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Summary:ASTERISK-20352: Incorrect handling of SIP 484 Address incomplete
Reporter:Alain RICHARD (arichard)Labels:
Date Opened:2012-09-03 05:13:11Date Closed:2012-09-04 09:34:24
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:1.8.15.0 1.8.15.1 Frequency of
Occurrence		Constant
Related
Issues:
Environment:Linux with asterisk 1.8.15.0 SIP provider trunk SIP consummer trunkAttachments:( 0) ASTERISK-20352_log
Description:In my setup, asterisk is used only as a SIP router between a provider and a consummer.

On an incoming call (from the provider to the consumer), if the consumer trunk return a SIP 484 Address incomplete (q.850 code 28), the error is forwarded by asterisk to the provider trunk as a SIP 503 Service unavail, although the included q.850 code is 28.

This is a violation of RFC 3398 :

  Response received                     Cause value in the REL
  -----------------                     ----------------------
...
  484 Address incomplete                28 Invalid Number Format (+)
...


The problem is 100% reproductible. With the following code :

sip.conf :

[general]
use_q850_reason=yes

[provider]
context=fromprovider

[consumer]


extensions.Conf :

[fromprovider]

exten => _123456,1,Nop(Call the consumer)
   same => n,Dial(SIP/${EXTEN}@comsumer)
   same => n,Verbose(2,DIALSTATUS = ${DIALSTATUS})
   same => n,Verbose(2,HANGUPCAUSE = ${HANGUPCAUSE})


When the call is coming, I get the following process :

1 Call is ringging from provider to consumer
2 Consumer returns a SIP 484 Address incomplete
3 The verbose output shows : DIALSTATUS=CHANUNAVAIL, HANGUPCAUSE=28
4 asterisk returns to the provider : 503 Service unavailable, q.850 code=28

So here it seams to be at least two error :

- On step 4, the returned code for HANGUPCAUSE 28 to the provider is SIP 503 instead of SIP 484 as requested by RFC 3398.
- On step 3, the DIAL app should have returned DIALSTATUS=CONGESTION instead of CHANUNAVAIL


I have succeed to implement a counter measure :

exten => _123456,1,Nop(Call the consumer)
   same => n,Dial(SIP/${EXTEN}@comsumer)
   same => n,Verbose(2,DIALSTATUS = ${DIALSTATUS})
   same => n,GotoIf($[${HANGUPCAUSE}=28] & $[${DIALSTATUS}!="CONGESTION"]?setcongestion:continue)
   same => n(setcongestion),Set(DIALSTATUS="CONGESTION")
   same => n,Verbose(2,DIALSTATUS = ${DIALSTATUS})
   same => n(continue),Verbose(2,HANGUPCAUSE = ${HANGUPCAUSE})

In that case, the provider will receive the correct SIP response (SIP 484 and q.850 code 28).


I don't know if the problem is within dial_app, chan_sip or on the generic handling of HANGUP CAUSE 28, but this violate the RFCs.

Regards,

Alain RICHARD

Comments:By: David Woolley (davidw) 2012-09-03 05:58:04.886-0500

The response you get depends on whether the dialplan allows more digits to be entered.  I think you need to follow this up further through support channels before declaring a bug.
By: Alain RICHARD (arichard) 2012-09-03 07:32:38.395-0500

Here asterisk is just used as a SIP router, and just forwards the reply from the customer PABX (in fact a patton gateway) to the provider trunk.

It is not supposed to interpret the numbers.

In fact the PABX generates a 484 reply because the SDA is not implemented on the PABX, but this is not the subject.

The patton gateway generate an RFC3398 compliant result :

<--- SIP read from UDP:3.3.3.251:5060 --->
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK4acf49d4;rport=5060;received=2.2.2.2
From: "987654" <sip:987654@2.2.2.2>;tag=as2eff6979
To: <sip:123456@3.3.3.251>;tag=3169669864
Call-ID: 27f7d0e862a52b933eeaf4c357b041bd@2.2.2.2:5060
CSeq: 102 INVITE
Server: Patton SN4638 5BIS 00A0BA06BAEF R6.1 2012-05-09 H323 SIP BRI M5T SIP Stack/4.0.30.30
Content-Length: 0


and asterisk change it to a non RFC3398 compliant result :

<--- Reliably Transmitting (NAT) to 1.1.1.104:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 1.1.1.104;branch=z9hG4bK1166.4a4d19f6.1;received=1.1.1.104;rport=5060
Via: SIP/2.0/UDP 1.1.1.116:5060;rport=5060;branch=z9hG4bK-77ef-1346673299-2623-361
From: "987654"<sip:987654@1.1.1.116>;tag=95ffcd055e0f78f7d5d397020e89288d9808247e
To: <sip:123456@1.1.1.104:5060>;tag=as4447fef1
Call-ID: 2c00-4b2-832012115459-ACKBAR-0-1.1.1.116
CSeq: 1 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Reason: Q.850;cause=28
X-Asterisk-HangupCause: Invalid number format
X-Asterisk-HangupCauseCode: 28
Content-Length: 0


You'll find enclosed the full log of the exchange.

By: Alain RICHARD (arichard) 2012-09-03 07:33:13.688-0500

Full log of the SIP problem.
By: David Woolley (davidw) 2012-09-04 05:31:35.952-0500

Asterisk is NOT a SIP router (i.e. proxy).  It is a back to back user agent, and terminates the protocol on each leg.
By: Matt Jordan (mjordan) 2012-09-04 09:34:17.375-0500

Exactly what David said.

You can relay a 484 back to the original UA by inspecting the HANGUPCAUSE and by careful usage of the Incomplete application.
By: Alain RICHARD (arichard) 2012-09-05 08:20:37.656-0500

Hi,

I have investigated a little bit more the problem and it seams it has nothing to do with Address incompletes or a bad usage of asterisk as a SIP router.

I have successfully reproduced the problem with the following scenario :

- asterisk is used as an IPBX
- phone1 is a SIP phone
- provider is a SIP provider

Process :

1 - phone1 calls 00444518779. The first 0 is for line access.
2 - asterisk send this to the SIP provider (Dial(SIP/${EXTEN:1}@provider)
3 - The provider tries to reach the called number, but this one is non existant and the provider return SIP 404 Not Found, q850 cause=1
4 - Asterisk returns to phone 1 a SIP/2.0 503 Service Unavailable; q850 cause=1

I think this is a perfect usage of asterisk as a PABX, and asterisk generates a SIP reply that is not compliant with RFC3398 :

SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 195.189.65.250:51023;branch=z9hG4bK-d8754z-b84080c4add209f5-1---d8754z-;received=192.168.240.34;rport=51023
From: "0000FFFF0002"<sip:0000FFFF0002@192.168.240.251;transport=UDP>;tag=eaa14d45
To: <sip:00444518779@192.168.240.251;transport=UDP>;tag=as10aad6d8
Call-ID: MTUzZjVlNGQ1YTNlNmU1OTU1ZGQ4YzE4Y2IxZWJlMTI.
CSeq: 2 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Reason: Q.850;cause=1
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1
Content-Length: 0


I am not accustomized with the asterisk developpment process, so please if this is not the right place to discuss this issue, on which list should i submit this ?

Regards,