[asterisk-bugs] [Asterisk 0011425]: Dailstatus says NOANSWER even if i pick the call

noreply at bugs.digium.com noreply at bugs.digium.com
Fri Nov 30 09:01:12 CST 2007 

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:              11-30-2007 09:01 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
====================================================================== 

---------------------------------------------------------------------- 
 file - 11-30-07 09:01  
---------------------------------------------------------------------- 
Interesting... well there is no 200 OK coming back from your SIP provider
to indicate the call was actually answered. The only message we get back
before cancelling the attempt is a 183 Session Progress to indicate the
call is progressing. Does it actually wait for 30 seconds before moving on
to those other dialplan steps? If so I don't know what to tell you,
Asterisk can't consider something answered until the other side says so. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
11-30-07 09:01  file           Note Added: 0074597                          
======================================================================

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