After 19sec phone hangs up


#1

Hi,

I have setup Asterisk for internal and external phone-calls. The internal works well. The external calls hangs up after approx 20 secs. I am using a VoIP Gateway for external calls. For right now I test it with X-Lite (softphone). There is no firewall between the external VoIP Gateway and the Asterisk Server.

My sip.conf Config:
[general]
port = 5060
bindaddr = 0.0.0.0
context = context
disallow=gsm
allow=ulaw
allow=alaw
register => XXXX:XXXX@sip.XXXXX.de/
;
[uuuu]
type=peer
username=uuuu
secret=XXXX
user=phone
host=sip.XXXXXX.de
fromuser=XXXX
authname=XXXX
fromdomain=sip.XXXXX.de
nat=no
canreinvite=no
insecure=very
dtmfmode=inband
dtmf=inband
context=context
qualify=200
;
[basty]
type=friend
username=basty
secret=XXXX
host=dynamic
;dtmfmode=rfc2833
callerid=“Bleh Blah” <6666>
qualify=200
insecure=yes

extensions.conf

[default]
exten => ,1,Dial(SIP/basty,60)
exten => ,2,Congestion
exten => ,102,Busy
;
[context]
include => default
exten => _0.,1,Dial(SIP/${EXTEN:1}@sip.XXXXX.de,60)
exten => _0.,2,Congestion
exten => _0.,102,Busy
;
exten => 6666,1,Dial(SIP/basty,60)
exten => 6666,2,Congestion
exten => 6666,102,Busy

Anyone has an idea ?

Thanks!

greetings
basty


#2

what does your asterisk console say as it hangs up ?


#3

Hi,

all I recive is:

== Spawn extension (context, 00XX1XXXXX21, 1) exited non-zero on ‘SIP/basty-1bb3’

even a tcpdump on the xl0 interface on src of the VoIP Gateway doesnt tell anything.


#4

Hi,

all I recive is:

== Spawn extension (context, 00XX1XXXXX21, 1) exited non-zero on ‘SIP/basty-1bb3’

even a tcpdump on the xl0 interface on src of the VoIP Gateway doesnt tell anything.[/quote]

I found out, that when I put “canreinvite=yes” into my sip.conf file, the phone call is open longer than 20sec. Only problem is, that I cant hear the remote side. Anyone got an idea ?


#5

I have the same problem, but I can perhaps give more info that might help both of us… !

You make a call, and you get “fast busy” after 20 seconds on the call? Well, is that 20 seconds of “call time”… AFTER dialing? I timed it and found that it took 10 seconds to MAKE the call + 20 seconds before the call dropped. 30 seconds.

I have SIP phones with a timer; if I make a call and the timer starts as soon as the other end “picks up” - then Asterisk acknowledges the call - and the line stays up. It’s when Asterisk does NOT realize the other end has picked up - and that’s when it “gives up” after 30 seconds… which is the same as picking up the phone, dialing nothing, listening to dialtone before it “times out” and plays fast busy.

I don’t have it with SIP - JUST with SIP to ZAP. With SIP, I turned on “qualify=yes” in each SIP extension and it fixed it. With ZAP… no solution yet.

Any help? Does that sound similar?