[Asterisk-Dev] Dial application invoked again and again

Tue Mar 1 22:56:50 MST 2005

hi all

Dial application is invoked again and again what is
the reason. i have tested it by printing some message
to debug. this application is invoked again and again

here is debug you can see lot of messages from
app_dial.c at the end. any one tell me what is the
reason is this a bug or what i am using CVS

Kamran Ahmad
------------------------------------------------------
*CLI> sip debug
SIP Debugging Enabled
*CLI>

Sip read:
INVITE sip:2000 at 192.168.0.203 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.117;branch=z9hG4bK2038176231
From:<sip:3000 at 192.168.0.203>;
To: <sip:2000 at 192.168.0.203>
Call-ID: 52 at 192.168.0.117
CSeq: 20 INVITE
Contact: <sip:3000 at 192.168.0.117>
Max-Forwards: 5
User-Agent:SKYPHONE/1.03
Subject: hello
Expires: 120
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS,
REFER,SUBSCRIBE, NOTIFY, MESSAGE
Content-Type: application/sdp
Content-Length:180

v=0
o=sibtay 2890844 842807 IN IP4 192.168.0.117
s=SDP Seminar
c=IN IP4 192.168.0.117
t=0 0
m=audio 13044 RTP/AVP 0 101
a=rtpmap:101 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:96 0-11,16

14 headers, 10 lines
Using latest request as basis request
Sending to 192.168.0.117 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.117:13044
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer
- audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4
(ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1
(g723), combined - 0x1 (g723)
Found user '3000'
Looking for 2000 in default
list_route: hop: <sip:3000 at 192.168.0.117>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.0.117;branch=z9hG4bK2038176231
From: <sip:3000 at 192.168.0.203>;
To: <sip:2000 at 192.168.0.203>;tag=as7a83cce0
Call-ID: 52 at 192.168.0.117
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2000 at 192.168.0.203>
Content-Length: 0

 to 192.168.0.117:5060
Mar  3 10:44:01 WARNING[6311]: app_dial.c:618
dial_exec_full: hello i am from app_dial
We're at 192.168.0.203 port 15344
Answering/Requesting with root capability 0x4 (ulaw)
Answering with capability 0x2 (gsm)
Answering with capability 0x8 (alaw)
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:2000 at 192.168.0.117 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.203:5060;branch=z9hG4bK56922e05
From: "3000" <sip:3000 at 192.168.0.203>;tag=as35d782e5
To: <sip:2000 at 192.168.0.117>
Contact: <sip:3000 at 192.168.0.203>
Call-ID:
3f6f2ff534398d411a2ca47b266ad133 at 192.168.0.203
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 03 Mar 2005 05:44:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 207

v=0
o=root 6311 6311 IN IP4 192.168.0.203
s=session
c=IN IP4 192.168.0.203
t=0 0
m=audio 15344 RTP/AVP 0 3 8
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
 (no NAT) to 192.168.0.117:5060

Sip read:
SIP/2.0 486 Busy Here
From:<sip:3000 at 192.168.0.203>
To: <sip:2000 at 192.168.0.117>
Contact:<3000 at 192.168.0.117>
Call-ID:
3f6f2ff534398d411a2ca47b266ad133 at 192.168.0.203
CSeq: 102 INVITE
User-Agent: SKYPHONE/1.03
via: SIP/2.0/UDP
192.168.0.203:5060;branch=z9hG4bK56922e05
Content-Length: 0

9 headers, 0 lines
Transmitting:
ACK sip:2000 at 192.168.0.117 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.203:5060;branch=z9hG4bK56922e05
From: "3000" <sip:3000 at 192.168.0.203>;tag=as35d782e5
To: <sip:2000 at 192.168.0.117>
Contact: <sip:3000 at 192.168.0.203>
Call-ID:
3f6f2ff534398d411a2ca47b266ad133 at 192.168.0.203
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 192.168.0.117:5060
Destroying call
'3f6f2ff534398d411a2ca47b266ad133 at 192.168.0.203'
Mar  3 10:44:11 NOTICE[6311]: rtp.c:452 ast_rtp_read:
RTP: Received packet with bad UDP checksum
Mar  3 10:44:11 WARNING[6311]: app_dial.c:618
dial_exec_full: hello i am from app_dial
Mar  3 10:44:11 WARNING[6311]: chan_sip.c:1345
create_addr: No such host: t
Destroying call
'647bf9bc0c12c2120f2d50cc1fb52ca6 at 192.168.0.203'
Mar  3 10:44:11 NOTICE[6311]: app_dial.c:918
dial_exec_full: Unable to create channel of type 'SIP'
(cause 3)
Mar  3 10:44:21 WARNING[6311]: app_dial.c:618
dial_exec_full: hello i am from app_dial
Mar  3 10:44:21 WARNING[6311]: chan_sip.c:1345
create_addr: No such host: t
Destroying call
'3b8f076e01b48b45119eebf3209161e2 at 192.168.0.203'
Mar  3 10:44:21 NOTICE[6311]: app_dial.c:918
dial_exec_full: Unable to create channel of type 'SIP'
(cause 3)
Mar  3 10:44:31 NOTICE[6311]: rtp.c:452 ast_rtp_read:
RTP: Received packet with bad UDP checksum
Mar  3 10:44:31 WARNING[6311]: app_dial.c:618
dial_exec_full: hello i am from app_dial
Mar  3 10:44:31 WARNING[6311]: chan_sip.c:1345
create_addr: No such host: t
Destroying call
'4f39a26e544cd15424b0a7ac03732faf at 192.168.0.203'
Mar  3 10:44:31 NOTICE[6311]: app_dial.c:918
dial_exec_full: Unable to create channel of type 'SIP'
(cause 3)
Mar  3 10:44:41 NOTICE[6311]: rtp.c:452 ast_rtp_read:
RTP: Received packet with bad UDP checksum
Mar  3 10:44:41 WARNING[6311]: app_dial.c:618
dial_exec_full: hello i am from app_dial
Mar  3 10:44:42 WARNING[6311]: chan_sip.c:1345
create_addr: No such host: t
Destroying call
'29a22a7170a0437263dd16a539127ac3 at 192.168.0.203'
Mar  3 10:44:42 NOTICE[6311]: app_dial.c:918
dial_exec_full: Unable to create channel of type 'SIP'
(cause 3)
Mar  3 10:44:52 WARNING[6311]: app_dial.c:618
dial_exec_full: hello i am from app_dial
Mar  3 10:44:52 WARNING[6311]: chan_sip.c:1345
create_addr: No such host: t
Destroying call
'056dce1835dca462689ec24840c7496f at 192.168.0.203'
Mar  3 10:44:52 NOTICE[6311]: app_dial.c:918
dial_exec_full: Unable to create channel of type 'SIP'
(cause 3)
Mar  3 10:45:02 WARNING[6311]: app_dial.c:618
dial_exec_full: hello i am from app_dial
Mar  3 10:45:03 WARNING[6311]: chan_sip.c:1345
create_addr: No such host: t
Destroying call
'733037df6865f27b61bf6f805871af96 at 192.168.0.203'
Mar  3 10:45:03 NOTICE[6311]: app_dial.c:918
dial_exec_full: Unable to create channel of type 'SIP'
(cause 3)
Mar  3 10:45:13 NOTICE[6311]: rtp.c:452 ast_rtp_read:
RTP: Received packet with bad UDP checksum
Mar  3 10:45:13 WARNING[6311]: app_dial.c:618
dial_exec_full: hello i am from app_dial
Mar  3 10:45:13 WARNING[6311]: chan_sip.c:1345
create_addr: No such host: t
Destroying call
'483dd02138e210726cb865003a56393b at 192.168.0.203'
Mar  3 10:45:13 NOTICE[6311]: app_dial.c:918
dial_exec_full: Unable to create channel of type 'SIP'
(cause 3)