Retransmitting #1 (NAT) to 1.23.214.105:51203

Asterisk Asterisk General
1

Hi

I have a problem on making calls from internet. We have two sites.

HQ has the Asterisk server in 192.168.x.x network.

Users on HQ don’t have any problem in registration, making and receiving calls.

However, on a remote site people are connecting via internet. We have 1 static IP address and we used NAT to route 5060 port.

Our setup is

Asterisk server <-> Beetel SOHO router <-> NAT <-> ISP with static IP <-> Remote site users via internet

We are able to register successfully. However, when we receive a call or make a call, the call gets hangup within 30 seconds. Sometimes it hangs on 10 seconds. (All calls for softphones via internet has this problem).

I can see similar retransmissions in the sip debug log.

[0KRetransmitting #1 (NAT) to 1.23.214.105:51203:

I have setup sip.conf with the following options

externip=
localnet=192.168.x.x/255.255.0.0

peers conf with
nat=yes
canreinvite=no
qualify=yes

rtp.conf

rtpstart=10000
rtpend=20000
Still no luck.

I believe the router is able to nat the 5060 SIP port but unable to NAT RTP ports. Correct me if I am wrong.

Below is the complete SIP debug log

[Kvoicecall*CLI> sip set debug on

voicecall*CLI>
[0KSIP Debugging enabled

[Kvoicecall*CLI>
[0K
<— SIP read from UDP:1.23.214.105:51203 —>
PUBLISH sip:8809@120.120.70.13:5080;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-a1ed0bc02cd1e130-1—d8754z-
Max-Forwards: 70
Contact: sip:8809@1.23.214.105:51203;transport=UDP
To: sip:8809@120.120.70.13:5080;transport=UDP
From: sip:8809@120.120.70.13:5080;transport=UDP;tag=5643a120
Call-ID: ZmMwY2Y2YThhMWVjZTIwMWNjODdhMDA5ZTNhMWQxYmM.
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Event: presence
Allow-Events: presence, kpml
Content-Length: 272

<?xml version="1.0" encoding="UTF-8"?>

open On the phone

<------------->
— (16 headers 3 lines) —

<— Transmitting (NAT) to 1.23.214.105:51203 —>
SIP/2.0 489 Bad Event

Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-a1ed0bc02cd1e130-1—d8754z-;received=1.23.214.105;rport=51203

From: sip:8809@120.120.70.13:5080;transport=UDP;tag=5643a120

To: sip:8809@120.120.70.13:5080;transport=UDP;tag=as0e513c2a

Call-ID: ZmMwY2Y2YThhMWVjZTIwMWNjODdhMDA5ZTNhMWQxYmM.

CSeq: 1 PUBLISH

Server: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Length: 0

<------------>
Really destroying SIP dialog ‘ZmMwY2Y2YThhMWVjZTIwMWNjODdhMDA5ZTNhMWQxYmM.’ Method: PUBLISH

[Kvoicecall*CLI>
[0K
<— SIP read from UDP:1.23.214.105:51203 —>
SUBSCRIBE sip:8809@120.120.70.13:5080;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-0ec3b93517239de5-1—d8754z-
Max-Forwards: 70
Contact: sip:8809@1.23.214.105:51203;transport=UDP
To: sip:8809@120.120.70.13:5080;transport=UDP
From: sip:8809@120.120.70.13:5080;transport=UDP;tag=8eb1ef0d
Call-ID: ZTU0ZmFhOWI3YTU0Njc0ZDIwNjAxMjY3ODk4NzRkNTQ.
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (16 headers 0 lines) —
Creating new subscription
Sending to 1.23.214.105:51203 (NAT)
list_route: hop: sip:8809@1.23.214.105:51203;transport=UDP
Found peer ‘8809’ for ‘8809’ from 1.23.214.105:51203

<— Transmitting (NAT) to 1.23.214.105:51203 —>
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-0ec3b93517239de5-1—d8754z-;received=1.23.214.105;rport=51203

From: sip:8809@120.120.70.13:5080;transport=UDP;tag=8eb1ef0d

To: sip:8809@120.120.70.13:5080;transport=UDP;tag=as3b805583

Call-ID: ZTU0ZmFhOWI3YTU0Njc0ZDIwNjAxMjY3ODk4NzRkNTQ.

CSeq: 1 SUBSCRIBE

Server: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“2e048830”

Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘ZTU0ZmFhOWI3YTU0Njc0ZDIwNjAxMjY3ODk4NzRkNTQ.’ in 12800 ms (Method: SUBSCRIBE)

[Kvoicecall*CLI>
[0K
<— SIP read from UDP:1.23.214.105:51203 —>
SUBSCRIBE sip:8809@120.120.70.13:5080;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-4bb1372c35f53944-1—d8754z-
Max-Forwards: 70
Contact: sip:8809@1.23.214.105:51203;transport=UDP
To: sip:8809@120.120.70.13:5080;transport=UDP
From: sip:8809@120.120.70.13:5080;transport=UDP;tag=8eb1ef0d
Call-ID: ZTU0ZmFhOWI3YTU0Njc0ZDIwNjAxMjY3ODk4NzRkNTQ.
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Authorization: Digest username=“8809”,realm=“asterisk”,nonce=“2e048830”,uri=“sip:8809@120.120.70.13:5080;transport=UDP”,response=“c983d8969d4dc22b6a09092c334f8f0f”,algorithm=MD5
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (17 headers 0 lines) —
Creating new subscription
Sending to 1.23.214.105:51203 (NAT)
Found peer ‘8809’ for ‘8809’ from 1.23.214.105:51203
Looking for 8809 in default (domain 120.120.70.13)

<— Transmitting (NAT) to 1.23.214.105:51203 —>
SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-4bb1372c35f53944-1—d8754z-;received=1.23.214.105;rport=51203

From: sip:8809@120.120.70.13:5080;transport=UDP;tag=8eb1ef0d

To: sip:8809@120.120.70.13:5080;transport=UDP;tag=as3b805583

Call-ID: ZTU0ZmFhOWI3YTU0Njc0ZDIwNjAxMjY3ODk4NzRkNTQ.

CSeq: 2 SUBSCRIBE

Server: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Length: 0

<------------>
Really destroying SIP dialog ‘ZTU0ZmFhOWI3YTU0Njc0ZDIwNjAxMjY3ODk4NzRkNTQ.’ Method: SUBSCRIBE

[Kvoicecall*CLI>
[0K
<— SIP read from UDP:1.23.214.105:51203 —>
INVITE sip:9942643483@120.120.70.13:5080;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-b5d2d075412ead47-1—d8754z-
Max-Forwards: 70
Contact: sip:8809@1.23.214.105:51203;transport=UDP
To: sip:9942643483@120.120.70.13:5080;transport=UDP
From: sip:8809@120.120.70.13:5080;transport=UDP;tag=5564e82c
Call-ID: OGEwM2MxOWYzNzQ2N2IxYzMzMTUyM2UzOWI3MzgzNDg.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Allow-Events: presence, kpml
Content-Length: 239

v=0
o=Z 0 0 IN IP4 1.23.214.105
s=Z
c=IN IP4 1.23.214.105
t=0 0
m=audio 8000 RTP/AVP 3 110 8 0 98 101
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (14 headers 12 lines) —
Sending to 1.23.214.105:51203 (NAT)
Using INVITE request as basis request - OGEwM2MxOWYzNzQ2N2IxYzMzMTUyM2UzOWI3MzgzNDg.
Found peer ‘8809’ for ‘8809’ from 1.23.214.105:51203

<— Reliably Transmitting (NAT) to 1.23.214.105:51203 —>
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-b5d2d075412ead47-1—d8754z-;received=1.23.214.105;rport=51203

From: sip:8809@120.120.70.13:5080;transport=UDP;tag=5564e82c

To: sip:9942643483@120.120.70.13:5080;transport=UDP;tag=as62020598

Call-ID: OGEwM2MxOWYzNzQ2N2IxYzMzMTUyM2UzOWI3MzgzNDg.

CSeq: 1 INVITE

Server: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“5012b0e2”

Content-Length: 0

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

[Kvoicecall*CLI>
[0KScheduling destruction of SIP dialog ‘OGEwM2MxOWYzNzQ2N2IxYzMzMTUyM2UzOWI3MzgzNDg.’ in 12800 ms (Method: INVITE)

[Kvoicecall*CLI>
[0KRetransmitting #1 (NAT) to 1.23.214.105:51203:
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-b5d2d075412ead47-1—d8754z-;received=1.23.214.105;rport=51203

From: sip:8809@120.120.70.13:5080;transport=UDP;tag=5564e82c

To: sip:9942643483@120.120.70.13:5080;transport=UDP;tag=as62020598

Call-ID: OGEwM2MxOWYzNzQ2N2IxYzMzMTUyM2UzOWI3MzgzNDg.

CSeq: 1 INVITE

Server: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“5012b0e2”

Content-Length: 0

[Kvoicecall*CLI>
[0K
<— SIP read from UDP:1.23.214.105:51203 —>
ACK sip:9942643483@120.120.70.13:5080;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-b5d2d075412ead47-1—d8754z-
Max-Forwards: 70
To: sip:9942643483@120.120.70.13:5080;transport=UDP;tag=as62020598
From: sip:8809@120.120.70.13:5080;transport=UDP;tag=5564e82c
Call-ID: OGEwM2MxOWYzNzQ2N2IxYzMzMTUyM2UzOWI3MzgzNDg.
CSeq: 1 ACK
Content-Length: 0

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

[Kvoicecall*CLI>
[0K
<— SIP read from UDP:1.23.214.105:51203 —>
INVITE sip:9942643483@120.120.70.13:5080;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-ede812f07fa6726a-1—d8754z-
Max-Forwards: 70
Contact: sip:8809@1.23.214.105:51203;transport=UDP
To: sip:9942643483@120.120.70.13:5080;transport=UDP
From: sip:8809@120.120.70.13:5080;transport=UDP;tag=5564e82c
Call-ID: OGEwM2MxOWYzNzQ2N2IxYzMzMTUyM2UzOWI3MzgzNDg.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Authorization: Digest username=“8809”,realm=“asterisk”,nonce=“5012b0e2”,uri=“sip:9942643483@120.120.70.13:5080;transport=UDP”,response=“cae1678b7e9bc565beda7b33d5c767bb”,algorithm=MD5
Allow-Events: presence, kpml
Content-Length: 239

v=0
o=Z 0 0 IN IP4 1.23.214.105
s=Z
c=IN IP4 1.23.214.105
t=0 0
m=audio 8000 RTP/AVP 3 110 8 0 98 101
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (15 headers 12 lines) —
Sending to 1.23.214.105:51203 (NAT)
Using INVITE request as basis request - OGEwM2MxOWYzNzQ2N2IxYzMzMTUyM2UzOWI3MzgzNDg.
Found peer ‘8809’ for ‘8809’ from 1.23.214.105:51203
== Using SIP RTP CoS mark 5
Found RTP audio format 3
Found RTP audio format 110
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 98
Found RTP audio format 101
Found audio description format speex for ID 110
Found audio description format iLBC for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x6 (gsm|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 1.23.214.105:8000
Looking for 9942643483 in default (domain 120.120.70.13)
list_route: hop: sip:8809@1.23.214.105:51203;transport=UDP

<— Transmitting (NAT) to 1.23.214.105:51203 —>
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-ede812f07fa6726a-1—d8754z-;received=1.23.214.105;rport=51203

From: sip:8809@120.120.70.13:5080;transport=UDP;tag=5564e82c

To: sip:9942643483@120.120.70.13:5080;transport=UDP

Call-ID: OGEwM2MxOWYzNzQ2N2IxYzMzMTUyM2UzOWI3MzgzNDg.

CSeq: 2 INVITE

Server: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: sip:9942643483@120.120.70.13:5060

Content-Length: 0

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

[Kvoicecall*CLI>
[0K – Executing [9942643483@default:1] [1;36mAGI [0m(" [1;35mSIP/8809-00000008 [0m", " [1;35magi-NVA_recording.agi,BOTH------Y—N---Y—N [0m") in new stack

[Kvoicecall*CLI>
[0K – Launched AGI Script /var/lib/asterisk/agi-bin/agi-NVA_recording.agi

[Kvoicecall*CLI>
[0K – AGI Script Executing Application: (Monitor) Options: (wav,/var/spool/asterisk/monitor/MIX/20140826222517_8809_9942643483)

[Kvoicecall*CLI>
[0K – <SIP/8809-00000008>AGI Script agi-NVA_recording.agi completed, returning 0

[Kvoicecall*CLI>
[0K – Executing [9942643483@default:2] [1;36mAGI [0m(" [1;35mSIP/8809-00000008 [0m", " [1;35magi://127.0.0.1:4577/call_log [0m") in new stack

[Kvoicecall*CLI>
[0K – <SIP/8809-00000008>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0

[Kvoicecall*CLI>
[0K – Executing [9942643483@default:3] [1;36mDial [0m(" [1;35mSIP/8809-00000008 [0m", " [1;35mDAHDI/i1/9942643483 [0m") in new stack

[Kvoicecall*CLI>
[0K – Requested transfer capability: 0x00 - SPEECH

[Kvoicecall*CLI>
[0K – Called DAHDI/i1/9942643483

[Kvoicecall*CLI>
[0K
<— SIP read from UDP:1.23.214.105:51203 —>
ACK sip:9942643483@120.120.70.13:5080;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-b5d2d075412ead47-1—d8754z-
Max-Forwards: 70
To: sip:9942643483@120.120.70.13:5080;transport=UDP;tag=as62020598
From: sip:8809@120.120.70.13:5080;transport=UDP;tag=5564e82c
Call-ID: OGEwM2MxOWYzNzQ2N2IxYzMzMTUyM2UzOWI3MzgzNDg.
CSeq: 1 ACK
Content-Length: 0

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

[Kvoicecall*CLI>
[0K – DAHDI/i1/9942643483-9 is proceeding passing it to SIP/8809-00000008

<— Transmitting (NAT) to 1.23.214.105:51203 —>
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-ede812f07fa6726a-1—d8754z-;received=1.23.214.105;rport=51203

From: sip:8809@120.120.70.13:5080;transport=UDP;tag=5564e82c

To: sip:9942643483@120.120.70.13:5080;transport=UDP

Call-ID: OGEwM2MxOWYzNzQ2N2IxYzMzMTUyM2UzOWI3MzgzNDg.

CSeq: 2 INVITE

Server: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: sip:9942643483@120.120.70.13:5060

Content-Length: 0

<------------>
– DAHDI/i1/9942643483-9 is making progress passing it to SIP/8809-00000008
Audio is at 13442
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (NAT) to 1.23.214.105:51203 —>
SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-ede812f07fa6726a-1—d8754z-;received=1.23.214.105;rport=51203

From: sip:8809@120.120.70.13:5080;transport=UDP;tag=5564e82c

To: sip:9942643483@120.120.70.13:5080;transport=UDP;tag=as33ef04ce

Call-ID: OGEwM2MxOWYzNzQ2N2IxYzMzMTUyM2UzOWI3MzgzNDg.

CSeq: 2 INVITE

Server: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: sip:9942643483@120.120.70.13:5060

Content-Type: application/sdp

Require: timer

Content-Length: 304

v=0

o=root 1027354955 1027354955 IN IP4 120.120.70.13

s=Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

c=IN IP4 120.120.70.13

t=0 0

m=audio 13442 RTP/AVP 0 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

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

[Kvoicecall*CLI>
[0K – DAHDI/i1/9942643483-9 is ringing
[Aug 26 22:25:20] [1;31mWARNING [0m[29947]: [1;37mtranslate.c [0m: [1;37m206 [0m [1;37mframein [0m: no samples for ulawtolin

[Kvoicecall*CLI>
[0K – DAHDI/i1/9942643483-9 answered SIP/8809-00000008
Audio is at 13442
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 1.23.214.105:51203 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-ede812f07fa6726a-1—d8754z-;received=1.23.214.105;rport=51203

From: sip:8809@120.120.70.13:5080;transport=UDP;tag=5564e82c

To: sip:9942643483@120.120.70.13:5080;transport=UDP;tag=as33ef04ce

Call-ID: OGEwM2MxOWYzNzQ2N2IxYzMzMTUyM2UzOWI3MzgzNDg.

CSeq: 2 INVITE

Server: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: sip:9942643483@120.120.70.13:5060

Content-Type: application/sdp

Require: timer

Content-Length: 304

v=0

o=root 1027354955 1027354956 IN IP4 120.120.70.13

s=Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

c=IN IP4 120.120.70.13

t=0 0

m=audio 13442 RTP/AVP 0 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

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

[Kvoicecall*CLI>
[0KReliably Transmitting (NAT) to 192.168.1.111:46837:
OPTIONS sip:8812@120.120.70.13:29396;rinstance=d9f7e1791518d75f;transport=UDP SIP/2.0

Via: SIP/2.0/UDP 192.168.1.113:5060;branch=z9hG4bK361837a8;rport

Max-Forwards: 70

From: “asterisk” sip:asterisk@192.168.1.113;tag=as21a22147

To: sip:8812@120.120.70.13:29396;rinstance=d9f7e1791518d75f;transport=UDP

Contact: sip:asterisk@192.168.1.113:5060

Call-ID: 164762516338f7e95022ff96173c1add@192.168.1.113:5060

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

Date: Tue, 26 Aug 2014 16:55:27 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Length: 0

[Kvoicecall*CLI>
[0K
<— SIP read from UDP:192.168.1.111:46837 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.113:5060;branch=z9hG4bK361837a8;rport=5060
Contact: sip:192.168.1.111:46837
To: sip:8812@120.120.70.13:29396;rinstance=d9f7e1791518d75f;transport=UDP;tag=e9013433
From: "asterisk"sip:asterisk@192.168.1.113;tag=as21a22147
Call-ID: 164762516338f7e95022ff96173c1add@192.168.1.113:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘164762516338f7e95022ff96173c1add@192.168.1.113:5060’ Method: OPTIONS

[Kvoicecall*CLI>
[0KRetransmitting #1 (NAT) to 1.23.214.105:51203:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-ede812f07fa6726a-1—d8754z-;received=1.23.214.105;rport=51203

From: sip:8809@120.120.70.13:5080;transport=UDP;tag=5564e82c

To: sip:9942643483@120.120.70.13:5080;transport=UDP;tag=as33ef04ce

Call-ID: OGEwM2MxOWYzNzQ2N2IxYzMzMTUyM2UzOWI3MzgzNDg.

CSeq: 2 INVITE

Server: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: sip:9942643483@120.120.70.13:5060

Content-Type: application/sdp

Require: timer

Content-Length: 304

v=0

o=root 1027354955 1027354956 IN IP4 120.120.70.13

s=Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

c=IN IP4 120.120.70.13

t=0 0

m=audio 13442 RTP/AVP 0 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

[Kvoicecall*CLI>
[0K
<— SIP read from UDP:1.23.214.105:51203 —>
PUBLISH sip:8809@120.120.70.13:5080;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-10109770098ab4c5-1—d8754z-
Max-Forwards: 70
Contact: sip:8809@1.23.214.105:51203;transport=UDP
To: sip:8809@120.120.70.13:5080;transport=UDP
From: sip:8809@120.120.70.13:5080;transport=UDP;tag=03ae787e
Call-ID: MjQxN2JlNTM0MTQzNWZjYzEyOTU4YjJlZjNkMWFlMzA.
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Event: presence
Allow-Events: presence, kpml
Content-Length: 272

<?xml version="1.0" encoding="UTF-8"?>

open On the phone

<------------->
— (16 headers 3 lines) —

<— Transmitting (NAT) to 1.23.214.105:51203 —>
SIP/2.0 489 Bad Event

Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-10109770098ab4c5-1—d8754z-;received=1.23.214.105;rport=51203

From: sip:8809@120.120.70.13:5080;transport=UDP;tag=03ae787e

To: sip:8809@120.120.70.13:5080;transport=UDP;tag=as1662abd7

Call-ID: MjQxN2JlNTM0MTQzNWZjYzEyOTU4YjJlZjNkMWFlMzA.

CSeq: 1 PUBLISH

Server: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Length: 0

<------------>
Really destroying SIP dialog ‘MjQxN2JlNTM0MTQzNWZjYzEyOTU4YjJlZjNkMWFlMzA.’ Method: PUBLISH

[Kvoicecall*CLI>
[0K
<— SIP read from UDP:1.23.214.105:51203 —>
SUBSCRIBE sip:8809@120.120.70.13:5080;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-b36908661a84dcdf-1—d8754z-
Max-Forwards: 70
Contact: sip:8809@1.23.214.105:51203;transport=UDP
To: sip:8809@120.120.70.13:5080;transport=UDP
From: sip:8809@120.120.70.13:5080;transport=UDP;tag=ff70426a
Call-ID: YWUzNWEyMjgxODM5NjU3OThlNWYxYzk1ZmEwYTNhYzg.
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (16 headers 0 lines) —
Creating new subscription
Sending to 1.23.214.105:51203 (NAT)
list_route: hop: sip:8809@1.23.214.105:51203;transport=UDP
Found peer ‘8809’ for ‘8809’ from 1.23.214.105:51203

<— Transmitting (NAT) to 1.23.214.105:51203 —>
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-b36908661a84dcdf-1—d8754z-;received=1.23.214.105;rport=51203

From: sip:8809@120.120.70.13:5080;transport=UDP;tag=ff70426a

To: sip:8809@120.120.70.13:5080;transport=UDP;tag=as339e0ad6

Call-ID: YWUzNWEyMjgxODM5NjU3OThlNWYxYzk1ZmEwYTNhYzg.

CSeq: 1 SUBSCRIBE

Server: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“3d0dad1a”

Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘YWUzNWEyMjgxODM5NjU3OThlNWYxYzk1ZmEwYTNhYzg.’ in 12800 ms (Method: SUBSCRIBE)

[Kvoicecall*CLI>
[0KReliably Transmitting (NAT) to 1.23.214.105:51203:
OPTIONS sip:8809@1.23.208.123:51203;rinstance=0750bf148e3b32dc;transport=UDP SIP/2.0

Via: SIP/2.0/UDP 120.120.70.13:5060;branch=z9hG4bK1b834f72;rport

Max-Forwards: 70

From: “asterisk” sip:asterisk@120.120.70.13;tag=as2e546941

To: sip:8809@1.23.208.123:51203;rinstance=0750bf148e3b32dc;transport=UDP

Contact: sip:asterisk@120.120.70.13:5060

Call-ID: 612d3d3e7e0c83f2662d509b3a51204b@120.120.70.13:5060

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

Date: Tue, 26 Aug 2014 16:55:27 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Length: 0

[Kvoicecall*CLI>
[0K
<— SIP read from UDP:1.23.214.105:51203 —>
SUBSCRIBE sip:8809@120.120.70.13:5080;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-97c198143090e8db-1—d8754z-
Max-Forwards: 70
Contact: sip:8809@1.23.214.105:51203;transport=UDP
To: sip:8809@120.120.70.13:5080;transport=UDP
From: sip:8809@120.120.70.13:5080;transport=UDP;tag=ff70426a
Call-ID: YWUzNWEyMjgxODM5NjU3OThlNWYxYzk1ZmEwYTNhYzg.
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Authorization: Digest username=“8809”,realm=“asterisk”,nonce=“3d0dad1a”,uri=“sip:8809@120.120.70.13:5080;transport=UDP”,response=“ecbdb33120fe3f6b727aa8f572d85825”,algorithm=MD5
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (17 headers 0 lines) —
Creating new subscription
Sending to 1.23.214.105:51203 (NAT)
Found peer ‘8809’ for ‘8809’ from 1.23.214.105:51203
Looking for 8809 in default (domain 120.120.70.13)

<— Transmitting (NAT) to 1.23.214.105:51203 —>
SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-97c198143090e8db-1—d8754z-;received=1.23.214.105;rport=51203

From: sip:8809@120.120.70.13:5080;transport=UDP;tag=ff70426a

To: sip:8809@120.120.70.13:5080;transport=UDP;tag=as339e0ad6

Call-ID: YWUzNWEyMjgxODM5NjU3OThlNWYxYzk1ZmEwYTNhYzg.

CSeq: 2 SUBSCRIBE

Server: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Length: 0

<------------>
Really destroying SIP dialog ‘YWUzNWEyMjgxODM5NjU3OThlNWYxYzk1ZmEwYTNhYzg.’ Method: SUBSCRIBE

[Kvoicecall*CLI>
[0K
<— SIP read from UDP:1.23.214.105:51203 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 120.120.70.13:5060;branch=z9hG4bK1b834f72;rport=5080
Contact: sip:10.26.167.60:51203
To: sip:8809@1.23.208.123:51203;rinstance=0750bf148e3b32dc;transport=UDP;tag=94c3d358
From: "asterisk"sip:asterisk@120.120.70.13;tag=as2e546941
Call-ID: 612d3d3e7e0c83f2662d509b3a51204b@120.120.70.13:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘612d3d3e7e0c83f2662d509b3a51204b@120.120.70.13:5060’ Method: OPTIONS

[Kvoicecall*CLI>
[0KRetransmitting #2 (NAT) to 1.23.214.105:51203:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-ede812f07fa6726a-1—d8754z-;received=1.23.214.105;rport=51203

From: sip:8809@120.120.70.13:5080;transport=UDP;tag=5564e82c

To: sip:9942643483@120.120.70.13:5080;transport=UDP;tag=as33ef04ce

Call-ID: OGEwM2MxOWYzNzQ2N2IxYzMzMTUyM2UzOWI3MzgzNDg.

CSeq: 2 INVITE

Server: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: sip:9942643483@120.120.70.13:5060

Content-Type: application/sdp

Require: timer

Content-Length: 304

v=0

o=root 1027354955 1027354956 IN IP4 120.120.70.13

s=Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

c=IN IP4 120.120.70.13

t=0 0

m=audio 13442 RTP/AVP 0 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

[Kvoicecall*CLI>
[0KRetransmitting #3 (NAT) to 1.23.214.105:51203:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-ede812f07fa6726a-1—d8754z-;received=1.23.214.105;rport=51203

From: sip:8809@120.120.70.13:5080;transport=UDP;tag=5564e82c

To: sip:9942643483@120.120.70.13:5080;transport=UDP;tag=as33ef04ce

Call-ID: OGEwM2MxOWYzNzQ2N2IxYzMzMTUyM2UzOWI3MzgzNDg.

CSeq: 2 INVITE

Server: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: sip:9942643483@120.120.70.13:5060

Content-Type: application/sdp

Require: timer

Content-Length: 304

v=0

o=root 1027354955 1027354956 IN IP4 120.120.70.13

s=Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

c=IN IP4 120.120.70.13

t=0 0

m=audio 13442 RTP/AVP 0 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

[Kvoicecall*CLI>
[0KRetransmitting #4 (NAT) to 1.23.214.105:51203:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-ede812f07fa6726a-1—d8754z-;received=1.23.214.105;rport=51203

From: sip:8809@120.120.70.13:5080;transport=UDP;tag=5564e82c

To: sip:9942643483@120.120.70.13:5080;transport=UDP;tag=as33ef04ce

Call-ID: OGEwM2MxOWYzNzQ2N2IxYzMzMTUyM2UzOWI3MzgzNDg.

CSeq: 2 INVITE

Server: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: sip:9942643483@120.120.70.13:5060

Content-Type: application/sdp

Require: timer

Content-Length: 304

v=0

o=root 1027354955 1027354956 IN IP4 120.120.70.13

s=Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

c=IN IP4 120.120.70.13

t=0 0

m=audio 13442 RTP/AVP 0 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

[Kvoicecall*CLI>
[0KRetransmitting #5 (NAT) to 1.23.214.105:51203:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-ede812f07fa6726a-1—d8754z-;received=1.23.214.105;rport=51203

From: sip:8809@120.120.70.13:5080;transport=UDP;tag=5564e82c

To: sip:9942643483@120.120.70.13:5080;transport=UDP;tag=as33ef04ce

Call-ID: OGEwM2MxOWYzNzQ2N2IxYzMzMTUyM2UzOWI3MzgzNDg.

CSeq: 2 INVITE

Server: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: sip:9942643483@120.120.70.13:5060

Content-Type: application/sdp

Require: timer

Content-Length: 304

v=0

o=root 1027354955 1027354956 IN IP4 120.120.70.13

s=Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

c=IN IP4 120.120.70.13

t=0 0

m=audio 13442 RTP/AVP 0 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

[Kvoicecall*CLI>
[0KRetransmitting #6 (NAT) to 1.23.214.105:51203:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-ede812f07fa6726a-1—d8754z-;received=1.23.214.105;rport=51203

From: sip:8809@120.120.70.13:5080;transport=UDP;tag=5564e82c

To: sip:9942643483@120.120.70.13:5080;transport=UDP;tag=as33ef04ce

Call-ID: OGEwM2MxOWYzNzQ2N2IxYzMzMTUyM2UzOWI3MzgzNDg.

CSeq: 2 INVITE

Server: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Session-Expires: 1800;refresher=uas

Contact: sip:9942643483@120.120.70.13:5060

Content-Type: application/sdp

Require: timer

Content-Length: 304

v=0

o=root 1027354955 1027354956 IN IP4 120.120.70.13

s=Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

c=IN IP4 120.120.70.13

t=0 0

m=audio 13442 RTP/AVP 0 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

[Kvoicecall*CLI>
[0KReliably Transmitting (NAT) to 120.120.70.2:15825:
OPTIONS sip:8806@120.120.70.2:15825;rinstance=d4c321314db12123;transport=UDP SIP/2.0

Via: SIP/2.0/UDP 120.120.70.13:5060;branch=z9hG4bK44010bd7;rport

Max-Forwards: 70

From: “asterisk” sip:asterisk@120.120.70.13;tag=as70308846

To: sip:8806@120.120.70.2:15825;rinstance=d4c321314db12123;transport=UDP

Contact: sip:asterisk@120.120.70.13:5060

Call-ID: 50ea592933b73c3b3e8d6a4a0d74aab2@120.120.70.13:5060

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

Date: Tue, 26 Aug 2014 16:55:39 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Length: 0

[Kvoicecall*CLI>
[0K
<— SIP read from UDP:120.120.70.2:15825 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 120.120.70.13:5060;branch=z9hG4bK44010bd7;rport=5080
Contact: sip:120.120.70.2:15825
To: sip:8806@120.120.70.2:15825;rinstance=d4c321314db12123;transport=UDP;tag=2a4b573e
From: "asterisk"sip:asterisk@120.120.70.13;tag=as70308846
Call-ID: 50ea592933b73c3b3e8d6a4a0d74aab2@120.120.70.13:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘50ea592933b73c3b3e8d6a4a0d74aab2@120.120.70.13:5060’ Method: OPTIONS

[Kvoicecall*CLI>
[0K[Aug 26 22:25:40] [1;31mWARNING [0m[18378]: [1;37mchan_sip.c [0m: [1;37m3983 [0m [1;37mretrans_pkt [0m: Retransmission timeout reached on transmission OGEwM2MxOWYzNzQ2N2IxYzMzMTUyM2UzOWI3MzgzNDg. for seqno 2 (Critical Response) – See wiki.asterisk.org/wiki/display/ … nsmissions
Packet timed out after 12801ms with no response
[Aug 26 22:25:40] [1;31mWARNING [0m[18378]: [1;37mchan_sip.c [0m: [1;37m4012 [0m [1;37mretrans_pkt [0m: Hanging up call OGEwM2MxOWYzNzQ2N2IxYzMzMTUyM2UzOWI3MzgzNDg. - no reply to our critical packet (see wiki.asterisk.org/wiki/display/ … nsmissions).

[Kvoicecall*CLI>
[0K – Executing [h@default:1] [1;36mAGI [0m(" [1;35mSIP/8809-00000008 [0m", " [1;35magi://127.0.0.1:4577/call_log–HVcauses–PRI-----NODEBUG-----18-----ANSWER-----23-----13 [0m") in new stack

[Kvoicecall*CLI>
[0K – <SIP/8809-00000008>AGI Script agi://127.0.0.1:4577/call_log–HVcauses … -23-----13 completed, returning 0
– Hungup ‘DAHDI/i1/9942643483-9’
== Spawn extension (default, 9942643483, 3) exited non-zero on 'SIP/8809-00000008’
Scheduling destruction of SIP dialog ‘OGEwM2MxOWYzNzQ2N2IxYzMzMTUyM2UzOWI3MzgzNDg.’ in 12800 ms (Method: INVITE)
set_destination: Parsing sip:8809@1.23.214.105:51203;transport=UDP for address/port to send to
set_destination: set destination to 1.23.214.105:51203
Reliably Transmitting (NAT) to 1.23.214.105:51203:
BYE sip:8809@1.23.214.105:51203;transport=UDP SIP/2.0

Via: SIP/2.0/UDP 120.120.70.13:5060;branch=z9hG4bK13ab0d4b;rport

Max-Forwards: 70

From: sip:9942643483@120.120.70.13:5080;transport=UDP;tag=as33ef04ce

To: sip:8809@120.120.70.13:5080;transport=UDP;tag=5564e82c

Call-ID: OGEwM2MxOWYzNzQ2N2IxYzMzMTUyM2UzOWI3MzgzNDg.

CSeq: 102 BYE

User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

Proxy-Authorization: Digest username=“8809”, realm=“asterisk”, algorithm=MD5, uri=“sip:120.120.70.13”, nonce="", response=“c054c8a511275c5b2621d96b6b4a7de9”

X-Asterisk-HangupCause: No user responding

X-Asterisk-HangupCauseCode: 18

Content-Length: 0

[Kvoicecall*CLI>
[0K
<— SIP read from UDP:1.23.214.105:51203 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 120.120.70.13:5060;branch=z9hG4bK13ab0d4b;rport=5080
Contact: sip:8809@1.23.214.105:51203;transport=UDP
To: sip:8809@120.120.70.13:5080;transport=UDP;tag=5564e82c
From: sip:9942643483@120.120.70.13:5080;transport=UDP;tag=as33ef04ce
Call-ID: OGEwM2MxOWYzNzQ2N2IxYzMzMTUyM2UzOWI3MzgzNDg.
CSeq: 102 BYE
User-Agent: Z 3.3.25608 r25552
Content-Length: 0

<------------->
— (9 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived

[Kvoicecall*CLI>
[0KReally destroying SIP dialog ‘OGEwM2MxOWYzNzQ2N2IxYzMzMTUyM2UzOWI3MzgzNDg.’ Method: INVITE

[Kvoicecall*CLI>
[0K
<— SIP read from UDP:1.23.214.105:51203 —>
PUBLISH sip:8809@120.120.70.13:5080;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-2cf68fedf738865a-1—d8754z-
Max-Forwards: 70
Contact: sip:8809@1.23.214.105:51203;transport=UDP
To: sip:8809@120.120.70.13:5080;transport=UDP
From: sip:8809@120.120.70.13:5080;transport=UDP;tag=71e11a26
Call-ID: N2ZiM2QwZDczZGM0OTU0NDc4Mjk0ZDk4ZmY2MjE1Njc.
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Event: presence
Allow-Events: presence, kpml
Content-Length: 266

<?xml version="1.0" encoding="UTF-8"?>

open Online

<------------->
— (16 headers 3 lines) —

<— Transmitting (NAT) to 1.23.214.105:51203 —>
SIP/2.0 489 Bad Event

Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-2cf68fedf738865a-1—d8754z-;received=1.23.214.105;rport=51203

From: sip:8809@120.120.70.13:5080;transport=UDP;tag=71e11a26

To: sip:8809@120.120.70.13:5080;transport=UDP;tag=as132f345b

Call-ID: N2ZiM2QwZDczZGM0OTU0NDc4Mjk0ZDk4ZmY2MjE1Njc.

CSeq: 1 PUBLISH

Server: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Length: 0

<------------>
Really destroying SIP dialog ‘N2ZiM2QwZDczZGM0OTU0NDc4Mjk0ZDk4ZmY2MjE1Njc.’ Method: PUBLISH

[Kvoicecall*CLI>
[0K
<— SIP read from UDP:1.23.214.105:51203 —>
SUBSCRIBE sip:8809@120.120.70.13:5080;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-4414a54c9adfae2c-1—d8754z-
Max-Forwards: 70
Contact: sip:8809@1.23.214.105:51203;transport=UDP
To: sip:8809@120.120.70.13:5080;transport=UDP
From: sip:8809@120.120.70.13:5080;transport=UDP;tag=afdca336
Call-ID: NGJmY2EyZmYyNjVhMzkzYTAzNGY1MGQzZjcwNDk0YmQ.
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (16 headers 0 lines) —
Creating new subscription
Sending to 1.23.214.105:51203 (NAT)
list_route: hop: sip:8809@1.23.214.105:51203;transport=UDP
Found peer ‘8809’ for ‘8809’ from 1.23.214.105:51203

<— Transmitting (NAT) to 1.23.214.105:51203 —>
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-4414a54c9adfae2c-1—d8754z-;received=1.23.214.105;rport=51203

From: sip:8809@120.120.70.13:5080;transport=UDP;tag=afdca336

To: sip:8809@120.120.70.13:5080;transport=UDP;tag=as167344df

Call-ID: NGJmY2EyZmYyNjVhMzkzYTAzNGY1MGQzZjcwNDk0YmQ.

CSeq: 1 SUBSCRIBE

Server: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“166803e7”

Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘NGJmY2EyZmYyNjVhMzkzYTAzNGY1MGQzZjcwNDk0YmQ.’ in 7232 ms (Method: SUBSCRIBE)

[Kvoicecall*CLI>
[0K
<— SIP read from UDP:1.23.214.105:51203 —>
SUBSCRIBE sip:8809@120.120.70.13:5080;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-2cba5aca8a75554e-1—d8754z-
Max-Forwards: 70
Contact: sip:8809@1.23.214.105:51203;transport=UDP
To: sip:8809@120.120.70.13:5080;transport=UDP
From: sip:8809@120.120.70.13:5080;transport=UDP;tag=afdca336
Call-ID: NGJmY2EyZmYyNjVhMzkzYTAzNGY1MGQzZjcwNDk0YmQ.
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Authorization: Digest username=“8809”,realm=“asterisk”,nonce=“166803e7”,uri=“sip:8809@120.120.70.13:5080;transport=UDP”,response=“c55f7ca95a65bdf9a7941d406c0661e0”,algorithm=MD5
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (17 headers 0 lines) —
Creating new subscription
Sending to 1.23.214.105:51203 (NAT)
Found peer ‘8809’ for ‘8809’ from 1.23.214.105:51203
Looking for 8809 in default (domain 120.120.70.13)

<— Transmitting (NAT) to 1.23.214.105:51203 —>
SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 1.23.214.105:51203;branch=z9hG4bK-d8754z-2cba5aca8a75554e-1—d8754z-;received=1.23.214.105;rport=51203

From: sip:8809@120.120.70.13:5080;transport=UDP;tag=afdca336

To: sip:8809@120.120.70.13:5080;transport=UDP;tag=as167344df

Call-ID: NGJmY2EyZmYyNjVhMzkzYTAzNGY1MGQzZjcwNDk0YmQ.

CSeq: 2 SUBSCRIBE

Server: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Length: 0

<------------>
Really destroying SIP dialog ‘NGJmY2EyZmYyNjVhMzkzYTAzNGY1MGQzZjcwNDk0YmQ.’ Method: SUBSCRIBE

[Kvoicecall*CLI>
[0KReliably Transmitting (NAT) to 120.120.70.2:22696:
OPTIONS sip:8802@120.120.70.2:28885;rinstance=1af426daea8f68c9;transport=UDP SIP/2.0

Via: SIP/2.0/UDP 120.120.70.13:5060;branch=z9hG4bK7e041f1d;rport

Max-Forwards: 70

From: “asterisk” sip:asterisk@120.120.70.13;tag=as5dda7258

To: sip:8802@120.120.70.2:28885;rinstance=1af426daea8f68c9;transport=UDP

Contact: sip:asterisk@120.120.70.13:5060

Call-ID: 23ca6f756df8cf374207abca5b15ad2a@120.120.70.13:5060

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com

Date: Tue, 26 Aug 2014 16:55:44 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Length: 0

[Kvoicecall*CLI>
[0K
<— SIP read from UDP:120.120.70.2:22696 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 120.120.70.13:5060;branch=z9hG4bK7e041f1d;rport=5080
Contact: sip:192.168.1.111:5060
To: sip:8802@120.120.70.2:28885;rinstance=1af426daea8f68c9;transport=UDP;tag=28491504
From: "asterisk"sip:asterisk@120.120.70.13;tag=as5dda7258
Call-ID: 23ca6f756df8cf374207abca5b15ad2a@120.120.70.13:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Allow-Events: presence, kpml
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘23ca6f756df8cf374207abca5b15ad2a@120.120.70.13:5060’ Method: OPTIONS

[Kvoicecall*CLI> !

]0;root@voicecall:/etc/asterisk [root@voicecall asterisk]#

Please help me to fix this problem.

2

Something between you and the Z 3.3.25608 r25552 is badly broken and is sending ACK to a non-final response. As there is no user agent, I would guess it is a broken router or other proxy.

For future reference, Asterisk General is a discussion forum; support questions should go in Asterisk Support.

3

Thank you David.

I also suspect this could be the inability of the router to translate from the server to softphone.

Could it be? How can we make this thing work?

And in future I will ensure I post questions on proper boards.

Thanks
Raja

4

Replace the broken component., or disable the function in the broken component that is causing the problem. (People generally report that enabling SIPALG in a router is a bad idea.