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

[Bryant Zimmerman]

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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