Asterisk does not recognize that the called party has hanged up

Matej86 · September 7, 2018, 11:36am

Hello,

I have a problem with outgoing calls through my provider SIP trunk.

I am able to establish an outgoing call and I don’t have any kind of problems during the conversation.

The problem is when the called party hangs up. It seems the Asterisk does not get the information that the called party has hanged up and the call proceeds. If I want to stop the call, I have to end the call.

This is my sip.conf:

[general]
allowguest=no
alwaysauthreject=yes
prematuremedia=no
progressinband=yes

register=> *************************************

[1000]
type=friend
host=dynamic
secret=**************
context=internal
allow=all
mailbox=XXXX@default
qualify=yes

[1001]
type=friend
host=dynamic
secret=**************
context=internal
allow=all
mailbox=XXXX@default
qualify=yes

[telecom]
defaultuser=+39**********
type=peer
secret=*******************
qualify=no
outboundproxy=5.97.52.7
insecure=invite
host=telecomitalia.it
fromuser=+39**********
fromdomain=telecomitalia.it
directmedia=no
nat=force_rport,comedia
context=incoming

This is the extensions.conf file

[internal]

exten => 1000,1,NoOp()
same => n,Dial(SIP/1000,60)
same => n,Hangup()

exten => 1001,1,NoOp()
same => n,Dial(SIP/1001,60)
same => n,Hangup()

exten => _0039X.,1,Log(NOTICE, Dialing out from ${CALLERID(all)} to ${EXTEN:1} through TIM)
same => n,Dial(SIP/telecom/0${EXTEN:1},60)
same => n,Playtones(congestion)
same => n,Hangup()

exten => _+39X.,1,Log(NOTICE, Dialing out from ${CALLERID(all)} to ${EXTEN:1} through TIM)
same => n,Dial(SIP/telecom/00${EXTEN:1},60)
same => n,Playtones(congestion)
same => n,Hangup()

[incoming]

exten => s,1,Log(NOTICE, Incoming call from ${CALLERID(all)})
exten => s,n,Dial(SIP/1000)
exten => s,n,Hangup()

It is funny that when I call an italian landline number though this SIP trunk, I am getting the mentioned problem. When I call an italian mobile number though the SIP trunk, I am getting the hangup message from the called number and asterisk ends the call.

This is what the standard asterisk log shows me when I call a landline number, the called party answers the call, we have a short conversation and the the called party hangs up.

== Using SIP RTP CoS mark 5
– Executing [0039030********@internal:1] Log(“SIP/1001-0000009f”, “NOTICE, Dialing out from “” <1001> to 039030******* through TIM”) in new stack
[Sep 7 13:50:37] NOTICE[5213][C-00000051]: Ext. 0039030*********:1 @ internal: Dialing out from “” <1001> to 039030******** through TIM
– Executing [0039030********@internal:2] Dial(“SIP/1001-0000009f”, “SIP/telecom/0039030********,60”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/telecom/0039030********
– SIP/telecom-000000a0 is ringing
– SIP/telecom-000000a0 is making progress passing it to SIP/1001-0000009f
– SIP/telecom-000000a0 answered SIP/1001-0000009f
– Channel SIP/telecom-000000a0 joined ‘simple_bridge’ basic-bridge
– Channel SIP/1001-0000009f joined ‘simple_bridge’ basic-bridge

And only if I end the call, the the log shows me the following missing lines

– Channel SIP/1001-0000009f left ‘simple_bridge’ basic-bridge
– Channel SIP/telecom-000000a0 left ‘simple_bridge’ basic-bridge
== Spawn extension (internal, 0039030********, 2) exited non-zero on ‘SIP/1001-0000009f’

What can I do?

Thanks

jcolp · September 7, 2018, 11:58am

If you provide a SIP trace using “pjsip set logger on” then it can be seen if they actually sent the message and we ignored it. If they didn’t then it’s an upstream problem with the provider - and there is nothing really you can do on your side.

Matej86 · September 7, 2018, 12:08pm

Here is what asterisk is showing me when I enable SIP SET DEBUG ON


<--- SIP read from UDP:192.168.2.6:8204 --->
INVITE sip:0039030820276@192.168.2.164 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.6:8204;branch=z9hG4bKGR1IHMdE94jpspai;rport
Contact: <sip:1001@192.168.2.6:8204>
Max-Forwards: 70
From: <sip:1001@192.168.2.164>;tag=06E716FB65F2B6298D1665E7FEB872F2
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
To: <sip:0039030820276@192.168.2.164>
Content-Type: application/sdp
Call-ID: AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
CSeq: 1 INVITE
User-Agent: Acrobits Softphone Business/3.8.2
Content-Length: 305

v=0
o=- 4858458339 26001 IN IP4 172.26.170.170
s=xfppwij
c=IN IP4 192.168.2.6
t=0 0
m=audio 18262 RTP/AVP 103 101
a=rtpmap:101 telephone-event/8000
a=rtpmap:103 opus/48000/2
a=fmtp:101 0-15
a=fmtp:103 maxplaybackrate=16000;maxaveragebitrate=24000;useinbandfec=1;usedtx=1
a=ptime:20
a=sendrecv
<------------->
--- (13 headers 12 lines) ---
Sending to 192.168.2.6:8204 (no NAT)
Sending to 192.168.2.6:8204 (no NAT)
Using INVITE request as basis request - AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
Found peer '1001' for '1001' from 192.168.2.6:8204

<--- Reliably Transmitting (no NAT) to 192.168.2.6:8204 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.6:8204;branch=z9hG4bKGR1IHMdE94jpspai;received=192.168.2.6;rport=8204
From: <sip:1001@192.168.2.164>;tag=06E716FB65F2B6298D1665E7FEB872F2
To: <sip:0039030820276@192.168.2.164>;tag=as322b6c5a
Call-ID: AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
CSeq: 1 INVITE
Server: Asterisk PBX 15.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7e9026d3"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893' in 12672 ms (Method: INVITE)
Retransmitting #1 (no NAT) to 192.168.2.6:8204:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.6:8204;branch=z9hG4bKGR1IHMdE94jpspai;received=192.168.2.6;rport=8204
From: <sip:1001@192.168.2.164>;tag=06E716FB65F2B6298D1665E7FEB872F2
To: <sip:0039030820276@192.168.2.164>;tag=as322b6c5a
Call-ID: AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
CSeq: 1 INVITE
Server: Asterisk PBX 15.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7e9026d3"
Content-Length: 0


---

<--- SIP read from UDP:192.168.2.6:8204 --->
ACK sip:0039030820276@192.168.2.164 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.6:8204;branch=z9hG4bKGR1IHMdE94jpspai;rport
Max-Forwards: 70
Call-ID: AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
From: <sip:1001@192.168.2.164>;tag=06E716FB65F2B6298D1665E7FEB872F2
To: <sip:0039030820276@192.168.2.164>;tag=as322b6c5a
CSeq: 1 ACK
User-Agent: Acrobits Softphone Business/3.8.2
Content-Length: 0

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

<--- SIP read from UDP:192.168.2.6:8204 --->
INVITE sip:0039030820276@192.168.2.164 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.6:8204;branch=z9hG4bKBgoWg8Jzoiuf06L6;rport
Contact: <sip:1001@192.168.2.6:8204>
Max-Forwards: 70
From: <sip:1001@192.168.2.164>;tag=06E716FB65F2B6298D1665E7FEB872F2
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
To: <sip:0039030820276@192.168.2.164>
Content-Type: application/sdp
Call-ID: AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
CSeq: 2 INVITE
Authorization: Digest username="1001",realm="asterisk",algorithm=MD5,uri="sip:0039030820276@192.168.2.164",nonce="7e9026d3",response="465a33f53c8d6a8042cded28a05306f6"
User-Agent: Acrobits Softphone Business/3.8.2
Content-Length: 305

v=0
o=- 4858458339 26001 IN IP4 172.26.170.170
s=xfppwij
c=IN IP4 192.168.2.6
t=0 0
m=audio 18262 RTP/AVP 103 101
a=rtpmap:101 telephone-event/8000
a=rtpmap:103 opus/48000/2
a=fmtp:101 0-15
a=fmtp:103 maxplaybackrate=16000;maxaveragebitrate=24000;useinbandfec=1;usedtx=1
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 192.168.2.6:8204 (no NAT)
Using INVITE request as basis request - AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
Found peer '1001' for '1001' from 192.168.2.6:8204
  == Using SIP RTP CoS mark 5
Found RTP audio format 103
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Found audio description format opus for ID 103
Capabilities: us - (ulaw|alaw|gsm|h263|codec2|g723|g726|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|g729|speex|speex|speex|ilbc|g722|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263p|h264|mpeg4|vp8|vp9|red|t140|t38|silk|silk|silk|silk), peer - audio=(opus)/video=(nothing)/text=(nothing), combined - (opus)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.2.6:18262
Looking for 0039030820276 in internal (domain 192.168.2.164)
sip_route_dump: route/path hop: <sip:1001@192.168.2.6:8204>

<--- Transmitting (no NAT) to 192.168.2.6:8204 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.6:8204;branch=z9hG4bKBgoWg8Jzoiuf06L6;received=192.168.2.6;rport=8204
From: <sip:1001@192.168.2.164>;tag=06E716FB65F2B6298D1665E7FEB872F2
To: <sip:0039030820276@192.168.2.164>
Call-ID: AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
CSeq: 2 INVITE
Server: Asterisk PBX 15.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:0039030820276@192.168.2.164:5060>
Content-Length: 0


<------------>
    -- Executing [0039030820276@internal:1] Log("SIP/1001-000000a3", "NOTICE, Dialing out from "" <1001> to 039030820276 through TIM") in new stack
[Sep  7 14:05:26] NOTICE[5257][C-00000053]: Ext. 0039030820276:1 @ internal:  Dialing out from "" <1001> to 039030820276 through TIM
    -- Executing [0039030820276@internal:2] Dial("SIP/1001-000000a3", "SIP/telecom/0039030820276,60") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 16692
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 (NAT) to 5.97.52.7:5060:
INVITE sip:0039030820276@telecomitalia.it SIP/2.0
Via: SIP/2.0/UDP 192.168.2.164:5060;branch=z9hG4bK60a042bd;rport
Max-Forwards: 70
From: <sip:+39040633324@telecomitalia.it>;tag=as3c495127
To: <sip:0039030820276@telecomitalia.it>
Contact: <sip:+39040633324@192.168.2.164:5060>
Call-ID: 4ce701190ceee0731a300e6d4abf6c83@telecomitalia.it
CSeq: 102 INVITE
User-Agent: Asterisk PBX 15.5.0
Date: Fri, 07 Sep 2018 12:05:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 953731005 953731005 IN IP4 192.168.2.164
s=Asterisk PBX 15.5.0
c=IN IP4 192.168.2.164
t=0 0
m=audio 16692 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/telecom/0039030820276

<--- SIP read from UDP:5.97.52.7:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.164:5060;received=87.4.160.7;branch=z9hG4bK60a042bd;rport=5060
From: <sip:+39040633324@telecomitalia.it>;tag=as3c495127
To: <sip:0039030820276@telecomitalia.it>
Call-ID: 4ce701190ceee0731a300e6d4abf6c83@telecomitalia.it
CSeq: 102 INVITE

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

<--- SIP read from UDP:5.97.52.7:5060 --->
SIP/2.0 407 Proxy Authentication Required 02035033D
Via: SIP/2.0/UDP 192.168.2.164:5060;received=87.4.160.7;branch=z9hG4bK60a042bd;rport=5060
From: <sip:+39040633324@telecomitalia.it>;tag=as3c495127
To: <sip:0039030820276@telecomitalia.it>;tag=ef3e51940fc70645a0d2c055267b02b2
Call-ID: 4ce701190ceee0731a300e6d4abf6c83@telecomitalia.it
CSeq: 102 INVITE
Content-Length: 0
Proxy-Authenticate: Digest nonce="B60D101C9269925B0000000084E41472",realm="telecomitalia.it",algorithm=MD5,qop="auth"

<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 5.97.52.7:5060:
ACK sip:0039030820276@telecomitalia.it SIP/2.0
Via: SIP/2.0/UDP 192.168.2.164:5060;branch=z9hG4bK60a042bd;rport
Max-Forwards: 70
From: <sip:+39040633324@telecomitalia.it>;tag=as3c495127
To: <sip:0039030820276@telecomitalia.it>;tag=ef3e51940fc70645a0d2c055267b02b2
Contact: <sip:+39040633324@192.168.2.164:5060>
Call-ID: 4ce701190ceee0731a300e6d4abf6c83@telecomitalia.it
CSeq: 102 ACK
User-Agent: Asterisk PBX 15.5.0
Content-Length: 0


---
Audio is at 16692
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 (NAT) to 5.97.52.7:5060:
INVITE sip:0039030820276@telecomitalia.it SIP/2.0
Via: SIP/2.0/UDP 192.168.2.164:5060;branch=z9hG4bK4c76155a;rport
Max-Forwards: 70
From: <sip:+39040633324@telecomitalia.it>;tag=as3c495127
To: <sip:0039030820276@telecomitalia.it>
Contact: <sip:+39040633324@192.168.2.164:5060>
Call-ID: 4ce701190ceee0731a300e6d4abf6c83@telecomitalia.it
CSeq: 103 INVITE
User-Agent: Asterisk PBX 15.5.0
Proxy-Authorization: Digest username="+39040633324", realm="telecomitalia.it", algorithm=MD5, uri="sip:0039030820276@telecomitalia.it", nonce="B60D101C9269925B0000000084E41472", response="fd232b6aba4e715cfbdf75c869274848", qop=auth, cnonce="751c6be8", nc=00000001
Date: Fri, 07 Sep 2018 12:05:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 953731005 953731006 IN IP4 192.168.2.164
s=Asterisk PBX 15.5.0
c=IN IP4 192.168.2.164
t=0 0
m=audio 16692 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

---

<--- SIP read from UDP:5.97.52.7:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.164:5060;received=87.4.160.7;branch=z9hG4bK4c76155a;rport=5060
From: <sip:+39040633324@telecomitalia.it>;tag=as3c495127
To: <sip:0039030820276@telecomitalia.it>
Call-ID: 4ce701190ceee0731a300e6d4abf6c83@telecomitalia.it
CSeq: 103 INVITE

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

<--- SIP read from UDP:192.168.2.6:8204 --->
ACK sip:0039030820276@192.168.2.164 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.6:8204;branch=z9hG4bKGR1IHMdE94jpspai;rport
Max-Forwards: 70
Call-ID: AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
From: <sip:1001@192.168.2.164>;tag=06E716FB65F2B6298D1665E7FEB872F2
To: <sip:0039030820276@192.168.2.164>;tag=as322b6c5a
CSeq: 1 ACK
User-Agent: Acrobits Softphone Business/3.8.2
Content-Length: 0

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

<--- SIP read from UDP:5.97.52.7:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.2.164:5060;received=87.4.160.7;branch=z9hG4bK4c76155a;rport=5060
From: <sip:+39040633324@telecomitalia.it>;tag=as3c495127
To: <sip:0039030820276@telecomitalia.it>;tag=p65545t1536321926m180505c1119068138s1_3534468296-1265672840
Call-ID: 4ce701190ceee0731a300e6d4abf6c83@telecomitalia.it
CSeq: 103 INVITE
Content-Length: 252
Contact: <sip:p65545t1536321926m180505c1119068138s1@5.97.52.7:5060;transport=udp>
Content-Type: application/sdp
Allow: UPDATE, PRACK, REFER, NOTIFY, OPTIONS, BYE, ACK, CANCEL, INVITE, REGISTER
Accept: application/sdp
Accept: application/isup
Accept: application/xml
Supported: timer
Server: Ericsson MTAS - CXP9020729/8 R8H01

v=0
o=- 1291177671 3535369597 IN IP4 5.97.52.7
s=IMSS
c=IN IP4 5.97.52.7
t=0 0
m=audio 54102 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=sqn:0
a=cdsc: 1 image udptl t38
a=sendrecv
a=ptime:20
<------------->
--- (15 headers 13 lines) ---
sip_route_dump: route/path hop: <sip:p65545t1536321926m180505c1119068138s1@5.97.52.7:5060;transport=udp>
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|h263), 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 5.97.52.7:54102
    -- SIP/telecom-000000a4 is ringing

<--- Transmitting (no NAT) to 192.168.2.6:8204 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.2.6:8204;branch=z9hG4bKBgoWg8Jzoiuf06L6;received=192.168.2.6;rport=8204
From: <sip:1001@192.168.2.164>;tag=06E716FB65F2B6298D1665E7FEB872F2
To: <sip:0039030820276@192.168.2.164>;tag=as3e6fab78
Call-ID: AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
CSeq: 2 INVITE
Server: Asterisk PBX 15.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:0039030820276@192.168.2.164:5060>
Content-Length: 0


<------------>
Audio is at 17550
Adding codec opus to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 192.168.2.6:8204 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.2.6:8204;branch=z9hG4bKBgoWg8Jzoiuf06L6;received=192.168.2.6;rport=8204
From: <sip:1001@192.168.2.164>;tag=06E716FB65F2B6298D1665E7FEB872F2
To: <sip:0039030820276@192.168.2.164>;tag=as3e6fab78
Call-ID: AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
CSeq: 2 INVITE
Server: Asterisk PBX 15.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:0039030820276@192.168.2.164:5060>
Content-Type: application/sdp
Content-Length: 327

v=0
o=root 956074400 956074400 IN IP4 192.168.2.164
s=Asterisk PBX 15.5.0
c=IN IP4 192.168.2.164
t=0 0
m=audio 17550 RTP/AVP 103 101
a=rtpmap:103 opus/48000/2
a=fmtp:103 maxplaybackrate=16000;maxaveragebitrate=24000;useinbandfec=1;usedtx=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=sendrecv

<------------>
    -- SIP/telecom-000000a4 is making progress passing it to SIP/1001-000000a3

<--- SIP read from UDP:5.97.52.7:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.164:5060;received=87.4.160.7;branch=z9hG4bK4c76155a;rport=5060
From: <sip:+39040633324@telecomitalia.it>;tag=as3c495127
To: <sip:0039030820276@telecomitalia.it>;tag=p65545t1536321926m180505c1119068138s1_3534468296-1265672840
Call-ID: 4ce701190ceee0731a300e6d4abf6c83@telecomitalia.it
CSeq: 103 INVITE
Content-Length: 252
Contact: <sip:p65545t1536321926m180505c1119068138s1@5.97.52.7:5060;transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Content-Type: application/sdp
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, INFO, PUBLISH, PRACK, UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE
Accept: application/sdp
Accept: application/isup
Accept: application/xml
Supported: timer
Session-Expires: 360;refresher=uas
Server: Ericsson MTAS - CXP9020729/8 R8H01
Authentication-Info: qop=auth,rspauth="a687c030a5f9797b62991fa50b9a87cf",cnonce="751c6be8",nc=00000001
Session-ID: 88e74e711d26b60ba6ca7fee3574a4da

v=0
o=- 1291177671 3535369597 IN IP4 5.97.52.7
s=IMSS
c=IN IP4 5.97.52.7
t=0 0
m=audio 54102 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=sqn:0
a=cdsc: 1 image udptl t38
a=sendrecv
a=ptime:20
<------------->
--- (18 headers 13 lines) ---
sip_route_dump: route/path hop: <sip:p65545t1536321926m180505c1119068138s1@5.97.52.7:5060;transport=udp>
Transmitting (NAT) to 5.97.52.7:5060:
ACK sip:p65545t1536321926m180505c1119068138s1@5.97.52.7:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.2.164:5060;branch=z9hG4bK20d2a25a;rport
Max-Forwards: 70
From: <sip:+39040633324@telecomitalia.it>;tag=as3c495127
To: <sip:0039030820276@telecomitalia.it>;tag=p65545t1536321926m180505c1119068138s1_3534468296-1265672840
Contact: <sip:+39040633324@192.168.2.164:5060>
Call-ID: 4ce701190ceee0731a300e6d4abf6c83@telecomitalia.it
CSeq: 103 ACK
User-Agent: Asterisk PBX 15.5.0
Content-Length: 0


---
    -- SIP/telecom-000000a4 answered SIP/1001-000000a3
Audio is at 17550
Adding codec opus to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.2.6:8204 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.6:8204;branch=z9hG4bKBgoWg8Jzoiuf06L6;received=192.168.2.6;rport=8204
From: <sip:1001@192.168.2.164>;tag=06E716FB65F2B6298D1665E7FEB872F2
To: <sip:0039030820276@192.168.2.164>;tag=as3e6fab78
Call-ID: AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
CSeq: 2 INVITE
Server: Asterisk PBX 15.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:0039030820276@192.168.2.164:5060>
Content-Type: application/sdp
Content-Length: 327

v=0
o=root 956074400 956074400 IN IP4 192.168.2.164
s=Asterisk PBX 15.5.0
c=IN IP4 192.168.2.164
t=0 0
m=audio 17550 RTP/AVP 103 101
a=rtpmap:103 opus/48000/2
a=fmtp:103 maxplaybackrate=16000;maxaveragebitrate=24000;useinbandfec=1;usedtx=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=sendrecv

<------------>
    -- Channel SIP/telecom-000000a4 joined 'simple_bridge' basic-bridge <3954216e-1805-43ef-aca6-d3f9b8c08a88>
    -- Channel SIP/1001-000000a3 joined 'simple_bridge' basic-bridge <3954216e-1805-43ef-aca6-d3f9b8c08a88>
Retransmitting #1 (no NAT) to 192.168.2.6:8204:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.6:8204;branch=z9hG4bKBgoWg8Jzoiuf06L6;received=192.168.2.6;rport=8204
From: <sip:1001@192.168.2.164>;tag=06E716FB65F2B6298D1665E7FEB872F2
To: <sip:0039030820276@192.168.2.164>;tag=as3e6fab78
Call-ID: AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
CSeq: 2 INVITE
Server: Asterisk PBX 15.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:0039030820276@192.168.2.164:5060>
Content-Type: application/sdp
Content-Length: 327

v=0
o=root 956074400 956074400 IN IP4 192.168.2.164
s=Asterisk PBX 15.5.0
c=IN IP4 192.168.2.164
t=0 0
m=audio 17550 RTP/AVP 103 101
a=rtpmap:103 opus/48000/2
a=fmtp:103 maxplaybackrate=16000;maxaveragebitrate=24000;useinbandfec=1;usedtx=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.2.6:8204 --->
ACK sip:0039030820276@192.168.2.164:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.6:8204;branch=z9hG4bKPbNz0PxfQtHksJES;rport
Max-Forwards: 70
To: <sip:0039030820276@192.168.2.164>;tag=as3e6fab78
From: <sip:1001@192.168.2.164>;tag=06E716FB65F2B6298D1665E7FEB872F2
Call-ID: AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
CSeq: 2 ACK
User-Agent: Acrobits Softphone Business/3.8.2
Content-Length: 0

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

<--- SIP read from UDP:192.168.2.6:8204 --->
ACK sip:0039030820276@192.168.2.164:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.6:8204;branch=z9hG4bKDPNdDfjBnNQ30KGv;rport
Max-Forwards: 70
To: <sip:0039030820276@192.168.2.164>;tag=as3e6fab78
From: <sip:1001@192.168.2.164>;tag=06E716FB65F2B6298D1665E7FEB872F2
Call-ID: AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
CSeq: 2 ACK
User-Agent: Acrobits Softphone Business/3.8.2
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '1544165243@192_168_2_82' Method: REGISTER
[Sep  7 14:05:37] NOTICE[1639]: chan_sip.c:15774 sip_reregister:    -- Re-registration for  +39040633324@5.97.52.7
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 5.97.52.7:5060:
REGISTER sip:telecomitalia.it SIP/2.0
Via: SIP/2.0/UDP 192.168.2.164:5060;branch=z9hG4bK180c2ed4
Max-Forwards: 70
From: <sip:+39040633324@telecomitalia.it>;tag=as048f48ed
To: <sip:+39040633324@telecomitalia.it>
Call-ID: 491e7b5953378a1c0a5c56bc0ea471d9@127.0.1.1
CSeq: 117 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 15.5.0
Authorization: Digest username="+39040633324@telecomitalia.it", realm="telecomitalia.it", algorithm=MD5, uri="sip:telecomitalia.it", nonce="8D8A8A799265925B00000000420F6648", response="834d968bee6ba9c8b5ecf206b527bcff", qop=auth, cnonce="3dc41c8a", nc=00000004
Expires: 120
Contact: <sip:s@192.168.2.164:5060>
Content-Length: 0


---

<--- SIP read from UDP:5.97.52.7:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.164:5060;received=87.4.160.7;branch=z9hG4bK180c2ed4;rport=5060
From: <sip:+39040633324@telecomitalia.it>;tag=as048f48ed
To: <sip:+39040633324@telecomitalia.it>;tag=aprqj9llhta6hpc00-adplhm00000a7
Call-ID: 491e7b5953378a1c0a5c56bc0ea471d9@127.0.1.1
CSeq: 117 REGISTER
P-Associated-URI: <sip:+39040633324@telecomitalia.it>
P-Associated-URI: <tel:+39040633324>
Contact: <sip:s@192.168.2.164:5060>;expires=360


=====here is when the called party hangs up =======
<------------->
--- (9 headers 0 lines) ---

Really destroying SIP dialog '491e7b5953378a1c0a5c56bc0ea471d9@127.0.1.1' Method: REGISTER
Really destroying SIP dialog '171ba9e3541c681b1e34f7c76a2e25c5@192.168.2.164:5060' Method: NOTIFY
Really destroying SIP dialog 'FDC60CD2A61461CCE7F191A6FB957E12A622AA29' Method: REGISTER

<--- SIP read from UDP:192.168.2.6:2180 --->

<------------->
Reliably Transmitting (no NAT) to 192.168.2.6:2180:
OPTIONS sip:1002@192.168.2.6:2180;rinstance=8F92FDA8 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.164:5060;branch=z9hG4bK6792ae4b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.2.164>;tag=as637eb4a8
To: <sip:1002@192.168.2.6:2180;rinstance=8F92FDA8>
Contact: <sip:asterisk@192.168.2.164:5060>
Call-ID: 144fedc803df14cd4737eb563491d224@192.168.2.164:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.5.0
Date: Fri, 07 Sep 2018 12:05:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.2.6:2180 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.164:5060;branch=z9hG4bK6792ae4b
Contact: <sip:1002@192.168.2.6:2180>
From: "asterisk" <sip:asterisk@192.168.2.164>;tag=as637eb4a8
Call-ID: 144fedc803df14cd4737eb563491d224@192.168.2.164:5060
CSeq: 102 OPTIONS
To: <sip:1002@192.168.2.6:2180;rinstance=8F92FDA8>
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
Accept: application/sdp
User-Agent: Acrobits Softphone Business/3.8.2
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '144fedc803df14cd4737eb563491d224@192.168.2.164:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.2.6:8204 --->

<------------->
Reliably Transmitting (no NAT) to 192.168.2.6:8204:
OPTIONS sip:1001@192.168.2.6:8204;rinstance=8F92FDA8 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.164:5060;branch=z9hG4bK6337c8d7
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.2.164>;tag=as3606b45f
To: <sip:1001@192.168.2.6:8204;rinstance=8F92FDA8>
Contact: <sip:asterisk@192.168.2.164:5060>
Call-ID: 0d93457537d788de6266d4be70a946dc@192.168.2.164:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.5.0
Date: Fri, 07 Sep 2018 12:06:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.2.6:8204 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.164:5060;branch=z9hG4bK6337c8d7
Contact: <sip:1001@192.168.2.6:8204>
From: "asterisk" <sip:asterisk@192.168.2.164>;tag=as3606b45f
Call-ID: 0d93457537d788de6266d4be70a946dc@192.168.2.164:5060
CSeq: 102 OPTIONS
To: <sip:1001@192.168.2.6:8204;rinstance=8F92FDA8>
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
Accept: application/sdp
User-Agent: Acrobits Softphone Business/3.8.2
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '0d93457537d788de6266d4be70a946dc@192.168.2.164:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.2.6:2180 --->

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

<--- SIP read from UDP:5.97.52.7:5060 --->
OPTIONS sip:+39040633324@192.168.2.164:5060 SIP/2.0
Via: SIP/2.0/UDP 5.97.52.7:5060;branch=z9hG4bKjgs0j80030aa4jmtakf0sh00000k1.1
To: <sip:+39040633324@telecomitalia.it>;tag=as3c495127
From: <sip:0039030820276@telecomitalia.it>;tag=p65545t1536321926m180505c1119068138s1_3534468296-1265672840
Call-ID: 4ce701190ceee0731a300e6d4abf6c83@telecomitalia.it
CSeq: 104 OPTIONS
Max-Forwards: 66
Content-Length: 0
Contact: <sip:p65545t1536321926m180505c1119068138s1@5.97.52.7:5060;transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, INFO, PUBLISH, PRACK, UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE
User-Agent: Ericsson MTAS - CXP9020729/8 R8H01

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

<--- Transmitting (NAT) to 5.97.52.7:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 5.97.52.7:5060;branch=z9hG4bKjgs0j80030aa4jmtakf0sh00000k1.1;received=5.97.52.7;rport=5060
From: <sip:0039030820276@telecomitalia.it>;tag=p65545t1536321926m180505c1119068138s1_3534468296-1265672840
To: <sip:+39040633324@telecomitalia.it>;tag=as3c495127
Call-ID: 4ce701190ceee0731a300e6d4abf6c83@telecomitalia.it
CSeq: 104 OPTIONS
Server: Asterisk PBX 15.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:+39040633324@192.168.2.164:5060>
Accept: application/sdp
Content-Length: 0


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

<--- SIP read from UDP:192.168.2.6:8204 --->

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

<--- SIP read from UDP:192.168.2.6:2180 --->

<------------->
Reliably Transmitting (no NAT) to 192.168.2.6:2180:
OPTIONS sip:1002@192.168.2.6:2180;rinstance=8F92FDA8 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.164:5060;branch=z9hG4bK3d596f0a
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.2.164>;tag=as27949d30
To: <sip:1002@192.168.2.6:2180;rinstance=8F92FDA8>
Contact: <sip:asterisk@192.168.2.164:5060>
Call-ID: 07a8c2e12aa71e4c401b089a58fb911b@192.168.2.164:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.5.0
Date: Fri, 07 Sep 2018 12:06:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.2.6:2180 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.164:5060;branch=z9hG4bK3d596f0a
Contact: <sip:1002@192.168.2.6:2180>
From: "asterisk" <sip:asterisk@192.168.2.164>;tag=as27949d30
Call-ID: 07a8c2e12aa71e4c401b089a58fb911b@192.168.2.164:5060
CSeq: 102 OPTIONS
To: <sip:1002@192.168.2.6:2180;rinstance=8F92FDA8>
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
Accept: application/sdp
User-Agent: Acrobits Softphone Business/3.8.2
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '07a8c2e12aa71e4c401b089a58fb911b@192.168.2.164:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.2.6:8204 --->

<------------->
Reliably Transmitting (no NAT) to 192.168.2.6:8204:
OPTIONS sip:1001@192.168.2.6:8204;rinstance=8F92FDA8 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.164:5060;branch=z9hG4bK328868b4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.2.164>;tag=as407723a8
To: <sip:1001@192.168.2.6:8204;rinstance=8F92FDA8>
Contact: <sip:asterisk@192.168.2.164:5060>
Call-ID: 7bf036be61e86c755b94c0f108804970@192.168.2.164:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.5.0
Date: Fri, 07 Sep 2018 12:07:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.2.6:8204 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.164:5060;branch=z9hG4bK328868b4
Contact: <sip:1001@192.168.2.6:8204>
From: "asterisk" <sip:asterisk@192.168.2.164>;tag=as407723a8
Call-ID: 7bf036be61e86c755b94c0f108804970@192.168.2.164:5060
CSeq: 102 OPTIONS
To: <sip:1001@192.168.2.6:8204;rinstance=8F92FDA8>
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
Accept: application/sdp
User-Agent: Acrobits Softphone Business/3.8.2
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '7bf036be61e86c755b94c0f108804970@192.168.2.164:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.2.6:2180 --->

Matej86 · September 7, 2018, 12:15pm

<— SIP read from UDP:192.168.2.6:8204 —>
INVITE sip:0039030820276@192.168.2.164 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.6:8204;branch=z9hG4bKGR1IHMdE94jpspai;rport
Contact: sip:1001@192.168.2.6:8204
Max-Forwards: 70
From: sip:1001@192.168.2.164;tag=06E716FB65F2B6298D1665E7FEB872F2
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
To: sip:0039030820276@192.168.2.164
Content-Type: application/sdp
Call-ID: AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
CSeq: 1 INVITE
User-Agent: Acrobits Softphone Business/3.8.2
Content-Length: 305

v=0
o=- 4858458339 26001 IN IP4 172.26.170.170
s=xfppwij
c=IN IP4 192.168.2.6
t=0 0
m=audio 18262 RTP/AVP 103 101
a=rtpmap:101 telephone-event/8000
a=rtpmap:103 opus/48000/2
a=fmtp:101 0-15
a=fmtp:103 maxplaybackrate=16000;maxaveragebitrate=24000;useinbandfec=1;usedtx=1
a=ptime:20
a=sendrecv
<------------->
— (13 headers 12 lines) —
Sending to 192.168.2.6:8204 (no NAT)
Sending to 192.168.2.6:8204 (no NAT)
Using INVITE request as basis request - AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
Found peer ‘1001’ for ‘1001’ from 192.168.2.6:8204

<— Reliably Transmitting (no NAT) to 192.168.2.6:8204 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.6:8204;branch=z9hG4bKGR1IHMdE94jpspai;received=192.168.2.6;rport=8204
From: sip:1001@192.168.2.164;tag=06E716FB65F2B6298D1665E7FEB872F2
To: sip:0039030820276@192.168.2.164;tag=as322b6c5a
Call-ID: AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
CSeq: 1 INVITE
Server: Asterisk PBX 15.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“7e9026d3”
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893’ in 12672 ms (Method: INVITE)
Retransmitting #1 (no NAT) to 192.168.2.6:8204:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.6:8204;branch=z9hG4bKGR1IHMdE94jpspai;received=192.168.2.6;rport=8204
From: sip:1001@192.168.2.164;tag=06E716FB65F2B6298D1665E7FEB872F2
To: sip:0039030820276@192.168.2.164;tag=as322b6c5a
Call-ID: AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
CSeq: 1 INVITE
Server: Asterisk PBX 15.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“7e9026d3”
Content-Length: 0

<— SIP read from UDP:192.168.2.6:8204 —>
ACK sip:0039030820276@192.168.2.164 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.6:8204;branch=z9hG4bKGR1IHMdE94jpspai;rport
Max-Forwards: 70
Call-ID: AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
From: sip:1001@192.168.2.164;tag=06E716FB65F2B6298D1665E7FEB872F2
To: sip:0039030820276@192.168.2.164;tag=as322b6c5a
CSeq: 1 ACK
User-Agent: Acrobits Softphone Business/3.8.2
Content-Length: 0

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

<— SIP read from UDP:192.168.2.6:8204 —>
INVITE sip:0039030820276@192.168.2.164 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.6:8204;branch=z9hG4bKBgoWg8Jzoiuf06L6;rport
Contact: sip:1001@192.168.2.6:8204
Max-Forwards: 70
From: sip:1001@192.168.2.164;tag=06E716FB65F2B6298D1665E7FEB872F2
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
To: sip:0039030820276@192.168.2.164
Content-Type: application/sdp
Call-ID: AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
CSeq: 2 INVITE
Authorization: Digest username=“1001”,realm=“asterisk”,algorithm=MD5,uri="sip:0039030820276@192.168.2.164",nonce=“7e9026d3”,response=“465a33f53c8d6a8042cded28a05306f6”
User-Agent: Acrobits Softphone Business/3.8.2
Content-Length: 305

<— Transmitting (no NAT) to 192.168.2.6:8204 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.6:8204;branch=z9hG4bKBgoWg8Jzoiuf06L6;received=192.168.2.6;rport=8204
From: sip:1001@192.168.2.164;tag=06E716FB65F2B6298D1665E7FEB872F2
To: sip:0039030820276@192.168.2.164
Call-ID: AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
CSeq: 2 INVITE
Server: Asterisk PBX 15.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:0039030820276@192.168.2.164:5060
Content-Length: 0

<------------>
– Executing [0039030820276@internal:1] Log(“SIP/1001-000000a3”, “NOTICE, Dialing out from “” <1001> to 039030820276 through TIM”) in new stack
[Sep 7 14:05:26] NOTICE[5257][C-00000053]: Ext. 0039030820276:1 @ internal: Dialing out from “” <1001> to 039030820276 through TIM
– Executing [0039030820276@internal:2] Dial(“SIP/1001-000000a3”, “SIP/telecom/0039030820276,60”) in new stack
== Using SIP RTP CoS mark 5
Audio is at 16692
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 (NAT) to 5.97.52.7:5060:
INVITE sip:0039030820276@telecomitalia.it SIP/2.0
Via: SIP/2.0/UDP 192.168.2.164:5060;branch=z9hG4bK60a042bd;rport
Max-Forwards: 70
From: sip:+39040633324@telecomitalia.it;tag=as3c495127
To: sip:0039030820276@telecomitalia.it
Contact: sip:+39040633324@192.168.2.164:5060
Call-ID: 4ce701190ceee0731a300e6d4abf6c83@telecomitalia.it
CSeq: 102 INVITE
User-Agent: Asterisk PBX 15.5.0
Date: Fri, 07 Sep 2018 12:05:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 953731005 953731005 IN IP4 192.168.2.164
s=Asterisk PBX 15.5.0
c=IN IP4 192.168.2.164
t=0 0
m=audio 16692 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/telecom/0039030820276

<— SIP read from UDP:5.97.52.7:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.164:5060;received=87.4.160.7;branch=z9hG4bK60a042bd;rport=5060
From: sip:+39040633324@telecomitalia.it;tag=as3c495127
To: sip:0039030820276@telecomitalia.it
Call-ID: 4ce701190ceee0731a300e6d4abf6c83@telecomitalia.it
CSeq: 102 INVITE

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

<— SIP read from UDP:5.97.52.7:5060 —>
SIP/2.0 407 Proxy Authentication Required 02035033D
Via: SIP/2.0/UDP 192.168.2.164:5060;received=87.4.160.7;branch=z9hG4bK60a042bd;rport=5060
From: sip:+39040633324@telecomitalia.it;tag=as3c495127
To: sip:0039030820276@telecomitalia.it;tag=ef3e51940fc70645a0d2c055267b02b2
Call-ID: 4ce701190ceee0731a300e6d4abf6c83@telecomitalia.it
CSeq: 102 INVITE
Content-Length: 0
Proxy-Authenticate: Digest nonce=“B60D101C9269925B0000000084E41472”,realm=“telecomitalia.it”,algorithm=MD5,qop=“auth”

<------------->
— (8 headers 0 lines) —
Transmitting (NAT) to 5.97.52.7:5060:
ACK sip:0039030820276@telecomitalia.it SIP/2.0
Via: SIP/2.0/UDP 192.168.2.164:5060;branch=z9hG4bK60a042bd;rport
Max-Forwards: 70
From: sip:+39040633324@telecomitalia.it;tag=as3c495127
To: sip:0039030820276@telecomitalia.it;tag=ef3e51940fc70645a0d2c055267b02b2
Contact: sip:+39040633324@192.168.2.164:5060
Call-ID: 4ce701190ceee0731a300e6d4abf6c83@telecomitalia.it
CSeq: 102 ACK
User-Agent: Asterisk PBX 15.5.0
Content-Length: 0

Audio is at 16692
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 (NAT) to 5.97.52.7:5060:
INVITE sip:0039030820276@telecomitalia.it SIP/2.0
Via: SIP/2.0/UDP 192.168.2.164:5060;branch=z9hG4bK4c76155a;rport
Max-Forwards: 70
From: sip:+39040633324@telecomitalia.it;tag=as3c495127
To: sip:0039030820276@telecomitalia.it
Contact: sip:+39040633324@192.168.2.164:5060
Call-ID: 4ce701190ceee0731a300e6d4abf6c83@telecomitalia.it
CSeq: 103 INVITE
User-Agent: Asterisk PBX 15.5.0
Proxy-Authorization: Digest username="+39040633324", realm=“telecomitalia.it”, algorithm=MD5, uri=“sip:0039030820276@telecomitalia.it”, nonce=“B60D101C9269925B0000000084E41472”, response=“fd232b6aba4e715cfbdf75c869274848”, qop=auth, cnonce=“751c6be8”, nc=00000001
Date: Fri, 07 Sep 2018 12:05:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 953731005 953731006 IN IP4 192.168.2.164
s=Asterisk PBX 15.5.0
c=IN IP4 192.168.2.164
t=0 0
m=audio 16692 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

<— SIP read from UDP:5.97.52.7:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.164:5060;received=87.4.160.7;branch=z9hG4bK4c76155a;rport=5060
From: sip:+39040633324@telecomitalia.it;tag=as3c495127
To: sip:0039030820276@telecomitalia.it
Call-ID: 4ce701190ceee0731a300e6d4abf6c83@telecomitalia.it
CSeq: 103 INVITE

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

<— SIP read from UDP:192.168.2.6:8204 —>
ACK sip:0039030820276@192.168.2.164 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.6:8204;branch=z9hG4bKGR1IHMdE94jpspai;rport
Max-Forwards: 70
Call-ID: AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
From: sip:1001@192.168.2.164;tag=06E716FB65F2B6298D1665E7FEB872F2
To: sip:0039030820276@192.168.2.164;tag=as322b6c5a
CSeq: 1 ACK
User-Agent: Acrobits Softphone Business/3.8.2
Content-Length: 0

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

<— SIP read from UDP:5.97.52.7:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.2.164:5060;received=87.4.160.7;branch=z9hG4bK4c76155a;rport=5060
From: sip:+39040633324@telecomitalia.it;tag=as3c495127
To: sip:0039030820276@telecomitalia.it;tag=p65545t1536321926m180505c1119068138s1_3534468296-1265672840
Call-ID: 4ce701190ceee0731a300e6d4abf6c83@telecomitalia.it
CSeq: 103 INVITE
Content-Length: 252
Contact: sip:p65545t1536321926m180505c1119068138s1@5.97.52.7:5060;transport=udp
Content-Type: application/sdp
Allow: UPDATE, PRACK, REFER, NOTIFY, OPTIONS, BYE, ACK, CANCEL, INVITE, REGISTER
Accept: application/sdp
Accept: application/isup
Accept: application/xml
Supported: timer
Server: Ericsson MTAS - CXP9020729/8 R8H01

v=0
o=- 1291177671 3535369597 IN IP4 5.97.52.7
s=IMSS
c=IN IP4 5.97.52.7
t=0 0
m=audio 54102 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=sqn:0
a=cdsc: 1 image udptl t38
a=sendrecv
a=ptime:20
<------------->
— (15 headers 13 lines) —
sip_route_dump: route/path hop: sip:p65545t1536321926m180505c1119068138s1@5.97.52.7:5060;transport=udp
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|h263), 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 5.97.52.7:54102
– SIP/telecom-000000a4 is ringing

<— Transmitting (no NAT) to 192.168.2.6:8204 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.2.6:8204;branch=z9hG4bKBgoWg8Jzoiuf06L6;received=192.168.2.6;rport=8204
From: sip:1001@192.168.2.164;tag=06E716FB65F2B6298D1665E7FEB872F2
To: sip:0039030820276@192.168.2.164;tag=as3e6fab78
Call-ID: AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
CSeq: 2 INVITE
Server: Asterisk PBX 15.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:0039030820276@192.168.2.164:5060
Content-Length: 0

<------------>
Audio is at 17550
Adding codec opus to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (no NAT) to 192.168.2.6:8204 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.2.6:8204;branch=z9hG4bKBgoWg8Jzoiuf06L6;received=192.168.2.6;rport=8204
From: sip:1001@192.168.2.164;tag=06E716FB65F2B6298D1665E7FEB872F2
To: sip:0039030820276@192.168.2.164;tag=as3e6fab78
Call-ID: AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
CSeq: 2 INVITE
Server: Asterisk PBX 15.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:0039030820276@192.168.2.164:5060
Content-Type: application/sdp
Content-Length: 327

v=0
o=root 956074400 956074400 IN IP4 192.168.2.164
s=Asterisk PBX 15.5.0
c=IN IP4 192.168.2.164
t=0 0
m=audio 17550 RTP/AVP 103 101
a=rtpmap:103 opus/48000/2
a=fmtp:103 maxplaybackrate=16000;maxaveragebitrate=24000;useinbandfec=1;usedtx=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=sendrecv

<------------>
– SIP/telecom-000000a4 is making progress passing it to SIP/1001-000000a3

<— SIP read from UDP:5.97.52.7:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.164:5060;received=87.4.160.7;branch=z9hG4bK4c76155a;rport=5060
From: sip:+39040633324@telecomitalia.it;tag=as3c495127
To: sip:0039030820276@telecomitalia.it;tag=p65545t1536321926m180505c1119068138s1_3534468296-1265672840
Call-ID: 4ce701190ceee0731a300e6d4abf6c83@telecomitalia.it
CSeq: 103 INVITE
Content-Length: 252
Contact: sip:p65545t1536321926m180505c1119068138s1@5.97.52.7:5060;transport=udp;+g.3gpp.icsi-ref=“urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel”
Content-Type: application/sdp
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, INFO, PUBLISH, PRACK, UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE
Accept: application/sdp
Accept: application/isup
Accept: application/xml
Supported: timer
Session-Expires: 360;refresher=uas
Server: Ericsson MTAS - CXP9020729/8 R8H01
Authentication-Info: qop=auth,rspauth=“a687c030a5f9797b62991fa50b9a87cf”,cnonce=“751c6be8”,nc=00000001
Session-ID: 88e74e711d26b60ba6ca7fee3574a4da

v=0
o=- 1291177671 3535369597 IN IP4 5.97.52.7
s=IMSS
c=IN IP4 5.97.52.7
t=0 0
m=audio 54102 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=sqn:0
a=cdsc: 1 image udptl t38
a=sendrecv
a=ptime:20
<------------->
— (18 headers 13 lines) —
sip_route_dump: route/path hop: sip:p65545t1536321926m180505c1119068138s1@5.97.52.7:5060;transport=udp
Transmitting (NAT) to 5.97.52.7:5060:
ACK sip:p65545t1536321926m180505c1119068138s1@5.97.52.7:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.2.164:5060;branch=z9hG4bK20d2a25a;rport
Max-Forwards: 70
From: sip:+39040633324@telecomitalia.it;tag=as3c495127
To: sip:0039030820276@telecomitalia.it;tag=p65545t1536321926m180505c1119068138s1_3534468296-1265672840
Contact: sip:+39040633324@192.168.2.164:5060
Call-ID: 4ce701190ceee0731a300e6d4abf6c83@telecomitalia.it
CSeq: 103 ACK
User-Agent: Asterisk PBX 15.5.0
Content-Length: 0

-- SIP/telecom-000000a4 answered SIP/1001-000000a3

Audio is at 17550
Adding codec opus to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 192.168.2.6:8204 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.6:8204;branch=z9hG4bKBgoWg8Jzoiuf06L6;received=192.168.2.6;rport=8204
From: sip:1001@192.168.2.164;tag=06E716FB65F2B6298D1665E7FEB872F2
To: sip:0039030820276@192.168.2.164;tag=as3e6fab78
Call-ID: AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
CSeq: 2 INVITE
Server: Asterisk PBX 15.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:0039030820276@192.168.2.164:5060
Content-Type: application/sdp
Content-Length: 327

v=0
o=root 956074400 956074400 IN IP4 192.168.2.164
s=Asterisk PBX 15.5.0
c=IN IP4 192.168.2.164
t=0 0
m=audio 17550 RTP/AVP 103 101
a=rtpmap:103 opus/48000/2
a=fmtp:103 maxplaybackrate=16000;maxaveragebitrate=24000;useinbandfec=1;usedtx=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=sendrecv

<------------>
– Channel SIP/telecom-000000a4 joined ‘simple_bridge’ basic-bridge <3954216e-1805-43ef-aca6-d3f9b8c08a88>
– Channel SIP/1001-000000a3 joined ‘simple_bridge’ basic-bridge <3954216e-1805-43ef-aca6-d3f9b8c08a88>
Retransmitting #1 (no NAT) to 192.168.2.6:8204:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.6:8204;branch=z9hG4bKBgoWg8Jzoiuf06L6;received=192.168.2.6;rport=8204
From: sip:1001@192.168.2.164;tag=06E716FB65F2B6298D1665E7FEB872F2
To: sip:0039030820276@192.168.2.164;tag=as3e6fab78
Call-ID: AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
CSeq: 2 INVITE
Server: Asterisk PBX 15.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:0039030820276@192.168.2.164:5060
Content-Type: application/sdp
Content-Length: 327

v=0
o=root 956074400 956074400 IN IP4 192.168.2.164
s=Asterisk PBX 15.5.0
c=IN IP4 192.168.2.164
t=0 0
m=audio 17550 RTP/AVP 103 101
a=rtpmap:103 opus/48000/2
a=fmtp:103 maxplaybackrate=16000;maxaveragebitrate=24000;useinbandfec=1;usedtx=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=sendrecv

<— SIP read from UDP:192.168.2.6:8204 —>
ACK sip:0039030820276@192.168.2.164:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.6:8204;branch=z9hG4bKPbNz0PxfQtHksJES;rport
Max-Forwards: 70
To: sip:0039030820276@192.168.2.164;tag=as3e6fab78
From: sip:1001@192.168.2.164;tag=06E716FB65F2B6298D1665E7FEB872F2
Call-ID: AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
CSeq: 2 ACK
User-Agent: Acrobits Softphone Business/3.8.2
Content-Length: 0

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

<— SIP read from UDP:192.168.2.6:8204 —>
ACK sip:0039030820276@192.168.2.164:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.6:8204;branch=z9hG4bKDPNdDfjBnNQ30KGv;rport
Max-Forwards: 70
To: sip:0039030820276@192.168.2.164;tag=as3e6fab78
From: sip:1001@192.168.2.164;tag=06E716FB65F2B6298D1665E7FEB872F2
Call-ID: AE22BD21BB065E75F3FB36E5D6B4DFB7AF27A893
CSeq: 2 ACK
User-Agent: Acrobits Softphone Business/3.8.2
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘1544165243@192_168_2_82’ Method: REGISTER
[Sep 7 14:05:37] NOTICE[1639]: chan_sip.c:15774 sip_reregister: – Re-registration for +39040633324@5.97.52.7
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 5.97.52.7:5060:
REGISTER sip:telecomitalia.it SIP/2.0
Via: SIP/2.0/UDP 192.168.2.164:5060;branch=z9hG4bK180c2ed4
Max-Forwards: 70
From: sip:+39040633324@telecomitalia.it;tag=as048f48ed
To: sip:+39040633324@telecomitalia.it
Call-ID: 491e7b5953378a1c0a5c56bc0ea471d9@127.0.1.1
CSeq: 117 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 15.5.0
Authorization: Digest username="+39040633324@telecomitalia.it", realm=“telecomitalia.it”, algorithm=MD5, uri=“sip:telecomitalia.it”, nonce=“8D8A8A799265925B00000000420F6648”, response=“834d968bee6ba9c8b5ecf206b527bcff”, qop=auth, cnonce=“3dc41c8a”, nc=00000004
Expires: 120
Contact: sip:s@192.168.2.164:5060
Content-Length: 0

<— SIP read from UDP:5.97.52.7:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.164:5060;received=87.4.160.7;branch=z9hG4bK180c2ed4;rport=5060
From: sip:+39040633324@telecomitalia.it;tag=as048f48ed
To: sip:+39040633324@telecomitalia.it;tag=aprqj9llhta6hpc00-adplhm00000a7
Call-ID: 491e7b5953378a1c0a5c56bc0ea471d9@127.0.1.1
CSeq: 117 REGISTER
P-Associated-URI: sip:+39040633324@telecomitalia.it
P-Associated-URI: tel:+39040633324
Contact: sip:s@192.168.2.164:5060;expires=360

=====here is when the called party hangs up =======
<------------->
— (9 headers 0 lines) —

Really destroying SIP dialog ‘491e7b5953378a1c0a5c56bc0ea471d9@127.0.1.1’ Method: REGISTER
Really destroying SIP dialog ‘171ba9e3541c681b1e34f7c76a2e25c5@192.168.2.164:5060’ Method: NOTIFY
Really destroying SIP dialog ‘FDC60CD2A61461CCE7F191A6FB957E12A622AA29’ Method: REGISTER

<— SIP read from UDP:192.168.2.6:2180 —>

<------------->
Reliably Transmitting (no NAT) to 192.168.2.6:2180:
OPTIONS sip:1002@192.168.2.6:2180;rinstance=8F92FDA8 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.164:5060;branch=z9hG4bK6792ae4b
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.2.164;tag=as637eb4a8
To: sip:1002@192.168.2.6:2180;rinstance=8F92FDA8
Contact: sip:asterisk@192.168.2.164:5060
Call-ID: 144fedc803df14cd4737eb563491d224@192.168.2.164:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.5.0
Date: Fri, 07 Sep 2018 12:05:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.2.6:2180 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.164:5060;branch=z9hG4bK6792ae4b
Contact: sip:1002@192.168.2.6:2180
From: “asterisk” sip:asterisk@192.168.2.164;tag=as637eb4a8
Call-ID: 144fedc803df14cd4737eb563491d224@192.168.2.164:5060
CSeq: 102 OPTIONS
To: sip:1002@192.168.2.6:2180;rinstance=8F92FDA8
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
Accept: application/sdp
User-Agent: Acrobits Softphone Business/3.8.2
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog ‘144fedc803df14cd4737eb563491d224@192.168.2.164:5060’ Method: OPTIONS

<— SIP read from UDP:192.168.2.6:8204 —>

<------------->
Reliably Transmitting (no NAT) to 192.168.2.6:8204:
OPTIONS sip:1001@192.168.2.6:8204;rinstance=8F92FDA8 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.164:5060;branch=z9hG4bK6337c8d7
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.2.164;tag=as3606b45f
To: sip:1001@192.168.2.6:8204;rinstance=8F92FDA8
Contact: sip:asterisk@192.168.2.164:5060
Call-ID: 0d93457537d788de6266d4be70a946dc@192.168.2.164:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.5.0
Date: Fri, 07 Sep 2018 12:06:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.2.6:8204 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.164:5060;branch=z9hG4bK6337c8d7
Contact: sip:1001@192.168.2.6:8204
From: “asterisk” sip:asterisk@192.168.2.164;tag=as3606b45f
Call-ID: 0d93457537d788de6266d4be70a946dc@192.168.2.164:5060
CSeq: 102 OPTIONS
To: sip:1001@192.168.2.6:8204;rinstance=8F92FDA8
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
Accept: application/sdp
User-Agent: Acrobits Softphone Business/3.8.2
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog ‘0d93457537d788de6266d4be70a946dc@192.168.2.164:5060’ Method: OPTIONS

<— SIP read from UDP:192.168.2.6:2180 —>

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

<— SIP read from UDP:5.97.52.7:5060 —>
OPTIONS sip:+39040633324@192.168.2.164:5060 SIP/2.0
Via: SIP/2.0/UDP 5.97.52.7:5060;branch=z9hG4bKjgs0j80030aa4jmtakf0sh00000k1.1
To: sip:+39040633324@telecomitalia.it;tag=as3c495127
From: sip:0039030820276@telecomitalia.it;tag=p65545t1536321926m180505c1119068138s1_3534468296-1265672840
Call-ID: 4ce701190ceee0731a300e6d4abf6c83@telecomitalia.it
CSeq: 104 OPTIONS
Max-Forwards: 66
Content-Length: 0
Contact: sip:p65545t1536321926m180505c1119068138s1@5.97.52.7:5060;transport=udp;+g.3gpp.icsi-ref=“urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel”
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, INFO, PUBLISH, PRACK, UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE
User-Agent: Ericsson MTAS - CXP9020729/8 R8H01

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

<— Transmitting (NAT) to 5.97.52.7:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 5.97.52.7:5060;branch=z9hG4bKjgs0j80030aa4jmtakf0sh00000k1.1;received=5.97.52.7;rport=5060
From: sip:0039030820276@telecomitalia.it;tag=p65545t1536321926m180505c1119068138s1_3534468296-1265672840
To: sip:+39040633324@telecomitalia.it;tag=as3c495127
Call-ID: 4ce701190ceee0731a300e6d4abf6c83@telecomitalia.it
CSeq: 104 OPTIONS
Server: Asterisk PBX 15.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:+39040633324@192.168.2.164:5060
Accept: application/sdp
Content-Length: 0

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

<— SIP read from UDP:192.168.2.6:8204 —>

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

<— SIP read from UDP:192.168.2.6:2180 —>

<------------->
Reliably Transmitting (no NAT) to 192.168.2.6:2180:
OPTIONS sip:1002@192.168.2.6:2180;rinstance=8F92FDA8 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.164:5060;branch=z9hG4bK3d596f0a
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.2.164;tag=as27949d30
To: sip:1002@192.168.2.6:2180;rinstance=8F92FDA8
Contact: sip:asterisk@192.168.2.164:5060
Call-ID: 07a8c2e12aa71e4c401b089a58fb911b@192.168.2.164:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.5.0
Date: Fri, 07 Sep 2018 12:06:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.2.6:2180 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.164:5060;branch=z9hG4bK3d596f0a
Contact: sip:1002@192.168.2.6:2180
From: “asterisk” sip:asterisk@192.168.2.164;tag=as27949d30
Call-ID: 07a8c2e12aa71e4c401b089a58fb911b@192.168.2.164:5060
CSeq: 102 OPTIONS
To: sip:1002@192.168.2.6:2180;rinstance=8F92FDA8
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
Accept: application/sdp
User-Agent: Acrobits Softphone Business/3.8.2
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog ‘07a8c2e12aa71e4c401b089a58fb911b@192.168.2.164:5060’ Method: OPTIONS

<— SIP read from UDP:192.168.2.6:8204 —>

<------------->
Reliably Transmitting (no NAT) to 192.168.2.6:8204:
OPTIONS sip:1001@192.168.2.6:8204;rinstance=8F92FDA8 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.164:5060;branch=z9hG4bK328868b4
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.2.164;tag=as407723a8
To: sip:1001@192.168.2.6:8204;rinstance=8F92FDA8
Contact: sip:asterisk@192.168.2.164:5060
Call-ID: 7bf036be61e86c755b94c0f108804970@192.168.2.164:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.5.0
Date: Fri, 07 Sep 2018 12:07:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.2.6:8204 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.164:5060;branch=z9hG4bK328868b4
Contact: sip:1001@192.168.2.6:8204
From: “asterisk” sip:asterisk@192.168.2.164;tag=as407723a8
Call-ID: 7bf036be61e86c755b94c0f108804970@192.168.2.164:5060
CSeq: 102 OPTIONS
To: sip:1001@192.168.2.6:8204;rinstance=8F92FDA8
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
Accept: application/sdp
User-Agent: Acrobits Softphone Business/3.8.2
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog ‘7bf036be61e86c755b94c0f108804970@192.168.2.164:5060’ Method: OPTIONS

<— SIP read from UDP:192.168.2.6:2180 —>

jcolp · September 7, 2018, 12:15pm

There is no indication from the remote side that the person has hung up.

Matej86 · September 7, 2018, 12:19pm

You are completely right, it is making me crazy. It seems that no indication from the landline is reaching my pbx, while it is if I call a mobile number using the same trunk :(((((

Moreover, if I can receive all calls from the landline and mobile numbers to my pbx, I suppose that the incoming connection is working and there is no NAT issues

david551 · September 7, 2018, 12:30pm

Mobile phones are, essentially, ISDN devices, so can generate both CLEAR and RELEASE indications. An analogue landline can only generate one type of indication, which is treated as CLEAR, to allow a subsequent REANSWER when an alternative phone is picked up.

Traditionally, the network wouldn’t infer a RELEASE until after about 3 minutes, although most UK lines now only use a few seconds.

SIP gateways generally only clear the call on RELEASE, although some seem to treat CLEAR as being a hold indications.

For someone receiving the call on an analogue line, there would be no technical indication of a CLEAR at all; it would have to be inferred from what the caller said, or from the sound of the handset being put down.

Mobile network air interface time is a scarce, shared resource, and mobile users can’t just pick up on another phone, so the mobile networks send RELEASE immediately.

Matej86 · September 7, 2018, 12:40pm

Thank you David for your explanation. I really didn’t how the mobile differs from analogue and how it all works.

So, what do you think, it is only a provider fault or is there something I can do on asterisk side? When the called party hangup, I cannot hear nothing on my side and after 2/3 minutes, the call automatically ends.

Blockquote
SIP gateways generally only clear the call on RELEASE, although some seem to treat CLEAR as being a hold indications.

david551 · September 7, 2018, 12:56pm

Only the provider is likely to be able to do anything, but I would say it was the nature of how the PSTN works, not a fault, and you should not design expecting to be able to detect calling party clears.

Matej86 · September 7, 2018, 1:02pm

Alright David, so it seems that there is nothing I can really do I will leave with that

Topic		Replies	Views
Problems hanging up from called Asterisk Support	3	192	December 10, 2013
SIP Trunk not disconnected after call hangup Asterisk Support	5	3092	April 21, 2011
SIP not detecting hangup Asterisk Support	2	237	April 22, 2007
External call not hanging up if caller hangs up while phone is ringing Asterisk SIP	3	1365	November 6, 2020
Hangup aren't detected Asterisk Support	7	247	March 11, 2011

Asterisk does not recognize that the called party has hanged up

Related topics