Hello,
I have been trying for a while to get my SIP trunks working without any luck and am hoping that someone can help me out. If there is a solution for this here already, sorry in advance for not finding it.
I am using the latest PIAF2 image with FreePBX 2.9 and Asterisk 10 on an Asus EEE PC.
My problem is that inbound and outbound calls on my Voice Network trunk don't connect. I get ringing, but when the call is answered the originating end keeps ringing. On inbound calls the Asterisk console shows the messages and I watch it sent to voicemail when the call is declined, where an empty voicemail is created.
I have the following ports forwarded in my Dlink DIR-615 router:
TCP: 5060-5061
UDP: 4659, 5000-5082, 10000-20000
I tried using the Dynamic IP/NAT settings in "Asterisk SIP Settings" (using no-ip.com and dlinkddns.com) without any luck, so I used the following in sip_custom.conf as per Ward's post on knol.google.com with his ip.sh script in /var/lib/asterisk/agi-bin:
externip=<my.external.ip.address>
localnet=192.168.0.0/255.255.255.0
nat=yes
Voice Network has a customer portal with recommended incoming and outgoing settings for Asterisk and Trixbox 2.6. I have tried the Trixbox config (since it uses FreePBX) and variations by adding from the Asterisk config without any luck. Currently I am using the following in my trunk config:
[Outgoing Settings]
host=sip.voicenetwork.ca
fromdomain=voicenetwork.ca
context=from-trunk
type=friend
username=<my.username>
fromuser=<my.username>
secret=<my.password>
dtmfmode=rfc2833
disallow=all
allow=ulaw
sendrpid=yes
qualify=yes
canreinvite=no
insecure=port,invite
[Incoming Settings]
type=peer
host=sip.voicenetwork.ca
fromuser=<my.username>
dtmfmode=rfc2833
disallow=all
allow=ulaw
qualify=yes
canreinvite=no
context=from-trunk
I have looked through the various configs and can't find reference to the "from-trunk" context.
Also, In the recommended Asterisk settings on the Voice Network portal there is this:
; To use VoiceNetwork.ca to termination your calls
; add the following line to your extensions.conf file
;
exten => _X.,1,Dial(SIP/voicenetwork-out/${EXTEN})
Where "voicenetwork-out" is the header of the outgoing settings in sip.conf:
[voicenetwork-out]
host= sip.voicenetwork.ca
type=friend
username=<my.username>
fromuser=<my.username>
fromdomain=voicenetwork.ca
sendrpid=yes
qualify=no
secret=<my.password>
dtmfmode=rfc2833
disallow=all
allow=ulaw
context=voicenetwork-incoming ; incoming DID calls will arrive in the voicenetwork-incoming context
insecure=port,invite
Thanks for your time,
Al
I have been trying for a while to get my SIP trunks working without any luck and am hoping that someone can help me out. If there is a solution for this here already, sorry in advance for not finding it.
I am using the latest PIAF2 image with FreePBX 2.9 and Asterisk 10 on an Asus EEE PC.
My problem is that inbound and outbound calls on my Voice Network trunk don't connect. I get ringing, but when the call is answered the originating end keeps ringing. On inbound calls the Asterisk console shows the messages and I watch it sent to voicemail when the call is declined, where an empty voicemail is created.
I have the following ports forwarded in my Dlink DIR-615 router:
TCP: 5060-5061
UDP: 4659, 5000-5082, 10000-20000
I tried using the Dynamic IP/NAT settings in "Asterisk SIP Settings" (using no-ip.com and dlinkddns.com) without any luck, so I used the following in sip_custom.conf as per Ward's post on knol.google.com with his ip.sh script in /var/lib/asterisk/agi-bin:
externip=<my.external.ip.address>
localnet=192.168.0.0/255.255.255.0
nat=yes
Voice Network has a customer portal with recommended incoming and outgoing settings for Asterisk and Trixbox 2.6. I have tried the Trixbox config (since it uses FreePBX) and variations by adding from the Asterisk config without any luck. Currently I am using the following in my trunk config:
[Outgoing Settings]
host=sip.voicenetwork.ca
fromdomain=voicenetwork.ca
context=from-trunk
type=friend
username=<my.username>
fromuser=<my.username>
secret=<my.password>
dtmfmode=rfc2833
disallow=all
allow=ulaw
sendrpid=yes
qualify=yes
canreinvite=no
insecure=port,invite
[Incoming Settings]
type=peer
host=sip.voicenetwork.ca
fromuser=<my.username>
dtmfmode=rfc2833
disallow=all
allow=ulaw
qualify=yes
canreinvite=no
context=from-trunk
I have looked through the various configs and can't find reference to the "from-trunk" context.
Also, In the recommended Asterisk settings on the Voice Network portal there is this:
; To use VoiceNetwork.ca to termination your calls
; add the following line to your extensions.conf file
;
exten => _X.,1,Dial(SIP/voicenetwork-out/${EXTEN})
Where "voicenetwork-out" is the header of the outgoing settings in sip.conf:
[voicenetwork-out]
host= sip.voicenetwork.ca
type=friend
username=<my.username>
fromuser=<my.username>
fromdomain=voicenetwork.ca
sendrpid=yes
qualify=no
secret=<my.password>
dtmfmode=rfc2833
disallow=all
allow=ulaw
context=voicenetwork-incoming ; incoming DID calls will arrive in the voicenetwork-incoming context
insecure=port,invite
Thanks for your time,
Al