Call drops after 10-15 secs (I checked other similiar issues here)

Hi,

When I use a remote extension with my RaspPBX server, the call drops after 10-15 secs. Ive set portforwarding of 5060 and RTP from 15000-20000 to the Pbx Server. The one thing i notice in the log is that the response sent from the pbx has Ip address 192.168.1.1 and not the IP Address of the remote extension.

Settings for Extension 500

context=from-internal
canreinvite=no
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=yes
port=5060
qualify=yes
qualifyfreq=60
transport=udp only
avpf=no
icesupport=no
encryption=no


Asterisk SIP Settings
NAT=yes
IP Configuration=DYNAMIC IP
DYNAMIC HOST = asteriskserverdynamichostname
Local Networkds = 192.168.1.0/255.255.255.0
Reinvite Behavious=NO
SRV Lookup=NO


root@raspbx:/var/log/asterisk# asterisk -r
Asterisk 11.6.0, Copyright © 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer [email protected]
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

Connected to Asterisk 11.6.0 currently running on raspbx (pid = 2861)
raspbx*CLI> sip set debug on
SIP Debugging re-enabled

<— SIP read from UDP:192.168.1.1:27652 —>
REGISTER sip:asteriskserverdynamichostname SIP/2.0
Via: SIP/2.0/UDP 125.17.166.115:32897;rport;branch=z9hG4bKPjZMydtTrPUrIfVfQjoZB1FhK.um5EkkNI
Route: sip:asteriskserverdynamichostname;transport=udp;lr
Max-Forwards: 70
From: sip:500@asteriskserverdynamichostname;tag=MnQUwZ1Jdbwot2jrp45A8pqwec22uh00
To: sip:500@asteriskserverdynamichostname
Call-ID: NJ.nsgfam-a99OB-vTU8iu76haQpVU1M
CSeq: 21332 REGISTER
User-Agent: CSipSimple_jflte-18/r2330
Contact: sip:[email protected]:32897;ob;+sip.ice
Expires: 900
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 192.168.1.1:27652 (NAT)
Sending to 192.168.1.1:27652 (NAT)

<— Transmitting (NAT) to 192.168.1.1:27652 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 125.17.166.115:32897;branch=z9hG4bKPjZMydtTrPUrIfVfQjoZB1FhK.um5EkkNI;received=192.168.1.1;rport=27652
From: sip:500@asteriskserverdynamichostname;tag=MnQUwZ1Jdbwot2jrp45A8pqwec22uh00
To: sip:500@asteriskserverdynamichostname;tag=as603a5fee
Call-ID: NJ.nsgfam-a99OB-vTU8iu76haQpVU1M
CSeq: 21332 REGISTER
Server: FPBX-2.11.0(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="7c3f7979"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘NJ.nsgfam-a99OB-vTU8iu76haQpVU1M’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.1.1:27652 —>
REGISTER sip:asteriskserverdynamichostname SIP/2.0
Via: SIP/2.0/UDP 125.17.166.115:32897;rport;branch=z9hG4bKPjHdsCzRLI.7nf1oQ.87bitXtQRbZcfhDf
Route: sip:asteriskserverdynamichostname;transport=udp;lr
Max-Forwards: 70
From: sip:500@asteriskserverdynamichostname;tag=MnQUwZ1Jdbwot2jrp45A8pqwec22uh00
To: sip:500@asteriskserverdynamichostname
Call-ID: NJ.nsgfam-a99OB-vTU8iu76haQpVU1M
CSeq: 21333 REGISTER
User-Agent: CSipSimple_jflte-18/r2330
Contact: sip:[email protected]:32897;ob;+sip.ice
Expires: 900
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username=“500”, realm=“asterisk”, nonce=“7c3f7979”, uri=“sip:asteriskserverdynamichostname”, response=“95cbd5810c2513906b60334b5c82ab84”, algorithm=MD5
Contact: sip:[email protected]:19735;ob;expires=0;+sip.ice
Content-Length: 0

<------------->
— (15 headers 0 lines) —
Sending to 192.168.1.1:27652 (NAT)
Reliably Transmitting (NAT) to 192.168.1.1:27652:
OPTIONS sip:[email protected]:32897;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK12b59b8c;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as2af144ef
To: sip:[email protected]:32897;ob
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.6.0)
Date: Fri, 21 Mar 2014 11:50:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<— Transmitting (NAT) to 192.168.1.1:27652 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 125.17.166.115:32897;branch=z9hG4bKPjHdsCzRLI.7nf1oQ.87bitXtQRbZcfhDf;received=192.168.1.1;rport=27652
From: sip:500@asteriskserverdynamichostname;tag=MnQUwZ1Jdbwot2jrp45A8pqwec22uh00
To: sip:500@asteriskserverdynamichostname;tag=as603a5fee
Call-ID: NJ.nsgfam-a99OB-vTU8iu76haQpVU1M
CSeq: 21333 REGISTER
Server: FPBX-2.11.0(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 900
Contact: sip:[email protected]:32897;ob;expires=900
Date: Fri, 21 Mar 2014 11:50:21 GMT
Content-Length: 0

<------------>
Reliably Transmitting (NAT) to 192.168.1.1:19735:
NOTIFY sip:[email protected]:19735;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK66497d1d;rport
Max-Forwards: 70
Route: sip:[email protected]:19735;ob
From: “Unknown” sip:[email protected];tag=as57741fd7
To: sip:[email protected]:19735;ob;tag=9y5gKgG.qgWohv01uB.BzxREXlBZZ9t8
Contact: sip:[email protected]:5060
Call-ID: Lx1dL0yjgM-A1nMx.1O2WB2BykS4mK0P
CSeq: 103 NOTIFY
User-Agent: FPBX-2.11.0(11.6.0)
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 88

Messages-Waiting: no
Message-Account: sip:*[email protected]
Voice-Message: 0/0 (0/0)


Scheduling destruction of SIP dialog ‘NJ.nsgfam-a99OB-vTU8iu76haQpVU1M’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.1.1:27652 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.147:5060;rport=5060;received=182.19.129.15;branch=z9hG4bK12b59b8c
Call-ID: [email protected]:5060
From: “Unknown” sip:[email protected];tag=as2af144ef
To: sip:[email protected]:32897;ob;tag=z9hG4bK12b59b8c
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: CSipSimple_jflte-18/r2330
Content-Type: application/sdp
Content-Length: 289

v=0
o=- 3604391419 3604391419 IN IP4 125.17.166.115
s=pjmedia
t=0 0
m=audio 33838 RTP/AVP 99 0 8 101
c=IN IP4 125.17.166.115
a=sendrecv
a=rtpmap:99 SILK/24000
a=fmtp:99 useinbandfec=0
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
— (13 headers 13 lines) —
[2014-03-21 19:50:21] NOTICE[2905]: chan_sip.c:23475 handle_response_peerpoke: Peer ‘500’ is now Reachable. (158ms / 2000ms)
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS

<— SIP read from UDP:192.168.1.1:27652 —>
REGISTER sip:asteriskserverdynamichostname SIP/2.0
Via: SIP/2.0/UDP 125.17.166.115:32897;rport;branch=z9hG4bKPjzjDEJlXbwFTFypVmQCaiIHQoJELPzQAq
Route: sip:asteriskserverdynamichostname;transport=udp;lr
Max-Forwards: 70
From: sip:500@asteriskserverdynamichostname;tag=KBsnwzr0QkJ35-CUkSjtcwplpxfd31Zj
To: sip:500@asteriskserverdynamichostname
Call-ID: NJ.nsgfam-a99OB-vTU8iu76haQpVU1M
CSeq: 21334 REGISTER
Authorization: Digest username=“500”, realm=“asterisk”, nonce=“7c3f7979”, uri=“sip:asteriskserverdynamichostname”, response=“95cbd5810c2513906b60334b5c82ab84”, algorithm=MD5
User-Agent: CSipSimple_jflte-18/r2330
Contact: sip:[email protected]:27652;ob;+sip.ice
Contact: sip:[email protected]:32897;ob;expires=0;+sip.ice
Expires: 900
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0

<------------->
— (15 headers 0 lines) —
Sending to 192.168.1.1:27652 (NAT)
[2014-03-21 19:50:21] NOTICE[2905]: chan_sip.c:16419 check_auth: Correct auth, but based on stale nonce received from ‘sip:500@asteriskserverdynamichostname;tag=KBsnwzr0QkJ35-CUkSjtcwplpxfd31Zj’

<— Transmitting (NAT) to 192.168.1.1:27652 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 125.17.166.115:32897;branch=z9hG4bKPjzjDEJlXbwFTFypVmQCaiIHQoJELPzQAq;received=192.168.1.1;rport=27652
From: sip:500@asteriskserverdynamichostname;tag=KBsnwzr0QkJ35-CUkSjtcwplpxfd31Zj
To: sip:500@asteriskserverdynamichostname;tag=as603a5fee
Call-ID: NJ.nsgfam-a99OB-vTU8iu76haQpVU1M
CSeq: 21334 REGISTER
Server: FPBX-2.11.0(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“27fa3e79”, stale=true
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘NJ.nsgfam-a99OB-vTU8iu76haQpVU1M’ in 32000 ms (Method: REGISTER)
Retransmitting #1 (NAT) to 192.168.1.1:19735:
NOTIFY sip:[email protected]:19735;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK66497d1d;rport
Max-Forwards: 70
Route: sip:[email protected]:19735;ob
From: “Unknown” sip:[email protected];tag=as57741fd7
To: sip:[email protected]:19735;ob;tag=9y5gKgG.qgWohv01uB.BzxREXlBZZ9t8
Contact: sip:[email protected]:5060
Call-ID: Lx1dL0yjgM-A1nMx.1O2WB2BykS4mK0P
CSeq: 103 NOTIFY
User-Agent: FPBX-2.11.0(11.6.0)
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 88

Messages-Waiting: no
Message-Account: sip:*[email protected]
Voice-Message: 0/0 (0/0)


<— SIP read from UDP:192.168.1.1:27652 —>
REGISTER sip:asteriskserverdynamichostname SIP/2.0
Via: SIP/2.0/UDP 125.17.166.115:32897;rport;branch=z9hG4bKPjM3eo91pCE2sjgY7B2v1p7loz0XrxY-7w
Route: sip:asteriskserverdynamichostname;transport=udp;lr
Max-Forwards: 70
From: sip:500@asteriskserverdynamichostname;tag=KBsnwzr0QkJ35-CUkSjtcwplpxfd31Zj
To: sip:500@asteriskserverdynamichostname
Call-ID: NJ.nsgfam-a99OB-vTU8iu76haQpVU1M
CSeq: 21335 REGISTER
User-Agent: CSipSimple_jflte-18/r2330
Contact: sip:[email protected]:27652;ob;+sip.ice
Contact: sip:[email protected]:32897;ob;expires=0;+sip.ice
Expires: 900
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username=“500”, realm=“asterisk”, nonce=“27fa3e79”, uri=“sip:asteriskserverdynamichostname”, response=“c23f74f9ec27e0fd9ed7eb14ec94ae81”, algorithm=MD5
Content-Length: 0

<------------->
— (15 headers 0 lines) —
Sending to 192.168.1.1:27652 (NAT)
Reliably Transmitting (NAT) to 192.168.1.1:27652:
OPTIONS sip:[email protected]:27652;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK74a87ac4;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as3a2bf0ed
To: sip:[email protected]:27652;ob
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.6.0)
Date: Fri, 21 Mar 2014 11:50:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<— Transmitting (NAT) to 192.168.1.1:27652 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 125.17.166.115:32897;branch=z9hG4bKPjM3eo91pCE2sjgY7B2v1p7loz0XrxY-7w;received=192.168.1.1;rport=27652
From: sip:500@asteriskserverdynamichostname;tag=KBsnwzr0QkJ35-CUkSjtcwplpxfd31Zj
To: sip:500@asteriskserverdynamichostname;tag=as603a5fee
Call-ID: NJ.nsgfam-a99OB-vTU8iu76haQpVU1M
CSeq: 21335 REGISTER
Server: FPBX-2.11.0(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 900
Contact: sip:[email protected]:27652;ob;expires=900
Date: Fri, 21 Mar 2014 11:50:21 GMT
Content-Length: 0

<------------>
Reliably Transmitting (NAT) to 192.168.1.1:19735:
NOTIFY sip:[email protected]:19735;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK66497d1d;rport
Max-Forwards: 70
Route: sip:[email protected]:19735;ob
From: “Unknown” sip:[email protected];tag=as57741fd7
To: sip:[email protected]:19735;ob;tag=9y5gKgG.qgWohv01uB.BzxREXlBZZ9t8
Contact: sip:[email protected]:5060
Call-ID: Lx1dL0yjgM-A1nMx.1O2WB2BykS4mK0P
CSeq: 104 NOTIFY
User-Agent: FPBX-2.11.0(11.6.0)
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 88

Messages-Waiting: no
Message-Account: sip:*[email protected]
Voice-Message: 0/0 (0/0)


Scheduling destruction of SIP dialog ‘NJ.nsgfam-a99OB-vTU8iu76haQpVU1M’ in 32000 ms (Method: REGISTER)
Retransmitting #1 (NAT) to 192.168.1.1:19735:
NOTIFY sip:[email protected]:19735;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK66497d1d;rport
Max-Forwards: 70
Route: sip:[email protected]:19735;ob
From: “Unknown” sip:[email protected];tag=as57741fd7
To: sip:[email protected]:19735;ob;tag=9y5gKgG.qgWohv01uB.BzxREXlBZZ9t8
Contact: sip:[email protected]:5060
Call-ID: Lx1dL0yjgM-A1nMx.1O2WB2BykS4mK0P
CSeq: 104 NOTIFY
User-Agent: FPBX-2.11.0(11.6.0)
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 88

Messages-Waiting: no
Message-Account: sip:*[email protected]
Voice-Message: 0/0 (0/0)


<— SIP read from UDP:192.168.1.1:27652 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.147:5060;rport=5060;received=182.19.129.15;branch=z9hG4bK74a87ac4
Call-ID: [email protected]:5060
From: “Unknown” sip:[email protected];tag=as3a2bf0ed
To: sip:[email protected]:27652;ob;tag=z9hG4bK74a87ac4
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: CSipSimple_jflte-18/r2330
Content-Type: application/sdp
Content-Length: 289

v=0
o=- 3604391419 3604391419 IN IP4 125.17.166.115
s=pjmedia
t=0 0
m=audio 33840 RTP/AVP 99 0 8 101
c=IN IP4 125.17.166.115
a=sendrecv
a=rtpmap:99 SILK/24000
a=fmtp:99 useinbandfec=0
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
— (13 headers 13 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS

<— SIP read from UDP:192.168.1.1:27652 —>
SUBSCRIBE sip:500@asteriskserverdynamichostname SIP/2.0
Via: SIP/2.0/UDP 125.17.166.115:32897;rport;branch=z9hG4bKPjV5.KfHJsXYXn9AFjR6qVKNNY.18XMq-I
Max-Forwards: 70
From: sip:500@asteriskserverdynamichostname;tag=XMu2ZdE3CYseup.1qrxRcq1iagrpQOGx
To: sip:500@asteriskserverdynamichostname
Contact: sip:[email protected]:27652;ob;+sip.ice
Call-ID: RVnuUQaQ3sudm5Qa2ffac9zSaoG.mIGc
CSeq: 8484 SUBSCRIBE
Route: sip:asteriskserverdynamichostname;transport=udp;lr
Event: message-summary
Expires: 3600
Supported: replaces, 100rel, timer, norefersub
Accept: application/simple-message-summary
Allow-Events: presence, message-summary, refer
User-Agent: CSipSimple_jflte-18/r2330
Content-Length: 0

<------------->
— (16 headers 0 lines) —
Sending to 192.168.1.1:27652 (NAT)
Creating new subscription
Sending to 192.168.1.1:27652 (NAT)
list_route: hop: sip:[email protected]:27652;ob
Found peer ‘500’ for ‘500’ from 192.168.1.1:27652

<— Transmitting (NAT) to 192.168.1.1:27652 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 125.17.166.115:32897;branch=z9hG4bKPjV5.KfHJsXYXn9AFjR6qVKNNY.18XMq-I;received=192.168.1.1;rport=27652
From: sip:500@asteriskserverdynamichostname;tag=XMu2ZdE3CYseup.1qrxRcq1iagrpQOGx
To: sip:500@asteriskserverdynamichostname;tag=as7f0acb2e
Call-ID: RVnuUQaQ3sudm5Qa2ffac9zSaoG.mIGc
CSeq: 8484 SUBSCRIBE
Server: FPBX-2.11.0(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="5c1f6217"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘RVnuUQaQ3sudm5Qa2ffac9zSaoG.mIGc’ in 13952 ms (Method: SUBSCRIBE)
Retransmitting #2 (NAT) to 192.168.1.1:19735:
NOTIFY sip:[email protected]:19735;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK66497d1d;rport
Max-Forwards: 70
Route: sip:[email protected]:19735;ob
From: “Unknown” sip:[email protected];tag=as57741fd7
To: sip:[email protected]:19735;ob;tag=9y5gKgG.qgWohv01uB.BzxREXlBZZ9t8
Contact: sip:[email protected]:5060
Call-ID: Lx1dL0yjgM-A1nMx.1O2WB2BykS4mK0P
CSeq: 103 NOTIFY
User-Agent: FPBX-2.11.0(11.6.0)
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 88

Messages-Waiting: no
Message-Account: sip:*[email protected]
Voice-Message: 0/0 (0/0)


<— SIP read from UDP:192.168.1.1:27652 —>
SUBSCRIBE sip:500@asteriskserverdynamichostname SIP/2.0
Via: SIP/2.0/UDP 125.17.166.115:32897;rport;branch=z9hG4bKPjDQ4xuDROFZwN2ZfFbV5Bvry4g3lNdbaU
Max-Forwards: 70
From: sip:500@asteriskserverdynamichostname;tag=XMu2ZdE3CYseup.1qrxRcq1iagrpQOGx
To: sip:500@asteriskserverdynamichostname
Contact: sip:[email protected]:27652;ob;+sip.ice
Call-ID: RVnuUQaQ3sudm5Qa2ffac9zSaoG.mIGc
CSeq: 8485 SUBSCRIBE
Route: sip:asteriskserverdynamichostname;transport=udp;lr
Event: message-summary
Expires: 3600
Supported: replaces, 100rel, timer, norefersub
Accept: application/simple-message-summary
Allow-Events: presence, message-summary, refer
User-Agent: CSipSimple_jflte-18/r2330
Authorization: Digest username=“500”, realm=“asterisk”, nonce=“5c1f6217”, uri=“sip:500@asteriskserverdynamichostname”, response=“774f842318a7b182b9b6c1d13970030b”, algorithm=MD5
Content-Length: 0

<------------->
— (17 headers 0 lines) —
Creating new subscription
Sending to 192.168.1.1:27652 (NAT)
Found peer ‘500’ for ‘500’ from 192.168.1.1:27652
Really destroying SIP dialog ‘Lx1dL0yjgM-A1nMx.1O2WB2BykS4mK0P’ Method: SUBSCRIBE
Scheduling destruction of SIP dialog ‘RVnuUQaQ3sudm5Qa2ffac9zSaoG.mIGc’ in 3610000 ms (Method: SUBSCRIBE)

<— Transmitting (NAT) to 192.168.1.1:27652 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 125.17.166.115:32897;branch=z9hG4bKPjDQ4xuDROFZwN2ZfFbV5Bvry4g3lNdbaU;received=192.168.1.1;rport=27652
From: sip:500@asteriskserverdynamichostname;tag=XMu2ZdE3CYseup.1qrxRcq1iagrpQOGx
To: sip:500@asteriskserverdynamichostname;tag=as7f0acb2e
Call-ID: RVnuUQaQ3sudm5Qa2ffac9zSaoG.mIGc
CSeq: 8485 SUBSCRIBE
Server: FPBX-2.11.0(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: sip:[email protected]:5060;expires=3600
Content-Length: 0

<------------>
Reliably Transmitting (NAT) to 192.168.1.1:27652:
NOTIFY sip:[email protected]:27652;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK21784998;rport
Max-Forwards: 70
Route: sip:[email protected]:27652;ob
From: “Unknown” sip:[email protected];tag=as7f0acb2e
To: sip:[email protected]:27652;ob;tag=XMu2ZdE3CYseup.1qrxRcq1iagrpQOGx
Contact: sip:[email protected]:5060
Call-ID: RVnuUQaQ3sudm5Qa2ffac9zSaoG.mIGc
CSeq: 102 NOTIFY
User-Agent: FPBX-2.11.0(11.6.0)
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 88

Messages-Waiting: no
Message-Account: sip:*[email protected]
Voice-Message: 0/0 (0/0)


<— SIP read from UDP:192.168.1.1:27652 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.147:5060;rport=5060;received=182.19.129.15;branch=z9hG4bK21784998
Call-ID: RVnuUQaQ3sudm5Qa2ffac9zSaoG.mIGc
From: “Unknown” sip:[email protected];tag=as7f0acb2e
To: sip:[email protected]:27652;ob;tag=XMu2ZdE3CYseup.1qrxRcq1iagrpQOGx
CSeq: 102 NOTIFY
Contact: sip:[email protected]:27652;ob;+sip.ice
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘4ZnBa0S3yAlQEp9zClSr7MaiSQlJF7xV’ Method: REGISTER

<— SIP read from UDP:192.168.1.1:27652 —>
INVITE sip:411@asteriskserverdynamichostname SIP/2.0
Via: SIP/2.0/UDP 125.17.166.115:27384;rport;branch=z9hG4bKPjBbWttwLpOEuyib7dJDGxvc4dSTPagccX
Max-Forwards: 70
From: sip:500@asteriskserverdynamichostname;tag=bA.G9XoZXtGoe8deW-AZpOP.0GvROtuS
To: sip:411@asteriskserverdynamichostname
Contact: sip:[email protected]:27652;ob;+sip.ice
Call-ID: -hQnvPh1Bz21l7eYtHBlbT2qb52skTe5
CSeq: 12430 INVITE
Route: sip:asteriskserverdynamichostname;transport=udp;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_jflte-18/r2330
Content-Type: application/sdp
Content-Length: 490

v=0
o=- 3604391426 3604391426 IN IP4 125.17.166.115
s=pjmedia
c=IN IP4 125.17.166.115
t=0 0
m=audio 33842 RTP/AVP 99 0 8 101
c=IN IP4 125.17.166.115
a=sendrecv
a=rtpmap:99 SILK/24000
a=fmtp:99 useinbandfec=0
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ice-ufrag:23de8217
a=ice-pwd:16f79195
a=candidate:Ha5b9a68 1 UDP 2130706431 10.91.154.104 4033 typ host
a=candidate:Ha5b9a68 2 UDP 2130706430 10.91.154.104 4022 typ host

<------------->
— (16 headers 18 lines) —
Sending to 192.168.1.1:27652 (NAT)
Sending to 192.168.1.1:27652 (NAT)
Using INVITE request as basis request - -hQnvPh1Bz21l7eYtHBlbT2qb52skTe5
Found peer ‘500’ for ‘500’ from 192.168.1.1:27652

<— Reliably Transmitting (NAT) to 192.168.1.1:27652 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 125.17.166.115:27384;branch=z9hG4bKPjBbWttwLpOEuyib7dJDGxvc4dSTPagccX;received=192.168.1.1;rport=27652
From: sip:500@asteriskserverdynamichostname;tag=bA.G9XoZXtGoe8deW-AZpOP.0GvROtuS
To: sip:411@asteriskserverdynamichostname;tag=as7391bec7
Call-ID: -hQnvPh1Bz21l7eYtHBlbT2qb52skTe5
CSeq: 12430 INVITE
Server: FPBX-2.11.0(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="605d096c"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘-hQnvPh1Bz21l7eYtHBlbT2qb52skTe5’ in 13952 ms (Method: INVITE)

<— SIP read from UDP:192.168.1.1:27652 —>
ACK sip:411@asteriskserverdynamichostname SIP/2.0
Via: SIP/2.0/UDP 125.17.166.115:27384;rport;branch=z9hG4bKPjBbWttwLpOEuyib7dJDGxvc4dSTPagccX
Max-Forwards: 70
From: sip:500@asteriskserverdynamichostname;tag=bA.G9XoZXtGoe8deW-AZpOP.0GvROtuS
To: sip:411@asteriskserverdynamichostname;tag=as7391bec7
Call-ID: -hQnvPh1Bz21l7eYtHBlbT2qb52skTe5
CSeq: 12430 ACK
Route: sip:asteriskserverdynamichostname;transport=udp;lr
Content-Length: 0

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

<— SIP read from UDP:192.168.1.1:27652 —>
INVITE sip:411@asteriskserverdynamichostname SIP/2.0
Via: SIP/2.0/UDP 125.17.166.115:27384;rport;branch=z9hG4bKPjQS0ih7ztkVMWynK3uHgQa0oSBFvzZGqK
Max-Forwards: 70
From: sip:500@asteriskserverdynamichostname;tag=bA.G9XoZXtGoe8deW-AZpOP.0GvROtuS
To: sip:411@asteriskserverdynamichostname
Contact: sip:[email protected]:27652;ob;+sip.ice
Call-ID: -hQnvPh1Bz21l7eYtHBlbT2qb52skTe5
CSeq: 12431 INVITE
Route: sip:asteriskserverdynamichostname;transport=udp;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_jflte-18/r2330
Authorization: Digest username=“500”, realm=“asterisk”, nonce=“605d096c”, uri=“sip:411@asteriskserverdynamichostname”, response=“512b6b8ff691475ed739cd8b9bce3384”, algorithm=MD5
Content-Type: application/sdp
Content-Length: 490

v=0
o=- 3604391426 3604391426 IN IP4 125.17.166.115
s=pjmedia
c=IN IP4 125.17.166.115
t=0 0
m=audio 33842 RTP/AVP 99 0 8 101
c=IN IP4 125.17.166.115
a=sendrecv
a=rtpmap:99 SILK/24000
a=fmtp:99 useinbandfec=0
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ice-ufrag:23de8217
a=ice-pwd:16f79195
a=candidate:Ha5b9a68 1 UDP 2130706431 10.91.154.104 4033 typ host
a=candidate:Ha5b9a68 2 UDP 2130706430 10.91.154.104 4022 typ host

<------------->
— (17 headers 18 lines) —
Sending to 192.168.1.1:27652 (NAT)
Using INVITE request as basis request - -hQnvPh1Bz21l7eYtHBlbT2qb52skTe5
Found peer ‘500’ for ‘500’ from 192.168.1.1:27652
Found RTP audio format 99
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format SILK for ID 99
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 - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw|silk24)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 125.17.166.115:33842
Looking for 411 in from-internal (domain asteriskserverdynamichostname)
list_route: hop: sip:[email protected]:27652;ob

<— Transmitting (NAT) to 192.168.1.1:27652 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 125.17.166.115:27384;branch=z9hG4bKPjQS0ih7ztkVMWynK3uHgQa0oSBFvzZGqK;received=192.168.1.1;rport=27652
From: sip:500@asteriskserverdynamichostname;tag=bA.G9XoZXtGoe8deW-AZpOP.0GvROtuS
To: sip:411@asteriskserverdynamichostname
Call-ID: -hQnvPh1Bz21l7eYtHBlbT2qb52skTe5
CSeq: 12431 INVITE
Server: FPBX-2.11.0(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Length: 0

<------------>
Audio is at 19454
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 192.168.1.1:27652 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 125.17.166.115:27384;branch=z9hG4bKPjQS0ih7ztkVMWynK3uHgQa0oSBFvzZGqK;received=192.168.1.1;rport=27652
From: sip:500@asteriskserverdynamichostname;tag=bA.G9XoZXtGoe8deW-AZpOP.0GvROtuS
To: sip:411@asteriskserverdynamichostname;tag=as4c8cc724
Call-ID: -hQnvPh1Bz21l7eYtHBlbT2qb52skTe5
CSeq: 12431 INVITE
Server: FPBX-2.11.0(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Require: timer
Content-Length: 286

v=0
o=root 252674008 252674008 IN IP4 192.168.1.147
s=Asterisk PBX 11.6.0
c=IN IP4 192.168.1.147
t=0 0
m=audio 19454 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
Retransmitting #1 (NAT) to 192.168.1.1:27652:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 125.17.166.115:27384;branch=z9hG4bKPjQS0ih7ztkVMWynK3uHgQa0oSBFvzZGqK;received=192.168.1.1;rport=27652
From: sip:500@asteriskserverdynamichostname;tag=bA.G9XoZXtGoe8deW-AZpOP.0GvROtuS
To: sip:411@asteriskserverdynamichostname;tag=as4c8cc724
Call-ID: -hQnvPh1Bz21l7eYtHBlbT2qb52skTe5
CSeq: 12431 INVITE
Server: FPBX-2.11.0(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Require: timer
Content-Length: 286

v=0
o=root 252674008 252674008 IN IP4 192.168.1.147
s=Asterisk PBX 11.6.0
c=IN IP4 192.168.1.147
t=0 0
m=audio 19454 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


Retransmitting #2 (NAT) to 192.168.1.1:27652:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 125.17.166.115:27384;branch=z9hG4bKPjQS0ih7ztkVMWynK3uHgQa0oSBFvzZGqK;received=192.168.1.1;rport=27652
From: sip:500@asteriskserverdynamichostname;tag=bA.G9XoZXtGoe8deW-AZpOP.0GvROtuS
To: sip:411@asteriskserverdynamichostname;tag=as4c8cc724
Call-ID: -hQnvPh1Bz21l7eYtHBlbT2qb52skTe5
CSeq: 12431 INVITE
Server: FPBX-2.11.0(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Require: timer
Content-Length: 286

v=0
o=root 252674008 252674008 IN IP4 192.168.1.147
s=Asterisk PBX 11.6.0
c=IN IP4 192.168.1.147
t=0 0
m=audio 19454 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


Retransmitting #3 (NAT) to 192.168.1.1:27652:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 125.17.166.115:27384;branch=z9hG4bKPjQS0ih7ztkVMWynK3uHgQa0oSBFvzZGqK;received=192.168.1.1;rport=27652
From: sip:500@asteriskserverdynamichostname;tag=bA.G9XoZXtGoe8deW-AZpOP.0GvROtuS
To: sip:411@asteriskserverdynamichostname;tag=as4c8cc724
Call-ID: -hQnvPh1Bz21l7eYtHBlbT2qb52skTe5
CSeq: 12431 INVITE
Server: FPBX-2.11.0(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Require: timer
Content-Length: 286

v=0
o=root 252674008 252674008 IN IP4 192.168.1.147
s=Asterisk PBX 11.6.0
c=IN IP4 192.168.1.147
t=0 0
m=audio 19454 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


Retransmitting #4 (NAT) to 192.168.1.1:27652:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 125.17.166.115:27384;branch=z9hG4bKPjQS0ih7ztkVMWynK3uHgQa0oSBFvzZGqK;received=192.168.1.1;rport=27652
From: sip:500@asteriskserverdynamichostname;tag=bA.G9XoZXtGoe8deW-AZpOP.0GvROtuS
To: sip:411@asteriskserverdynamichostname;tag=as4c8cc724
Call-ID: -hQnvPh1Bz21l7eYtHBlbT2qb52skTe5
CSeq: 12431 INVITE
Server: FPBX-2.11.0(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Require: timer
Content-Length: 286

v=0
o=root 252674008 252674008 IN IP4 192.168.1.147
s=Asterisk PBX 11.6.0
c=IN IP4 192.168.1.147
t=0 0
m=audio 19454 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


Reliably Transmitting (no NAT) to 192.168.1.148:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK1f854a1f
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as50e35b8a
To: sip:[email protected]:5060
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.6.0)
Date: Fri, 21 Mar 2014 11:50:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:192.168.1.148:5060 —>
SIP/2.0 200 OK
To: sip:[email protected]:5060;tag=a3b98ff578395be2i0
From: “Unknown” sip:[email protected];tag=as50e35b8a
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK1f854a1f
Server: Linksys/SPA3102-5.1.10(GW)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
Retransmitting #5 (NAT) to 192.168.1.1:27652:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 125.17.166.115:27384;branch=z9hG4bKPjQS0ih7ztkVMWynK3uHgQa0oSBFvzZGqK;received=192.168.1.1;rport=27652
From: sip:500@asteriskserverdynamichostname;tag=bA.G9XoZXtGoe8deW-AZpOP.0GvROtuS
To: sip:411@asteriskserverdynamichostname;tag=as4c8cc724
Call-ID: -hQnvPh1Bz21l7eYtHBlbT2qb52skTe5
CSeq: 12431 INVITE
Server: FPBX-2.11.0(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Require: timer
Content-Length: 286

v=0
o=root 252674008 252674008 IN IP4 192.168.1.147
s=Asterisk PBX 11.6.0
c=IN IP4 192.168.1.147
t=0 0
m=audio 19454 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


<— SIP read from UDP:192.168.1.148:5061 —>
REGISTER sip:192.168.1.147 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.148:5061;branch=z9hG4bK-ce4f311d
From: PSTN sip:[email protected];tag=ad157c81388578c6o1
To: PSTN sip:[email protected]
Call-ID: [email protected]
CSeq: 46319 REGISTER
Max-Forwards: 70
Authorization: Digest username=“1-pstn”,realm=“asterisk”,nonce=“75fa5b06”,uri=“sip:192.168.1.147”,algorithm=MD5,response="1e6d5544b4d3ac2d4e50f3cd3b7ec925"
Contact: PSTN sip:[email protected]:5061;expires=300
User-Agent: Linksys/SPA3102-5.1.10(GW)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
— (13 headers 0 lines) —
Sending to 192.168.1.148:5061 (NAT)
Sending to 192.168.1.148:5061 (NAT)

<— Transmitting (no NAT) to 192.168.1.148:5061 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.148:5061;branch=z9hG4bK-ce4f311d;received=192.168.1.148
From: PSTN sip:[email protected];tag=ad157c81388578c6o1
To: PSTN sip:[email protected];tag=as1262e4e3
Call-ID: [email protected]
CSeq: 46319 REGISTER
Server: FPBX-2.11.0(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="303e44b8"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.1.148:5061 —>
REGISTER sip:192.168.1.147 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.148:5061;branch=z9hG4bK-41cbcf1e
From: PSTN sip:[email protected];tag=ad157c81388578c6o1
To: PSTN sip:[email protected]
Call-ID: [email protected]
CSeq: 46320 REGISTER
Max-Forwards: 70
Authorization: Digest username=“1-pstn”,realm=“asterisk”,nonce=“303e44b8”,uri=“sip:192.168.1.147”,algorithm=MD5,response="ebc45f59bef92b81bf88424c1350a302"
Contact: PSTN sip:[email protected]:5061;expires=300
User-Agent: Linksys/SPA3102-5.1.10(GW)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
— (13 headers 0 lines) —
Sending to 192.168.1.148:5061 (no NAT)
Reliably Transmitting (no NAT) to 192.168.1.148:5061:
OPTIONS sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK2b7e0f14
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as4e5142ea
To: sip:[email protected]:5061
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.6.0)
Date: Fri, 21 Mar 2014 11:50:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<— Transmitting (no NAT) to 192.168.1.148:5061 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.148:5061;branch=z9hG4bK-41cbcf1e;received=192.168.1.148
From: PSTN sip:[email protected];tag=ad157c81388578c6o1
To: PSTN sip:[email protected];tag=as1262e4e3
Call-ID: [email protected]
CSeq: 46320 REGISTER
Server: FPBX-2.11.0(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 300
Contact: sip:[email protected]:5061;expires=300
Date: Fri, 21 Mar 2014 11:50:37 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.1.148:5061 —>
SIP/2.0 200 OK
To: sip:[email protected]:5061;tag=8b98f75da1bedb9ai1
From: “Unknown” sip:[email protected];tag=as4e5142ea
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK2b7e0f14
Server: Linksys/SPA3102-5.1.10(GW)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
Retransmitting #6 (NAT) to 192.168.1.1:27652:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 125.17.166.115:27384;branch=z9hG4bKPjQS0ih7ztkVMWynK3uHgQa0oSBFvzZGqK;received=192.168.1.1;rport=27652
From: sip:500@asteriskserverdynamichostname;tag=bA.G9XoZXtGoe8deW-AZpOP.0GvROtuS
To: sip:411@asteriskserverdynamichostname;tag=as4c8cc724
Call-ID: -hQnvPh1Bz21l7eYtHBlbT2qb52skTe5
CSeq: 12431 INVITE
Server: FPBX-2.11.0(11.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Require: timer
Content-Length: 286

v=0
o=root 252674008 252674008 IN IP4 192.168.1.147
s=Asterisk PBX 11.6.0
c=IN IP4 192.168.1.147
t=0 0
m=audio 19454 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


Scheduling destruction of SIP dialog ‘-hQnvPh1Bz21l7eYtHBlbT2qb52skTe5’ in 13952 ms (Method: INVITE)
set_destination: Parsing sip:[email protected]:27652;ob for address/port to send to
set_destination: set destination to 192.168.1.1:27652
Reliably Transmitting (NAT) to 192.168.1.1:27652:
BYE sip:[email protected]:27652;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK469d39be;rport
Max-Forwards: 70
From: sip:411@asteriskserverdynamichostname;tag=as4c8cc724
To: sip:500@asteriskserverdynamichostname;tag=bA.G9XoZXtGoe8deW-AZpOP.0GvROtuS
Call-ID: -hQnvPh1Bz21l7eYtHBlbT2qb52skTe5
CSeq: 102 BYE
User-Agent: FPBX-2.11.0(11.6.0)
Proxy-Authorization: Digest username=“500”, realm=“asterisk”, algorithm=MD5, uri=“sip:asteriskserverdynamichostname”, nonce=“605d096c”, response="1764b21412592410e50e9f98276ee593"
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0


<— SIP read from UDP:192.168.1.1:27652 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.147:5060;rport=5060;received=182.19.129.15;branch=z9hG4bK469d39be
Call-ID: -hQnvPh1Bz21l7eYtHBlbT2qb52skTe5
From: sip:411@asteriskserverdynamichostname;tag=as4c8cc724
To: sip:500@asteriskserverdynamichostname;tag=bA.G9XoZXtGoe8deW-AZpOP.0GvROtuS
CSeq: 102 BYE
Content-Length: 0

<------------->
— (7 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘-hQnvPh1Bz21l7eYtHBlbT2qb52skTe5’ Method: INVITE
raspbx*CLI> exit
Asterisk cleanly ending (0).
Executing last minute cleanups
root@raspbx:/var/log/asterisk#

For starters we could have done without the full sip debug. The generic asterisk debug is usually fine for a first pass.

Also why did you select ports 15000 to 20000? it should be 10000-20000.

Sorry about that,will keep note next time.Im using port 15000 onwards, because 14000 series, i use it for other purposes. ive set up the same port range in asterisk as well.