Call Disconnecting after 20 secs

pokhraj_d · February 26, 2019, 2:30pm

Hello,
I am facing a situation where the outgoing call has been disconnecting after 20 secs. It is telling “408- Request Timeout”
I am using the below SIP configuration

[general]
context=public
allowguest=no
srvlookup=yes
udpbindaddr=0.0.0.0
tcpenable=no
transport=udp
;nat=force_rport
nat=route
qualify=yes
canreinvite=no
disallow=all
alwaysauthreject = yes
deny=0.0.0.0/0.0.0.0
permit=88.99.xxx.xxx/255.255.255.0
externip=88.99.xxx.xxx [Asterik server IP running on cloud]
localnet=192.168.1.5/255.255.255.0 [My Computer Local Ip address]


[Pokhraj]
;type=friend
type=peer
context=incoming
allow=ulaw,alaw
secret=xxxx
host=dynamic

[Madhumita]
;type=friend
type=peer
context=incoming
allow=ulaw,alaw
secret=xxxx
host=dynamic

Please auggest. Am I performing any wrong configuration at SIP.conf?

pokhraj_d · February 26, 2019, 2:34pm

My SIP Debug is as follow

Reliably Transmitting (no NAT) to 223.191.16.147:7098:
OPTIONS sip:Pokhraj@223.191.16.147:7098;rinstance=fa1038c95bd79575 SIP/2.0
Via: SIP/2.0/UDP 88.99.245.202:5060;branch=z9hG4bK183bf5a0
Max-Forwards: 70
From: "asterisk" <sip:asterisk@88.99.245.202>;tag=as34e6f35b
To: <sip:Pokhraj@223.191.16.147:7098;rinstance=fa1038c95bd79575>
Contact: <sip:asterisk@88.99.245.202:5060>
Call-ID: 659b0395694a3e22528ce20c48131b37@88.99.245.202:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.7.1
Date: Tue, 26 Feb 2019 14:31:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:223.191.16.147:7098 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 88.99.245.202:5060;branch=z9hG4bK183bf5a0
Contact: <sip:192.168.43.100:8000>
To: <sip:Pokhraj@223.191.16.147:7098;rinstance=fa1038c95bd79575>;tag=f84e9303
From: "asterisk" <sip:asterisk@88.99.245.202>;tag=as34e6f35b
Call-ID: 659b0395694a3e22528ce20c48131b37@88.99.245.202:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
Supported: replaces
User-Agent: X-Lite release 5.4.0 stamp 94388
Allow-Events: talk, hold
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '659b0395694a3e22528ce20c48131b37@88.99.245.202:5060' Method: OPTIONS

<--- SIP read from UDP:223.191.16.147:7090 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 88.99.245.202:5060;received=88.99.245.202;branch=z9hG4bK16eb32d3
Call-ID: 0206e7e8511ba41419ec4b373ed97857@88.99.245.202:5060
From: "Pokhraj" <sip:Pokhraj@88.99.245.202>;tag=as696e9844
To: <sip:Madhumita@192.168.43.100;ob>;tag=a50e61b5ebb44f38baf55115d7adf88d
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: <sip:Madhumita@192.168.43.100:50418;ob>
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 322

v=0
o=- 3760200043 3760200044 IN IP4 192.168.43.100
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4008 RTP/AVP 0 101
c=IN IP4 192.168.43.100
b=TIAS:64000
a=rtcp:4009 IN IP4 192.168.43.100
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1170553184 cname:54d60ea93f0b3087
<------------->
--- (11 headers 15 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.43.100:4008
sip_route_dump: route/path hop: <sip:Madhumita@192.168.43.100:50418;ob>
set_destination: Parsing <sip:Madhumita@192.168.43.100:50418;ob> for address/port to send to
set_destination: set destination to 192.168.43.100:50418
Transmitting (no NAT) to 192.168.43.100:50418:
ACK sip:Madhumita@192.168.43.100:50418;ob SIP/2.0
Via: SIP/2.0/UDP 88.99.245.202:5060;branch=z9hG4bK1bf082fa
Max-Forwards: 70
From: "Pokhraj" <sip:Pokhraj@88.99.245.202>;tag=as696e9844
To: <sip:Madhumita@192.168.43.100:50418;ob>;tag=a50e61b5ebb44f38baf55115d7adf88d
Contact: <sip:Pokhraj@88.99.245.202:5060>
Call-ID: 0206e7e8511ba41419ec4b373ed97857@88.99.245.202:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 15.7.1
Content-Length: 0


---
Audio is at 11452
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 223.191.16.147:7098 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.43.100:8000;branch=z9hG4bK-524287-1---b8e8f0610b2a1c05;received=223.191.16.147;rport=7098
From: "Pokhraj"<sip:Pokhraj@88.99.245.202>;tag=d2296915
To: <sip:200@88.99.245.202>;tag=as10efdb26
Call-ID: 94388NGQ3ODI2NTc2ODUyMDdiZGYyZDM1MzI5ZjYxZmQxYjE
CSeq: 2 INVITE
Server: Asterisk PBX 15.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:200@88.99.245.202:5060>
Content-Type: application/sdp
Content-Length: 277

v=0
o=root 1289480310 1289480310 IN IP4 88.99.245.202
s=Asterisk PBX 15.7.1
c=IN IP4 88.99.245.202
t=0 0
m=audio 11452 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:223.191.16.147:7098 --->
ACK sip:200@88.99.245.202:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.43.100:8000;branch=z9hG4bK-524287-1---88ffc61e79d7056f;rport
Max-Forwards: 70
Contact: <sip:Pokhraj@223.191.16.147:7098;rinstance=fa1038c95bd79575>
To: <sip:200@88.99.245.202>;tag=as10efdb26
From: "Pokhraj"<sip:Pokhraj@88.99.245.202>;tag=d2296915
Call-ID: 94388NGQ3ODI2NTc2ODUyMDdiZGYyZDM1MzI5ZjYxZmQxYjE
CSeq: 2 ACK
User-Agent: X-Lite release 5.4.0 stamp 94388
Content-Length: 0

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

<--- SIP read from UDP:223.191.16.147:7090 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 88.99.245.202:5060;received=88.99.245.202;branch=z9hG4bK16eb32d3
Call-ID: 0206e7e8511ba41419ec4b373ed97857@88.99.245.202:5060
From: "Pokhraj" <sip:Pokhraj@88.99.245.202>;tag=as696e9844
To: <sip:Madhumita@192.168.43.100;ob>;tag=a50e61b5ebb44f38baf55115d7adf88d
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: <sip:Madhumita@192.168.43.100:50418;ob>
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 322

v=0
o=- 3760200043 3760200044 IN IP4 192.168.43.100
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4008 RTP/AVP 0 101
c=IN IP4 192.168.43.100
b=TIAS:64000
a=rtcp:4009 IN IP4 192.168.43.100
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1170553184 cname:54d60ea93f0b3087
<------------->
--- (11 headers 15 lines) ---
set_destination: Parsing <sip:Madhumita@192.168.43.100:50418;ob> for address/port to send to
set_destination: set destination to 192.168.43.100:50418
Transmitting (no NAT) to 192.168.43.100:50418:
ACK sip:Madhumita@192.168.43.100:50418;ob SIP/2.0
Via: SIP/2.0/UDP 88.99.245.202:5060;branch=z9hG4bK715f7f9d
Max-Forwards: 70
From: "Pokhraj" <sip:Pokhraj@88.99.245.202>;tag=as696e9844
To: <sip:Madhumita@192.168.43.100:50418;ob>;tag=a50e61b5ebb44f38baf55115d7adf88d
Contact: <sip:Pokhraj@88.99.245.202:5060>
Call-ID: 0206e7e8511ba41419ec4b373ed97857@88.99.245.202:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 15.7.1
Content-Length: 0


---

<--- SIP read from UDP:223.191.16.147:7090 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 88.99.245.202:5060;received=88.99.245.202;branch=z9hG4bK16eb32d3
Call-ID: 0206e7e8511ba41419ec4b373ed97857@88.99.245.202:5060
From: "Pokhraj" <sip:Pokhraj@88.99.245.202>;tag=as696e9844
To: <sip:Madhumita@192.168.43.100;ob>;tag=a50e61b5ebb44f38baf55115d7adf88d
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: <sip:Madhumita@192.168.43.100:50418;ob>
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 322

v=0
o=- 3760200043 3760200044 IN IP4 192.168.43.100
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4008 RTP/AVP 0 101
c=IN IP4 192.168.43.100
b=TIAS:64000
a=rtcp:4009 IN IP4 192.168.43.100
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1170553184 cname:54d60ea93f0b3087
<------------->
--- (11 headers 15 lines) ---
set_destination: Parsing <sip:Madhumita@192.168.43.100:50418;ob> for address/port to send to
set_destination: set destination to 192.168.43.100:50418
Transmitting (no NAT) to 192.168.43.100:50418:
ACK sip:Madhumita@192.168.43.100:50418;ob SIP/2.0
Via: SIP/2.0/UDP 88.99.245.202:5060;branch=z9hG4bK3456865a
Max-Forwards: 70
From: "Pokhraj" <sip:Pokhraj@88.99.245.202>;tag=as696e9844
To: <sip:Madhumita@192.168.43.100:50418;ob>;tag=a50e61b5ebb44f38baf55115d7adf88d
Contact: <sip:Pokhraj@88.99.245.202:5060>
Call-ID: 0206e7e8511ba41419ec4b373ed97857@88.99.245.202:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 15.7.1
Content-Length: 0


---

<--- SIP read from UDP:223.191.16.147:7090 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 88.99.245.202:5060;received=88.99.245.202;branch=z9hG4bK16eb32d3
Call-ID: 0206e7e8511ba41419ec4b373ed97857@88.99.245.202:5060
From: "Pokhraj" <sip:Pokhraj@88.99.245.202>;tag=as696e9844
To: <sip:Madhumita@192.168.43.100;ob>;tag=a50e61b5ebb44f38baf55115d7adf88d
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: <sip:Madhumita@192.168.43.100:50418;ob>
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 322

v=0
o=- 3760200043 3760200044 IN IP4 192.168.43.100
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4008 RTP/AVP 0 101
c=IN IP4 192.168.43.100
b=TIAS:64000
a=rtcp:4009 IN IP4 192.168.43.100
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1170553184 cname:54d60ea93f0b3087
<------------->
--- (11 headers 15 lines) ---
set_destination: Parsing <sip:Madhumita@192.168.43.100:50418;ob> for address/port to send to
set_destination: set destination to 192.168.43.100:50418
Transmitting (no NAT) to 192.168.43.100:50418:
ACK sip:Madhumita@192.168.43.100:50418;ob SIP/2.0
Via: SIP/2.0/UDP 88.99.245.202:5060;branch=z9hG4bK5a949519
Max-Forwards: 70
From: "Pokhraj" <sip:Pokhraj@88.99.245.202>;tag=as696e9844
To: <sip:Madhumita@192.168.43.100:50418;ob>;tag=a50e61b5ebb44f38baf55115d7adf88d
Contact: <sip:Pokhraj@88.99.245.202:5060>
Call-ID: 0206e7e8511ba41419ec4b373ed97857@88.99.245.202:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 15.7.1
Content-Length: 0


---

<--- SIP read from UDP:223.191.16.147:7090 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 88.99.245.202:5060;received=88.99.245.202;branch=z9hG4bK16eb32d3
Call-ID: 0206e7e8511ba41419ec4b373ed97857@88.99.245.202:5060
From: "Pokhraj" <sip:Pokhraj@88.99.245.202>;tag=as696e9844
To: <sip:Madhumita@192.168.43.100;ob>;tag=a50e61b5ebb44f38baf55115d7adf88d
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: <sip:Madhumita@192.168.43.100:50418;ob>
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 322

v=0
o=- 3760200043 3760200044 IN IP4 192.168.43.100
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4008 RTP/AVP 0 101
c=IN IP4 192.168.43.100
b=TIAS:64000
a=rtcp:4009 IN IP4 192.168.43.100
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1170553184 cname:54d60ea93f0b3087
<------------->
--- (11 headers 15 lines) ---
set_destination: Parsing <sip:Madhumita@192.168.43.100:50418;ob> for address/port to send to
set_destination: set destination to 192.168.43.100:50418
Transmitting (no NAT) to 192.168.43.100:50418:
ACK sip:Madhumita@192.168.43.100:50418;ob SIP/2.0
Via: SIP/2.0/UDP 88.99.245.202:5060;branch=z9hG4bK20c14289
Max-Forwards: 70
From: "Pokhraj" <sip:Pokhraj@88.99.245.202>;tag=as696e9844
To: <sip:Madhumita@192.168.43.100:50418;ob>;tag=a50e61b5ebb44f38baf55115d7adf88d
Contact: <sip:Pokhraj@88.99.245.202:5060>
Call-ID: 0206e7e8511ba41419ec4b373ed97857@88.99.245.202:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 15.7.1
Content-Length: 0


---

<--- SIP read from UDP:223.191.16.147:7090 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 88.99.245.202:5060;received=88.99.245.202;branch=z9hG4bK16eb32d3
Call-ID: 0206e7e8511ba41419ec4b373ed97857@88.99.245.202:5060
From: "Pokhraj" <sip:Pokhraj@88.99.245.202>;tag=as696e9844
To: <sip:Madhumita@192.168.43.100;ob>;tag=a50e61b5ebb44f38baf55115d7adf88d
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: <sip:Madhumita@192.168.43.100:50418;ob>
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 322

v=0
o=- 3760200043 3760200044 IN IP4 192.168.43.100
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4008 RTP/AVP 0 101
c=IN IP4 192.168.43.100
b=TIAS:64000
a=rtcp:4009 IN IP4 192.168.43.100
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1170553184 cname:54d60ea93f0b3087
<------------->
--- (11 headers 15 lines) ---
set_destination: Parsing <sip:Madhumita@192.168.43.100:50418;ob> for address/port to send to
set_destination: set destination to 192.168.43.100:50418
Transmitting (no NAT) to 192.168.43.100:50418:
ACK sip:Madhumita@192.168.43.100:50418;ob SIP/2.0
Via: SIP/2.0/UDP 88.99.245.202:5060;branch=z9hG4bK4499c724
Max-Forwards: 70
From: "Pokhraj" <sip:Pokhraj@88.99.245.202>;tag=as696e9844
To: <sip:Madhumita@192.168.43.100:50418;ob>;tag=a50e61b5ebb44f38baf55115d7adf88d
Contact: <sip:Pokhraj@88.99.245.202:5060>
Call-ID: 0206e7e8511ba41419ec4b373ed97857@88.99.245.202:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 15.7.1
Content-Length: 0


---

<--- SIP read from UDP:223.191.16.147:7090 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 88.99.245.202:5060;received=88.99.245.202;branch=z9hG4bK16eb32d3
Call-ID: 0206e7e8511ba41419ec4b373ed97857@88.99.245.202:5060
From: "Pokhraj" <sip:Pokhraj@88.99.245.202>;tag=as696e9844
To: <sip:Madhumita@192.168.43.100;ob>;tag=a50e61b5ebb44f38baf55115d7adf88d
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: <sip:Madhumita@192.168.43.100:50418;ob>
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 322

v=0
o=- 3760200043 3760200044 IN IP4 192.168.43.100
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4008 RTP/AVP 0 101
c=IN IP4 192.168.43.100
b=TIAS:64000
a=rtcp:4009 IN IP4 192.168.43.100
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1170553184 cname:54d60ea93f0b3087
<------------->
--- (11 headers 15 lines) ---
set_destination: Parsing <sip:Madhumita@192.168.43.100:50418;ob> for address/port to send to
set_destination: set destination to 192.168.43.100:50418
Transmitting (no NAT) to 192.168.43.100:50418:
ACK sip:Madhumita@192.168.43.100:50418;ob SIP/2.0
Via: SIP/2.0/UDP 88.99.245.202:5060;branch=z9hG4bK558000d5
Max-Forwards: 70
From: "Pokhraj" <sip:Pokhraj@88.99.245.202>;tag=as696e9844
To: <sip:Madhumita@192.168.43.100:50418;ob>;tag=a50e61b5ebb44f38baf55115d7adf88d
Contact: <sip:Pokhraj@88.99.245.202:5060>
Call-ID: 0206e7e8511ba41419ec4b373ed97857@88.99.245.202:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 15.7.1
Content-Length: 0


---

<--- SIP read from UDP:223.191.16.147:7090 --->

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

<--- SIP read from UDP:223.191.16.147:7090 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 88.99.245.202:5060;received=88.99.245.202;branch=z9hG4bK16eb32d3
Call-ID: 0206e7e8511ba41419ec4b373ed97857@88.99.245.202:5060
From: "Pokhraj" <sip:Pokhraj@88.99.245.202>;tag=as696e9844
To: <sip:Madhumita@192.168.43.100;ob>;tag=a50e61b5ebb44f38baf55115d7adf88d
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: <sip:Madhumita@192.168.43.100:50418;ob>
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 322

v=0
o=- 3760200043 3760200044 IN IP4 192.168.43.100
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4008 RTP/AVP 0 101
c=IN IP4 192.168.43.100
b=TIAS:64000
a=rtcp:4009 IN IP4 192.168.43.100
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1170553184 cname:54d60ea93f0b3087
<------------->
--- (11 headers 15 lines) ---
set_destination: Parsing <sip:Madhumita@192.168.43.100:50418;ob> for address/port to send to
set_destination: set destination to 192.168.43.100:50418
Transmitting (no NAT) to 192.168.43.100:50418:
ACK sip:Madhumita@192.168.43.100:50418;ob SIP/2.0
Via: SIP/2.0/UDP 88.99.245.202:5060;branch=z9hG4bK45d22c80
Max-Forwards: 70
From: "Pokhraj" <sip:Pokhraj@88.99.245.202>;tag=as696e9844
To: <sip:Madhumita@192.168.43.100:50418;ob>;tag=a50e61b5ebb44f38baf55115d7adf88d
Contact: <sip:Pokhraj@88.99.245.202:5060>
Call-ID: 0206e7e8511ba41419ec4b373ed97857@88.99.245.202:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 15.7.1
Content-Length: 0


---

<--- SIP read from UDP:223.191.16.147:7090 --->
BYE sip:Pokhraj@88.99.245.202:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.43.100:50418;rport;branch=z9hG4bKPj9ea4713a1fd74efab048229ffa8fdeaa
Max-Forwards: 70
From: <sip:Madhumita@192.168.43.100;ob>;tag=a50e61b5ebb44f38baf55115d7adf88d
To: "Pokhraj" <sip:Pokhraj@88.99.245.202>;tag=as696e9844
Call-ID: 0206e7e8511ba41419ec4b373ed97857@88.99.245.202:5060
CSeq: 25996 BYE
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Sending to 223.191.16.147:7090 (no NAT)
Scheduling destruction of SIP dialog '0206e7e8511ba41419ec4b373ed97857@88.99.245.202:5060' in 98624 ms (Method: BYE)

<--- Transmitting (no NAT) to 223.191.16.147:7090 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.43.100:50418;branch=z9hG4bKPj9ea4713a1fd74efab048229ffa8fdeaa;received=223.191.16.147;rport=7090
From: <sip:Madhumita@192.168.43.100;ob>;tag=a50e61b5ebb44f38baf55115d7adf88d
To: "Pokhraj" <sip:Pokhraj@88.99.245.202>;tag=as696e9844
Call-ID: 0206e7e8511ba41419ec4b373ed97857@88.99.245.202:5060
CSeq: 25996 BYE
Server: Asterisk PBX 15.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '94388NGQ3ODI2NTc2ODUyMDdiZGYyZDM1MzI5ZjYxZmQxYjE' in 109952 ms (Method: ACK)
set_destination: Parsing <sip:Pokhraj@223.191.16.147:7098;rinstance=fa1038c95bd79575> for address/port to send to
set_destination: set destination to 223.191.16.147:7098
Reliably Transmitting (no NAT) to 223.191.16.147:7098:
BYE sip:Pokhraj@223.191.16.147:7098;rinstance=fa1038c95bd79575 SIP/2.0
Via: SIP/2.0/UDP 88.99.245.202:5060;branch=z9hG4bK61c884c1;rport
Max-Forwards: 70
From: <sip:200@88.99.245.202>;tag=as10efdb26
To: "Pokhraj"<sip:Pokhraj@88.99.245.202>;tag=d2296915
Call-ID: 94388NGQ3ODI2NTc2ODUyMDdiZGYyZDM1MzI5ZjYxZmQxYjE
CSeq: 102 BYE
User-Agent: Asterisk PBX 15.7.1
Proxy-Authorization: Digest username="Pokhraj", realm="asterisk", algorithm=MD5, uri="sip:88.99.245.202", nonce="7c2258df", response="d7fa6ae738923855be0a01747c06936f"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:223.191.16.147:7098 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 88.99.245.202:5060;branch=z9hG4bK61c884c1;rport=5060
Contact: <sip:Pokhraj@223.191.16.147:7098;rinstance=fa1038c95bd79575>
To: "Pokhraj"<sip:Pokhraj@88.99.245.202>;tag=d2296915
From: <sip:200@88.99.245.202>;tag=as10efdb26
Call-ID: 94388NGQ3ODI2NTc2ODUyMDdiZGYyZDM1MzI5ZjYxZmQxYjE
CSeq: 102 BYE
User-Agent: X-Lite release 5.4.0 stamp 94388
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '94388NGQ3ODI2NTc2ODUyMDdiZGYyZDM1MzI5ZjYxZmQxYjE' Method: ACK

<--- SIP read from UDP:223.191.16.147:7090 --->

<------------->
Reliably Transmitting (no NAT) to 223.191.16.147:7090:
OPTIONS sip:Madhumita@192.168.43.100:50418;ob SIP/2.0
Via: SIP/2.0/UDP 88.99.245.202:5060;branch=z9hG4bK75aef7a5
Max-Forwards: 70
From: "asterisk" <sip:asterisk@88.99.245.202>;tag=as7be8590f
To: <sip:Madhumita@192.168.43.100:50418;ob>
Contact: <sip:asterisk@88.99.245.202:5060>
Call-ID: 01b841cf612d4e8f43e3c4cf1ae93c2c@88.99.245.202:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.7.1
Date: Tue, 26 Feb 2019 14:31:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #1 (no NAT) to 223.191.16.147:7090:
OPTIONS sip:Madhumita@192.168.43.100:50418;ob SIP/2.0
Via: SIP/2.0/UDP 88.99.245.202:5060;branch=z9hG4bK75aef7a5
Max-Forwards: 70
From: "asterisk" <sip:asterisk@88.99.245.202>;tag=as7be8590f
To: <sip:Madhumita@192.168.43.100:50418;ob>
Contact: <sip:asterisk@88.99.245.202:5060>
Call-ID: 01b841cf612d4e8f43e3c4cf1ae93c2c@88.99.245.202:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.7.1
Date: Tue, 26 Feb 2019 14:31:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #2 (no NAT) to 223.191.16.147:7090:
OPTIONS sip:Madhumita@192.168.43.100:50418;ob SIP/2.0
Via: SIP/2.0/UDP 88.99.245.202:5060;branch=z9hG4bK75aef7a5
Max-Forwards: 70
From: "asterisk" <sip:asterisk@88.99.245.202>;tag=as7be8590f
To: <sip:Madhumita@192.168.43.100:50418;ob>
Contact: <sip:asterisk@88.99.245.202:5060>
Call-ID: 01b841cf612d4e8f43e3c4cf1ae93c2c@88.99.245.202:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.7.1
Date: Tue, 26 Feb 2019 14:31:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:223.191.16.147:7098 --->


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

<--- SIP read from UDP:223.191.16.147:7090 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 88.99.245.202:5060;received=88.99.245.202;branch=z9hG4bK75aef7a5
Call-ID: 01b841cf612d4e8f43e3c4cf1ae93c2c@88.99.245.202:5060
From: "asterisk" <sip:asterisk@88.99.245.202>;tag=as7be8590f
To: <sip:Madhumita@192.168.43.100;ob>;tag=z9hG4bK75aef7a5
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: MicroSIP/3.19.10
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
[Feb 26 09:31:54] NOTICE[13718]: chan_sip.c:24711 handle_response_peerpoke: Peer 'Madhumita' is now Lagged. (2893ms / 2000ms)
Really destroying SIP dialog '01b841cf612d4e8f43e3c4cf1ae93c2c@88.99.245.202:5060' Method: OPTIONS
srv*CLI> sip set desip set debug off
No such command 'sip set desip set debug off' (type 'core show help sip set desip' for other possible commands)
Reliably Transmitting (no NAT) to 223.191.16.147:7090:
OPTIONS sip:Madhumita@192.168.43.100:50418;ob SIP/2.0
Via: SIP/2.0/UDP 88.99.245.202:5060;branch=z9hG4bK4f0502d2
Max-Forwards: 70
From: "asterisk" <sip:asterisk@88.99.245.202>;tag=as35c72248
To: <sip:Madhumita@192.168.43.100:50418;ob>
Contact: <sip:asterisk@88.99.245.202:5060>
Call-ID: 244c166542b845ee17bb8c9b5583fe02@88.99.245.202:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.7.1
Date: Tue, 26 Feb 2019 14:32:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:223.191.16.147:7090 --->

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

<--- SIP read from UDP:223.191.16.147:7090 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 88.99.245.202:5060;received=88.99.245.202;branch=z9hG4bK4f0502d2
Call-ID: 244c166542b845ee17bb8c9b5583fe02@88.99.245.202:5060
From: "asterisk" <sip:asterisk@88.99.245.202>;tag=as35c72248
To: <sip:Madhumita@192.168.43.100;ob>;tag=z9hG4bK4f0502d2
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: MicroSIP/3.19.10
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
[Feb 26 09:32:04] NOTICE[13718]: chan_sip.c:24711 handle_response_peerpoke: Peer 'Madhumita' is now Reachable. (199ms / 2000ms)
Really destroying SIP dialog '244c166542b845ee17bb8c9b5583fe02@88.99.245.202:5060' Method: OPTIONS
Reliably Transmitting (no NAT) to 223.191.16.147:7098:
OPTIONS sip:Pokhraj@223.191.16.147:7098;rinstance=fa1038c95bd79575 SIP/2.0
Via: SIP/2.0/UDP 88.99.245.202:5060;branch=z9hG4bK5ac7d884
Max-Forwards: 70
From: "asterisk" <sip:asterisk@88.99.245.202>;tag=as3c9293cc
To: <sip:Pokhraj@223.191.16.147:7098;rinstance=fa1038c95bd79575>
Contact: <sip:asterisk@88.99.245.202:5060>
Call-ID: 632e398b287422c306a511070c7edc0e@88.99.245.202:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.7.1
Date: Tue, 26 Feb 2019 14:32:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:223.191.16.147:7098 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 88.99.245.202:5060;branch=z9hG4bK5ac7d884
Contact: <sip:192.168.43.100:8000>
To: <sip:Pokhraj@223.191.16.147:7098;rinstance=fa1038c95bd79575>;tag=d5eb9952
From: "asterisk" <sip:asterisk@88.99.245.202>;tag=as3c9293cc
Call-ID: 632e398b287422c306a511070c7edc0e@88.99.245.202:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
Supported: replaces
User-Agent: X-Lite release 5.4.0 stamp 94388
Allow-Events: talk, hold
Content-Length: 0

jcolp · February 26, 2019, 2:42pm

Your NAT setting is incorrect, so it’s not being applied. You want:

nat=force_rport,comedia

pokhraj_d · February 26, 2019, 3:17pm

Hello jcolp,
Awesome. Its working as expected. But one more confusion.
As per the above configuration, I am able to call and the receiving party is able to hear my voice.
Now when the receiving party is saying something, I am not able to hear anything. Please note his system (Laptop) is also behind NAT, as he is also using router at his point for internet.

Is there is any more configuration I need to perform from my side or his side?

Please advice.

jcolp · February 26, 2019, 3:26pm

No, but you need to isolate the problem further. For example “rtp set debug on” will show you media flowing through Asterisk. You should also check port forwarding to ensure that the RTP range is forwarded to Asterisk if it is behind a router or something.

pokhraj_d · February 26, 2019, 5:53pm

Are you talking about the remote receiver RTP range? Is my understanding correct?
Also could you please put some light about the default RTP range for Asterisk.

Thanks

jcolp · February 26, 2019, 5:54pm

No, I was referring to if Asterisk is behind a router. You did not specify the network layout - so I can only guess.

The default port range for Asterisk RTP is 10000 - 20000 and is configured in rtp.conf

EkFudrek · February 26, 2019, 6:10pm

Maybe it is a stupid question. Why do you need NAT routing on the Hetzner box? Wouldn’t a normal firewall do the same thing in your case?

If so, why don’t you run something like pfSense and let Asterisk also run under this system?

If you have several SIP devices within your LAN, I think it makes sense to consider running a SIP proxy on the home router, but that depends on which router you are using (Lancom has an additional module, but with a Fritzbox you are out of luck, I think).

pokhraj_d · February 28, 2019, 8:34am

Hello,
The below is my current configuration…

I have a private IP which provided by my ISP. [192.168.10.24]
I am using router for internet.
My asterisk is running on cloud server which have a ip as [88.29.xx.xxxx].
I have port forward at my router the default port for asterisk [5060] for outgoing.

So, Do I need GLOBAL_IP for connecting x-lite from my desktop ?
Also Do I need port forward [10000-20000] for incoming call?

david551 · February 28, 2019, 9:45am

Please explain this as ISPs generally provide only public IPs and it is the local router that doe any NATting to private ones.

(I’m also curious as to why so many people insist on putting low processing high traffic volume servers on remote (cloud) sites, but that is more an audio quality issue, although in this case it is adding complications, by by exacerbating the NAT complexity. I’d seriously consider running NAT from your private network to the cloud server, for security as well as to simplify the IP address hackery.)

johnkiniston · February 28, 2019, 4:38pm

Sadly with the exhaustion of IPv4 address space this is no longer true in many markets.

‘Carrier Grade’ NAT is popping up everywhere.

Call Disconnecting after 20 secs

Hello, I am facing a situation where the outgoing call has been disconnecting after 20 secs. It is telling “408- Request Timeout” I am using the below SIP configuration

My SIP Debug is as follow

Hello,
I am facing a situation where the outgoing call has been disconnecting after 20 secs. It is telling “408- Request Timeout”
I am using the below SIP configuration