[Solved] Dropped calls after ~30 seconds

Asterisk Asterisk Support
1

Hey everyone.

Been bashing my head on this one for several days. I have been searching high and low, been trying all manner of configuration changes, even put in a support call with my router’s manufacturer.

Overall, Asterisk is running great so long as the clients are connected internally. However, when a client registers externally, calls are dropping within 30 seconds regardless of where they are going (another extension, outside #, or an outside # reaching in to that extension). The odd thing is there is two way audio here - everything is working as it should except for that dropped call.

Here’s the setup:
Asterisk PBX v1.8.7.0 with two NIC’s
ETH0: Internal (192.168.1.60)
ETH1: External (x.x.x.x)

Business VoIP SIP provider has a local box of their own on 192.168.1.62 that Asterisk registers with.

Router is a SonicWall TZ200W with Consistent NAT enabled and sip transformations are disabled.
ETH1 is on a direct interface to the sonic wall (no NAT).
Firewall is open for UDP ports 5060, 5061, and 10000-20000

Dialplan requires 9 to be dialed prior to making an outbound call.

sip.conf:

[general]
context=default
directmedia=no
allowoverlap=no
alwaysauthreject=yes
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
allowguest=no
srvlookup=yes
tos_sip=cs3
tos_audio=ef
relaxdtmf=yes
trustrpid=no
sendrpid=yes
externip=x.x.x.x
localnet=192.168.1.0/255.255.255.0


register => mydid:mydidSecret:myvoipprovideraccount@192.168.1.62/myvoidprovideraccount~3600

[authentication]
[myvoipprovider-trunk]
type=friend
username=myvoipprovider
fromdomain=192.168.1.62
realm=192.168.1.62
host=192.168.1.62
dtmfmode=rfc2833
secret=myvoipprovider_password
nat=yes
canreinvite=no
insecure=invite,port
qualify=yes
disallow=all
allow=ulaw
amaflags=default
trustrpid=no
sendrpid=yes
sendrpid=pai
context=cox_in
directmedia=no

[phones](!)
type=friend
host=dynamic
context=default
canreinvite=no
directmedia=no
disallow=all
allow=ulaw
nat=yes
call-limit=5
qualify=yes

[140](phones)
username=140
regexten=140
secret=my_secret_password
mailbox=140@default
callerid="John Pels" <140>

SIP Debug where
x.x.x.x = asterisk external IP
y.y.y.y = outside line

<--- SIP read from UDP:174.252.121.19:4665 --->
REGISTER sip:x.x.x.x SIP/2.0
Via: SIP/2.0/UDP 174.252.121.19:4665;rport;branch=z9hG4bK+cec8e5640d0ecd576dcfdf2d4131ca071+s36+1
From: <sip:140@x.x.x.x>;tag=s36+1+4275000b+15cec39b
Max-Forwards: 70
To: <sip:140@x.x.x.x>
Call-ID: 530399074390@10.243.79.87
CSeq: 1 REGISTER
Contact: <sip:140@174.252.121.19:4665;transport=udp>
Expires: 3600
User-Agent: Sipdroid/2.7 beta/DROID2
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 174.252.121.19:4665 (no NAT)

<--- Transmitting (NAT) to 174.252.121.19:4665 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 174.252.121.19:4665;branch=z9hG4bK+cec8e5640d0ecd576dcfdf2d4131ca071+s36+1;received=174.252.121.19;rport=4665
From: <sip:140@x.x.x.x>;tag=s36+1+4275000b+15cec39b
To: <sip:140@x.x.x.x>;tag=as76934f6c
Call-ID: 530399074390@10.243.79.87
CSeq: 1 REGISTER
Server: Asterisk PBX 1.8.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="06b06b01"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '530399074390@10.243.79.87' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:174.252.121.19:4665 --->
INVITE sip:9y.y.y.y@x.x.x.x SIP/2.0
Via: SIP/2.0/UDP 174.252.121.19:4665;rport;branch=z9hG4bK+8f069052534fe723dce656947a39a0ad1+s36+1
From: <sip:140@x.x.x.x>;tag=s36+1+4272000b+4f798f96
To: <sip:9y.y.y.y@x.x.x.x>
Call-ID: 581844940246@10.243.79.87-S
Max-Forwards: 70
CSeq: 1 INVITE
Contact: <sip:140@174.252.121.19:4665;transport=udp>
Expires: 3600
User-Agent: Sipdroid/2.7 beta/DROID2
Content-Length: 400
Content-Type: application/sdp

v=0
o=140@x.x.x.x 0 0 IN IP4 174.252.121.19
s=Session SIP/SDP
c=IN IP4 174.252.121.19
t=0 0
m=audio 4648 RTP/AVP 9 8 0 97 3 106 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:97 speex/8000
a=rtpmap:3 GSM/8000
a=rtpmap:106 BV16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:4643
m=video 4650 RTP/AVP 103
a=rtpmap:103 h263-1998/90000
<------------->
--- (12 headers 17 lines) ---
Sending to 174.252.121.19:4665 (no NAT)
Using INVITE request as basis request - 581844940246@10.243.79.87-S
Found peer '140' for '140' from 174.252.121.19:4665

<--- Reliably Transmitting (NAT) to 174.252.121.19:4665 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 174.252.121.19:4665;branch=z9hG4bK+8f069052534fe723dce656947a39a0ad1+s36+1;received=174.252.121.19;rport=4665
From: <sip:140@x.x.x.x>;tag=s36+1+4272000b+4f798f96
To: <sip:9y.y.y.y@x.x.x.x>;tag=as56f84a56
Call-ID: 581844940246@10.243.79.87-S
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="240e3e71"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '581844940246@10.243.79.87-S' in 25024 ms (Method: INVITE)

<--- SIP read from UDP:174.252.121.19:4665 --->
ACK sip:9y.y.y.y@x.x.x.x SIP/2.0
Via: SIP/2.0/UDP 174.252.121.19:4665;rport;branch=z9hG4bK+8f069052534fe723dce656947a39a0ad1+s36+1
From: <sip:140@x.x.x.x>;tag=s36+1+4272000b+4f798f96
To: <sip:9y.y.y.y@x.x.x.x>;tag=as56f84a56
Call-ID: 581844940246@10.243.79.87-S
Max-Forwards: 70
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:174.252.121.19:4665 --->
REGISTER sip:x.x.x.x SIP/2.0
Via: SIP/2.0/UDP 174.252.121.19:4665;rport;branch=z9hG4bK+8eb90deded81e5c151b0e33b9da5a9f71+s36+1
From: <sip:140@x.x.x.x>;tag=s36+1+4275000b+5feb9c9b
Max-Forwards: 70
To: <sip:140@x.x.x.x>
Call-ID: 530399074390@10.243.79.87
CSeq: 2 REGISTER
Contact: <sip:140@174.252.121.19:4665;transport=udp>
Expires: 3600
User-Agent: Sipdroid/2.7 beta/DROID2
Authorization: Digest username="140", realm="asterisk", nonce="06b06b01", uri="sip:x.x.x.x", algorithm=MD5, response="e207e75bfa2fd1485279a216625c0e71"
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 174.252.121.19:4665 (NAT)
Reliably Transmitting (NAT) to 174.252.121.19:4665:
OPTIONS sip:140@174.252.121.19:4665;transport=udp SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK2f3138eb;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@x.x.x.x>;tag=as712e3dd1
To: <sip:140@174.252.121.19:4665;transport=udp>
Contact: <sip:asterisk@x.x.x.x:5060>
Call-ID: 2b4f30c23ada40133dd8befd3b4e13b4@x.x.x.x:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.7.0
Date: Mon, 20 Aug 2012 20:16:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (NAT) to 174.252.121.19:4665 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 174.252.121.19:4665;branch=z9hG4bK+8eb90deded81e5c151b0e33b9da5a9f71+s36+1;received=174.252.121.19;rport=4665
From: <sip:140@x.x.x.x>;tag=s36+1+4275000b+5feb9c9b
To: <sip:140@x.x.x.x>;tag=as76934f6c
Call-ID: 530399074390@10.243.79.87
CSeq: 2 REGISTER
Server: Asterisk PBX 1.8.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: <sip:140@174.252.121.19:4665;transport=udp>;expires=3600
Date: Mon, 20 Aug 2012 20:16:01 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '530399074390@10.243.79.87' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:174.252.121.19:4665 --->
INVITE sip:9y.y.y.y@x.x.x.x SIP/2.0
Via: SIP/2.0/UDP 174.252.121.19:4665;rport;branch=z9hG4bK+f89443b9aff42fa263984a680e57a84c1+s36+1
From: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
To: <sip:9y.y.y.y@x.x.x.x>
Call-ID: 581844940246@10.243.79.87-S
Max-Forwards: 70
CSeq: 2 INVITE
Contact: <sip:140@174.252.121.19:4665;transport=udp>
Expires: 3600
User-Agent: Sipdroid/2.7 beta/DROID2
Authorization: Digest username="140", realm="asterisk", nonce="240e3e71", uri="sip:9y.y.y.y@x.x.x.x", algorithm=MD5, response="b0a796d41440a0ef635e1d421019d07b"
Content-Length: 400
Content-Type: application/sdp

v=0
o=140@x.x.x.x 0 0 IN IP4 174.252.121.19
s=Session SIP/SDP
c=IN IP4 174.252.121.19
t=0 0
m=audio 4648 RTP/AVP 9 8 0 97 3 106 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:97 speex/8000
a=rtpmap:3 GSM/8000
a=rtpmap:106 BV16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:4667
m=video 4668 RTP/AVP 103
a=rtpmap:103 h263-1998/90000
<------------->
--- (13 headers 17 lines) ---
Sending to 174.252.121.19:4665 (NAT)
Using INVITE request as basis request - 581844940246@10.243.79.87-S
Found peer '140' for '140' from 174.252.121.19:4665
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 97
Found RTP audio format 3
Found RTP audio format 106
Found RTP audio format 101
Found audio description format G722 for ID 9
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format speex for ID 97
Found audio description format GSM for ID 3
Found unknown media description format BV16 for ID 106
Found audio description format telephone-event for ID 101
Found RTP video format 103
Found video description format h263-1998 for ID 103
Capabilities: us - 0x4 (ulaw), peer - audio=0x120e (gsm|ulaw|alaw|speex|g722)/video=0x100000 (h263p)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 174.252.121.19:4648
Looking for 9y.y.y.y in default (domain x.x.x.x)
list_route: hop: <sip:140@174.252.121.19:4665;transport=udp>

<--- Transmitting (NAT) to 174.252.121.19:4665 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 174.252.121.19:4665;branch=z9hG4bK+f89443b9aff42fa263984a680e57a84c1+s36+1;received=174.252.121.19;rport=4665
From: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
To: <sip:9y.y.y.y@x.x.x.x>
Call-ID: 581844940246@10.243.79.87-S
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:9y.y.y.y@x.x.x.x:5060>
Content-Length: 0


<------------>
    -- Executing [9y.y.y.y@default:1] Set("SIP/140-0000005d", "CALLERID(all)="Insystech, Inc" <mydid>") in new stack
    -- Executing [9y.y.y.y@default:2] Dial("SIP/140-0000005d", "SIP/y.y.y.y@myvoipprovideraccount,30,r") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.1.62:5060:
INVITE sip:y.y.y.y@192.168.1.62 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.60:5060;branch=z9hG4bK50d195fd;rport
Max-Forwards: 70
From: "Insystech, Inc" <sip:mydid@192.168.1.62>;tag=as0ecbe613
To: <sip:y.y.y.y@192.168.1.62>
Contact: <sip:mydid@192.168.1.60:5060>
Call-ID: 61422484218e8cf71b03bf636ee52f89@192.168.1.62
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.7.0
Date: Mon, 20 Aug 2012 20:16:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
P-Asserted-Identity: "Insystech, Inc" <sip:mydid@192.168.1.62>
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 579120403 579120403 IN IP4 192.168.1.60
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.60
t=0 0
m=audio 12608 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called SIP/y.y.y.y@myvoipprovideraccount

<--- Transmitting (NAT) to 174.252.121.19:4665 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 174.252.121.19:4665;branch=z9hG4bK+f89443b9aff42fa263984a680e57a84c1+s36+1;received=174.252.121.19;rport=4665
From: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
To: <sip:9y.y.y.y@x.x.x.x>;tag=as3d87b2f9
Call-ID: 581844940246@10.243.79.87-S
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:9y.y.y.y@x.x.x.x:5060>
Content-Length: 0


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

<--- SIP read from UDP:192.168.1.62:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.60:5060;received=184.184.24.166;branch=z9hG4bK50d195fd;rport=5060
From: "Insystech, Inc" <sip:mydid@192.168.1.62>;tag=as0ecbe613
To: <sip:y.y.y.y@192.168.1.62>
Call-ID: 61422484218e8cf71b03bf636ee52f89@192.168.1.62
CSeq: 102 INVITE
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.62:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.60:5060;received=184.184.24.166;branch=z9hG4bK50d195fd;rport=5060
From: "Insystech, Inc" <sip:mydid@192.168.1.62>;tag=as0ecbe613
To: <sip:y.y.y.y@192.168.1.62>;tag=SDl9rs399-1578343328-1345493672110
Call-ID: 61422484218e8cf71b03bf636ee52f89@192.168.1.62
CSeq: 102 INVITE
WWW-Authenticate: DIGEST realm="BroadWorks", nonce="BroadWorksXh6406xtaT5jcb1bBW", algorithm=MD5, qop="auth"
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 192.168.1.62:5060:
ACK sip:y.y.y.y@192.168.1.62 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.60:5060;branch=z9hG4bK50d195fd;rport
Max-Forwards: 70
From: "Insystech, Inc" <sip:mydid@192.168.1.62>;tag=as0ecbe613
To: <sip:y.y.y.y@192.168.1.62>;tag=SDl9rs399-1578343328-1345493672110
Contact: <sip:mydid@192.168.1.60:5060>
Call-ID: 61422484218e8cf71b03bf636ee52f89@192.168.1.62
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.7.0
Content-Length: 0


---
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.1.62:5060:
INVITE sip:y.y.y.y@192.168.1.62 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.60:5060;branch=z9hG4bK2b13fb91;rport
Max-Forwards: 70
From: "Insystech, Inc" <sip:mydid@192.168.1.62>;tag=as0ecbe613
To: <sip:y.y.y.y@192.168.1.62>
Contact: <sip:mydid@192.168.1.60:5060>
Call-ID: 61422484218e8cf71b03bf636ee52f89@192.168.1.62
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.7.0
Authorization: Digest username="477049014802", realm="BroadWorks", algorithm=MD5, uri="sip:y.y.y.y@192.168.1.62", nonce="BroadWorksXh6406xtaT5jcb1bBW", response="e1f396b27054fe654af36bbce5815df2", qop=auth, cnonce="69fb9094", nc=00000001
Date: Mon, 20 Aug 2012 20:16:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
P-Asserted-Identity: "Insystech, Inc" <sip:mydid@192.168.1.62>
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 579120403 579120404 IN IP4 192.168.1.60
s=Asterisk PBX 1.8.7.0
c=IN IP4 192.168.1.60
t=0 0
m=audio 12608 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.1.62:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.60:5060;received=184.184.24.166;branch=z9hG4bK2b13fb91;rport=5060
From: "Insystech, Inc" <sip:mydid@192.168.1.62>;tag=as0ecbe613
To: <sip:y.y.y.y@192.168.1.62>
Call-ID: 61422484218e8cf71b03bf636ee52f89@192.168.1.62
CSeq: 103 INVITE
Content-Length: 0

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

<--- SIP read from UDP:174.252.121.19:4665 --->
SIP/2.0 200 OK
Call-ID: 2b4f30c23ada40133dd8befd3b4e13b4@x.x.x.x:5060
CSeq: 102 OPTIONS
From: "asterisk" <sip:asterisk@x.x.x.x>;tag=as712e3dd1
To: <sip:140@174.252.121.19:4665;transport=udp>;tag=s36+1+4270000b+63cc0d31
Via: SIP/2.0/UDP x.x.x.x:5060;received=x.x.x.x;rport=5060;branch=z9hG4bK2f3138eb
Contact: <sip:140@174.252.121.19:4665;transport=udp>
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '2b4f30c23ada40133dd8befd3b4e13b4@x.x.x.x:5060' Method: OPTIONS

<--- SIP read from UDP:174.252.121.19:4665 --->
SUBSCRIBE sip:140@x.x.x.x SIP/2.0
Via: SIP/2.0/UDP 174.252.121.19:4665;rport;branch=z9hG4bK+09aef22680140dce2249ceb7d156a5d31+s36+1
From: <sip:140@x.x.x.x>;tag=s36+1+4267000b+409aabef
To: <sip:140@x.x.x.x>
Call-ID: 604050037420@10.243.79.87-S
Max-Forwards: 70
CSeq: 1 SUBSCRIBE
Contact: <sip:140@174.252.121.19:4665;transport=udp>
Expires: 184000
User-Agent: Sipdroid/2.7 beta/DROID2
Event: message-summary
Accept: application/simple-message-summary
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Creating new subscription
Sending to 174.252.121.19:4665 (no NAT)
list_route: hop: <sip:140@174.252.121.19:4665;transport=udp>
Found peer '140' for '140' from 174.252.121.19:4665

<--- Transmitting (NAT) to 174.252.121.19:4665 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 174.252.121.19:4665;branch=z9hG4bK+09aef22680140dce2249ceb7d156a5d31+s36+1;received=174.252.121.19;rport=4665
From: <sip:140@x.x.x.x>;tag=s36+1+4267000b+409aabef
To: <sip:140@x.x.x.x>;tag=as3280e478
Call-ID: 604050037420@10.243.79.87-S
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 1.8.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="27f4e829"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '604050037420@10.243.79.87-S' in 32064 ms (Method: SUBSCRIBE)

<--- SIP read from UDP:192.168.1.62:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.60:5060;received=184.184.24.166;branch=z9hG4bK2b13fb91;rport=5060
Record-Route: <sip:Asterisks@192.168.1.62;lr>
From: "Insystech, Inc" <sip:mydid@192.168.1.62>;tag=as0ecbe613
To: <sip:y.y.y.y@192.168.1.62>;tag=SDl9rs399-1720459397-1345493673623
Call-ID: 61422484218e8cf71b03bf636ee52f89@192.168.1.62
CSeq: 103 INVITE
Contact: <sip:192.168.1.62:5060;transport=udp>
Session: Media
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE
Content-Type: application/sdp
Content-Length: 227

v=0
o=BroadWorks 50419338 1 IN IP4 192.168.1.62
s=-
c=IN IP4 192.168.1.62
t=0 0
m=audio 16840 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
<------------->
--- (12 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.62:16840
    -- SIP/myvoipprovideraccount-0000005e is making progress passing it to SIP/140-0000005d
Really destroying SIP dialog '824573088552@10.243.79.87-S' Method: SUBSCRIBE

<--- SIP read from UDP:192.168.1.62:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.60:5060;received=184.184.24.166;branch=z9hG4bK2b13fb91;rport=5060
Record-Route: <sip:Asterisks@192.168.1.62;lr>
From: "Insystech, Inc" <sip:mydid@192.168.1.62>;tag=as0ecbe613
To: <sip:y.y.y.y@192.168.1.62>;tag=SDl9rs399-1720459397-1345493673623
Call-ID: 61422484218e8cf71b03bf636ee52f89@192.168.1.62
CSeq: 103 INVITE
Contact: <sip:192.168.1.62:5060;transport=udp>
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE
Content-Type: application/sdp
Accept: multipart/mixed, application/media_control+xml, application/sdp
Content-Length: 227

v=0
o=BroadWorks 50419338 1 IN IP4 192.168.1.62
s=-
c=IN IP4 192.168.1.62
t=0 0
m=audio 16840 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
<------------->
--- (12 headers 11 lines) ---
list_route: hop: <sip:Asterisks@192.168.1.62;lr>
set_destination: Parsing <sip:Asterisks@192.168.1.62;lr> for address/port to send to
set_destination: set destination to 192.168.1.62:5060
Transmitting (NAT) to 192.168.1.62:5060:
ACK sip:192.168.1.62:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.60:5060;branch=z9hG4bK77857718;rport
Route: <sip:Asterisks@192.168.1.62;lr>
Max-Forwards: 70
From: "Insystech, Inc" <sip:mydid@192.168.1.62>;tag=as0ecbe613
To: <sip:y.y.y.y@192.168.1.62>;tag=SDl9rs399-1720459397-1345493673623
Contact: <sip:mydid@192.168.1.60:5060>
Call-ID: 61422484218e8cf71b03bf636ee52f89@192.168.1.62
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.7.0
Content-Length: 0


---
    -- SIP/myvoipprovideraccount-0000005e answered SIP/140-0000005d
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 174.252.121.19:4665 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 174.252.121.19:4665;branch=z9hG4bK+f89443b9aff42fa263984a680e57a84c1+s36+1;received=174.252.121.19;rport=4665
From: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
To: <sip:9y.y.y.y@x.x.x.x>;tag=as3d87b2f9
Call-ID: 581844940246@10.243.79.87-S
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:9y.y.y.y@x.x.x.x:5060>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 869302929 869302929 IN IP4 x.x.x.x
s=Asterisk PBX 1.8.7.0
c=IN IP4 x.x.x.x
t=0 0
m=audio 14114 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 103

<------------>
    -- Locally bridging SIP/140-0000005d and SIP/myvoipprovideraccount-0000005e

<--- SIP read from UDP:174.252.121.19:4665 --->
ACK sip:9y.y.y.y@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP 174.252.121.19:4665;branch=z9hG4bK+72ef2efb186307b2dced56b7d4ea8ecf1+s36+1
Call-ID: 581844940246@10.243.79.87-S
From: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
To: <sip:9y.y.y.y@x.x.x.x>;tag=as3d87b2f9
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:140@174.252.121.19:4665;transport=udp>
Expires: 3600
User-Agent: Sipdroid/2.7 beta/DROID2
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Retransmitting #1 (NAT) to 174.252.121.19:4665:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 174.252.121.19:4665;branch=z9hG4bK+f89443b9aff42fa263984a680e57a84c1+s36+1;received=174.252.121.19;rport=4665
From: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
To: <sip:9y.y.y.y@x.x.x.x>;tag=as3d87b2f9
Call-ID: 581844940246@10.243.79.87-S
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:9y.y.y.y@x.x.x.x:5060>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 869302929 869302929 IN IP4 x.x.x.x
s=Asterisk PBX 1.8.7.0
c=IN IP4 x.x.x.x
t=0 0
m=audio 14114 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 103

---

<--- SIP read from UDP:174.252.121.19:4665 --->
ACK sip:9y.y.y.y@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP 174.252.121.19:4665;branch=z9hG4bK+72ef2efb186307b2dced56b7d4ea8ecf1+s36+1
Call-ID: 581844940246@10.243.79.87-S
From: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
To: <sip:9y.y.y.y@x.x.x.x>;tag=as3d87b2f9
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:140@174.252.121.19:4665;transport=udp>
Expires: 3600
User-Agent: Sipdroid/2.7 beta/DROID2
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Retransmitting #2 (NAT) to 174.252.121.19:4665:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 174.252.121.19:4665;branch=z9hG4bK+f89443b9aff42fa263984a680e57a84c1+s36+1;received=174.252.121.19;rport=4665
From: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
To: <sip:9y.y.y.y@x.x.x.x>;tag=as3d87b2f9
Call-ID: 581844940246@10.243.79.87-S
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:9y.y.y.y@x.x.x.x:5060>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 869302929 869302929 IN IP4 x.x.x.x
s=Asterisk PBX 1.8.7.0
c=IN IP4 x.x.x.x
t=0 0
m=audio 14114 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 103

---

<--- SIP read from UDP:174.252.121.19:4665 --->
ACK sip:9y.y.y.y@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP 174.252.121.19:4665;branch=z9hG4bK+72ef2efb186307b2dced56b7d4ea8ecf1+s36+1
Call-ID: 581844940246@10.243.79.87-S
From: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
To: <sip:9y.y.y.y@x.x.x.x>;tag=as3d87b2f9
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:140@174.252.121.19:4665;transport=udp>
Expires: 3600
User-Agent: Sipdroid/2.7 beta/DROID2
Content-Length: 0

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

Retransmitting #3 (NAT) to 174.252.121.19:4665:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 174.252.121.19:4665;branch=z9hG4bK+f89443b9aff42fa263984a680e57a84c1+s36+1;received=174.252.121.19;rport=4665
From: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
To: <sip:9y.y.y.y@x.x.x.x>;tag=as3d87b2f9
Call-ID: 581844940246@10.243.79.87-S
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:9y.y.y.y@x.x.x.x:5060>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 869302929 869302929 IN IP4 x.x.x.x
s=Asterisk PBX 1.8.7.0
c=IN IP4 x.x.x.x
t=0 0
m=audio 14114 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 103

---

<--- SIP read from UDP:174.252.121.19:4665 --->
ACK sip:9y.y.y.y@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP 174.252.121.19:4665;branch=z9hG4bK+72ef2efb186307b2dced56b7d4ea8ecf1+s36+1
Call-ID: 581844940246@10.243.79.87-S
From: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
To: <sip:9y.y.y.y@x.x.x.x>;tag=as3d87b2f9
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:140@174.252.121.19:4665;transport=udp>
Expires: 3600
User-Agent: Sipdroid/2.7 beta/DROID2
Content-Length: 0

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

Retransmitting #4 (NAT) to 174.252.121.19:4665:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 174.252.121.19:4665;branch=z9hG4bK+f89443b9aff42fa263984a680e57a84c1+s36+1;received=174.252.121.19;rport=4665
From: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
To: <sip:9y.y.y.y@x.x.x.x>;tag=as3d87b2f9
Call-ID: 581844940246@10.243.79.87-S
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:9y.y.y.y@x.x.x.x:5060>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 869302929 869302929 IN IP4 x.x.x.x
s=Asterisk PBX 1.8.7.0
c=IN IP4 x.x.x.x
t=0 0
m=audio 14114 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 103

---

<--- SIP read from UDP:174.252.121.19:4665 --->
ACK sip:9y.y.y.y@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP 174.252.121.19:4665;branch=z9hG4bK+72ef2efb186307b2dced56b7d4ea8ecf1+s36+1
Call-ID: 581844940246@10.243.79.87-S
From: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
To: <sip:9y.y.y.y@x.x.x.x>;tag=as3d87b2f9
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:140@174.252.121.19:4665;transport=udp>
Expires: 3600
User-Agent: Sipdroid/2.7 beta/DROID2
Content-Length: 0

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

Retransmitting #5 (NAT) to 174.252.121.19:4665:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 174.252.121.19:4665;branch=z9hG4bK+f89443b9aff42fa263984a680e57a84c1+s36+1;received=174.252.121.19;rport=4665
From: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
To: <sip:9y.y.y.y@x.x.x.x>;tag=as3d87b2f9
Call-ID: 581844940246@10.243.79.87-S
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:9y.y.y.y@x.x.x.x:5060>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 869302929 869302929 IN IP4 x.x.x.x
s=Asterisk PBX 1.8.7.0
c=IN IP4 x.x.x.x
t=0 0
m=audio 14114 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 103

---

<--- SIP read from UDP:174.252.121.19:4665 --->
ACK sip:9y.y.y.y@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP 174.252.121.19:4665;branch=z9hG4bK+72ef2efb186307b2dced56b7d4ea8ecf1+s36+1
Call-ID: 581844940246@10.243.79.87-S
From: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
To: <sip:9y.y.y.y@x.x.x.x>;tag=as3d87b2f9
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:140@174.252.121.19:4665;transport=udp>
Expires: 3600
User-Agent: Sipdroid/2.7 beta/DROID2
Content-Length: 0

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

<--- SIP read from UDP:174.252.121.19:4665 --->
SUBSCRIBE sip:140@x.x.x.x SIP/2.0
Via: SIP/2.0/UDP 174.252.121.19:4665;rport;branch=z9hG4bK+282c0df077fa25654043dfe3910c53da1+s36+1
From: <sip:140@x.x.x.x>;tag=s36+1+4262000b+52ccc389
To: <sip:140@x.x.x.x>
Call-ID: 824573088552@10.243.79.87-S
Max-Forwards: 70
CSeq: 3 SUBSCRIBE
Contact: <sip:140@174.252.121.19:4665;transport=udp>
Expires: 184000
User-Agent: Sipdroid/2.7 beta/DROID2
Event: message-summary
Accept: application/simple-message-summary
Authorization: Digest username="140", realm="asterisk", nonce="0d9e1e0e", uri="sip:140@x.x.x.x", algorithm=MD5, response="24e99c6da183e6bbebe3e925c7d1ed5b"
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Creating new subscription
Sending to 174.252.121.19:4665 (no NAT)
list_route: hop: <sip:140@174.252.121.19:4665;transport=udp>
Found peer '140' for '140' from 174.252.121.19:4665

<--- Transmitting (NAT) to 174.252.121.19:4665 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 174.252.121.19:4665;branch=z9hG4bK+282c0df077fa25654043dfe3910c53da1+s36+1;received=174.252.121.19;rport=4665
From: <sip:140@x.x.x.x>;tag=s36+1+4262000b+52ccc389
To: <sip:140@x.x.x.x>;tag=as77e3c7ac
Call-ID: 824573088552@10.243.79.87-S
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX 1.8.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2e3683d4"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '824573088552@10.243.79.87-S' in 32064 ms (Method: SUBSCRIBE)
Retransmitting #6 (NAT) to 174.252.121.19:4665:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 174.252.121.19:4665;branch=z9hG4bK+f89443b9aff42fa263984a680e57a84c1+s36+1;received=174.252.121.19;rport=4665
From: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
To: <sip:9y.y.y.y@x.x.x.x>;tag=as3d87b2f9
Call-ID: 581844940246@10.243.79.87-S
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:9y.y.y.y@x.x.x.x:5060>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 869302929 869302929 IN IP4 x.x.x.x
s=Asterisk PBX 1.8.7.0
c=IN IP4 x.x.x.x
t=0 0
m=audio 14114 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 103

---

<--- SIP read from UDP:174.252.121.19:4665 --->
ACK sip:9y.y.y.y@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP 174.252.121.19:4665;branch=z9hG4bK+72ef2efb186307b2dced56b7d4ea8ecf1+s36+1
Call-ID: 581844940246@10.243.79.87-S
From: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
To: <sip:9y.y.y.y@x.x.x.x>;tag=as3d87b2f9
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:140@174.252.121.19:4665;transport=udp>
Expires: 3600
User-Agent: Sipdroid/2.7 beta/DROID2
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '080a1bc257264e90051e2b015832b97a@127.0.1.1' Method: REGISTER
Retransmitting #7 (NAT) to 174.252.121.19:4665:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 174.252.121.19:4665;branch=z9hG4bK+f89443b9aff42fa263984a680e57a84c1+s36+1;received=174.252.121.19;rport=4665
From: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
To: <sip:9y.y.y.y@x.x.x.x>;tag=as3d87b2f9
Call-ID: 581844940246@10.243.79.87-S
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:9y.y.y.y@x.x.x.x:5060>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 869302929 869302929 IN IP4 x.x.x.x
s=Asterisk PBX 1.8.7.0
c=IN IP4 x.x.x.x
t=0 0
m=audio 14114 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 103

---

<--- SIP read from UDP:174.252.121.19:4665 --->
ACK sip:9y.y.y.y@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP 174.252.121.19:4665;branch=z9hG4bK+72ef2efb186307b2dced56b7d4ea8ecf1+s36+1
Call-ID: 581844940246@10.243.79.87-S
From: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
To: <sip:9y.y.y.y@x.x.x.x>;tag=as3d87b2f9
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:140@174.252.121.19:4665;transport=udp>
Expires: 3600
User-Agent: Sipdroid/2.7 beta/DROID2
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Retransmitting #8 (NAT) to 174.252.121.19:4665:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 174.252.121.19:4665;branch=z9hG4bK+f89443b9aff42fa263984a680e57a84c1+s36+1;received=174.252.121.19;rport=4665
From: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
To: <sip:9y.y.y.y@x.x.x.x>;tag=as3d87b2f9
Call-ID: 581844940246@10.243.79.87-S
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:9y.y.y.y@x.x.x.x:5060>
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 869302929 869302929 IN IP4 x.x.x.x
s=Asterisk PBX 1.8.7.0
c=IN IP4 x.x.x.x
t=0 0
m=audio 14114 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 103

---

<--- SIP read from UDP:174.252.121.19:4665 --->
ACK sip:9y.y.y.y@x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP 174.252.121.19:4665;branch=z9hG4bK+72ef2efb186307b2dced56b7d4ea8ecf1+s36+1
Call-ID: 581844940246@10.243.79.87-S
From: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
To: <sip:9y.y.y.y@x.x.x.x>;tag=as3d87b2f9
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:140@174.252.121.19:4665;transport=udp>
Expires: 3600
User-Agent: Sipdroid/2.7 beta/DROID2
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
[Aug 20 16:16:33] WARNING[1039]: chan_sip.c:3622 retrans_pkt: Retransmission timeout reached on transmission 581844940246@10.243.79.87-S for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 25023ms with no response
[Aug 20 16:16:33] WARNING[1039]: chan_sip.c:3651 retrans_pkt: Hanging up call 581844940246@10.243.79.87-S - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
Scheduling destruction of SIP dialog '61422484218e8cf71b03bf636ee52f89@192.168.1.62' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:Asterisks@192.168.1.62;lr> for address/port to send to
set_destination: set destination to 192.168.1.62:5060
Reliably Transmitting (NAT) to 192.168.1.62:5060:
BYE sip:192.168.1.62:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.60:5060;branch=z9hG4bK07c5bc4c;rport
Route: <sip:Asterisks@192.168.1.62;lr>
Max-Forwards: 70
From: "Insystech, Inc" <sip:mydid@192.168.1.62>;tag=as0ecbe613
To: <sip:y.y.y.y@192.168.1.62>;tag=SDl9rs399-1720459397-1345493673623
Call-ID: 61422484218e8cf71b03bf636ee52f89@192.168.1.62
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.8.7.0
Authorization: Digest username="477049014802", realm="BroadWorks", algorithm=MD5, uri="sip:192.168.1.62:5060", nonce="BroadWorksXh6406xtaT5jcb1bBW", response="ead5d31336b732b19166b86f5b66dee6", qop=auth, cnonce="20614a5a", nc=00000002
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (default, 9y.y.y.y, 2) exited non-zero on 'SIP/140-0000005d'
Scheduling destruction of SIP dialog '581844940246@10.243.79.87-S' in 25024 ms (Method: INVITE)
set_destination: Parsing <sip:140@174.252.121.19:4665;transport=udp> for address/port to send to
set_destination: set destination to 174.252.121.19:4665
Reliably Transmitting (NAT) to 174.252.121.19:4665:
BYE sip:140@174.252.121.19:4665;transport=udp SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK39ee4ac4;rport
Max-Forwards: 70
From: <sip:9y.y.y.y@x.x.x.x>;tag=as3d87b2f9
To: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
Call-ID: 581844940246@10.243.79.87-S
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.7.0
Proxy-Authorization: Digest username="140", realm="asterisk", algorithm=MD5, uri="sip:x.x.x.x", nonce="", response="641a0f9ebe179a4e15a0ec46b711a464"
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.62:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.60:5060;received=184.184.24.166;branch=z9hG4bK07c5bc4c;rport=5060
Record-Route: <sip:Asterisks@192.168.1.62;lr>
From: "Insystech, Inc" <sip:mydid@192.168.1.62>;tag=as0ecbe613
To: <sip:y.y.y.y@192.168.1.62>;tag=SDl9rs399-1720459397-1345493673623
Call-ID: 61422484218e8cf71b03bf636ee52f89@192.168.1.62
CSeq: 104 BYE
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '61422484218e8cf71b03bf636ee52f89@192.168.1.62' Method: INVITE
Really destroying SIP dialog '530399074390@10.243.79.87' Method: REGISTER

<--- SIP read from UDP:174.252.121.19:4665 --->
SIP/2.0 200 OK
Call-ID: 581844940246@10.243.79.87-S
CSeq: 102 BYE
From: <sip:9y.y.y.y@x.x.x.x>;tag=as3d87b2f9
To: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
Via: SIP/2.0/UDP x.x.x.x:5060;received=x.x.x.x;rport=5060;branch=z9hG4bK39ee4ac4
Server: Sipdroid/2.7 beta/DROID2
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Retransmitting #1 (NAT) to 174.252.121.19:4665:
BYE sip:140@174.252.121.19:4665;transport=udp SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK39ee4ac4;rport
Max-Forwards: 70
From: <sip:9y.y.y.y@x.x.x.x>;tag=as3d87b2f9
To: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
Call-ID: 581844940246@10.243.79.87-S
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.7.0
Proxy-Authorization: Digest username="140", realm="asterisk", algorithm=MD5, uri="sip:x.x.x.x", nonce="", response="641a0f9ebe179a4e15a0ec46b711a464"
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0


---

<--- SIP read from UDP:174.252.121.19:4665 --->
SIP/2.0 200 OK
Call-ID: 581844940246@10.243.79.87-S
CSeq: 102 BYE
From: <sip:9y.y.y.y@x.x.x.x>;tag=as3d87b2f9
To: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
Via: SIP/2.0/UDP x.x.x.x:5060;received=x.x.x.x;rport=5060;branch=z9hG4bK39ee4ac4
Server: Sipdroid/2.7 beta/DROID2
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '604050037420@10.243.79.87-S' Method: SUBSCRIBE

<------------>
Scheduling destruction of SIP dialog 'MTFjNzE0MjZjNjljMjVmMmEwMTVkMzc5MmFhY2Y4NDU.' in 32000 ms (Method: REGISTER)

<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '495d9b43554058843a22a7f01821cdab@192.168.1.60:5060' Method: OPTIONS
Retransmitting #2 (NAT) to 174.252.121.19:4665:
BYE sip:140@174.252.121.19:4665;transport=udp SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK39ee4ac4;rport
Max-Forwards: 70
From: <sip:9y.y.y.y@x.x.x.x>;tag=as3d87b2f9
To: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
Call-ID: 581844940246@10.243.79.87-S
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.7.0
Proxy-Authorization: Digest username="140", realm="asterisk", algorithm=MD5, uri="sip:x.x.x.x", nonce="", response="641a0f9ebe179a4e15a0ec46b711a464"
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0


---

<--- SIP read from UDP:174.252.121.19:4665 --->
SIP/2.0 200 OK
Call-ID: 581844940246@10.243.79.87-S
CSeq: 102 BYE
From: <sip:9y.y.y.y@x.x.x.x>;tag=as3d87b2f9
To: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
Via: SIP/2.0/UDP x.x.x.x:5060;received=x.x.x.x;rport=5060;branch=z9hG4bK39ee4ac4
Server: Sipdroid/2.7 beta/DROID2
Content-Length: 0

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

Retransmitting #3 (NAT) to 174.252.121.19:4665:
BYE sip:140@174.252.121.19:4665;transport=udp SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK39ee4ac4;rport
Max-Forwards: 70
From: <sip:9y.y.y.y@x.x.x.x>;tag=as3d87b2f9
To: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
Call-ID: 581844940246@10.243.79.87-S
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.7.0
Proxy-Authorization: Digest username="140", realm="asterisk", algorithm=MD5, uri="sip:x.x.x.x", nonce="", response="641a0f9ebe179a4e15a0ec46b711a464"
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0


---

<--- SIP read from UDP:174.252.121.19:4665 --->
SIP/2.0 200 OK
Call-ID: 581844940246@10.243.79.87-S
CSeq: 102 BYE
From: <sip:9y.y.y.y@x.x.x.x>;tag=as3d87b2f9
To: <sip:140@x.x.x.x>;tag=s36+1+4272000b+6fc35e44
Via: SIP/2.0/UDP x.x.x.x:5060;received=x.x.x.x;rport=5060;branch=z9hG4bK39ee4ac4
Server: Sipdroid/2.7 beta/DROID2
Content-Length: 0

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

<--- SIP read from UDP:174.252.121.19:4665 --->
SUBSCRIBE sip:140@x.x.x.x SIP/2.0
Via: SIP/2.0/UDP 174.252.121.19:4665;rport;branch=z9hG4bK+9f04a184dadf74685a1404e5ed0705391+s36+1
From: <sip:140@x.x.x.x>;tag=s36+1+425c000b+72b0d1e2
To: <sip:140@x.x.x.x>
Call-ID: 604050037420@10.243.79.87-S
Max-Forwards: 70
CSeq: 2 SUBSCRIBE
Contact: <sip:140@174.252.121.19:4665;transport=udp>
Expires: 184000
User-Agent: Sipdroid/2.7 beta/DROID2
Event: message-summary
Accept: application/simple-message-summary
Authorization: Digest username="140", realm="asterisk", nonce="27f4e829", uri="sip:140@x.x.x.x", algorithm=MD5, response="f3e04512a2b154930cde9a8d64abbf35"
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Creating new subscription
Sending to 174.252.121.19:4665 (no NAT)
list_route: hop: <sip:140@174.252.121.19:4665;transport=udp>
Found peer '140' for '140' from 174.252.121.19:4665

<--- Transmitting (NAT) to 174.252.121.19:4665 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 174.252.121.19:4665;branch=z9hG4bK+9f04a184dadf74685a1404e5ed0705391+s36+1;received=174.252.121.19;rport=4665
From: <sip:140@x.x.x.x>;tag=s36+1+425c000b+72b0d1e2
To: <sip:140@x.x.x.x>;tag=as3fa57e5f
Call-ID: 604050037420@10.243.79.87-S
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 1.8.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7f060eab"
Content-Length: 0


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

Any and all help is greatly appreciated, thanks.

-John

—updated to include Asterisk version

2

Branch ID on ACK doesn’t match the branch ID on the OK that is acking.

3

Ok… I fully understand what you’re saying, but my question is how do I fix it without borking the rest of the system? In such a way that a user can switch from being being external to internal without an issue?

4

You really need to fix the broken remote device. However, there is a SIP option that reduces the strictness with which Asterisk imlements SIP. You may find that if you find that option and set it to the less compliant mode, that it will tolerate the error.

5

The remote device being used for testing is using a very basic SIP client (SIPDroid) with almost no settings to it. All it does it take the username/password and authenticates to the server. I’m testing using this because the end hope being that my non-technical users who will connect remotely can just plugin the information and be good to go.

I’m looking for that SIP option in Asterisk (any clue what it might be?) in the meantime and I’ll give that a try.

Any other clues or suggestions absolutely welcome.

6

sip.conf.sample:

;pedantic=yes ; Enable checking of tags in headers, ; international character conversions in URIs ; and multiline formatted headers for strict ; SIP compatibility (defaults to "no")
This would be the first choice for working wround broken SIP implementations. Especially if this works, you should raise a bug report against the remote device.

The default may have changed.

7 
pedantic=no

This fixed it! I found it very odd that I had to specify it as no, considering that according to the documentation the default value should have already been no.

Regardless, I have already had an external client connect using X-Lite and have maintained multiple calls for over 15 minutes each (well above and beyond the 30 seconds happening before!) and according to my sip debugging the packets are all going where they should be.

Thank you very much for the help!

8

The default changed. That quote was from an old version.

Please submit the bug report for the broken device that is sending the bad tags.