[asterisk-bugs] [Asterisk 0011425]: Dailstatus says NOANSWER even if i pick the call
noreply at bugs.digium.com
noreply at bugs.digium.com
Wed Feb 6 07:23:43 CST 2008
A NOTE has been added to this issue.
======================================================================
http://bugs.digium.com/view.php?id=11425
======================================================================
Reported By: naveenpalani
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 11425
Category: Channels/chan_oss
Reproducibility: always
Severity: major
Priority: normal
Status: feedback
Asterisk Version: 1.4.11
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
======================================================================
Date Submitted: 11-30-2007 04:12 CST
Last Modified: 02-06-2008 07:23 CST
======================================================================
Summary: Dailstatus says NOANSWER even if i pick the call
Description:
I am making an outgoing call using a sip provider.
I Could make calls to the required numbers and deliver the intended audio
speech. However when i pick the call, Dail status doesnt give me "ANSWER"
as the status back, I always get NOANSWER as the reposnse back.
Gives out the message in my Asterisk cli prompt:
No one is available to answer at this time (1:0/0/0)
Can someone suggest me why do i get this message and the dialstatus does
not give me answer even i pick up.
My sip debug is as given below:
*CLI> Really destroying SIP dialog
'40205cd06ecc959a5c3fd83b27881379 at 10.1.1.68' Method: REGISTER
-- Attempting call on Local/outbound at dialout for
outbound-handler at dialout:1 (Retry 1)
-- Executing [outbound at dialout:1]
Answer("Local/outbound at dialout-e3ed,2", "") in new stack
-- Executing [outbound at dialout:2]
Wait("Local/outbound at dialout-e3ed,2", "30") in new stack
-- Executing [outbound-handler at dialout:1]
Dial("Local/outbound at dialout-e3ed,1",
"SIP/011919960466622 at proxy2.bandtel.com|120") in new stack
Audio is at 10.1.1.68 port 32392
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 66.237.65.67:5060:
INVITE sip:011919960466622 at 65.175.129.149 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK44e0a2fb;rport
From: "Linux" <sip:2068200001 at proxy2.bandtel.com>;tag=as0f8dfdfa
To: <sip:011919960466622 at 65.175.129.149>
Contact: <sip:2068200001 at 10.1.1.68>
Call-ID: 3c4ec9614a41dea303fd8a57192cb180 at proxy2.bandtel.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Linux"
<sip:Linux at proxy2.bandtel.com>;privacy=off;screen=no
Date: Fri, 30 Nov 2007 09:36:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 234
v=0
o=root 31524 31524 IN IP4 10.1.1.68
s=session
c=IN IP4 10.1.1.68
t=0 0
m=audio 32392 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called 011919960466622 at proxy2.bandtel.com
<--- SIP read from 66.237.65.67:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK44e0a2fb;rport=5060
From: "Linux" <sip:2068200001 at proxy2.bandtel.com>;tag=as0f8dfdfa
To: <sip:011919960466622 at 65.175.129.149>
Call-ID: 3c4ec9614a41dea303fd8a57192cb180 at proxy2.bandtel.com
CSeq: 102 INVITE
Server: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from 66.237.65.67:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK44e0a2fb;rport=5060
Record-Route: <sip:66.237.65.67;ftag=as0f8dfdfa;lr>
From: Linux <sip:2068200001 at proxy2.bandtel.com>;tag=as0f8dfdfa
To: <sip:011919960466622 at 65.175.129.149>
Call-ID: 3c4ec9614a41dea303fd8a57192cb180 at proxy2.bandtel.com
CSeq: 102 INVITE
Server: Sippy
WWW-Authenticate: Digest
realm="66.237.65.67",nonce="30c0770bab595318a6961a00a640fdc7474fdc26"
<------------->
--- (9 headers 0 lines) ---
Transmitting (NAT) to 66.237.65.67:5060:
ACK sip:011919960466622 at 65.175.129.149 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK44e0a2fb;rport
From: "Linux" <sip:2068200001 at proxy2.bandtel.com>;tag=as0f8dfdfa
To: <sip:011919960466622 at 65.175.129.149>
Contact: <sip:2068200001 at 10.1.1.68>
Call-ID: 3c4ec9614a41dea303fd8a57192cb180 at proxy2.bandtel.com
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Linux"
<sip:Linux at proxy2.bandtel.com>;privacy=off;screen=no
Content-Length: 0
---
Audio is at 10.1.1.68 port 32392
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 66.237.65.67:5060:
INVITE sip:011919960466622 at 65.175.129.149 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK10fd7dab;rport
From: "Linux" <sip:2068200001 at proxy2.bandtel.com>;tag=as0f8dfdfa
To: <sip:011919960466622 at 65.175.129.149>
Contact: <sip:2068200001 at 10.1.1.68>
Call-ID: 3c4ec9614a41dea303fd8a57192cb180 at proxy2.bandtel.com
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Linux"
<sip:Linux at proxy2.bandtel.com>;privacy=off;screen=no
Authorization: Digest username="2068200001", realm="66.237.65.67",
algorithm=MD5, uri="sip:011919960466622 at 65.175.129.149",
nonce="30c0770bab595318a6961a00a640fdc7474fdc26",
response="09cb003eac8cacc93ff4fbfec2605f6a", opaque=""
Date: Fri, 30 Nov 2007 09:36:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 234
v=0
o=root 31524 31525 IN IP4 10.1.1.68
s=session
c=IN IP4 10.1.1.68
t=0 0
m=audio 32392 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
<--- SIP read from 66.237.65.67:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK10fd7dab;rport=5060
From: "Linux" <sip:2068200001 at proxy2.bandtel.com>;tag=as0f8dfdfa
To: <sip:011919960466622 at 65.175.129.149>
Call-ID: 3c4ec9614a41dea303fd8a57192cb180 at proxy2.bandtel.com
CSeq: 103 INVITE
Server: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Reliably Transmitting (NAT) to 66.237.65.67:5060:
OPTIONS sip:65.175.129.149 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK692c47be;rport
From: "asterisk" <sip:asterisk at 10.1.1.68>;tag=as013c2cf1
To: <sip:65.175.129.149>
Contact: <sip:asterisk at 10.1.1.68>
Call-ID: 0fd272674df0d814509669360caf1f25 at 10.1.1.68
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 30 Nov 2007 09:37:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
<--- SIP read from 66.237.65.67:5060 --->
SIP/2.0 501 Not Implemented
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK692c47be;rport=5060
From: asterisk <sip:asterisk at 10.1.1.68>;tag=as013c2cf1
To: <sip:65.175.129.149>
Call-ID: 0fd272674df0d814509669360caf1f25 at 10.1.1.68
CSeq: 102 OPTIONS
Server: Sippy
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '0fd272674df0d814509669360caf1f25 at 10.1.1.68'
Method: OPTIONS
<--- SIP read from 66.237.65.67:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK10fd7dab;rport=5060
Record-Route: <sip:66.237.65.67;ftag=as0f8dfdfa;lr>
From: Linux <sip:2068200001 at proxy2.bandtel.com>;tag=as0f8dfdfa
To:
<sip:011919960466622 at 65.175.129.149>;tag=16907dc5b05da4b49d991e769bce1862
Call-ID: 3c4ec9614a41dea303fd8a57192cb180 at proxy2.bandtel.com
CSeq: 103 INVITE
Server: Sippy
Content-Length: 116
Content-Type: application/sdp
v=0
o=GK-ATSI-SAT1 0 0 IN IP4 64.194.200.100
s=sip call
t=0 0
m=audio 62378 RTP/AVP 0
c=IN IP4 64.194.200.120
<------------->
--- (10 headers 6 lines) ---
Found RTP audio format 0
Peer audio RTP is at port 64.194.200.120:62378
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0
(nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0
(nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 64.194.200.120:62378
-- SIP/proxy2.bandtel.com-08b5ec28 is making progress passing it to
Local/outbound at dialout-e3ed,1
-- Executing [outbound at dialout:3]
NoOp("Local/outbound at dialout-e3ed,2", "status=") in new stack
-- Executing [outbound at dialout:4] AGI("Local/outbound at dialout-e3ed,2",
"agi://10.1.1.68/ivr/unanswered") in new stack
-- AGI Script agi://10.1.1.68/ivr/unanswered completed, returning 0
-- Executing [outbound at dialout:5]
Hangup("Local/outbound at dialout-e3ed,2", "") in new stack
== Spawn extension (dialout, outbound, 5) exited non-zero on
'Local/outbound at dialout-e3ed,2'
Scheduling destruction of SIP dialog
'3c4ec9614a41dea303fd8a57192cb180 at proxy2.bandtel.com' in 6400 ms (Method:
INVITE)
Reliably Transmitting (NAT) to 66.237.65.67:5060:
CANCEL sip:011919960466622 at 65.175.129.149 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK10fd7dab;rport
From: "Linux" <sip:2068200001 at proxy2.bandtel.com>;tag=as0f8dfdfa
To: <sip:011919960466622 at 65.175.129.149>
Call-ID: 3c4ec9614a41dea303fd8a57192cb180 at proxy2.bandtel.com
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Linux"
<sip:Linux at proxy2.bandtel.com>;privacy=off;screen=no
Content-Length: 0
---
Scheduling destruction of SIP dialog
'3c4ec9614a41dea303fd8a57192cb180 at proxy2.bandtel.com' in 6400 ms (Method:
INVITE)
== Spawn extension (dialout, outbound-handler, 1) exited non-zero on
'Local/outbound at dialout-e3ed,1'
[Nov 30 03:37:23] NOTICE[32020]: pbx_spool.c:351 attempt_thread: Call
completed to Local/outbound at dialout
<--- SIP read from 66.237.65.67:5060 --->
SIP/2.0 200 ok -- no more pending branches
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK10fd7dab;rport=5060
From: "Linux" <sip:2068200001 at proxy2.bandtel.com>;tag=as0f8dfdfa
To:
<sip:011919960466622 at 65.175.129.149>;tag=52c7b1d5444c5b44ef4d77f6a6c80dc0-24c4
Call-ID: 3c4ec9614a41dea303fd8a57192cb180 at proxy2.bandtel.com
CSeq: 103 CANCEL
Server: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog
'3c4ec9614a41dea303fd8a57192cb180 at proxy2.bandtel.com' Method: INVITE
<--- SIP read from 66.237.65.67:5060 --->
BYE sip:2068200001 at 10.1.1.68 SIP/2.0
Via: SIP/2.0/UDP
66.237.65.67;branch=z9hG4bKcf21.6973be264b9fb08855ecde4425d56ab2.0
Via: SIP/2.0/UDP
66.237.65.67:5061;branch=z9hG4bK82d7124b92daf3dc7a36f7c11bfdcdd1;rport=5061
Max-Forwards: 16
From:
<sip:011919960466622 at 65.175.129.149>;tag=16907dc5b05da4b49d991e769bce1862
To: Linux <sip:2068200001 at proxy2.bandtel.com>;tag=as0f8dfdfa
Call-ID: 3c4ec9614a41dea303fd8a57192cb180 at proxy2.bandtel.com
CSeq: 100 BYE
Contact: Anonymous <sip:66.237.65.67:5061>
Expires: 300
User-Agent: Sippy
cisco-GUID: 1336583865-2834834141-1419345930-365715115
h323-conf-id: 1336583865-2834834141-1419345930-365715115
<------------->
--- (13 headers 0 lines) ---
<--- Transmitting (no NAT) to 66.237.65.67:5060 --->
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP
66.237.65.67;branch=z9hG4bKcf21.6973be264b9fb08855ecde4425d56ab2.0;received=66.237.65.67
Via: SIP/2.0/UDP
66.237.65.67:5061;branch=z9hG4bK82d7124b92daf3dc7a36f7c11bfdcdd1;rport=5061
From:
<sip:011919960466622 at 65.175.129.149>;tag=16907dc5b05da4b49d991e769bce1862
To: Linux <sip:2068200001 at proxy2.bandtel.com>;tag=as0f8dfdfa
Call-ID: 3c4ec9614a41dea303fd8a57192cb180 at proxy2.bandtel.com
CSeq: 100 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
Reliably Transmitting (NAT) to 66.237.65.67:5060:
OPTIONS sip:65.175.129.149 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK0929dddd;rport
From: "asterisk" <sip:asterisk at 10.1.1.68>;tag=as752b1d75
To: <sip:65.175.129.149>
Contact: <sip:asterisk at 10.1.1.68>
Call-ID: 3b7f41405076bfdd00edf8807cdfb387 at 10.1.1.68
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 30 Nov 2007 09:38:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
<--- SIP read from 66.237.65.67:5060 --->
SIP/2.0 501 Not Implemented
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK0929dddd;rport=5060
From: asterisk <sip:asterisk at 10.1.1.68>;tag=as752b1d75
To: <sip:65.175.129.149>
Call-ID: 3b7f41405076bfdd00edf8807cdfb387 at 10.1.1.68
CSeq: 102 OPTIONS
Server: Sippy
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '3b7f41405076bfdd00edf8807cdfb387 at 10.1.1.68'
Method: OPTIONS
*CLI> Reliably Transmitting (NAT) to 66.237.65.67:5060:
OPTIONS sip:65.175.129.149 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK1e127475;rport
From: "asterisk" <sip:asterisk at 10.1.1.68>;tag=as5bbf1bfd
To: <sip:65.175.129.149>
Contact: <sip:asterisk at 10.1.1.68>
Call-ID: 1823801e5843dcee4710728823504ac3 at 10.1.1.68
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 30 Nov 2007 09:39:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
<--- SIP read from 66.237.65.67:5060 --->
SIP/2.0 501 Not Implemented
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK1e127475;rport=5060
From: asterisk <sip:asterisk at 10.1.1.68>;tag=as5bbf1bfd
To: <sip:65.175.129.149>
Call-ID: 1823801e5843dcee4710728823504ac3 at 10.1.1.68
CSeq: 102 OPTIONS
Server: Sippy
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '1823801e5843dcee4710728823504ac3 at 10.1.1.68'
Method: OPTIONS
======================================================================
----------------------------------------------------------------------
naveenpalani - 02-06-08 07:23
----------------------------------------------------------------------
Yes, i could resolve the issue by replacing the firewall setup. I was
initially using the Cisco Pix firewall which was not allowing me to get the
200 ok signal. Now replaced it with sonicwall firewall, could now get the
200 ok from the sip provider.
It was primary the natting issues with Cisco Pix.
Thanks for all your support.
Issue History
Date Modified Username Field Change
======================================================================
02-06-08 07:23 naveenpalani Note Added: 0081783
======================================================================
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