[Asterisk-Users] SIP connections do not hang up

Fri Jul 30 13:58:51 MST 2004

Hi everybody,

I have strange problem:

I'm calling from inside (either X-Lite using SIP channel or a ISDN telephone
using Zap Channel) using sipgate to a number in public network.
When I'm hanging up before the other person picked up the phone, the line is
not closed correctly.
The phone keeps on ringing until timeout (of Sipgate I assume) and it even
costs my money, if the other person picks up the ringing phone, even if I
already hung up.

The register entries come when I reject the call on the public network
phone.

I hope anyone can help me!

Thanx in advance,

Florian

PS: Hanging up when using CAPI instead of SIP works perfectly...

====================================
sip.conf:

[general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0   ; Address to bind SIP channel to
context = psin          ; Default context for incoming calls
tos=lowdelay                    ; IP QoS parameter, either keyword or value

maxexpirey=3600         ; Max length of incoming registration we allow
defaultexpirey=120              ; Default length of incoming/outoing
registration

disallow=all                    ; Disallow all codecs
allow=alaw
allow=ulaw                      ; Allow codecs in order of preference
allow=g726
allow=gsm
allow=ilbc
musicclass=random
externip = rauserv.dyndns.org ; Address that we're going to put in outbound
SIP messages
localnet=192.168.0.0/255.255.255.0; All RFC 1918 addresses are local
networks

language=de
register => 8888888:aaaaaa at 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
;pickupgroup=1
;callgroup=1
;dtmfmode=rfc2833

==============================

See sip debug output:

============
To: <sip:07141220856 at sipgate.de>
Contact: <sip:8888888 at 82.83.56.91>
Call-ID: 4fd908680d46bcd35ff3901c51bc0120 at 82.83.56.91
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 30 Jul 2004 20:40:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 314

v=0
o=root 32131 32131 IN IP4 82.83.56.91
s=session
c=IN IP4 82.83.56.91
t=0 0
m=audio 15080 RTP/AVP 8 0 2 3 97 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 (no NAT) to 217.10.79.9:5060
    -- Called 07141220856 at sipgate
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 2
Found RTP audio format 101
Peer RTP is at port 217.10.64.78:0
Found description format PCMA
Found description format PCMU
Found description format GSM
Found description format iLBC
Found description format G726-32
Found description format telephone-event
Capabilities: us - 0x41e(GSM|ULAW|ALAW|G726|ILBC), peer -
audio=0x41e(GSM|ULAW|ALAW|G726|ILBC)/video=0x0(EMPTY), combined -
0x41e(GSM|ULAW|ALAW|G726|ILBC)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined -
0x1(G723)
    -- SIP/sipgate-76e5 is making progress passing it to SIP/2112-495b
Reliably Transmitting:
CANCEL sip:497141220856 at 217.10.64.78 SIP/2.0
Via: SIP/2.0/UDP 82.83.56.91:5060;branch=z9hG4bK0b9e12cc
From: "Florian" <sip:8888888 at sipgate.de>;tag=as1c3c2263
To: <sip:07141220856 at sipgate.de>
Contact: <sip:8888888 at 82.83.56.91>
Call-ID: 4fd908680d46bcd35ff3901c51bc0120 at 82.83.56.91
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 217.10.79.9:5060
Scheduling destruction of call
'4fd908680d46bcd35ff3901c51bc0120 at 82.83.56.91' in 15000 ms
  == Spawn extension (out, 907141220856, 3) exited non-zero on
'SIP/2112-495b'
Transmitting:
ACK sip:497141220856 at 217.10.64.78 SIP/2.0
Via: SIP/2.0/UDP 82.83.56.91:5060;branch=z9hG4bK0b9e12cc
From: "Florian" <sip:8888888 at sipgate.de>;tag=as1c3c2263
To: <sip:07141220856 at sipgate.de>;tag=cbf5cb1d0d4e31526039b4f3671ccf51-3d18
Contact: <sip:8888888 at 82.83.56.91>
Call-ID: 4fd908680d46bcd35ff3901c51bc0120 at 82.83.56.91
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 217.10.79.9:5060
Destroying call '4fd908680d46bcd35ff3901c51bc0120 at 82.83.56.91'
11 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:217.10.79.9 SIP/2.0
Via: SIP/2.0/UDP 82.83.56.91:5060;branch=z9hG4bK2eefa4de
From: "asterisk" <sip:asterisk at 82.83.56.91>;tag=as7c4322e5
To: <sip:217.10.79.9>
Contact: <sip:asterisk at 82.83.56.91>
Call-ID: 6d3771aa1377fbf249c866b5767becfe at 82.83.56.91
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Fri, 30 Jul 2004 20:41:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0

 (no NAT) to 217.10.79.9:5060
Destroying call '6d3771aa1377fbf249c866b5767becfe at 82.83.56.91'
11 headers, 0 lines
Reliably Transmitting:
REGISTER sip:sipgate.de SIP/2.0
Via: SIP/2.0/UDP 82.83.56.91:5060;branch=z9hG4bK388496a5
From: <sip:8888888 at sipgate.de>;tag=as7cc829e2
To: <sip:8888888 at sipgate.de>
Call-ID: 6ed4586d1befa53f5f12e16a6d22e667 at 192.168.99.11
CSeq: 108 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:2001 at 82.83.56.91>
Event: registration
Content-Length: 0

 (no NAT) to 217.10.79.9:5060
12 headers, 0 lines
Reliably Transmitting:
REGISTER sip:sipgate.de SIP/2.0
Via: SIP/2.0/UDP 82.83.56.91:5060;branch=z9hG4bK0b70e7c4
From: <sip:8888888 at sipgate.de>;tag=as7cc829e2
To: <sip:8888888 at sipgate.de>
Call-ID: 6ed4586d1befa53f5f12e16a6d22e667 at 192.168.99.11
CSeq: 109 REGISTER
User-Agent: Asterisk PBX
Authorization: Digest username="8888888", realm="sipgate.de", algorithm=MD5,
uri="sip:sipgate.de", nonce="410ab3a95b427f0b044db27cdada37d268edcfee",
response="f4b21e167ad3b177beddceac6aea0567", opaque=""
Expires: 120
Contact: <sip:2001 at 82.83.56.91>
Event: registration
Content-Length: 0

 (no NAT) to 217.10.79.9:5060
Destroying call '6ed4586d1befa53f5f12e16a6d22e667 at 192.168.99.11'
Destroying call '4fd908680d46bcd35ff3901c51bc0120 at 82.83.56.91'
Destroying call '4fd908680d46bcd35ff3901c51bc0120 at 82.83.56.91'
Destroying call '4fd908680d46bcd35ff3901c51bc0120 at 82.83.56.91'
Destroying call '4fd908680d46bcd35ff3901c51bc0120 at 82.83.56.91'
Destroying call '4fd908680d46bcd35ff3901c51bc0120 at 82.83.56.91'
Destroying call '4fd908680d46bcd35ff3901c51bc0120 at 82.83.56.91'

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