Kamran Ahmad
2005-Mar-02 10:04 UTC
[Asterisk-Users] Dial application invoked again and again
hi all
i am using CVS with Realtime mysql on backend. 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
Kamran Ahmad
------------------------------------------------------
*CLI> sip debug
SIP Debugging Enabled
*CLI>
Sip read:
INVITE sip:2000@192.168.0.203 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.117;branch=z9hG4bK2038176231
From:<sip:3000@192.168.0.203>;
To: <sip:2000@192.168.0.203>
Call-ID: 52@192.168.0.117
CSeq: 20 INVITE
Contact: <sip:3000@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@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@192.168.0.203>;
To: <sip:2000@192.168.0.203>;tag=as7a83cce0
Call-ID: 52@192.168.0.117
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2000@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@192.168.0.117 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.203:5060;branch=z9hG4bK56922e05
From: "3000" <sip:3000@192.168.0.203>;tag=as35d782e5
To: <sip:2000@192.168.0.117>
Contact: <sip:3000@192.168.0.203>
Call-ID:
3f6f2ff534398d411a2ca47b266ad133@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@192.168.0.203>
To: <sip:2000@192.168.0.117>
Contact:<3000@192.168.0.117>
Call-ID:
3f6f2ff534398d411a2ca47b266ad133@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@192.168.0.117 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.203:5060;branch=z9hG4bK56922e05
From: "3000" <sip:3000@192.168.0.203>;tag=as35d782e5
To: <sip:2000@192.168.0.117>
Contact: <sip:3000@192.168.0.203>
Call-ID:
3f6f2ff534398d411a2ca47b266ad133@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@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@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@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@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@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@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@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@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)
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Kamran Ahmad
2005-Mar-03 02:14 UTC
[Asterisk-Users] Re: Dial application invoked again and again
hi If i remove "_." from my dialplan(extensions.conf). application is invoked only once. otherwise application is invoked again and again. any one know what is the problem and how to make (global) dialplan for all user agents. thanks Kamran __________________________________ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web http://birthday.yahoo.com/netrospective/