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Summary:ASTERISK-17077: SIP response don't worked $(HASH(SIP_CAUSE,<slave-channel-name>)}
Reporter:dogonovmax (dogonovmax)Labels:
Date Opened:2010-12-08 06:16:12.000-0600Date Closed:2011-06-07 14:05:23
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/CallCompletionSupplementaryServices
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:SIP response in asterisk 1.8.0 don't worked for me!!!!


****** ADDITIONAL INFORMATION ******

SIP response in asterisk 1.8.0 don't worked for me!!!!

extension.conf

[default]

exten => _X.,1,Dial(SIP/212.33.19.20/123)
exten => _X.,n,Hangup(${HASH(SIP_CAUSE,<channel-name>)})
exten => _X.,n,NoOP(${HASH(SIP_CAUSE,<channel-name>)})


Asterisk logs


<--- Transmitting (NAT) to 217.23.69.182:1072 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.23.69.182:1069;branch=z9hG4bK-d8754z-8889c2da84cde9d2-1---d8754z-;received=217.23.69.182;rport=1072
From: <sip:xlite1@91.201.21.181;transport=UDP>;tag=74196722
To: <sip:1234@91.201.21.181;transport=UDP>
Call-ID: OWJiNTEyNmY0ZTE3YTUzMjUxOTI1NTZjMzM0ZDdiNTU.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1234@91.201.21.181:5060>
Content-Length: 0


<------------>
   -- Executing [1234@default:1] Dial("SIP/xlite1-00000012", "SIP/212.33.19.20/123") in new stack
 == Using SIP RTP CoS mark 5
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x800000000000 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 212.33.19.20:5060:
INVITE sip:123@212.33.19.20 SIP/2.0
Via: SIP/2.0/UDP 91.201.21.181:5060;branch=z9hG4bK689bc2f6
Max-Forwards: 70
From: "xlite1" <sip:xlite1@91.201.21.181>;tag=as07c2ff4a
To: <sip:123@212.33.19.20>
Contact: <sip:xlite1@91.201.21.181:5060>
Call-ID: 5d2bb0b60537a9d530548c0210592a29@91.201.21.181:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.0
Date: Wed, 08 Dec 2010 11:26:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 310

v=0
o=root 1351194758 1351194758 IN IP4 91.201.21.181
s=Asterisk PBX 1.8.0
c=IN IP4 91.201.21.181
t=0 0
m=audio 13276 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
   -- Called 212.33.19.20/123

<--- SIP read from UDP:212.33.19.20:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 91.201.21.181:5060;branch=z9hG4bK689bc2f6;received=91.201.21.181
From: "xlite1" <sip:xlite1@91.201.21.181>;tag=as07c2ff4a
To: <sip:123@212.33.19.20>;tag=as54a57c33
Call-ID: 5d2bb0b60537a9d530548c0210592a29@91.201.21.181:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 212.33.19.20:5060:
ACK sip:123@212.33.19.20 SIP/2.0
Via: SIP/2.0/UDP 91.201.21.181:5060;branch=z9hG4bK689bc2f6
Max-Forwards: 70
From: "xlite1" <sip:xlite1@91.201.21.181>;tag=as07c2ff4a
To: <sip:123@212.33.19.20>;tag=as54a57c33
Contact: <sip:xlite1@91.201.21.181:5060>
Call-ID: 5d2bb0b60537a9d530548c0210592a29@91.201.21.181:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.0
Content-Length: 0


---
   -- SIP/212.33.19.20-00000013 is circuit-busy
 == Everyone is busy/congested at this time (1:0/1/0)
   -- Executing [1234@default:2] Hangup("SIP/xlite1-00000012", "") in new stack
 == Spawn extension (default, 1234, 2) exited non-zero on 'SIP/xlite1-00000012'
   -- Executing [h@default:1] NoOp("SIP/xlite1-00000012", "") in new stack
Scheduling destruction of SIP dialog 'OWJiNTEyNmY0ZTE3YTUzMjUxOTI1NTZjMzM0ZDdiNTU.' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (NAT) to 217.23.69.182:1072 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 217.23.69.182:1069;branch=z9hG4bK-d8754z-8889c2da84cde9d2-1---d8754z-;received=217.23.69.182;rport=1072
From: <sip:xlite1@91.201.21.181;transport=UDP>;tag=74196722
To: <sip:1234@91.201.21.181;transport=UDP>;tag=as358b0db1
Call-ID: OWJiNTEyNmY0ZTE3YTUzMjUxOTI1NTZjMzM0ZDdiNTU.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog '5d2bb0b60537a9d530548c0210592a29@91.201.21.181:5060' Method: INVITE

<--- SIP read from UDP:217.23.69.182:1072 --->
ACK sip:1234@91.201.21.181;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 217.23.69.182:1069;branch=z9hG4bK-d8754z-8889c2da84cde9d2-1---d8754z-;rport
Max-Forwards: 70
To: <sip:1234@91.201.21.181;transport=UDP>;tag=as358b0db1
From: <sip:xlite1@91.201.21.181;transport=UDP>;tag=74196722
Call-ID: OWJiNTEyNmY0ZTE3YTUzMjUxOTI1NTZjMzM0ZDdiNTU.
CSeq: 2 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:217.23.69.182:1072 --->


<------------->
Really destroying SIP dialog 'MDEyNmEzMjQ0MmMxZjMzZTYzNjJlNTE1Mjk5OTZlZDQ.' Method: REGISTER
Really destroying SIP dialog 'YmQ3MjU1YjQ1OWNkNWU3Mjk2ZmRlMGNlNmU4YWViOGE.' Method: REGISTER
Really destroying SIP dialog 'OGNlMjQ4Mjg2NzIwOThmZDFjZTNhMmQ4MWE2NDA4NTI.' Method: REGISTER
Really destroying SIP dialog 'OWJiNTEyNmY0ZTE3YTUzMjUxOTI1NTZjMzM0ZDdiNTU.' Method: ACK
s13*CLI>
s13*CLI>
s13*CLI>
s13*CLI>
s13*CLI>
s13*CLI> core show version
Asterisk 1.8.0 built by root @ s13 on a i686 running Linux on 2010-12-08 09:24:32 UTC

<--- SIP read from UDP:217.23.69.182:1072 --->

Comments:By: Leif Madsen (lmadsen) 2010-12-16 09:26:36.000-0600

Looks like a support issue to me. Incorrect usage of HASH() or a dialplan that wasn't reloaded.

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