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Summary:ASTERISK-02168: Asterisk does not hang up SIP call
Reporter:goofer22 (goofer22)Labels:
Date Opened:2004-08-01 05:11:33Date Closed:2004-09-25 02:12:16
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) sipdebug.txt
( 1) sipdebug2.txt
Description:I have an ISDN telephone connected to a HFC ISDN card on an asterisk server and using an internal SIP Client X-Lite connected to asterisk. I am using the German SIP provider Sipgate.de. The sip commands show that I am registered properly with Sipgate.

My problem is that when I want to call (using ISDN phone or internal SIP client) via the Sip provider a real phone number (ISDN phone or internal SIP >> Asterisk >> SIP ), I get a ring tone. When I now decide to hang up (e.g. if nobody answers), the called telephone continues to ring almost forever. This error is reproducable everytime. The following error message shows up since the newest CVS Version:
"app_dial.c : 362 wait_for_answer: Unable to forward frame"

If the other party answers and I am the first one to hang up, the call sometimes does not get cancelled as well. The called party has to hang up first to really disconnect the call. This error is not yet reproducable, as I said, sometimes it works and asterisk hangs up correctly.

SIP.conf:

[general]
port = 5060      
bindaddr = 0.0.0.0
context = psin    
tos=lowdelay
maxexpirey=3600  
defaultexpirey=120
disallow=all              
allow=alaw
allow=ulaw
allow=g726
allow=gsm
allow=ilbc
musicclass=random
externip = xxx.dyndns.org
localnet=192.168.0.0/255.255.255.0
language=de
register => 8888888:aaaaaa@sipgate.de/2001

[sipgate]
type = friend
username = 8888888
canreinvite=no
secret = aaaaaaa
host = sipgate.de
context=psin
fromuser = 8888888
fromdomain = sipgate.de
nat = no
qualify = yes
insecure=very

..some client definitions follow...

Extensions.conf:
exten => _9.,1,Wait(1)
exten => _9.,2,Answer
exten => _9.,3,Dial(SIP/${EXTEN:1}@sipgate,20)
exten => _9.,4,Hangup

exten => h,1,Hangup

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

When using the local SIP client using X-Lite directly with sipgate instead thru asterisk, everything works perfectly. So this does not seem to be a sipgate issue.
Comments:By: twisted (twisted) 2004-08-01 05:17:31

please provide a more detailed sip debug....   perhaps from the start of the call until after the call has been hung up.... with your trace, it looks like the normal invite + registration stuff going on here..
By: goofer22 (goofer22) 2004-08-01 05:32:37

I hope, sipdebug2.txt is better
By: Mark Spencer (markster) 2004-08-01 10:34:13

This bug has already been fixed in CVS as of just a few days ago, please update to latest CVS head and try again.
By: goofer22 (goofer22) 2004-08-01 11:23:03

I tried it with CVS Date of July 26th as this is latest date the bri-stuff download script provides. I'll try again with today's date. Hopefully the bri-stuff patch still works
By: Mark Spencer (markster) 2004-08-01 14:19:52

I'm closing this one out since it has already been fixed but feel free to reopen if your problem isn't solved.