP 0 8 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (12 headers 12 lines) —
Got SDP version 8000 and unique parts [707 8000 IN IP4 192.168.1.141]
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.141:5056
sip_route_dump: route/path hop: <sip:707 @ 192.168.1.141:5060>
Transmitting (NAT) to 192.168.1.141:5060:
ACK sip:707 @ 192.168.1.141:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.120:5060;branch=z9hG4bK15e9c626;rport
Max-Forwards: 70
From: “87773300792” <sip:87773300792 @ 192.168.1.120>;tag=as5def761e
To: <sip:707 @ 192.168.1.141:5060>;tag=616664531
Contact: <sip:87773300792 @ 192.168.1.120:5060>
Call-ID: 08944d22387d40a3499258a478837dfb @ 192.168.1.120:5060
CSeq: 102 ACK
User-Agent: FPBX-16.0.33(18.16.0)
Content-Length: 0

Audio is at 18300
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 10.40.0.1:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.40.0.1:5060;branch=z9hG4bK-p36ko-c0sfk;received=10.40.0.1;rport=5060
From: “87773300792” <sip:87773300792 @ 10.40.0.1:5060;user=phone>;tag=gfhospxx0t
To: “7232492078” <sip:7232492078 @ 10.40.0.1:5060;user=phone>;tag=as1c8a3329
Call-ID: 4662966fipd390bpl @ 10.40.0.1
CSeq: 137 INVITE
Server: FPBX-16.0.33(18.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:7232492078 @ 10.224.33.10:5060>
Content-Type: application/sdp
Content-Length: 276

v=0
o=root 1696347143 1696347143 IN IP4 10.224.33.10
s=Asterisk PBX 18.16.0
c=IN IP4 10.224.33.10
t=0 0
m=audio 18300 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>

<— SIP read from UDP:10.40.0.1:5060 —>
ACK sip:7232492078 @ 10.224.33.10:5060 SIP/2.0
Call-ID: 4662966fipd390bpl @ 10.40.0.1
CSeq: 137 ACK
From: “87773300792” <sip:87773300792 @ 10.40.0.1:5060;user=phone>;tag=gfhospxx0t
To: “7232492078” <sip:7232492078 @ 10.40.0.1:5060;user=phone>;tag=as1c8a3329
Via: SIP/2.0/UDP 10.40.0.1:5060;branch=z9hG4bK-ucc6t-9ludx
Max-Forwards: 70
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:10.40.0.1:5060 —>
INVITE sip:7232492078 @ 10.224.33.10:5060 SIP/2.0
Accept: application/sdp
Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS,REGISTER,REFER,SUBSCRIBE,MESSAGE,PUBLISH
Call-ID: 4662966fipd390bpl @ 10.40.0.1
Contact: “87773300792” <sip:87773300792 @ 10.40.0.1:5060;transport=UDP;user=phone>
CSeq: 138 INVITE
Expires: 3600
From: “87773300792” <sip:87773300792 @ 10.40.0.1:5060;user=phone>;tag=gfhospxx0t
To: “7232492078” <sip:7232492078 @ 10.40.0.1:5060;user=phone>;tag=as1c8a3329
Organization: IskraTel
User-Agent: SI3000
Via: SIP/2.0/UDP 10.40.0.1:5060;branch=z9hG4bK-shkdr-n8bsh
Max-Forwards: 70
Subject: Call from CS6111
Content-Length: 211
Content-Type: application/sdp
Content-Disposition: session;handling=required

v=0
o=- 2127603 7513106 IN IP4 10.40.0.8
s=-
c=IN IP4 10.40.0.8
b=AS:64
t=0 0
m=audio 17954 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
— (17 headers 12 lines) —
Sending to 10.40.0.1:5060 (NAT)
Comparing SDP version 7513105 → 7513106 and unique parts [- 2127603 IN IP4 10.40.0.8] → [- 2127603 IN IP4 10.40.0.8]
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.40.0.8:17954

<— Transmitting (NAT) to 10.40.0.1:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.40.0.1:5060;branch=z9hG4bK-shkdr-n8bsh;received=10.40.0.1;rport=5060
From: “87773300792” <sip:87773300792 @ 10.40.0.1:5060;user=phone>;tag=gfhospxx0t
To: “7232492078” <sip:7232492078 @ 10.40.0.1:5060;user=phone>;tag=as1c8a3329
Call-ID: 4662966fipd390bpl @ 10.40.0.1
CSeq: 138 INVITE
Server: FPBX-16.0.33(18.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:7232492078 @ 10.224.33.10:5060>
Content-Length: 0

<------------>
Audio is at 18300
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 10.40.0.1:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.40.0.1:5060;branch=z9hG4bK-shkdr-n8bsh;received=10.40.0.1;rport=5060
From: “87773300792” <sip:87773300792 @ 10.40.0.1:5060;user=phone>;tag=gfhospxx0t
To: “7232492078” <sip:7232492078 @ 10.40.0.1:5060;user=phone>;tag=as1c8a3329
Call-ID: 4662966fipd390bpl @ 10.40.0.1
CSeq: 138 INVITE
Server: FPBX-16.0.33(18.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:7232492078 @ 10.224.33.10:5060>
Content-Type: application/sdp
Content-Length: 252

v=0
o=root 1696347143 1696347144 IN IP4 10.224.33.10
s=Asterisk PBX 18.16.0
c=IN IP4 10.224.33.10
t=0 0
m=audio 18300 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>

<— SIP read from UDP:10.40.0.1:5060 —>
ACK sip:7232492078 @ 10.224.33.10:5060 SIP/2.0
Call-ID: 4662966fipd390bpl @ 10.40.0.1
CSeq: 138 ACK
From: “87773300792” <sip:87773300792 @ 10.40.0.1:5060;user=phone>;tag=gfhospxx0t
To: “7232492078” <sip:7232492078 @ 10.40.0.1:5060;user=phone>;tag=as1c8a3329
Via: SIP/2.0/UDP 10.40.0.1:5060;branch=z9hG4bK-el5yh-km7i3
Max-Forwards: 70
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Reliably Transmitting (NAT) to 192.168.1.125:5060:
OPTIONS sip:215 @ 192.168.1.125:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.120:5060;branch=z9hG4bK420bbb8e;rport
Max-Forwards: 70
From: “Unknown” <sip:Unknown @ 192.168.1.120>;tag=as2e3ceea7
To: <sip:215 @ 192.168.1.125:5060>
Contact: <sip:Unknown @ 192.168.1.120:5060>
Call-ID: 146de6f527a569697ebea6b42cd4a7d2 @ 192.168.1.120:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-16.0.33(18.16.0)
Date: Thu, 25 Apr 2024 16:14:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.1.125:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.120:5060;branch=z9hG4bK420bbb8e;rport=5060
From: “Unknown” <sip:Unknown @ 192.168.1.120>;tag=as2e3ceea7
To: <sip:215 @ 192.168.1.125:5060>;tag=1230339868
Call-ID: 146de6f527a569697ebea6b42cd4a7d2 @ 192.168.1.120:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1625 1.0.7.13
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘146de6f527a569697ebea6b42cd4a7d2 @ 192.168.1.120:5060’ Method: OPTIONS

<— SIP read from UDP:192.168.1.141:5060 —>
BYE sip:87773300792 @ 192.168.1.120:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.141:5060;branch=z9hG4bK633771940;rport
From: <sip:707 @ 192.168.1.141:5060>;tag=616664531
To: “87773300792” <sip:87773300792 @ 192.168.1.120>;tag=as5def761e
Call-ID: 08944d22387d40a3499258a478837dfb @ 192.168.1.120:5060
CSeq: 103 BYE
Contact: <sip:707 @ 192.168.1.141:5060>
Max-Forwards: 70
Supported: replaces, path, timer
User-Agent: Grandstream GXP1625 1.0.7.13
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 192.168.1.141:5060 (NAT)
Scheduling destruction of SIP dialog ‘08944d22387d40a3499258a478837dfb @ 192.168.1.120:5060’ in 6400 ms (Method: BYE)

<— Transmitting (NAT) to 192.168.1.141:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.141:5060;branch=z9hG4bK633771940;received=192.168.1.141;rport=5060
From: <sip:707 @ 192.168.1.141:5060>;tag=616664531
To: “87773300792” <sip:87773300792 @ 192.168.1.120>;tag=as5def761e
Call-ID: 08944d22387d40a3499258a478837dfb @ 192.168.1.120:5060
CSeq: 103 BYE
Server: FPBX-16.0.33(18.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘4662966fipd390bpl @ 10.40.0.1’ in 6400 ms (Method: ACK)
Reliably Transmitting (NAT) to 10.40.0.1:5060:
BYE sip:87773300792 @ 10.40.0.1:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.224.33.10:5060;branch=z9hG4bK1ca9c3c3;rport
Max-Forwards: 70
From: “7232492078” <sip:7232492078 @ 10.40.0.1:5060;user=phone>;tag=as1c8a3329
To: “87773300792” <sip:87773300792 @ 10.40.0.1:5060;user=phone>;tag=gfhospxx0t
Call-ID: 4662966fipd390bpl @ 10.40.0.1
CSeq: 102 BYE
User-Agent: FPBX-16.0.33(18.16.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<— SIP read from UDP:10.40.0.1:5060 —>
SIP/2.0 200 OK
Call-ID: 4662966fipd390bpl @ 10.40.0.1
CSeq: 102 BYE
From: “7232492078” <sip:7232492078 @ 10.40.0.1:5060;user=phone>;tag=as1c8a3329
To: “87773300792” <sip:87773300792 @ 10.40.0.1:5060;user=phone>;tag=gfhospxx0t
Via: SIP/2.0/UDP 10.224.33.10:5060;received=10.224.33.10;branch=z9hG4bK1ca9c3c3;rport=5060
Content-Length: 0

<------------->
— (7 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘4662966fipd390bpl @ 10.40.0.1’ Method: ACK
freepbx*CLI> sip set debug off
SIP Debugging Disabled

4

What’s the difference between 10.40.0.1 and 10.40.0.8?

You are sending audio to 10.40.0.8, as instructed and you have told them to send 10.224.33.10, which appears to be valid for the signalling, from 10.40.0.1, but you are not getting any incoming audio, on that side. Is any firewall open for the correct port range. It needs to be open for 18300/UDP in the second trace. One can’t tell what was used in the first case, but generally the range is 10,000 to 20,000.

5

My Ip is 10.224.33.10, directly connected to operator equipment 10.224.33.9.
10.40.0.1 is operator PBX. Any other ip in the subnet 10.40.0.0/24 I dont know.

6

You will need a route between 10.40.0.0/24 and whatever subnet 10.224.33.10/9 are in. (So the question might be better formatted as 'what is the subnet of 10.224.33.* and how is it physically associated with 10.40.0.0/24

7

there is nothing wrong with the route, because trunk registration is taking place.

judging by the provider’s response 10.40.0.8 is the IP address of the signal board through which the connection occurs.

Perhaps there is a problem with the peer settings
In logs I have

From: “87773300792” <sip:87773300792 @ 10.40.0.1:5060;user=phone>;tag=gfhospxx0t
To: “7232492078” <sip:7232492078 @ 10.40.0.1:5060;user=phone>

But as I think it have to be like:

From: “87773300792” <sip:87773300792 @ 10.40.0.1:5060;user=phone>;tag=gfhospxx0t
To: “7232492078” <sip:7232492078 @ 10.224.33.10:5060;user=phone>

couse peer 7232492078 located at my PBX with ip 10.224.33.10

8

This is not a problem, because Asterisk ignores the domain part of the To header.

I suspect that audio to the provider’s media server at 10.40.0.8 may not be routed correctly.

At a shell prompt, type
ip route
and post the output.

1 Like
9

You are exactly right…
The route was only for host 10.40.0.1, not for subnet

route add 10.40.0.0/24 via 10.224.33.9 - solved the problem…

Thank you all!!!

10

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