Thanks Bryant,
As I mentioned in my post - I forced everything on alaw - to make sure
it is not a codec problem. All ends support alaw.
Also, I've used:
directmedia=no
caninvite=no
canreinvite=no
to make sure the Asterisk stays in the media path.
At the moment it seems like a Linux firewall problem - Linux just
doesn't like the UDP packets from sipgate.co.uk - I'll have to figure
out why - as the Netgear ADSL router let's them through.
Thanks,
Sebastian
On 02/10/11 20:15, Bryant Zimmerman wrote:
verify your codec are the same on both trunks. make sure the both trunks
are using the same codec. make sure you have the correct ports open.
make sure you force all udp traffic to flow through your astrisk switch
as well.
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
------------------------------------------------------------------------
*From*: "Sebastian Arcus" <[email protected]>
*Sent*: Sunday, October 02, 2011 11:20 AM
*To*: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[email protected]>
*Subject*: [asterisk-users] Sipgate trunk doesn't bridge with other
trunk, but works with local extensions
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:[email protected]/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@voipcheap)
exten => 6012878,n,HangUp()
Any suggestions would be appreciated
Sebastian
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