[asterisk-users] Sipgate trunk doesn't bridge with other trunk, but works with local extensions

Sebastian Arcus shop at open-t.co.uk
Sun Oct 2 10:20:33 CDT 2011


Hello list,

My setup is as follows:

Trunks: 2 sip trunks, one with voipcheap.co.uk, one with sipgate.co.uk
Extensions: 1 hardware sip phone
Asterisk: 1.8.7.0

Everything is working fine, except bridging between the sipgate and 
voipcheap trunks. I'll explain:

1. If I call from an external phone my sipgate landline number, it 
connects to my internal hardware sip phone/extension and works fine.
2. If I use my hardware sip phone to make outgoing calls through the 
voipcheap.co.uk trunk - it all works fine.
3. However, I want the call coming in through the sipgate trunk to call 
my mobile phone through the voipcheap trunk - this is not working. It 
will ring the mobile number, but when I answer there is no sound at 
either end.

I assume it is not:

1. A NAT problem, otherwise it would cause problems when making calls 
through voipcheap, or receiving through sipgate (but I could be wrong).
2. A codec problem - as I've forced everything on alaw

I can't see any errors in the console either. Please find below my 
sip.conf, extensions.conf:

#/etc/asterisk/sip.conf

[general]

canreinvite=no
disallow=all
allow=alaw
allowguest=no
externip=111.222.333.444
localnet=192.168.16.0/255.255.255.0

tos_sip=cs3                    ; Sets TOS for SIP packets.
tos_audio=ef                   ; Sets TOS for RTP audio packets.

registerattempts=0

register => 1234567:my_password at sipgate.co.uk/1234567

[sipgate]
type = friend
host=sipgate.co.uk
fromdomain=sipgate.co.uk
disallow=all
allow=alaw
qualify=yes
nat=yes
canreinvite=no

[voipcheap]
type=peer
username=my_username
fromdomain=sip.voipcheap.co.uk
realm=sip.voipcheap.co.uk
secret=my_password
host=sip.voipcheap.co.uk
disallow=all
allow=alaw
canreinvite=no

[20]
type=friend
username=20
secret=my_password
host=dynamic
context=from_internal_sip
qualify=yes



#/etc/asterisk/extensions.conf
[general]
static=yes
writeprotect=yes
autofallthrough=yes
priorityjumping=no


[from_internal_sip]
exten => _9.,1,Dial(SIP/${EXTEN:1}@voipcheap)
exten => _9.,n,HangUp()


[from_sipgate]

exten => 6012878,1,Dial(SIP/0794012345 at voipcheap)
exten => 6012878,n,HangUp()

Any suggestions would be appreciated

Sebastian



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