[asterisk-users] Problems with calls dropping on Android.

Mon Oct 14 01:59:15 CDT 2019

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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