Sebastian Nielsen
2019-Oct-14 06:59 UTC
[asterisk-users] Problems with calls dropping on Android.
Hello.
I have the following in sip.conf
[sip09]
type=peer
defaultuser=sip09
nat=yes
qualify=no
secret=sip09
host=dynamic
context=outgoing
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
allow=h263p
deny=0.0.0.0/0.0.0.0
permit=192.168.2.2/255.255.255.255
jbenable = yes
jbforce = yes
jbmaxsize = 100
jbresyncthreshold = 200
jbimpl = fixed
transport=tcp
sendrpid=yes
And these settings in Android native client.
Username: sip09
Password: sip09
Server: 192.168.1.10
Username at authentication: sip09
Display name: Same as username
Outgoing proxy: 192.168.1.10
Port: 5060
Transport: TCP
Send keep alive: Always
However, if I make a call FROM android phone, call is dropped after 30
seconds, regardless of answer or not. If I make call TO android phone, it
works normally.
No NAT problems inbetween, there is a VPN between the phone and SIP server
with full access.
I guess I need to do some trick to have it work with Android. Apparently the
packets are received in both ends - else audio wouldn't work, but guess the
stock native SIP client on android ignores certain packets right?
This is an Android 9 phone.
Additionally, I wonder if its possible to change the callerid shown in
display when calling out? Like RPID. It works on my desktop phones, if I
enter a short code, the full name and number is shown on display, but on the
Android phone, it doesn't work, only the dialled shortnumber is shown.
Also I wonder if its possible to have asterisk send the remote callerid
(when receiving a call) in such a way it gets stored in call log with full
names and such - without having to resort to using phonebook.
SIP debug log:
*CLI> sip set debug ip 192.168.2.2
SIP Debugging Enabled for IP: 192.168.2.2
*CLI> Really destroying SIP dialog
'6f9956035553ab1b79ca057f5dffe0ac at 192.168.2.2' Method: OPTIONS
Really destroying SIP dialog 'fc3307059c816094a6c6ce100cf383e5 at
192.168.2.2'
Method: OPTIONS
<--- SIP read from TCP:192.168.2.2:51729 --->
OPTIONS sip:192.168.1.10 SIP/2.0
Call-ID: e65234cb818a143bc3c167a782b98e96 at 192.168.2.2
CSeq: 3984 OPTIONS
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=3997716169
To: "sip09" <sip:sip09 at 192.168.1.10>
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK105b3648c13a72f8fbe7ce3049df71aa3130;rport
Max-Forwards: 70
User-Agent: SIPAUA/0.1.001
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.2.2:51729 (no NAT)
Looking for s in cellip (domain 192.168.1.10)
<--- Transmitting (no NAT) to 192.168.2.2:51729 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK105b3648c13a72f8fbe7ce3049df71aa3130;receive
d=192.168.2.2;rport=51729
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=3997716169
To: "sip09" <sip:sip09 at 192.168.1.10>;tag=as4c9bb00e
Call-ID: e65234cb818a143bc3c167a782b98e96 at 192.168.2.2
CSeq: 3984 OPTIONS
Server: Asterisk PBX 13.21.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces,timer
Contact: <sip:192.168.1.10:5060;transport=tcp>
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'e65234cb818a143bc3c167a782b98e96 at 192.168.2.2' in 32000 ms (Method:
OPTIONS)
<--- SIP read from TCP:192.168.2.2:51729 --->
INVITE sip:02 at 192.168.1.10 SIP/2.0
Call-ID: fcaad738faee2d0250d0cf2366139979 at 192.168.2.2
CSeq: 9116 INVITE
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=3432177901
To: <sip:02 at 192.168.1.10>
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bKf8bf8138c906000bc3f8601a5df558943130;rport
Max-Forwards: 70
Contact: "sip09" <sip:sip09 at 192.168.2.2:56334;transport=tcp>
Content-Type: application/sdp
Content-Length: 295
v=0
o=- 1571035683065 1571035683066 IN IP4 192.168.2.2
s=-
c=IN IP4 192.168.2.2
t=0 0
m=audio 26726 RTP/AVP 96 97 3 0 8 127
a=rtpmap:96 GSM-EFR/8000
a=rtpmap:97 AMR/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:127 telephone-event/8000
a=fmtp:127 0-15
<------------->
--- (10 headers 13 lines) ---
Sending to 192.168.2.2:51729 (no NAT)
Sending to 192.168.2.2:51729 (no NAT)
Using INVITE request as basis request -
fcaad738faee2d0250d0cf2366139979 at 192.168.2.2
Found peer 'sip09' for 'sip09' from 192.168.2.2:51729
<--- Reliably Transmitting (NAT) to 192.168.2.2:51729 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bKf8bf8138c906000bc3f8601a5df558943130;receive
d=192.168.2.2;rport=51729
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=3432177901
To: <sip:02 at 192.168.1.10>;tag=as4d53b5f5
Call-ID: fcaad738faee2d0250d0cf2366139979 at 192.168.2.2
CSeq: 9116 INVITE
Server: Asterisk PBX 13.21.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="6dc98e50"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'fcaad738faee2d0250d0cf2366139979 at 192.168.2.2' in 32000 ms (Method:
INVITE)
<--- SIP read from TCP:192.168.2.2:51729 --->
ACK sip:02 at 192.168.1.10 SIP/2.0
Call-ID: fcaad738faee2d0250d0cf2366139979 at 192.168.2.2
Max-Forwards: 70
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=3432177901
To: <sip:02 at 192.168.1.10>;tag=as4d53b5f5
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bKf8bf8138c906000bc3f8601a5df558943130;rport
CSeq: 9116 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from TCP:192.168.2.2:51729 --->
INVITE sip:02 at 192.168.1.10:5060 SIP/2.0
Call-ID: fcaad738faee2d0250d0cf2366139979 at 192.168.2.2
CSeq: 9117 INVITE
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=3432177901
To: <sip:02 at 192.168.1.10>
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK904a02d6a0fd6260129bcfdff3c18d343130;rport
Max-Forwards: 70
Contact: "sip09" <sip:sip09 at 192.168.2.2:56334;transport=tcp>
Content-Type: application/sdp
Authorization: Digest
username="sip09",realm="asterisk",nonce="6dc98e50",uri="sip:02
at 192.168.1.10:
5060",response="acc3dc6bebc31320467ebccd1bfe19b5",algorithm=MD5
Content-Length: 295
v=0
o=- 1571035683065 1571035683066 IN IP4 192.168.2.2
s=-
c=IN IP4 192.168.2.2
t=0 0
m=audio 26726 RTP/AVP 96 97 3 0 8 127
a=rtpmap:96 GSM-EFR/8000
a=rtpmap:97 AMR/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:127 telephone-event/8000
a=fmtp:127 0-15
<------------->
--- (11 headers 13 lines) ---
Sending to 192.168.2.2:51729 (no NAT)
Sending to 192.168.2.2:51729 (no NAT)
Using INVITE request as basis request -
fcaad738faee2d0250d0cf2366139979 at 192.168.2.2
Found peer 'sip09' for 'sip09' from 192.168.2.2:51729
<--- Reliably Transmitting (NAT) to 192.168.2.2:51729 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK904a02d6a0fd6260129bcfdff3c18d343130;receive
d=192.168.2.2;rport=51729
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=3432177901
To: <sip:02 at 192.168.1.10>;tag=as5bed3900
Call-ID: fcaad738faee2d0250d0cf2366139979 at 192.168.2.2
CSeq: 9117 INVITE
Server: Asterisk PBX 13.21.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="4e4178ea"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'fcaad738faee2d0250d0cf2366139979 at 192.168.2.2' in 32000 ms (Method:
INVITE)
<--- SIP read from TCP:192.168.2.2:51729 --->
ACK sip:02 at 192.168.1.10:5060 SIP/2.0
Call-ID: fcaad738faee2d0250d0cf2366139979 at 192.168.2.2
Max-Forwards: 70
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=3432177901
To: <sip:02 at 192.168.1.10>;tag=as5bed3900
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK904a02d6a0fd6260129bcfdff3c18d343130;rport
CSeq: 9117 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from TCP:192.168.2.2:51729 --->
INVITE sip:02 at 192.168.1.10:5060 SIP/2.0
Call-ID: fcaad738faee2d0250d0cf2366139979 at 192.168.2.2
CSeq: 9118 INVITE
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=3432177901
To: <sip:02 at 192.168.1.10>
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK3c0e21b640ac5c94489220a97aa992c63130;rport
Max-Forwards: 70
Contact: "sip09" <sip:sip09 at 192.168.2.2:56334;transport=tcp>
Content-Type: application/sdp
Authorization: Digest
username="sip09",realm="asterisk",nonce="4e4178ea",uri="sip:02
at 192.168.1.10:
5060",response="6af8fb169df3518374a93ab990c1048c",algorithm=MD5
Content-Length: 295
v=0
o=- 1571035683065 1571035683066 IN IP4 192.168.2.2
s=-
c=IN IP4 192.168.2.2
t=0 0
m=audio 26726 RTP/AVP 96 97 3 0 8 127
a=rtpmap:96 GSM-EFR/8000
a=rtpmap:97 AMR/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:127 telephone-event/8000
a=fmtp:127 0-15
<------------->
--- (11 headers 13 lines) ---
Sending to 192.168.2.2:51729 (NAT)
Using INVITE request as basis request -
fcaad738faee2d0250d0cf2366139979 at 192.168.2.2
Found peer 'sip09' for 'sip09' from 192.168.2.2:51729
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 127
Found unknown media description format GSM-EFR for ID 96
Found unknown media description format AMR for ID 97
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 127
Capabilities: us - (ulaw|alaw|h263p), peer -
audio=(ulaw|gsm|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
> 0x7f4fac02c9f0 -- Strict RTP learning after remote address set to:
192.168.2.2:26726
Peer audio RTP is at port 192.168.2.2:26726
Peer doesn't provide video
Looking for 02 in outgoing (domain 192.168.1.10)
sip_route_dump: route/path hop: <sip:sip09 at
192.168.2.2:56334;transport=tcp>
<--- Transmitting (NAT) to 192.168.2.2:51729 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK3c0e21b640ac5c94489220a97aa992c63130;receive
d=192.168.2.2;rport=51729
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=3432177901
To: <sip:02 at 192.168.1.10>
Call-ID: fcaad738faee2d0250d0cf2366139979 at 192.168.2.2
CSeq: 9118 INVITE
Server: Asterisk PBX 13.21.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces,timer
Contact: <sip:02 at 192.168.1.10:5060;transport=tcp>
Content-Length: 0
<------------>
-- Executing [02 at outgoing:1] Set("SIP/sip09-00000004",
"oex=02") in new
stack
-- Executing [02 at outgoing:2] Goto("SIP/sip09-00000004",
"noblf,s,1") in
new stack
-- Goto (noblf,s,1)
-- Executing [s at noblf:1] Set("SIP/sip09-00000004",
"clid=567169") in new
stack
-- Executing [s at noblf:2] Answer("SIP/sip09-00000004",
"") in new stack
Audio is at 5180
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 192.168.2.2:51729 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK3c0e21b640ac5c94489220a97aa992c63130;receive
d=192.168.2.2;rport=51729
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=3432177901
To: <sip:02 at 192.168.1.10>;tag=as6255d020
Call-ID: fcaad738faee2d0250d0cf2366139979 at 192.168.2.2
CSeq: 9118 INVITE
Server: Asterisk PBX 13.21.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces,timer
Contact: <sip:02 at 192.168.1.10:5060;transport=tcp>
Content-Type: application/sdp
Content-Length: 239
v=0
o=root 1088448975 1088448975 IN IP4 192.168.1.10
s=Asterisk PBX 13.21.1
c=IN IP4 192.168.1.10
t=0 0
m=audio 5180 RTP/AVP 0 127
a=rtpmap:0 PCMU/8000
a=rtpmap:127 telephone-event/8000
a=fmtp:127 0-16
a=maxptime:150
a=sendrecv
<------------>
<--- SIP read from TCP:192.168.2.2:51729 --->
ACK sip:02 at 192.168.1.10:5060;transport=tcp SIP/2.0
Call-ID: fcaad738faee2d0250d0cf2366139979 at 192.168.2.2
CSeq: 9118 ACK
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK3f24f8893ceaaa5fefe03c60346550eb3130
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=3432177901
To: <sip:02 at 192.168.1.10>;tag=as6255d020
Max-Forwards: 70
Authorization: Digest
username="sip09",realm="asterisk",nonce="4e4178ea",uri="sip:02
at 192.168.1.10:
5060",response="6af8fb169df3518374a93ab990c1048c",algorithm=MD5
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
> 0x7f4fac02c9f0 -- Strict RTP switching to RTP target address
192.168.2.2:26726 as source
-- Executing [s at noblf:3] GotoIf("SIP/sip09-00000004",
"0?invalidnumber,s,1") in new stack
-- Executing [s at noblf:4] Set("SIP/sip09-00000004",
"orignum=02") in new
stack
-- Executing [s at noblf:5] GotoIf("SIP/sip09-00000004",
"0?invalidnumber,s,1") in new stack
-- Executing [s at noblf:6] GotoIf("SIP/sip09-00000004",
"1?intercom,s,1")
in new stack
-- Goto (intercom,s,1)
-- Executing [s at intercom:1] Set("SIP/sip09-00000004",
"FILE(/var/secure_files/voicelog.txt,,,al,u)=ic,567169,20191014084803,02,02,
") in new stack
-- Executing [s at intercom:2] MixMonitor("SIP/sip09-00000004",
"/var/secure_files/recordings/ic-567169-20191014084803-02-02.wav") in
new
stack
-- Executing [s at intercom:3] Set("SIP/sip09-00000004",
"dialstring=SIP/sip01&SIP/sip02&SIP/sip03&SIP/sip04&SIP/sip05&SIP/sip06&SIP/
sip07&SIP/sip08&SIP/sip09") in new stack
== Begin MixMonitor Recording SIP/sip09-00000004
-- Executing [s at intercom:4] ExecIf("SIP/sip09-00000004",
"1?Set(dialstring=SIP/sip01&SIP/sip02&SIP/sip03&SIP/sip04&SIP/sip05&SIP/sip0
6&SIP/sip07&SIP/sip08&SIP/sip10)") in new stack
-- Executing [s at intercom:5] Set("SIP/sip09-00000004",
"CONNECTEDLINE(number,i)=02") in new stack
-- Executing [s at intercom:6] Set("SIP/sip09-00000004",
"CONNECTEDLINE(name,i)=Internsamtal") in new stack
-- Executing [s at intercom:7] Set("SIP/sip09-00000004",
"CONNECTEDLINE(num-presn,i)=allowed") in new stack
-- Executing [s at intercom:8] Set("SIP/sip09-00000004",
"CONNECTEDLINE(name-pres)=allowed") in new stack
Reliably Transmitting (NAT) to 192.168.2.2:51729:
UPDATE sip:sip09 at 192.168.2.2:56334;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.10:5060;branch=z9hG4bK7cb2f0d0;rport
Max-Forwards: 70
From: <sip:02 at 192.168.1.10>;tag=as6255d020
To: "sip09" <sip:sip09 at 192.168.1.10>;tag=3432177901
Contact: <sip:02 at 192.168.1.10:5060;transport=tcp>
Call-ID: fcaad738faee2d0250d0cf2366139979 at 192.168.2.2
CSeq: 101 UPDATE
User-Agent: Asterisk PBX 13.21.1
Remote-Party-ID: "Internsamtal"
<sip:02 at 192.168.1.10>;party=called;privacy=off;screen=yes
X-Asterisk-rpid-update: Yes
Content-Length: 0
---
-- Executing [s at intercom:9] ExecIf("SIP/sip09-00000004",
"0?Dial(SIP/sip01&SIP/sip02&SIP/sip03&SIP/sip04&SIP/sip05&SIP/sip06&SIP/sip0
7&SIP/sip08&SIP/sip10,60,mcI)") in new stack
-- Executing [s at intercom:10] ExecIf("SIP/sip09-00000004",
"0?Dial(SIP/sip01&SIP/sip02&SIP/sip03&SIP/sip04&SIP/sip05&SIP/sip06&SIP/sip0
7&SIP/sip08&SIP/sip10,60,mcI)") in new stack
-- Executing [s at intercom:11] ExecIf("SIP/sip09-00000004",
"1?Dial(SIP/sip02,60,mcI)") in new stack
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
-- Called SIP/sip02
-- Started music on hold, class 'default', on channel
'SIP/sip09-00000004'
<--- SIP read from TCP:192.168.2.2:51729 --->
OPTIONS sip:02 at 192.168.1.10 SIP/2.0
Call-ID: 31833826f012f172357c88a7a0fba06b at 192.168.2.2
CSeq: 3089 OPTIONS
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=4073710845
To: <sip:02 at 192.168.1.10>
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK481f30810acaa6dc88c891b0a4d5187f3130;rport
Max-Forwards: 70
Contact: "sip09" <sip:sip09 at 192.168.2.2:56334;transport=tcp>
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.2.2:51729 (no NAT)
Looking for 02 in cellip (domain 192.168.1.10)
<--- Transmitting (no NAT) to 192.168.2.2:51729 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK481f30810acaa6dc88c891b0a4d5187f3130;receive
d=192.168.2.2;rport=51729
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=4073710845
To: <sip:02 at 192.168.1.10>;tag=as5931d38a
Call-ID: 31833826f012f172357c88a7a0fba06b at 192.168.2.2
CSeq: 3089 OPTIONS
Server: Asterisk PBX 13.21.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces,timer
Contact: <sip:192.168.1.10:5060;transport=tcp>
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'31833826f012f172357c88a7a0fba06b at 192.168.2.2' in 32000 ms (Method:
OPTIONS)
-- SIP/sip02-00000005 is ringing
Really destroying SIP dialog '2f04334307b9c7dedab01938ce28ffcf at
192.168.2.2'
Method: OPTIONS
Really destroying SIP dialog '89f514d93dccf52cdd4b1f25d4dbda21 at
192.168.2.2'
Method: OPTIONS
> 0x7f4f98017f60 -- Strict RTP learning after remote address set to:
192.168.1.22:5266
-- SIP/sip02-00000005 answered SIP/sip09-00000004
-- Stopped music on hold on SIP/sip09-00000004
-- Channel SIP/sip02-00000005 joined 'simple_bridge' basic-bridge
<6bd8c07b-69ec-41a7-848a-0ccd163d8cf8>
-- Channel SIP/sip09-00000004 joined 'simple_bridge' basic-bridge
<6bd8c07b-69ec-41a7-848a-0ccd163d8cf8>
> 0x7f4f98017f60 -- Strict RTP switching to RTP target address
192.168.1.22:5266 as source
> 0x7f4fac02c9f0 -- Strict RTP learning complete - Locking on source
address 192.168.2.2:26726
<--- SIP read from TCP:192.168.2.2:51729 --->
OPTIONS sip:02 at 192.168.1.10 SIP/2.0
Call-ID: bdc61f1da890c32d00fe4b40ba7c4b56 at 192.168.2.2
CSeq: 3534 OPTIONS
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=1285764150
To: <sip:02 at 192.168.1.10>
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK8e4724c7eca97670bb0ff197934398d63130;rport
Max-Forwards: 70
Contact: "sip09" <sip:sip09 at 192.168.2.2:56334;transport=tcp>
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.2.2:51729 (no NAT)
Looking for 02 in cellip (domain 192.168.1.10)
<--- Transmitting (no NAT) to 192.168.2.2:51729 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK8e4724c7eca97670bb0ff197934398d63130;receive
d=192.168.2.2;rport=51729
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=1285764150
To: <sip:02 at 192.168.1.10>;tag=as7e6ba334
Call-ID: bdc61f1da890c32d00fe4b40ba7c4b56 at 192.168.2.2
CSeq: 3534 OPTIONS
Server: Asterisk PBX 13.21.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces,timer
Contact: <sip:192.168.1.10:5060;transport=tcp>
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'bdc61f1da890c32d00fe4b40ba7c4b56 at 192.168.2.2' in 32000 ms (Method:
OPTIONS)
<--- SIP read from TCP:192.168.2.2:51729 --->
OPTIONS sip:192.168.1.10 SIP/2.0
Call-ID: 7cc0203ea86166ac8288e3bc8eb017e9 at 192.168.2.2
CSeq: 5344 OPTIONS
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=1364106611
To: "sip09" <sip:sip09 at 192.168.1.10>
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK28f368161fff7bc74e034bb5cd20cac63130;rport
Max-Forwards: 70
User-Agent: SIPAUA/0.1.001
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.2.2:51729 (no NAT)
Looking for s in cellip (domain 192.168.1.10)
<--- Transmitting (no NAT) to 192.168.2.2:51729 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK28f368161fff7bc74e034bb5cd20cac63130;receive
d=192.168.2.2;rport=51729
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=1364106611
To: "sip09" <sip:sip09 at 192.168.1.10>;tag=as47fe0a4b
Call-ID: 7cc0203ea86166ac8288e3bc8eb017e9 at 192.168.2.2
CSeq: 5344 OPTIONS
Server: Asterisk PBX 13.21.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces,timer
Contact: <sip:192.168.1.10:5060;transport=tcp>
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'7cc0203ea86166ac8288e3bc8eb017e9 at 192.168.2.2' in 32000 ms (Method:
OPTIONS)
> 0x7f4f98017f60 -- Strict RTP learning complete - Locking on source
address 192.168.1.22:5266
Really destroying SIP dialog '3a0d80c61a957724e379ffca75290a02 at
192.168.2.2'
Method: OPTIONS
<--- SIP read from TCP:192.168.2.2:51729 --->
OPTIONS sip:02 at 192.168.1.10 SIP/2.0
Call-ID: f8428aea9ed98c84ac02f7c81fa8e828 at 192.168.2.2
CSeq: 7700 OPTIONS
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=226904208
To: <sip:02 at 192.168.1.10>
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bKc0070d0f230b884578e362436cdbc2853130;rport
Max-Forwards: 70
Contact: "sip09" <sip:sip09 at 192.168.2.2:56334;transport=tcp>
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.2.2:51729 (no NAT)
Looking for 02 in cellip (domain 192.168.1.10)
<--- Transmitting (no NAT) to 192.168.2.2:51729 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bKc0070d0f230b884578e362436cdbc2853130;receive
d=192.168.2.2;rport=51729
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=226904208
To: <sip:02 at 192.168.1.10>;tag=as7cb5eb33
Call-ID: f8428aea9ed98c84ac02f7c81fa8e828 at 192.168.2.2
CSeq: 7700 OPTIONS
Server: Asterisk PBX 13.21.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces,timer
Contact: <sip:192.168.1.10:5060;transport=tcp>
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'f8428aea9ed98c84ac02f7c81fa8e828 at 192.168.2.2' in 32000 ms (Method:
OPTIONS)
<--- SIP read from TCP:192.168.2.2:51729 --->
OPTIONS sip:192.168.1.10 SIP/2.0
Call-ID: 62f09d536c1345bd0a4fc2a371b8fe46 at 192.168.2.2
CSeq: 6954 OPTIONS
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=3993661396
To: "sip09" <sip:sip09 at 192.168.1.10>
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK9870e8b2490b6d7fc850c788b00ac4f73130;rport
Max-Forwards: 70
User-Agent: SIPAUA/0.1.001
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.2.2:51729 (no NAT)
Looking for s in cellip (domain 192.168.1.10)
<--- Transmitting (no NAT) to 192.168.2.2:51729 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK9870e8b2490b6d7fc850c788b00ac4f73130;receive
d=192.168.2.2;rport=51729
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=3993661396
To: "sip09" <sip:sip09 at 192.168.1.10>;tag=as4ff4a50f
Call-ID: 62f09d536c1345bd0a4fc2a371b8fe46 at 192.168.2.2
CSeq: 6954 OPTIONS
Server: Asterisk PBX 13.21.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces,timer
Contact: <sip:192.168.1.10:5060;transport=tcp>
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'62f09d536c1345bd0a4fc2a371b8fe46 at 192.168.2.2' in 32000 ms (Method:
OPTIONS)
Really destroying SIP dialog '4105301f775b66ff375fe8f4c3d77352 at
192.168.2.2'
Method: OPTIONS
<--- SIP read from TCP:192.168.2.2:51729 --->
OPTIONS sip:02 at 192.168.1.10 SIP/2.0
Call-ID: 9b4c51ae1a4872730e3cdc87501593d2 at 192.168.2.2
CSeq: 6762 OPTIONS
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=2468372686
To: <sip:02 at 192.168.1.10>
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bKb442e2855ec60c0eb870cc5dda5032ea3130;rport
Max-Forwards: 70
Contact: "sip09" <sip:sip09 at 192.168.2.2:56334;transport=tcp>
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.2.2:51729 (no NAT)
Looking for 02 in cellip (domain 192.168.1.10)
<--- Transmitting (no NAT) to 192.168.2.2:51729 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bKb442e2855ec60c0eb870cc5dda5032ea3130;receive
d=192.168.2.2;rport=51729
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=2468372686
To: <sip:02 at 192.168.1.10>;tag=as6b59e2fd
Call-ID: 9b4c51ae1a4872730e3cdc87501593d2 at 192.168.2.2
CSeq: 6762 OPTIONS
Server: Asterisk PBX 13.21.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces,timer
Contact: <sip:192.168.1.10:5060;transport=tcp>
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'9b4c51ae1a4872730e3cdc87501593d2 at 192.168.2.2' in 32000 ms (Method:
OPTIONS)
<--- SIP read from TCP:192.168.2.2:51729 --->
OPTIONS sip:192.168.1.10 SIP/2.0
Call-ID: 165a86636c144e6e65a9ab3bb9bd2bec at 192.168.2.2
CSeq: 5505 OPTIONS
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=3400830189
To: "sip09" <sip:sip09 at 192.168.1.10>
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bKdc71e57dc0844c04950db3da3d3936c83130;rport
Max-Forwards: 70
User-Agent: SIPAUA/0.1.001
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.2.2:51729 (no NAT)
Looking for s in cellip (domain 192.168.1.10)
<--- Transmitting (no NAT) to 192.168.2.2:51729 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bKdc71e57dc0844c04950db3da3d3936c83130;receive
d=192.168.2.2;rport=51729
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=3400830189
To: "sip09" <sip:sip09 at 192.168.1.10>;tag=as12192756
Call-ID: 165a86636c144e6e65a9ab3bb9bd2bec at 192.168.2.2
CSeq: 5505 OPTIONS
Server: Asterisk PBX 13.21.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces,timer
Contact: <sip:192.168.1.10:5060;transport=tcp>
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'165a86636c144e6e65a9ab3bb9bd2bec at 192.168.2.2' in 32000 ms (Method:
OPTIONS)
Really destroying SIP dialog 'e65234cb818a143bc3c167a782b98e96 at
192.168.2.2'
Method: OPTIONS
Really destroying SIP dialog 'fcaad738faee2d0250d0cf2366139979 at
192.168.2.2'
Method: ACK
-- Channel SIP/sip09-00000004 left 'simple_bridge' basic-bridge
<6bd8c07b-69ec-41a7-848a-0ccd163d8cf8>
-- Channel SIP/sip02-00000005 left 'simple_bridge' basic-bridge
<6bd8c07b-69ec-41a7-848a-0ccd163d8cf8>
== Spawn extension (intercom, s, 11) exited non-zero on
'SIP/sip09-00000004'
Really destroying SIP dialog 'fcaad738faee2d0250d0cf2366139979 at
192.168.2.2'
Method: ACK
Huh? Child handler, but nobody there?
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/sip09-00000004
Really destroying SIP dialog '31833826f012f172357c88a7a0fba06b at
192.168.2.2'
Method: OPTIONS
<--- SIP read from TCP:192.168.2.2:51729 --->
OPTIONS sip:02 at 192.168.1.10 SIP/2.0
Call-ID: d27f284f6c940648ac9405564677e149 at 192.168.2.2
CSeq: 5699 OPTIONS
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=1832850858
To: <sip:02 at 192.168.1.10>
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK3407e3da5861681dc4042a90356fa7f23130;rport
Max-Forwards: 70
Contact: "sip09" <sip:sip09 at 192.168.2.2:56334;transport=tcp>
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.2.2:51729 (no NAT)
Looking for 02 in cellip (domain 192.168.1.10)
<--- Transmitting (no NAT) to 192.168.2.2:51729 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK3407e3da5861681dc4042a90356fa7f23130;receive
d=192.168.2.2;rport=51729
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=1832850858
To: <sip:02 at 192.168.1.10>;tag=as06642c34
Call-ID: d27f284f6c940648ac9405564677e149 at 192.168.2.2
CSeq: 5699 OPTIONS
Server: Asterisk PBX 13.21.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces,timer
Contact: <sip:192.168.1.10:5060;transport=tcp>
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'd27f284f6c940648ac9405564677e149 at 192.168.2.2' in 32000 ms (Method:
OPTIONS)
<--- SIP read from TCP:192.168.2.2:51729 --->
OPTIONS sip:192.168.1.10 SIP/2.0
Call-ID: 2ed69e1e14b5c0317ca97b43b70db9e2 at 192.168.2.2
CSeq: 9505 OPTIONS
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=2427345124
To: "sip09" <sip:sip09 at 192.168.1.10>
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bKb39951cedef63174ee8730fee4e16edc3130;rport
Max-Forwards: 70
User-Agent: SIPAUA/0.1.001
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.2.2:51729 (no NAT)
Looking for s in cellip (domain 192.168.1.10)
<--- Transmitting (no NAT) to 192.168.2.2:51729 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bKb39951cedef63174ee8730fee4e16edc3130;receive
d=192.168.2.2;rport=51729
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=2427345124
To: "sip09" <sip:sip09 at 192.168.1.10>;tag=as70377f26
Call-ID: 2ed69e1e14b5c0317ca97b43b70db9e2 at 192.168.2.2
CSeq: 9505 OPTIONS
Server: Asterisk PBX 13.21.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces,timer
Contact: <sip:192.168.1.10:5060;transport=tcp>
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'2ed69e1e14b5c0317ca97b43b70db9e2 at 192.168.2.2' in 32000 ms (Method:
OPTIONS)
<--- SIP read from TCP:192.168.2.2:51729 --->
BYE sip:02 at 192.168.1.10:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK539d18ee436e89701260bba837f7a7043130
CSeq: 9119 BYE
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=3432177901
To: <sip:02 at 192.168.1.10>;tag=as6255d020
Call-ID: fcaad738faee2d0250d0cf2366139979 at 192.168.2.2
Allow:
INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
Max-Forwards: 70
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.2.2:56334 (no NAT)
<--- Transmitting (no NAT) to 192.168.2.2:56334 --->
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK539d18ee436e89701260bba837f7a7043130;receive
d=192.168.2.2
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=3432177901
To: <sip:02 at 192.168.1.10>;tag=as6255d020
Call-ID: fcaad738faee2d0250d0cf2366139979 at 192.168.2.2
CSeq: 9119 BYE
Server: Asterisk PBX 13.21.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces,timer
Content-Length: 0
<------------>
Really destroying SIP dialog 'bdc61f1da890c32d00fe4b40ba7c4b56 at
192.168.2.2'
Method: OPTIONS
Really destroying SIP dialog '7cc0203ea86166ac8288e3bc8eb017e9 at
192.168.2.2'
Method: OPTIONS
<--- SIP read from TCP:192.168.2.2:51729 --->
OPTIONS sip:192.168.1.10 SIP/2.0
Call-ID: 62e2882fccac509bb685f2deeee30d09 at 192.168.2.2
CSeq: 8317 OPTIONS
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=1993294406
To: "sip09" <sip:sip09 at 192.168.1.10>
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK8b76efb259330fc01da907cc23380df43130;rport
Max-Forwards: 70
User-Agent: SIPAUA/0.1.001
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.2.2:51729 (no NAT)
Looking for s in cellip (domain 192.168.1.10)
<--- Transmitting (no NAT) to 192.168.2.2:51729 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK8b76efb259330fc01da907cc23380df43130;receive
d=192.168.2.2;rport=51729
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=1993294406
To: "sip09" <sip:sip09 at 192.168.1.10>;tag=as08e85e53
Call-ID: 62e2882fccac509bb685f2deeee30d09 at 192.168.2.2
CSeq: 8317 OPTIONS
Server: Asterisk PBX 13.21.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces,timer
Contact: <sip:192.168.1.10:5060;transport=tcp>
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'62e2882fccac509bb685f2deeee30d09 at 192.168.2.2' in 32000 ms (Method:
OPTIONS)
Really destroying SIP dialog 'f8428aea9ed98c84ac02f7c81fa8e828 at
192.168.2.2'
Method: OPTIONS
Really destroying SIP dialog '62f09d536c1345bd0a4fc2a371b8fe46 at
192.168.2.2'
Method: OPTIONS
sip set debug
<--- SIP read from TCP:192.168.2.2:51729 --->
OPTIONS sip:192.168.1.10 SIP/2.0
Call-ID: 765fc9aff7309f1f67809701da1257d4 at 192.168.2.2
CSeq: 3549 OPTIONS
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=367223790
To: "sip09" <sip:sip09 at 192.168.1.10>
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK2ec533f3f5eafaea23c77230b12d4c3c3130;rport
Max-Forwards: 70
User-Agent: SIPAUA/0.1.001
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.2.2:51729 (no NAT)
Looking for s in cellip (domain 192.168.1.10)
<--- Transmitting (no NAT) to 192.168.2.2:51729 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK2ec533f3f5eafaea23c77230b12d4c3c3130;receive
d=192.168.2.2;rport=51729
From: "sip09" <sip:sip09 at 192.168.1.10>;tag=367223790
To: "sip09" <sip:sip09 at 192.168.1.10>;tag=as7a8acd4c
Call-ID: 765fc9aff7309f1f67809701da1257d4 at 192.168.2.2
CSeq: 3549 OPTIONS
Server: Asterisk PBX 13.21.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces,timer
Contact: <sip:192.168.1.10:5060;transport=tcp>
Accept: application/sdp
Content-Length: 0
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