DID Number Incoming Call not working

Hi guys,

I am new to asterisk. I can make outbound calls. But problem in incoming calls. When i make call to my did number nothing happen in asterisk. I forward the port 5060 to asterisk installed system and port 5060 is open.

My sip.conf

[general]

bindport = 5060
bindaddr = 0.0.0.0
externip = 59.99.239.180
tcpbindaddr = 0.0.0.0
udpbindadr=0.0.0.0
tcpenable = yes
srvlookup = yes
localnet=192.168.1.50/255.255.255.0

register => 88812XXXXX:XXXXXXX@sip.pennytel.com/88812XXXXX

[Lismore-sip-account]
defaultuser=88812XXXXX
type=peer
secret=XXXXX
qualify=yes
host=sip.pennytel.com
port=5060
disallow=all
allow=ulaw
allow=alaw
allow=g729
fromuser=88812XXXXX
fromdomain=sip.pennytel.com
canreinvite=no
canredirect=no
context=from-voip-provider
insecure=invite
nat=yes

[2001]
callerid=2001
type=friend
context=my-phones
secret=XXXXX
host=dynamic
disallow=all
allow=ulaw
allow=alaw
allow=g729

When i enable sip debug i get this

vserve-Ubuntu*CLI> sip set debug on
SIP Debugging enabled

<— SIP read from UDP:192.168.1.6:3359 —>

<------------->
Really destroying SIP dialog ‘NmJkYjYxZjVmMzQ1ZjdiZTE3YzljNTZkZmYxYTdkODk.’ Method: REGISTER
Really destroying SIP dialog ‘35ab62ef594f14681d549922712c02db@127.0.1.1’ Method: REGISTER
Reliably Transmitting (NAT) to 202.85.243.105:5060:
OPTIONS sip:sip.pennytel.com SIP/2.0
Via: SIP/2.0/UDP 59.99.239.180:5060;branch=z9hG4bK6ccb1448;rport
Max-Forwards: 70
From: “asterisk” sip:8881213187@59.99.239.180;tag=as4da6df69
To: sip:sip.pennytel.com
Contact: sip:8881213187@59.99.239.180:5060
Call-ID: 670571e3791f977833284df2164116df@59.99.239.180:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.2.1
Date: Thu, 10 May 2012 07:00:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:202.85.243.105:5060 —>
SIP/2.0 501 Unsupported Method
Via: SIP/2.0/UDP 59.99.239.180:5060;branch=z9hG4bK6ccb1448;rport=5060
To: sip:sip.pennytel.com;tag=3979cc58
From: “asterisk” sip:8881213187@59.99.239.180;tag=as4da6df69
Call-ID: 670571e3791f977833284df2164116df@59.99.239.180:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘670571e3791f977833284df2164116df@59.99.239.180:5060’ Method: OPTIONS

<— SIP read from UDP:192.168.1.6:3359 —>

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

<— SIP read from UDP:192.168.1.6:3359 —>

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

<— SIP read from UDP:192.168.1.6:3359 —>
REGISTER sip:192.168.1.50:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:3359;branch=z9hG4bK-d8754z-fd534d71c961a230-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:2001@192.168.1.6:3359;rinstance=c7716597dc067da8
To: sip:2001@192.168.1.50:5060
From: sip:2001@192.168.1.50:5060;tag=92482017
Call-ID: NmJkYjYxZjVmMzQ1ZjdiZTE3YzljNTZkZmYxYTdkODk.
CSeq: 161 REGISTER
Expires: 120
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
User-Agent: 3CXPhone 4.0.13527.0
Authorization: Digest username=“2001”,realm=“asterisk”,nonce=“203005b0”,uri=“sip:192.168.1.50:5060”,response=“be93511a0504ad498fff1f10db37f4da”,algorithm=MD5
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Sending to 192.168.1.6:3359 (NAT)

<— Transmitting (NAT) to 192.168.1.6:3359 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.6:3359;branch=z9hG4bK-d8754z-fd534d71c961a230-1—d8754z-;received=192.168.1.6;rport=3359
From: sip:2001@192.168.1.50:5060;tag=92482017
To: sip:2001@192.168.1.50:5060;tag=as485e2a75
Call-ID: NmJkYjYxZjVmMzQ1ZjdiZTE3YzljNTZkZmYxYTdkODk.
CSeq: 161 REGISTER
Server: Asterisk PBX 10.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="6dfb5c28"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘NmJkYjYxZjVmMzQ1ZjdiZTE3YzljNTZkZmYxYTdkODk.’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.1.6:3359 —>
REGISTER sip:192.168.1.50:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:3359;branch=z9hG4bK-d8754z-756f833987682d5a-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:2001@192.168.1.6:3359;rinstance=c7716597dc067da8
To: sip:2001@192.168.1.50:5060
From: sip:2001@192.168.1.50:5060;tag=92482017
Call-ID: NmJkYjYxZjVmMzQ1ZjdiZTE3YzljNTZkZmYxYTdkODk.
CSeq: 162 REGISTER
Expires: 120
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
User-Agent: 3CXPhone 4.0.13527.0
Authorization: Digest username=“2001”,realm=“asterisk”,nonce=“6dfb5c28”,uri=“sip:192.168.1.50:5060”,response=“40da0bcbb11bf13361b352a3ee88becb”,algorithm=MD5
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Sending to 192.168.1.6:3359 (NAT)

<— Transmitting (NAT) to 192.168.1.6:3359 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.6:3359;branch=z9hG4bK-d8754z-756f833987682d5a-1—d8754z-;received=192.168.1.6;rport=3359
From: sip:2001@192.168.1.50:5060;tag=92482017
To: sip:2001@192.168.1.50:5060;tag=as485e2a75
Call-ID: NmJkYjYxZjVmMzQ1ZjdiZTE3YzljNTZkZmYxYTdkODk.
CSeq: 162 REGISTER
Server: Asterisk PBX 10.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: sip:2001@192.168.1.6:3359;rinstance=c7716597dc067da8;expires=120
Date: Thu, 10 May 2012 07:01:35 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘NmJkYjYxZjVmMzQ1ZjdiZTE3YzljNTZkZmYxYTdkODk.’ in 32000 ms (Method: REGISTER)
Reliably Transmitting (NAT) to 202.85.243.105:5060:
OPTIONS sip:sip.pennytel.com SIP/2.0
Via: SIP/2.0/UDP 59.99.239.180:5060;branch=z9hG4bK00291324;rport
Max-Forwards: 70
From: “asterisk” sip:8881213187@59.99.239.180;tag=as363eed8d
To: sip:sip.pennytel.com
Contact: sip:8881213187@59.99.239.180:5060
Call-ID: 1fecf37919f773cf1fd344b435828941@59.99.239.180:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 10.2.1
Date: Thu, 10 May 2012 07:01:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:202.85.243.105:5060 —>
SIP/2.0 501 Unsupported Method
Via: SIP/2.0/UDP 59.99.239.180:5060;branch=z9hG4bK00291324;rport=5060
To: sip:sip.pennytel.com;tag=7c264a1c
From: “asterisk” sip:8881213187@59.99.239.180;tag=as363eed8d
Call-ID: 1fecf37919f773cf1fd344b435828941@59.99.239.180:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘1fecf37919f773cf1fd344b435828941@59.99.239.180:5060’ Method: OPTIONS

<— SIP read from UDP:192.168.1.6:3359 —>

<------------->
Really destroying SIP dialog ‘NmJkYjYxZjVmMzQ1ZjdiZTE3YzljNTZkZmYxYTdkODk.’ Method: REGISTER

<— SIP read from UDP:192.168.1.6:3359 —>

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

To be useful, the trace would need to include a complete outbound register sequence, and if that succeeded, an incoming call attempt. It includes neither.

I would suggest setting qualify=no whilst debugging, as it generates a lot of noise on the trace.

Thank you for your reply @david55

i change qualify=no then i make outbound call and get following details in sip debug

– Executing [919840470484@my-phones:1] Monitor(“SIP/2001-00000000”, “wav,20120510-153909-2001-919840470484,mb”) in new stack
– Executing [919840470484@my-phones:2] Dial(“SIP/2001-00000000”, “SIP/919840470484@Lismore-sip-account”) in new stack
== Using SIP RTP CoS mark 5
Audio is at 14244
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 202.85.243.105:5060:
INVITE sip:919840470484@sip.pennytel.com:5060 SIP/2.0
Via: SIP/2.0/UDP 59.99.239.180:5060;branch=z9hG4bK072bae57;rport
Max-Forwards: 70
From: “2001” sip:8881213187@sip.pennytel.com;tag=as6b98b743
To: sip:919840470484@sip.pennytel.com:5060
Contact: sip:8881213187@59.99.239.180:5060
Call-ID: 363d75375b5c5e7b222be1387686fda8@sip.pennytel.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.2.1
Date: Thu, 10 May 2012 10:09:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 329

v=0
o=root 129248110 129248110 IN IP4 59.99.239.180
s=Asterisk PBX 10.2.1
c=IN IP4 59.99.239.180
t=0 0
m=audio 14244 RTP/AVP 0 8 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


-- Called SIP/919840470484@Lismore-sip-account

<— SIP read from UDP:202.85.243.105:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 59.99.239.180:5060;branch=z9hG4bK072bae57;rport=5060
To: sip:919840470484@sip.pennytel.com:5060
From: "2001"sip:8881213187@sip.pennytel.com;tag=as6b98b743
Call-ID: 363d75375b5c5e7b222be1387686fda8@sip.pennytel.com
CSeq: 102 INVITE
Content-Length: 0

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

<— SIP read from UDP:202.85.243.105:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 59.99.239.180:5060;branch=z9hG4bK072bae57;rport=5060
Record-Route: sip:202.85.243.105:5077;lr
Record-Route: sip:202.85.243.105;lr
To: sip:919840470484@sip.pennytel.com:5060
From: "2001"sip:8881213187@sip.pennytel.com;tag=as6b98b743
Call-ID: 363d75375b5c5e7b222be1387686fda8@sip.pennytel.com
CSeq: 102 INVITE
Server: Sippy
WWW-Authenticate: Digest realm=“sip.pennytel.com”,nonce="f57287bdecbb8ad25e51800e7cf969428162"
Content-Length: 0

<------------->
— (11 headers 0 lines) —
set_destination: Parsing sip:919840470484@sip.pennytel.com:5060 for address/port to send to
set_destination: set destination to 202.85.243.105:5060
Transmitting (NAT) to 202.85.243.105:5060:
ACK sip:919840470484@sip.pennytel.com:5060 SIP/2.0
Via: SIP/2.0/UDP 59.99.239.180:5060;branch=z9hG4bK072bae57;rport
Max-Forwards: 70
From: “2001” sip:8881213187@sip.pennytel.com;tag=as6b98b743
To: sip:919840470484@sip.pennytel.com:5060
Contact: sip:8881213187@59.99.239.180:5060
Call-ID: 363d75375b5c5e7b222be1387686fda8@sip.pennytel.com
CSeq: 102 ACK
User-Agent: Asterisk PBX 10.2.1
Content-Length: 0


Audio is at 14244
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 202.85.243.105:5060:
INVITE sip:919840470484@sip.pennytel.com:5060 SIP/2.0
Via: SIP/2.0/UDP 59.99.239.180:5060;branch=z9hG4bK5f8d5e99;rport
Max-Forwards: 70
From: “2001” sip:8881213187@sip.pennytel.com;tag=as6b98b743
To: sip:919840470484@sip.pennytel.com:5060
Contact: sip:8881213187@59.99.239.180:5060
Call-ID: 363d75375b5c5e7b222be1387686fda8@sip.pennytel.com
CSeq: 103 INVITE
User-Agent: Asterisk PBX 10.2.1
Authorization: Digest username=“8881213187”, realm=“sip.pennytel.com”, algorithm=MD5, uri=“sip:919840470484@sip.pennytel.com:5060”, nonce=“f57287bdecbb8ad25e51800e7cf969428162”, response="e08e00e4b6aaa4103cdb30c9791c1bed"
Date: Thu, 10 May 2012 10:09:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 329

v=0
o=root 129248110 129248111 IN IP4 59.99.239.180
s=Asterisk PBX 10.2.1
c=IN IP4 59.99.239.180
t=0 0
m=audio 14244 RTP/AVP 0 8 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


<— SIP read from UDP:202.85.243.105:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 59.99.239.180:5060;branch=z9hG4bK5f8d5e99;rport=5060
To: sip:919840470484@sip.pennytel.com:5060
From: "2001"sip:8881213187@sip.pennytel.com;tag=as6b98b743
Call-ID: 363d75375b5c5e7b222be1387686fda8@sip.pennytel.com
CSeq: 103 INVITE
Content-Length: 0

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

<— SIP read from UDP:202.85.243.105:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 59.99.239.180:5060;branch=z9hG4bK5f8d5e99;rport=5060
Record-Route: sip:202.85.243.105:5077;lr
Record-Route: sip:202.85.243.105;lr
To: sip:919840470484@sip.pennytel.com:5060;tag=yj6pjfmlwwmjx3t7.i
From: "2001"sip:8881213187@sip.pennytel.com;tag=as6b98b743
Call-ID: 363d75375b5c5e7b222be1387686fda8@sip.pennytel.com
CSeq: 103 INVITE
Content-Type: application/sdp
Server: Sippy
Content-Length: 194

v=0
o=Sippy 739701752 1 IN IP4 202.85.243.105
s=SIP Call
t=0 0
m=audio 19186 RTP/AVP 0 101
c=IN IP4 202.85.241.91
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
<------------->
— (11 headers 9 lines) —
list_route: hop: sip:202.85.243.105;lr
list_route: hop: sip:202.85.243.105:5077;lr
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g729), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 202.85.241.91:19186
– SIP/Lismore-sip-account-00000001 is making progress passing it to SIP/2001-00000000
Audio is at 25078
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (NAT) to 192.168.1.6:19770 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.6:19770;branch=z9hG4bK-d8754z-a203e73b530c1136-1—d8754z-;received=192.168.1.6;rport=19770
From: sip:2001@192.168.1.50:5060;tag=60785e1e
To: sip:919840470484@192.168.1.50:5060;tag=as09b0a456
Call-ID: YjhjZGI5NDMwM2NmZjA0NGZhM2FlYWMyZTg0YzZjOTQ.
CSeq: 1 INVITE
Server: Asterisk PBX 10.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:919840470484@192.168.1.50:5060
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 1845864864 1845864864 IN IP4 192.168.1.50
s=Asterisk PBX 10.2.1
c=IN IP4 192.168.1.50
t=0 0
m=audio 25078 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 34

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

<— SIP read from UDP:192.168.1.6:19770 —>
REGISTER sip:192.168.1.50:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:19770;branch=z9hG4bK-d8754z-5e554b59fd668017-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:2001@192.168.1.6:19770;rinstance=e9158996eaa117f0
To: sip:2001@192.168.1.50:5060
From: sip:2001@192.168.1.50:5060;tag=4b667103
Call-ID: N2Y4NDM5OTNhOWNlMWQ3ZTEwNmRmOWE2YjczZmY5OWU.
CSeq: 13 REGISTER
Expires: 120
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
User-Agent: 3CXPhone 4.0.13527.0
Authorization: Digest username=“2001”,realm=“asterisk”,nonce=“1f9c7156”,uri=“sip:192.168.1.50:5060”,response=“a6d61c5967a52a5ad01d93381adb320d”,algorithm=MD5
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Sending to 192.168.1.6:19770 (NAT)

<— Transmitting (NAT) to 192.168.1.6:19770 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.6:19770;branch=z9hG4bK-d8754z-5e554b59fd668017-1—d8754z-;received=192.168.1.6;rport=19770
From: sip:2001@192.168.1.50:5060;tag=4b667103
To: sip:2001@192.168.1.50:5060;tag=as0e7b2753
Call-ID: N2Y4NDM5OTNhOWNlMWQ3ZTEwNmRmOWE2YjczZmY5OWU.
CSeq: 13 REGISTER
Server: Asterisk PBX 10.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="55873397"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘N2Y4NDM5OTNhOWNlMWQ3ZTEwNmRmOWE2YjczZmY5OWU.’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.1.6:19770 —>
REGISTER sip:192.168.1.50:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:19770;branch=z9hG4bK-d8754z-c0220b4c9e4db14b-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:2001@192.168.1.6:19770;rinstance=e9158996eaa117f0
To: sip:2001@192.168.1.50:5060
From: sip:2001@192.168.1.50:5060;tag=4b667103
Call-ID: N2Y4NDM5OTNhOWNlMWQ3ZTEwNmRmOWE2YjczZmY5OWU.
CSeq: 14 REGISTER
Expires: 120
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
User-Agent: 3CXPhone 4.0.13527.0
Authorization: Digest username=“2001”,realm=“asterisk”,nonce=“55873397”,uri=“sip:192.168.1.50:5060”,response=“6e7cf1a29ad35689ec80059c9182a808”,algorithm=MD5
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Sending to 192.168.1.6:19770 (NAT)
> Saved useragent “3CXPhone 4.0.13527.0” for peer 2001

<— Transmitting (NAT) to 192.168.1.6:19770 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.6:19770;branch=z9hG4bK-d8754z-c0220b4c9e4db14b-1—d8754z-;received=192.168.1.6;rport=19770
From: sip:2001@192.168.1.50:5060;tag=4b667103
To: sip:2001@192.168.1.50:5060;tag=as0e7b2753
Call-ID: N2Y4NDM5OTNhOWNlMWQ3ZTEwNmRmOWE2YjczZmY5OWU.
CSeq: 14 REGISTER
Server: Asterisk PBX 10.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: sip:2001@192.168.1.6:19770;rinstance=e9158996eaa117f0;expires=120
Date: Thu, 10 May 2012 10:09:18 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘N2Y4NDM5OTNhOWNlMWQ3ZTEwNmRmOWE2YjczZmY5OWU.’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.1.6:19770 —>
CANCEL sip:919840470484@192.168.1.50:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:19770;branch=z9hG4bK-d8754z-a203e73b530c1136-1—d8754z-;rport
Max-Forwards: 70
To: sip:919840470484@192.168.1.50:5060
From: sip:2001@192.168.1.50:5060;tag=60785e1e
Call-ID: YjhjZGI5NDMwM2NmZjA0NGZhM2FlYWMyZTg0YzZjOTQ.
CSeq: 1 CANCEL
User-Agent: 3CXPhone 4.0.13527.0
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Sending to 192.168.1.6:19770 (NAT)

<— Reliably Transmitting (NAT) to 192.168.1.6:19770 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.6:19770;branch=z9hG4bK-d8754z-a203e73b530c1136-1—d8754z-;received=192.168.1.6;rport=19770
From: sip:2001@192.168.1.50:5060;tag=60785e1e
To: sip:919840470484@192.168.1.50:5060;tag=as09b0a456
Call-ID: YjhjZGI5NDMwM2NmZjA0NGZhM2FlYWMyZTg0YzZjOTQ.
CSeq: 1 INVITE
Server: Asterisk PBX 10.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

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

<— Transmitting (NAT) to 192.168.1.6:19770 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.6:19770;branch=z9hG4bK-d8754z-a203e73b530c1136-1—d8754z-;received=192.168.1.6;rport=19770
From: sip:2001@192.168.1.50:5060;tag=60785e1e
To: sip:919840470484@192.168.1.50:5060;tag=as09b0a456
Call-ID: YjhjZGI5NDMwM2NmZjA0NGZhM2FlYWMyZTg0YzZjOTQ.
CSeq: 1 CANCEL
Server: Asterisk PBX 10.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '363d75375b5c5e7b222be1387686fda8@sip.pennytel.com’ in 32000 ms (Method: INVITE)
set_destination: Parsing sip:919840470484@sip.pennytel.com:5060 for address/port to send to

<— SIP read from UDP:192.168.1.6:19770 —>
ACK sip:919840470484@192.168.1.50:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:19770;branch=z9hG4bK-d8754z-a203e73b530c1136-1—d8754z-;rport
Max-Forwards: 70
To: sip:919840470484@192.168.1.50:5060;tag=as09b0a456
From: sip:2001@192.168.1.50:5060;tag=60785e1e
Call-ID: YjhjZGI5NDMwM2NmZjA0NGZhM2FlYWMyZTg0YzZjOTQ.
CSeq: 1 ACK
Content-Length: 0

<------------->
— (8 headers 0 lines) —
set_destination: set destination to 202.85.243.105:5060
Reliably Transmitting (NAT) to 202.85.243.105:5060:
CANCEL sip:919840470484@sip.pennytel.com:5060 SIP/2.0
Via: SIP/2.0/UDP 59.99.239.180:5060;branch=z9hG4bK5f8d5e99;rport
Max-Forwards: 70
From: “2001” sip:8881213187@sip.pennytel.com;tag=as6b98b743
To: sip:919840470484@sip.pennytel.com:5060
Call-ID: 363d75375b5c5e7b222be1387686fda8@sip.pennytel.com
CSeq: 103 CANCEL
User-Agent: Asterisk PBX 10.2.1
Content-Length: 0


Scheduling destruction of SIP dialog '363d75375b5c5e7b222be1387686fda8@sip.pennytel.com’ in 32000 ms (Method: INVITE)
== Spawn extension (my-phones, 919840470484, 2) exited non-zero on 'SIP/2001-00000000’
Really destroying SIP dialog ‘YjhjZGI5NDMwM2NmZjA0NGZhM2FlYWMyZTg0YzZjOTQ.’ Method: ACK

<— SIP read from UDP:202.85.243.105:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 59.99.239.180:5060;branch=z9hG4bK5f8d5e99;rport=5060
To: sip:919840470484@sip.pennytel.com:5060
From: "2001"sip:8881213187@sip.pennytel.com;tag=as6b98b743
Call-ID: 363d75375b5c5e7b222be1387686fda8@sip.pennytel.com
CSeq: 103 CANCEL
Content-Length: 0

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

<— SIP read from UDP:202.85.243.105:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 59.99.239.180:5060;branch=z9hG4bK5f8d5e99;rport=5060
Record-Route: sip:202.85.243.105:5077;lr
Record-Route: sip:202.85.243.105;lr
To: sip:919840470484@sip.pennytel.com:5060
From: "2001"sip:8881213187@sip.pennytel.com;tag=as6b98b743
Call-ID: 363d75375b5c5e7b222be1387686fda8@sip.pennytel.com
CSeq: 103 INVITE
Server: Sippy
Content-Length: 0

<------------->
— (10 headers 0 lines) —
set_destination: Parsing sip:919840470484@sip.pennytel.com:5060 for address/port to send to
set_destination: set destination to 202.85.243.105:5060
Transmitting (NAT) to 202.85.243.105:5060:
ACK sip:919840470484@sip.pennytel.com:5060 SIP/2.0
Via: SIP/2.0/UDP 59.99.239.180:5060;branch=z9hG4bK5f8d5e99;rport
Max-Forwards: 70
From: “2001” sip:8881213187@sip.pennytel.com;tag=as6b98b743
To: sip:919840470484@sip.pennytel.com:5060
Contact: sip:8881213187@59.99.239.180:5060
Call-ID: 363d75375b5c5e7b222be1387686fda8@sip.pennytel.com
CSeq: 103 ACK
User-Agent: Asterisk PBX 10.2.1
Content-Length: 0


Scheduling destruction of SIP dialog '363d75375b5c5e7b222be1387686fda8@sip.pennytel.com’ in 32000 ms (Method: INVITE)
Really destroying SIP dialog ‘3e5c13f2085b6a5d4570be037e52d02f@127.0.1.1’ Method: REGISTER

<— SIP read from UDP:192.168.1.6:19770 —>

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

You need to provide a trace with a failing outgoing REGISTER or a failing incoming call. There are no outbound REGISTERs or relevant incoming calls.

You might want to check the logs from when sip.conf was parsed. Maybe it didn’t like the Register line. Also use sip show registry (I think).

<— SIP read from UDP:202.85.243.105:5060 —>
SIP/2.0 501 Unsupported Method
Via: SIP/2.0/UDP 59.99.239.180:5060;branch=z9hG4bK7842365b;rport=5060
To: sip:sip.pennytel.com;tag=51c99f63
From: “asterisk” sip:8881213187@59.99.239.180;tag=as3da9cb3d
Call-ID: 291314ad75f627ed5f5818ea7d5f4c93@59.99.239.180:5060
CSeq: 102 OPTIONS
Content-Length: 0

I get 501 Unsupported Method response from my provider. is this problem?

No. In any case, setting qualify=no should get rid of these. Asterisk doesn’t care what response it gets to those OPTIONS, only that it gets a response. If it didn’t get one, you wouldn’t even be able to make outgoing calls.

If you were to get an unsupported method with REGISTER in the CSeq line, that would be a problem, but you haven’t provided any evidence that you are even sending the REGISTER request, let alone what the response was.

Asterisk successfully registered in Pennytel.

I Check Pennytel website it show my externel ip with port 5060 and asterisk version correctly.

Ubuntu*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
sip.pennytel.com:5060 N 8881213187 285 Registered Fri, 11 May 2012 09:58:57
1 SIP registrations.

As you don’t seem to be receiving the INVITE, assuming that your test calls were to yourself, I would look at your firewall/NAT configuration, to make sure that it can correctly route unsolicited 5060/UDP packets.