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

[Sebastian Arcus]

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" <shop at open-t.co.uk>
> *Sent*: Sunday, October 02, 2011 11:20 AM
> *To*: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> *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: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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