One way audio, no NAT

Furious_George · September 24, 2019, 12:49am

I’m using a Grandstream HT813, which includes an FXO port. Irrespective of what client I use for the other leg, called parties on the PSTN cannot hear callers.

My sip.conf settings are pretty standard:

[102]
type=friend
context=door                     ; Where to start in the dialplan when this phone calls
secret=shhhhh
host=dynamic
callerid=John Doe <102>        ; Full caller ID, to override the phones config
dtmfmode=inband                  ; either RFC2833 or INFO for the BudgeTone
call-limit=1                     ; permit only 1 outgoing call and 1 incoming call at a time
disallow=all                     ; need to disallow=all before we can use allow=
allow=ulaw                       ; Note: In user sections the order of codecs

[99]
type=peer
context=door                     ; Where to start in the dialplan when this phone calls
qualify=yes
secret=NEClamChowda
host=192.168.1.11
port=5062
insecure=port
dtmfmode=inband                  ; either RFC2833 or INFO for the BudgeTone
call-limit=1                     ; permit only 1 outgoing call and 1 incoming call at a time
disallow=all                     ; need to disallow=all before we can use allow=
allow=ulaw                       ; Note: In user sections the order of codecs

99 is the FXO gateway, and 102 is x-lite, which I’m using to test. In addition, I’ve also defined localnet. The rest of sip.conf is pretty much at the default.

Everything looks OK when attempting a call:

Using SIP RTP CoS mark 5
       > 0x9c90040 -- Strict RTP learning after remote address set to: 192.168.1.152:63786
    -- Executing [106@door:1] Dial("SIP/102-000000db", "sip/18882874637@192.168.1.11:5062,45") in new stack
  == Using SIP RTP CoS mark 5
    -- Called sip/18882874637@192.168.1.11:5062
    -- SIP/192.168.1.11:5062-000000dc is ringing
       > 0xb6e10230 -- Strict RTP learning after remote address set to: 192.168.1.11:5012
    -- SIP/192.168.1.11:5062-000000dc answered SIP/102-000000db
    -- Channel SIP/192.168.1.11:5062-000000dc joined 'simple_bridge' basic-bridge <aa0bd643-7219-4863-b729-fcb5f04a8d13>
    -- Channel SIP/102-000000db joined 'simple_bridge' basic-bridge <aa0bd643-7219-4863-b729-fcb5f04a8d13>
       > 0xb6e10230 -- Strict RTP switching to RTP target address 192.168.1.11:5012 as source
       > 0xb6e10230 -- Strict RTP learning complete - Locking on source address 192.168.1.11:5012
    -- Channel SIP/102-000000db left 'simple_bridge' basic-bridge <aa0bd643-7219-4863-b729-fcb5f04a8d13>
    -- Channel SIP/192.168.1.11:5062-000000dc left 'simple_bridge' basic-bridge <aa0bd643-7219-4863-b729-fcb5f04a8d13>
  == Spawn extension (door, 106, 1) exited non-zero on 'SIP/102-000000db'

For this call, the gateway is on 192.168.1.11, and listening on 5062. X-lite is on 192.168.1.152. RTP is being sent to 5012, which is correct, according to the port settings for the FXO.

I made sure that ulaw/pcmu is enabled and preferred on both clients. X-Lite is basically at the default settings, aside from the account I’ve added, and, in addition, I’ve disabled all non-ulaw codecs. Unfortunately, there’s no human-readable way to export the FXO’s settings, so I’ve taken some screenshots of the FXO section:

My assumption is that the problem lies somewhere in there.

Lastly, here’s a sip dump of a call:

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

<--- SIP read from UDP:192.168.1.152:50315 --->
INVITE sip:106@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.152:50315;branch=z9hG4bK-524287-1---25b91a459844d907;rport
Max-Forwards: 70
Contact: <sip:102@192.168.1.152:50315;rinstance=97600bd6c85b065a>
To: <sip:106@192.168.1.10>
From: <sip:102@192.168.1.10>;tag=7350dd1f
Call-ID: 99142Y2QwMWFkZjhkNmMyYmIwMDI5NGRmZDJmZTMxYjhiMDQ
CSeq: 1 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 5.6.1 stamp 99142
Content-Length: 155

v=0
o=- 13213763185156838 1 IN IP4 192.168.1.152
s=X-Lite release 5.6.1 stamp 99142
c=IN IP4 192.168.1.152
t=0 0
m=audio 59010 RTP/AVP 0
a=sendrecv
<------------->
--- (13 headers 7 lines) ---
Sending to 192.168.1.152:50315 (no NAT)
Sending to 192.168.1.152:50315 (no NAT)
Using INVITE request as basis request - 99142Y2QwMWFkZjhkNmMyYmIwMDI5NGRmZDJmZTMxYjhiMDQ
Found peer '102' for '102' from 192.168.1.152:50315

<--- Reliably Transmitting (no NAT) to 192.168.1.152:50315 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.152:50315;branch=z9hG4bK-524287-1---25b91a459844d907;received=192.168.1.152;rport=50315
From: <sip:102@192.168.1.10>;tag=7350dd1f
To: <sip:106@192.168.1.10>;tag=as49225482
Call-ID: 99142Y2QwMWFkZjhkNmMyYmIwMDI5NGRmZDJmZTMxYjhiMDQ
CSeq: 1 INVITE
Server: Asterisk PBX 13.28.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="19cf8217"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '99142Y2QwMWFkZjhkNmMyYmIwMDI5NGRmZDJmZTMxYjhiMDQ' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.152:50315 --->
ACK sip:106@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.152:50315;branch=z9hG4bK-524287-1---25b91a459844d907;rport
Max-Forwards: 70
To: <sip:106@192.168.1.10>;tag=as49225482
From: <sip:102@192.168.1.10>;tag=7350dd1f
Call-ID: 99142Y2QwMWFkZjhkNmMyYmIwMDI5NGRmZDJmZTMxYjhiMDQ
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.152:50315 --->
INVITE sip:106@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.152:50315;branch=z9hG4bK-524287-1---4099b726ccdb345f;rport
Max-Forwards: 70
Contact: <sip:102@192.168.1.152:50315;rinstance=97600bd6c85b065a>
To: <sip:106@192.168.1.10>
From: <sip:102@192.168.1.10>;tag=7350dd1f
Call-ID: 99142Y2QwMWFkZjhkNmMyYmIwMDI5NGRmZDJmZTMxYjhiMDQ
CSeq: 2 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 5.6.1 stamp 99142
Authorization: Digest username="102",realm="asterisk",nonce="19cf8217",uri="sip:106@192.168.1.10",response="b71e8d49bd77649b3cfd1adbeb127f59",algorithm=MD5
Content-Length: 155

v=0
o=- 13213763185156838 1 IN IP4 192.168.1.152
s=X-Lite release 5.6.1 stamp 99142
c=IN IP4 192.168.1.152
t=0 0
m=audio 59010 RTP/AVP 0
a=sendrecv
<------------->
--- (14 headers 7 lines) ---
Sending to 192.168.1.152:50315 (no NAT)
Using INVITE request as basis request - 99142Y2QwMWFkZjhkNmMyYmIwMDI5NGRmZDJmZTMxYjhiMDQ
Found peer '102' for '102' from 192.168.1.152:50315
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
       > 0x9c90040 -- Strict RTP learning after remote address set to: 192.168.1.152:59010
Peer audio RTP is at port 192.168.1.152:59010
Looking for 106 in door (domain 192.168.1.10)
sip_route_dump: route/path hop: <sip:102@192.168.1.152:50315;rinstance=97600bd6c85b065a>

<--- Transmitting (no NAT) to 192.168.1.152:50315 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.152:50315;branch=z9hG4bK-524287-1---4099b726ccdb345f;received=192.168.1.152;rport=50315
From: <sip:102@192.168.1.10>;tag=7350dd1f
To: <sip:106@192.168.1.10>
Call-ID: 99142Y2QwMWFkZjhkNmMyYmIwMDI5NGRmZDJmZTMxYjhiMDQ
CSeq: 2 INVITE
Server: Asterisk PBX 13.28.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:106@192.168.1.10:5060>
Content-Length: 0


<------------>
    -- Executing [106@door:1] Dial("SIP/102-000000dd", "sip/18882874637@192.168.1.11:5062,45") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 10378
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.11:5062:
INVITE sip:18882874637@192.168.1.11:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47583d0a
Max-Forwards: 70
From: "John Doe" <sip:102@192.168.1.10>;tag=as520a1b24
To: <sip:18882874637@192.168.1.11:5062>
Contact: <sip:102@192.168.1.10:5060>
Call-ID: 0202d55467ced0460287036b7afb56c9@192.168.1.10:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.28.1
Date: Tue, 24 Sep 2019 00:45:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 2003094717 2003094717 IN IP4 192.168.1.10
s=Asterisk PBX 13.28.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 10378 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

---
    -- Called sip/18882874637@192.168.1.11:5062

<--- SIP read from UDP:192.168.1.11:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47583d0a;rport=5060
From: "John Doe" <sip:102@192.168.1.10>;tag=as520a1b24
To: <sip:18882874637@192.168.1.11:5062>
Call-ID: 0202d55467ced0460287036b7afb56c9@192.168.1.10:5060
CSeq: 102 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT813 1.0.1.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Reliably Transmitting (no NAT) to 192.168.1.11:5062:
OPTIONS sip:99@192.168.1.11:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK0ac4b484
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.10>;tag=as66397b34
To: <sip:99@192.168.1.11:5062>
Contact: <sip:asterisk@192.168.1.10:5060>
Call-ID: 55ad552432bd1a0d5fcfb0df583594ba@192.168.1.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.28.1
Date: Tue, 24 Sep 2019 00:45:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.11:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK0ac4b484;rport=5060
From: "asterisk" <sip:asterisk@192.168.1.10>;tag=as66397b34
To: <sip:99@192.168.1.11:5062>;tag=566461090
Call-ID: 55ad552432bd1a0d5fcfb0df583594ba@192.168.1.10:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT813 1.0.1.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '55ad552432bd1a0d5fcfb0df583594ba@192.168.1.10:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.1.152:5060 --->


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

<--- SIP read from UDP:192.168.1.11:5062 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47583d0a;rport=5060
From: "John Doe" <sip:102@192.168.1.10>;tag=as520a1b24
To: <sip:18882874637@192.168.1.11:5062>;tag=1498542591
Call-ID: 0202d55467ced0460287036b7afb56c9@192.168.1.10:5060
CSeq: 102 INVITE
Contact: <sip:192.168.1.11:5062>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT813 1.0.1.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:192.168.1.11:5062>
    -- SIP/192.168.1.11:5062-000000de is ringing

<--- Transmitting (no NAT) to 192.168.1.152:50315 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.152:50315;branch=z9hG4bK-524287-1---4099b726ccdb345f;received=192.168.1.152;rport=50315
From: <sip:102@192.168.1.10>;tag=7350dd1f
To: <sip:106@192.168.1.10>;tag=as09e8aeb5
Call-ID: 99142Y2QwMWFkZjhkNmMyYmIwMDI5NGRmZDJmZTMxYjhiMDQ
CSeq: 2 INVITE
Server: Asterisk PBX 13.28.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:106@192.168.1.10:5060>
Content-Length: 0


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

<--- SIP read from UDP:192.168.1.11:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK47583d0a;rport=5060
From: "John Doe" <sip:102@192.168.1.10>;tag=as520a1b24
To: <sip:18882874637@192.168.1.11:5062>;tag=1498542591
Call-ID: 0202d55467ced0460287036b7afb56c9@192.168.1.10:5060
CSeq: 102 INVITE
Contact: <sip:192.168.1.11:5062>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT813 1.0.1.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 241

v=0
o=- 8002 8000 IN IP4 192.168.1.11
s=SIP Call
c=IN IP4 192.168.1.11
t=0 0
m=audio 5012 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 0-16,32-36,54
<------------->
--- (12 headers 12 lines) ---
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|gsm|h263), 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|)
       > 0x9c922c0 -- Strict RTP learning after remote address set to: 192.168.1.11:5012
Peer audio RTP is at port 192.168.1.11:5012
sip_route_dump: route/path hop: <sip:192.168.1.11:5062>
set_destination: Parsing <sip:192.168.1.11:5062> for address/port to send to
set_destination: set destination to 192.168.1.11:5062
Transmitting (no NAT) to 192.168.1.11:5062:
ACK sip:192.168.1.11:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK1a913ab8
Max-Forwards: 70
From: "John Doe" <sip:102@192.168.1.10>;tag=as520a1b24
To: <sip:18882874637@192.168.1.11:5062>;tag=1498542591
Contact: <sip:102@192.168.1.10:5060>
Call-ID: 0202d55467ced0460287036b7afb56c9@192.168.1.10:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.28.1
Content-Length: 0


---
    -- SIP/192.168.1.11:5062-000000de answered SIP/102-000000dd
Audio is at 17496
Adding codec ulaw to SDP

<--- Reliably Transmitting (no NAT) to 192.168.1.152:50315 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.152:50315;branch=z9hG4bK-524287-1---4099b726ccdb345f;received=192.168.1.152;rport=50315
From: <sip:102@192.168.1.10>;tag=7350dd1f
To: <sip:106@192.168.1.10>;tag=as09e8aeb5
Call-ID: 99142Y2QwMWFkZjhkNmMyYmIwMDI5NGRmZDJmZTMxYjhiMDQ
CSeq: 2 INVITE
Server: Asterisk PBX 13.28.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:106@192.168.1.10:5060>
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 32151317 32151317 IN IP4 192.168.1.10
s=Asterisk PBX 13.28.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 17496 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=maxptime:150
a=sendrecv

<------------>
    -- Channel SIP/192.168.1.11:5062-000000de joined 'simple_bridge' basic-bridge <97e4e4d3-4a8e-481b-8b0c-8b51acb9a4f6>
    -- Channel SIP/102-000000dd joined 'simple_bridge' basic-bridge <97e4e4d3-4a8e-481b-8b0c-8b51acb9a4f6>
       > 0x9c922c0 -- Strict RTP switching to RTP target address 192.168.1.11:5012 as source

<--- SIP read from UDP:192.168.1.152:50315 --->
ACK sip:106@192.168.1.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.152:50315;branch=z9hG4bK-524287-1---9c59b75f7487d842;rport
Max-Forwards: 70
Contact: <sip:102@192.168.1.152:50315;rinstance=97600bd6c85b065a>
To: <sip:106@192.168.1.10>;tag=as09e8aeb5
From: <sip:102@192.168.1.10>;tag=7350dd1f
Call-ID: 99142Y2QwMWFkZjhkNmMyYmIwMDI5NGRmZDJmZTMxYjhiMDQ
CSeq: 2 ACK
User-Agent: X-Lite release 5.6.1 stamp 99142
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
       > 0x9c922c0 -- Strict RTP learning complete - Locking on source address 192.168.1.11:5012

<--- SIP read from UDP:192.168.1.152:5060 --->


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

<--- SIP read from UDP:192.168.1.152:50315 --->


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

<--- SIP read from UDP:192.168.1.152:50315 --->
BYE sip:106@192.168.1.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.152:50315;branch=z9hG4bK-524287-1---1d6f5f77e89fda39;rport
Max-Forwards: 70
Contact: <sip:102@192.168.1.152:50315;rinstance=97600bd6c85b065a>
To: <sip:106@192.168.1.10>;tag=as09e8aeb5
From: <sip:102@192.168.1.10>;tag=7350dd1f
Call-ID: 99142Y2QwMWFkZjhkNmMyYmIwMDI5NGRmZDJmZTMxYjhiMDQ
CSeq: 3 BYE
User-Agent: X-Lite release 5.6.1 stamp 99142
Authorization: Digest username="102",realm="asterisk",nonce="19cf8217",uri="sip:106@192.168.1.10:5060",response="6d3c190457bf91ef2668a2f4ce03471f",algorithm=MD5
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.1.152:50315 (no NAT)
Scheduling destruction of SIP dialog '99142Y2QwMWFkZjhkNmMyYmIwMDI5NGRmZDJmZTMxYjhiMDQ' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.1.152:50315 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.152:50315;branch=z9hG4bK-524287-1---1d6f5f77e89fda39;received=192.168.1.152;rport=50315
From: <sip:102@192.168.1.10>;tag=7350dd1f
To: <sip:106@192.168.1.10>;tag=as09e8aeb5
Call-ID: 99142Y2QwMWFkZjhkNmMyYmIwMDI5NGRmZDJmZTMxYjhiMDQ
CSeq: 3 BYE
Server: Asterisk PBX 13.28.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
    -- Channel SIP/102-000000dd left 'simple_bridge' basic-bridge <97e4e4d3-4a8e-481b-8b0c-8b51acb9a4f6>
  == Spawn extension (door, 106, 1) exited non-zero on 'SIP/102-000000dd'
    -- Channel SIP/192.168.1.11:5062-000000de left 'simple_bridge' basic-bridge <97e4e4d3-4a8e-481b-8b0c-8b51acb9a4f6>
Scheduling destruction of SIP dialog '0202d55467ced0460287036b7afb56c9@192.168.1.10:5060' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:192.168.1.11:5062> for address/port to send to
set_destination: set destination to 192.168.1.11:5062
Reliably Transmitting (no NAT) to 192.168.1.11:5062:
BYE sip:192.168.1.11:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK59b4b4e7
Max-Forwards: 70
From: "John Doe" <sip:102@192.168.1.10>;tag=as520a1b24
To: <sip:18882874637@192.168.1.11:5062>;tag=1498542591
Call-ID: 0202d55467ced0460287036b7afb56c9@192.168.1.10:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 13.28.1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.11:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK59b4b4e7;rport=5060
From: "John Doe" <sip:102@192.168.1.10>;tag=as520a1b24
To: <sip:18882874637@192.168.1.11:5062>;tag=1498542591
Call-ID: 0202d55467ced0460287036b7afb56c9@192.168.1.10:5060
CSeq: 103 BYE
Contact: <sip:192.168.1.11:5062>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT813 1.0.1.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '0202d55467ced0460287036b7afb56c9@192.168.1.10:5060' Method: INVITE

<--- SIP read from UDP:192.168.1.152:5060 --->


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

<--- SIP read from UDP:192.168.1.152:5060 --->


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

<--- SIP read from UDP:192.168.1.152:5060 --->


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

<--- SIP read from UDP:192.168.1.152:50315 --->


<------------->
Really destroying SIP dialog '99142Y2QwMWFkZjhkNmMyYmIwMDI5NGRmZDJmZTMxYjhiMDQ' Method: BYE

<--- SIP read from UDP:192.168.1.152:5060 --->


<------------->
Reliably Transmitting (no NAT) to 192.168.1.11:5062:
OPTIONS sip:99@192.168.1.11:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK21d48fc0
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.1.10>;tag=as3c406e07
To: <sip:99@192.168.1.11:5062>
Contact: <sip:asterisk@192.168.1.10:5060>
Call-ID: 4e7c2bc1056354c10c3198222f4e52d5@192.168.1.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.28.1
Date: Tue, 24 Sep 2019 00:46:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.11:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK21d48fc0;rport=5060
From: "asterisk" <sip:asterisk@192.168.1.10>;tag=as3c406e07
To: <sip:99@192.168.1.11:5062>;tag=1007988402
Call-ID: 4e7c2bc1056354c10c3198222f4e52d5@192.168.1.10:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT813 1.0.1.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '4e7c2bc1056354c10c3198222f4e52d5@192.168.1.10:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.1.152:5060 --->

For reference:

192.168.1.10 is the asterisk server, listening on 5060
192.168.1.11 is the FXO, listening on 5062
192,168,1,152 is X-Lite

Thanks in advance.

penguinpbx · September 24, 2019, 6:02pm

On the Asterisk CLI, can you try “rtp set debug on” and watching for packets / posting that log ?

Furious_George · September 24, 2019, 6:51pm

Hi, and thanks for the reply.

The output of RTP debug appears normal.


ent RTP packet to      192.168.1.152:57568 (type 00, seq 021124, ts 1586944944, len 000160)
Got  RTP packet from    192.168.1.11:5012 (type 00, seq 006779, ts 1586945105, len 000160)
Sent RTP packet to      192.168.1.152:57568 (type 00, seq 021125, ts 1586945104, len 000160)
Got  RTP packet from    192.168.1.11:5012 (type 00, seq 006780, ts 1586945265, len 000160)
Sent RTP packet to      192.168.1.152:57568 (type 00, seq 021126, ts 1586945264, len 000160)
Got  RTP packet from    192.168.1.11:5012 (type 00, seq 006781, ts 1586945425, len 000160)
Sent RTP packet to      192.168.1.152:57568 (type 00, seq 021127, ts 1586945424, len 000160)
Got  RTP packet from    192.168.1.11:5012 (type 00, seq 006782, ts 1586945585, len 000160)
Sent RTP packet to      192.168.1.152:57568 (type 00, seq 021128, ts 1586945584, len 000160)
Got  RTP packet from    192.168.1.11:5012 (type 00, seq 006783, ts 1586945745, len 000160)
Sent RTP packet to      192.168.1.152:57568 (type 00, seq 021129, ts 1586945744, len 000160)
Got  RTP packet from    192.168.1.11:5012 (type 00, seq 006784, ts 1586945905, len 000160)
Sent RTP packet to      192.168.1.152:57568 (type 00, seq 021130, ts 1586945904, len 000160)
Got  RTP packet from    192.168.1.11:5012 (type 00, seq 006785, ts 1586946065, len 000160)
Sent RTP packet to      192.168.1.152:57568 (type 00, seq 021131, ts 1586946064, len 000160)
Got  RTP packet from    192.168.1.11:5012 (type 00, seq 006786, ts 1586946225, len 000160)
Sent RTP packet to      192.168.1.152:57568 (type 00, seq 021132, ts 1586946224, len 000160)
Got  RTP packet from    192.168.1.11:5012 (type 00, seq 006787, ts 1586946385, len 000160)
Sent RTP packet to      192.168.1.152:57568 (type 00, seq 021133, ts 1586946384, len 000160)
Got  RTP packet from    192.168.1.11:5012 (type 00, seq 006788, ts 1586946545, len 000160)
Sent RTP packet to      192.168.1.152:57568 (type 00, seq 021134, ts 1586946544, len 000160)
Got  RTP packet from    192.168.1.11:5012 (type 00, seq 006789, ts 1586946705, len 000160)
Sent RTP packet to      192.168.1.152:57568 (type 00, seq 021135, ts 1586946704, len 000160)
Got  RTP packet from    192.168.1.11:5012 (type 00, seq 006790, ts 1586946865, len 000160)
Sent RTP packet to      192.168.1.152:57568 (type 00, seq 021136, ts 1586946864, len 000160)
Got  RTP packet from    192.168.1.11:5012 (type 00, seq 006791, ts 1586947025, len 000160)
Sent RTP packet to      192.168.1.152:57568 (type 00, seq 021137, ts 1586947024, len 000160)
Got  RTP packet from    192.168.1.11:5012 (type 00, seq 006792, ts 1586947185, len 000160)
Sent RTP packet to      192.168.1.152:57568 (type 00, seq 021138, ts 1586947184, len 000160)
Got  RTP packet from    192.168.1.11:5012 (type 00, seq 006793, ts 1586947345, len 000160)
Sent RTP packet to      192.168.1.152:57568 (type 00, seq 021139, ts 1586947344, len 000160)
Got  RTP packet from    192.168.1.11:5012 (type 00, seq 006794, ts 1586947505, len 000160)
Sent RTP packet to      192.168.1.152:57568 (type 00, seq 021140, ts 1586947504, len 000160)
Got  RTP packet from    192.168.1.11:5012 (type 00, seq 006795, ts 1586947665, len 000160)
Sent RTP packet to      192.168.1.152:57568 (type 00, seq 021141, ts 1586947664, len 000160)
Got  RTP packet from    192.168.1.11:5012 (type 00, seq 006796, ts 1586947825, len 000160)
Sent RTP packet to      192.168.1.152:57568 (type 00, seq 021142, ts 1586947824, len 000160)
    -- Channel SIP/102-000000df left 'simple_bridge' basic-bridge <426a5507-c624-4887-b769-3b483ae82a32>
  == Spawn extension (door, 106, 1) exited non-zero on 'SIP/102-000000df'
    -- Channel SIP/192.168.1.11:5062-000000e0 left 'simple_bridge' basic-bridge <426a5507-c624-4887-b769-3b483ae82a32>

jcolp · September 24, 2019, 7:04pm

X-Lite is not sending audio to Asterisk (or it’s getting blocked), Asterisk is forwarding the media received from the gateway to X-Lite at the IP address and port it gave. Your problem is likely external to Asterisk, be it a firewall or something else.

Furious_George · September 24, 2019, 7:33pm

There’s no microphone attached to x-lite. I’m only trying to use inband DTMF to establish that audio is working in the other direction. I disabled Windows Defender Firewall and tried again, but I saw no change in the RTP output.

(If I set the DTMF mode to SIP INFO, then I do get DTMF on the other side.)

Is it possible that my test – specifically, using inband DTMF from x-lite to establish that there is bidirectional audio – is invalid?

I know that it was not working similarly from the analog phone connected to the FXS on the same HT813, i.e. there was no audio making it out to the called party.

That’s what inspired me to log into a local workstation and test using X-Lite and inband DTMF. When I saw that this was not working, I assumed it was for the same reason that audio was not reaching the called party from the analog phone on the FXS, but perhaps this test is invalid.

jcolp · September 24, 2019, 7:43pm

I can’t speak for X-Lite or its behavior in such circumstances. I can only state what is shown from the RTP debug.

penguinpbx · September 25, 2019, 2:38pm

Are there two instances of X-Lite running on your computer ? Maybe the first one seized your mic so the second has no audio to give to Asterisk ?

benphone · September 25, 2019, 2:54pm

usually this could happen with 2 user on computer using the same ext and not shuting down

Furious_George · September 26, 2019, 3:03am

Benphone and PenguinPBX,

Thanks for the replies. I had linphone running on the same workstation, so not 2 instances of x-lite. Closing linphone results in no change.

I’m just going to have to go out there and try stuff, because the epistemological problems seem functionally insurmountable at this point. I will report back irrespective of outcome.

Meanwhile, thank you guys for following up!

-Brian

system · October 26, 2019, 3:03am

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