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 —>
<------------->