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Summary:ASTERISK-03395: sip conversation falls after 17 seconds...
Reporter:Francesco Facco de Lagarda (fdl333)Labels:
Date Opened:2005-01-31 10:10:14.000-0600Date Closed:2011-06-07 14:10:02
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) asterisk-call-failed.rar
( 1) asterisk-call-failed.txt
Description:I use Univoice Italy as a SIP Provider.. everything works PERFECTLY with v.1.0.3 ... I compiled 1.0.5 and all outgoing calls fall after approx 17 seconds.. I recompiled 1.0.3 and all is fine again...

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

here is the relative section in the sip.conf file:

I have substituted the real password with "password"...

register=> 390240039504:password@sip.uni.it/390240039504

[uni2]
username=390240039504
type=peer
secret=password
realm=sip.uni.it
nat=yes
insecure=very
host=sip.uni.it
fromdomain=sip.uni.it
context=uni-in
canreinvite=no

in extensions.conf the outgoing is handled as follows:


exten => _9.,1,SetGroup(${CALLERIDNUM})
exten => _9.,2,Dial(SIP/uni2/${EXTEN:1},,)
exten => _9.,3,Congestion
exten => _9.,103,Macro(outisbusy)

Comments:By: Andrey S Pankov (casper) 2005-01-31 10:28:58.000-0600

You need to capture sip debug for both calls and upload them here as per bug posting guidelines.
By: Francesco Facco de Lagarda (fdl333) 2005-01-31 14:32:22.000-0600

I use sip debug extensively, but HOW do I capture it to file?
(In this case I would use "sip debug peer uni2"..)
By: Brian West (bkw918) 2005-01-31 15:26:44.000-0600

you copy and paste it.

bkw
By: Brian West (bkw918) 2005-01-31 15:27:37.000-0600

Not major and when did realm become something that could be done per peer?  I thought it was only in the general section?

bkw
By: Clod Patry (junky) 2005-01-31 22:03:50.000-0600

ya can capture it via:
linux# script
linux# asterisk -rvvvv
[blah]
asterisk CLI> exit
linux# exit

and then, cat typescript.
By: Mark Spencer (markster) 2005-02-02 14:22:50.000-0600

Also, can you please try latest CVS and see if the problem occurs in CVS head?
By: Francesco Facco de Lagarda (fdl333) 2005-02-04 12:18:29.000-0600

Here's the transcript you asked: it's a call from iax ext. 205 (i think) to 0102543967 via sip provider uni-voice..

the call fell on the POTS side after 17 secs, but the iax side gave no indication...
By: Mark Spencer (markster) 2005-02-04 14:33:52.000-0600

Again, as per the bug guidelines, please attach a *text file* of the debug resulting from "sip debug" on Asterisk.  Also please see if latest CVS fixes the problem, as there was an earlier bug fixed that may be relevant.
By: Francesco Facco de Lagarda (fdl333) 2005-02-05 03:20:14.000-0600

1) yes, this is the latest CVS - Downloaded yesterday.
2) I rared the file 'cause it's large
3) Here it is again, unrared..
By: Mark Spencer (markster) 2005-02-05 08:58:10.000-0600

NVITE sip:0102543967@sip.uni.it SIP/2.0
Via: SIP/2.0/UDP 10.0.0.99:5060;branch=z9hG4bK694fec49;rport
From: "205" <sip:205@sip.uni.it>;tag=as1cf84adc
To: <sip:0102543967@sip.uni.it>
Contact: <sip:205@10.0.0.99>
Call-ID: 23af95bf6f065e6a691bccc50811d510@sip.uni.it
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 04 Feb 2005 17:54:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 234

The presense of the incorrect Via line could suggest a configuration problem (e.g. you are behind a NAT and are not using externip), further the debug shows that while we do send ACK, they continue to send 200 OK suggesting they are not receiving the ACK we are transmitting. Lastly, because you have "nat=yes" for this peer, you are transmitting to the original 5060 address and *not* the new 5061 address that they have requested further communication should be sent to.

Try disabling NAT mode for this peer and try turning on a proper externip and let us know if that makes it work.
By: Francesco Facco de Lagarda (fdl333) 2005-02-05 09:39:27.000-0600

Fine, thanks...Can you please give me precise instructions on how to do that..

Please give me ANY help you can on this peer config, as it took 3 days of trial and error to get it to work! Funny thing that on the incomming calls, I can't use wildcard matching "_.,"

In my sip-additional.conf I have:

[390240039504]
username=390240039504
type=user
realm=sip.uni.it
insecure=very
host=sip.uni.it
fromuser=390240039504
fromdomain=sip.uni.it
context=uni-in

[uni2]
username=390240039504
type=peer
secret=password
realm=sip.uni.it
nat=yes
insecure=very
host=sip.uni.it
fromdomain=sip.uni.it
context=uni-in
canreinvite=no


and in my extensions.conf I have:

[uni-in]
include => ext-did
exten => 390240039504,1,SetVar(intype=${INCOMING})
exten => 390240039504,2,Cut(intype=intype,-,1)
exten => 390240039504,3,GotoIf($[${intype} = EXT]?4:5)
exten => 390240039504,4,Goto(ext-local,${INCOMING:4},1)
exten => 390240039504,5,GotoIf($[${intype} = GRP]?6:7)
exten => 390240039504,6,Goto(ext-group,${INCOMING:4},1)
exten => 390240039504,7,Answer
exten => 390240039504,8,Wait(1)
exten => 390240039504,9,Goto(${INCOMING},s,1)
exten => fax,1,Goto(ext-fax,in_fax,1)
exten => h,1,Hangup

'cause exten => _.,1, etc.. would NOT work!
By: Francesco Facco de Lagarda (fdl333) 2005-02-05 09:44:35.000-0600

Sorry, One more point, guys..
I just installed an E1 card on my machine, and the telco's enabling the PRI any day now (at the moment the channels aren't open their end yet).. They were unable to give me details regarding setup parameters. the notes on the zapata conf say:

# The framing is one of "cas" or "ccs" for E1
# The coding is one of "ami" or "hdb3" for E1
# E1's may have the additional keyword "crc4" to enable CRC4 checking

What software can I use to "debug" the ISDN PRI to understand If my parameters are ok (educated guesses say ccs,hdb3 for Italy).. anyway it's "EuroISDN"..

The CLI only give me "zap show channels", and "zap show channel xx", and
the card just has two leds on the back (but no one can tell me what they should do) At the moment one is off, and one is flashing red (to be expected, the telco hasn't set up the channels their end yet)..
By: Mark Spencer (markster) 2005-02-05 10:31:06.000-0600

This is not the place for technical support.  At this point your issues are only configuration issues.  You may use the lists or call Digium for technical support for your E1 card.
By: Francesco Facco de Lagarda (fdl333) 2005-02-06 05:16:50.000-0600

"Try disabling NAT mode for this peer and try turning on a proper externip and let us know if that makes it work."

I haven't done it yet, so HOW can you say that the bug is resloved? The fact remains that with v.1.0.3 the calls do NOT fall after 17 secs whereas both with v1.0.5 and with the latest CSV they DO.

I asked you for precise instructions on how to "..diable nat and turning on a proper extern ip.." and I'm still waiting. I also posted my complete config files to see if anything else could cause the problem!
By: Mark Spencer (markster) 2005-02-06 16:56:40.000-0600

The bug tracker is not the place to get questions like "how do I turn off nat" or "how do I configure externip" answered.  Perhaps by comparing the traces of the 1.0.3 SIP debug to the CVS head SIP debug you can determine more clearly what is different between the two and thus come up with a more specific, meaningful bug report.

I have given you already an analysis of your SIP trace indicating that your selection of nat=yes is preventing Asterisk from sending the SIP packets to the requested position so until that is resolved, this appears to be a configuration issue.  We cannot use the bug tracker to provide you with technical support, so please do not reopen this bug until you have solved the technical support portion of your issue and at *least* can attach a broken trace with nat=no and the working trace from whatever version of asterisk last worked for you.