Hi,
Firsty, I’m a bit of a newbie at this, so bare with me.
The problem I am facing is that whenever I call my sipgate.co.uk from a landline phone, about 15-30 seconds into the call, the phone will cut off. However, if I call from a sip softphone on my network, the call works fine.
The output that I have included at the bottom seems to me that somewhere along the line, asterisk is being told that I have hung up the phone, when I haven’t.
If anyone could shed light on whats happening, I would be extremely grateful.
Below are the details of my setup.
Asterisk 1.4 (from source) on Ubuntu
I have UDP/TCP ports 5060, 5004 and 10000 all set to forward to my asterisk computer (192.168.0.2)
sip.config
[general]
qualify=yes
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
register => XXXXX:YYYYY@sipgate.co.uk/XXXXX
externip = 82.43.XXX.XX
localnet=192.168.0.2/255.255.248.0
nat=yes
[sipgate.co.uk]
type=peer
secret=YYYYY
username=XXXXX
host=sipgate.co.uk
fromuser=XXXXX
outgoingproxy=sipgate.co.uk
canreinvite=no
dtmfmode=rfc2833
insecure=very
context=default
disallow=all
allow=gsm
allow=ulaw
allow=alaw
extensions.conf
[general]
static=yes
writeprotect=no
autofallthrough=no
clearglobalvars=no
priorityjumping=no
[default]
exten => s,1,Answer()
exten => s,n,SayPhonetic(“a”)
exten => s,n,MusicOnHold()
exten => t,1,SayPhonetic(“b”)
Output when called from landline
[…]
<------------->
— (13 headers 0 lines) —
nic-desktop*CLI>
<— SIP read from 217.10.79.23:5060 —>
BYE sip:XXXXX@82.43.XXX.XX SIP/2.0
Via: SIP/2.0/UDP 217.10.79.23:5060;branch=z9hG4bKac7e.c61668b5.0
Via: SIP/2.0/UDP 172.20.40.4;rport=5060;branch=z9hG4bKac7e.c61668b5.0
Via: SIP/2.0/UDP 217.10.79.23:5060;received=217.10.68.2;branch=z9hG4bK2efec6a6
Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK2efec6a6;rport=5060
From: “020ZZZZZZZZ” sip:020ZZZZZZZZ@sipgate.co.uk;tag=as453fc14e
To: sip:XXXXX@sipgate.co.uk;tag=njgmj
Call-ID: 29b5d010333840863f3f40b4621f6d15@sipgate.co.uk
CSeq: 103 BYE
Max-Forwards: 67
Content-Length: 0
X-hint: rr-enforced
<------------->
— (12 headers 0 lines) —
Sending to 217.10.79.23 : 5060 (NAT)
nic-desktop*CLI>
<— Transmitting (NAT) to 217.10.79.23:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.79.23:5060;branch=z9hG4bKac7e.c61668b5.0;received=217.10.79.23
Via: SIP/2.0/UDP 172.20.40.4;rport=5060;branch=z9hG4bKac7e.c61668b5.0
Via: SIP/2.0/UDP 217.10.79.23:5060;received=217.10.68.2;branch=z9hG4bK2efec6a6
Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK2efec6a6;rport=5060
From: “020ZZZZZZZZ” sip:020ZZZZZZZZ@sipgate.co.uk;tag=as453fc14e
To: sip:XXXXX@sipgate.co.uk;tag=njgmj
Call-ID: 29b5d010333840863f3f40b4621f6d15@sipgate.co.uk
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:XXXXX@82.43.XXX.XX
Content-Length: 0
[…]