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Summary:ASTERISK-10936: Dailstatus says NOANSWER even if i pick the call
Reporter:Naveen (naveenpalani)Labels:
Date Opened:2007-11-30 04:12:43.000-0600Date Closed:2011-06-07 14:03:16
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_oss
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) extensions.txt
( 1) sipdebug.txt
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@10.1.1.68' Method: REGISTER
   -- Attempting call on Local/outbound@dialout for outbound-handler@dialout:1 (Retry 1)
   -- Executing [outbound@dialout:1] Answer("Local/outbound@dialout-e3ed,2", "") in new stack
   -- Executing [outbound@dialout:2] Wait("Local/outbound@dialout-e3ed,2", "30") in new stack
   -- Executing [outbound-handler@dialout:1] Dial("Local/outbound@dialout-e3ed,1", "SIP/011919960466622@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@65.175.129.149 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK44e0a2fb;rport
From: "Linux" <sip:2068200001@proxy2.bandtel.com>;tag=as0f8dfdfa
To: <sip:011919960466622@65.175.129.149>
Contact: <sip:2068200001@10.1.1.68>
Call-ID: 3c4ec9614a41dea303fd8a57192cb180@proxy2.bandtel.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Linux" <sip:Linux@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@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@proxy2.bandtel.com>;tag=as0f8dfdfa
To: <sip:011919960466622@65.175.129.149>
Call-ID: 3c4ec9614a41dea303fd8a57192cb180@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@proxy2.bandtel.com>;tag=as0f8dfdfa
To: <sip:011919960466622@65.175.129.149>
Call-ID: 3c4ec9614a41dea303fd8a57192cb180@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@65.175.129.149 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK44e0a2fb;rport
From: "Linux" <sip:2068200001@proxy2.bandtel.com>;tag=as0f8dfdfa
To: <sip:011919960466622@65.175.129.149>
Contact: <sip:2068200001@10.1.1.68>
Call-ID: 3c4ec9614a41dea303fd8a57192cb180@proxy2.bandtel.com
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Linux" <sip:Linux@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@65.175.129.149 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK10fd7dab;rport
From: "Linux" <sip:2068200001@proxy2.bandtel.com>;tag=as0f8dfdfa
To: <sip:011919960466622@65.175.129.149>
Contact: <sip:2068200001@10.1.1.68>
Call-ID: 3c4ec9614a41dea303fd8a57192cb180@proxy2.bandtel.com
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Linux" <sip:Linux@proxy2.bandtel.com>;privacy=off;screen=no
Authorization: Digest username="2068200001", realm="66.237.65.67", algorithm=MD5, uri="sip:011919960466622@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@proxy2.bandtel.com>;tag=as0f8dfdfa
To: <sip:011919960466622@65.175.129.149>
Call-ID: 3c4ec9614a41dea303fd8a57192cb180@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@10.1.1.68>;tag=as013c2cf1
To: <sip:65.175.129.149>
Contact: <sip:asterisk@10.1.1.68>
Call-ID: 0fd272674df0d814509669360caf1f25@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@10.1.1.68>;tag=as013c2cf1
To: <sip:65.175.129.149>
Call-ID: 0fd272674df0d814509669360caf1f25@10.1.1.68
CSeq: 102 OPTIONS
Server: Sippy


<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '0fd272674df0d814509669360caf1f25@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@proxy2.bandtel.com>;tag=as0f8dfdfa
To: <sip:011919960466622@65.175.129.149>;tag=16907dc5b05da4b49d991e769bce1862
Call-ID: 3c4ec9614a41dea303fd8a57192cb180@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@dialout-e3ed,1
   -- Executing [outbound@dialout:3] NoOp("Local/outbound@dialout-e3ed,2", "status=") in new stack
   -- Executing [outbound@dialout:4] AGI("Local/outbound@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@dialout:5] Hangup("Local/outbound@dialout-e3ed,2", "") in new stack
 == Spawn extension (dialout, outbound, 5) exited non-zero on 'Local/outbound@dialout-e3ed,2'
Scheduling destruction of SIP dialog '3c4ec9614a41dea303fd8a57192cb180@proxy2.bandtel.com' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to 66.237.65.67:5060:
CANCEL sip:011919960466622@65.175.129.149 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK10fd7dab;rport
From: "Linux" <sip:2068200001@proxy2.bandtel.com>;tag=as0f8dfdfa
To: <sip:011919960466622@65.175.129.149>
Call-ID: 3c4ec9614a41dea303fd8a57192cb180@proxy2.bandtel.com
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Linux" <sip:Linux@proxy2.bandtel.com>;privacy=off;screen=no
Content-Length: 0


---
Scheduling destruction of SIP dialog '3c4ec9614a41dea303fd8a57192cb180@proxy2.bandtel.com' in 6400 ms (Method: INVITE)
 == Spawn extension (dialout, outbound-handler, 1) exited non-zero on 'Local/outbound@dialout-e3ed,1'
[Nov 30 03:37:23] NOTICE[32020]: pbx_spool.c:351 attempt_thread: Call completed to Local/outbound@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@proxy2.bandtel.com>;tag=as0f8dfdfa
To: <sip:011919960466622@65.175.129.149>;tag=52c7b1d5444c5b44ef4d77f6a6c80dc0-24c4
Call-ID: 3c4ec9614a41dea303fd8a57192cb180@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@proxy2.bandtel.com' Method: INVITE

<--- SIP read from 66.237.65.67:5060 --->
BYE sip:2068200001@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@65.175.129.149>;tag=16907dc5b05da4b49d991e769bce1862
To: Linux <sip:2068200001@proxy2.bandtel.com>;tag=as0f8dfdfa
Call-ID: 3c4ec9614a41dea303fd8a57192cb180@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@65.175.129.149>;tag=16907dc5b05da4b49d991e769bce1862
To: Linux <sip:2068200001@proxy2.bandtel.com>;tag=as0f8dfdfa
Call-ID: 3c4ec9614a41dea303fd8a57192cb180@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@10.1.1.68>;tag=as752b1d75
To: <sip:65.175.129.149>
Contact: <sip:asterisk@10.1.1.68>
Call-ID: 3b7f41405076bfdd00edf8807cdfb387@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@10.1.1.68>;tag=as752b1d75
To: <sip:65.175.129.149>
Call-ID: 3b7f41405076bfdd00edf8807cdfb387@10.1.1.68
CSeq: 102 OPTIONS
Server: Sippy


<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '3b7f41405076bfdd00edf8807cdfb387@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@10.1.1.68>;tag=as5bbf1bfd
To: <sip:65.175.129.149>
Contact: <sip:asterisk@10.1.1.68>
Call-ID: 1823801e5843dcee4710728823504ac3@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@10.1.1.68>;tag=as5bbf1bfd
To: <sip:65.175.129.149>
Call-ID: 1823801e5843dcee4710728823504ac3@10.1.1.68
CSeq: 102 OPTIONS
Server: Sippy


<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '1823801e5843dcee4710728823504ac3@10.1.1.68' Method: OPTIONS
Comments:By: Joshua C. Colp (jcolp) 2007-11-30 08:24:19.000-0600

Please remove the Answer from your dialout context and see if that fixes your issue. The call is being prematurely answered and the other dialplan logic is being executed, which includes Hangup.
By: Naveen (naveenpalani) 2007-11-30 08:53:52.000-0600

If i remove Answer command from dialout context, the control does not switch to outbound-handler extension and execute the dial command. So i used it as Answer() (with the brackets) which now dials me the number and once i pick the call the dialstatus gives me as "NOANSWER" in the outbound-handler extension.Does not go to outbound extension again.

Gives the message as:

Nobody picked up in 120000 ms

Attached is my extensions.txt file with my dialplan.


By: Joshua C. Colp (jcolp) 2007-11-30 09:01:11.000-0600

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.
By: Naveen (naveenpalani) 2007-11-30 09:14:27.000-0600

Well our sip provider says that we are not responding to thier 200 ok on answer. How can we respond to it.

yes it waits for the 30 secs and only then executes the other commands.
By: Joshua C. Colp (jcolp) 2007-11-30 09:17:41.000-0600

There is no 200 OK from the provider in the sip debug provided, it was never received.
By: Naveen (naveenpalani) 2007-11-30 09:45:05.000-0600

Please check on my sip debug attached. I made few changes to the sip.conf file. Notice whether the 200 ok is now recieved from my sip provider because they say that we are not responding to their 200 ok on answer.

file attached - sipdebug.txt

Can you provide me a sample outbound call dialplan to check on my server.
By: Joshua C. Colp (jcolp) 2007-11-30 09:52:01.000-0600

I still see no 200 OK back from them and it looks fine. As for a sample... try calling SIP/1000@neutrino.joshua-colp.com - it should answered and play back the demo congratulations audio.
By: Naveen (naveenpalani) 2007-11-30 10:24:36.000-0600

It works great with the sample you gave me. How can i recieve the 200 ok from my sip provider. Do i need to do any configuration in my sip.conf file.
By: Joshua C. Colp (jcolp) 2007-11-30 10:29:22.000-0600

No - there is no extra configuration and in fact you are receiving other messages from them fine. Confirm with ethereal/tcpdump that the 200 OK is being received by your machine... if not there's nothing you can do, it's not getting to you and it is on their side.
By: Naveen (naveenpalani) 2007-11-30 12:46:01.000-0600

Iam new to linux, installed the tcpdump on my machine and its running from a long time. Could you let me know how can i confirm in tcpdump if i have received 200 ok or not. Do we have any specific command to do that and where can i find the results.
By: Naveen (naveenpalani) 2007-12-01 09:56:06.000-0600

Also, Is there something we need to do after completing the registration?. Like complete the handshake with the acknowledge from our side?.

Also will there be any issues if dont have the SIP providers DNS details in our host file?.

Appreciate your help on this.
By: Joshua C. Colp (jcolp) 2008-01-08 14:27:23.000-0600

No, there is nothing else you need to do. As for tcpdump:

tcpdump port 5060

It will show all traffic on port 5060 (SIP). Try to place a call and see if a 200 OK comes back from the provider.
By: jmls (jmls) 2008-02-06 04:18:32.000-0600

naveenpalani, is this still an issue ? If so, did you do what file last suggested ?
By: Naveen (naveenpalani) 2008-02-06 07:23:43.000-0600

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.
By: jmls (jmls) 2008-02-06 09:06:38.000-0600

configuration issue