Hi everyone,
First off thank you for the wonderful work you're doing here to bring lower cost telephony to all! I setup my pbxiaf installation on a raspberrypi B device. Used the link for the raspberrypi installation (Incredible PBX 3.11 for the Raspberry Pi featuring the very latest Asterisk 11 and FreePBX 2.11 versions for a near perfect telephony platform) put it on a sdcard it boots everythings up. I used the google motif to add a google account to the system and it worked after a bit of fidgeting. Now I ported my house number to vitelity and its complete. I don't know exactly what I'm suposed to do to add it to my system and have it all work out. This is the only info I get from Vitelity
Please add the following configuration to your /etc/asterisk/sip.conf
Add the following at the bottom of your sip.conf
Please add the following configuration to your /etc/asterisk/extensions.conf
Add the following to the bottom of your extensions.conf
...I looked in these files wasnt sure whether i should manually add these settings or not, and there is no mention of usernames or passwords in that configuration offered by vitelity. So I used the GUI in PBXIAF to add an extension 702, trunk, inbound, outbound route. I'm not sure the settings I added there were correct. I'm also not sure the settings on the vitelity portal are correct either. I do have a subaccount created on vitelity and it says it is online, I tried it out with YATE and it connects and dials out to my verizon cellphone, however I get no audio either way. I have opened up a ton of those ports as it says in a pdf i found on pbxiaf website that said to open 5060-5082 and 10000:20000. In other places it says to not open any ports at all to the device, so I'm confused on what I'm supposed to do. I would welcome any help from someone with experience on vitelity or pbxiaf for raspberry or pbx in general.
Thank you all in advance, hope to hear from you soon!
Edit ----
I added this information straight into the files it asks for and saved/restarted but none of the info is pulled up in the gui.
At the top of your /etc/asterisk/sip.conf right after the general settings add:
register =>myusername:[email protected]:5060
Near the bottom add:
First off thank you for the wonderful work you're doing here to bring lower cost telephony to all! I setup my pbxiaf installation on a raspberrypi B device. Used the link for the raspberrypi installation (Incredible PBX 3.11 for the Raspberry Pi featuring the very latest Asterisk 11 and FreePBX 2.11 versions for a near perfect telephony platform) put it on a sdcard it boots everythings up. I used the google motif to add a google account to the system and it worked after a bit of fidgeting. Now I ported my house number to vitelity and its complete. I don't know exactly what I'm suposed to do to add it to my system and have it all work out. This is the only info I get from Vitelity
Please add the following configuration to your /etc/asterisk/sip.conf
Add the following at the bottom of your sip.conf
Code:
[inbound]
type=friend
dtmfmode=auto
host=inbound29.vitelity.net
context=inbound
allow=all
insecure=port,invite
[outbound]
type=friend
dtmfmode=auto
host=outbound.vitelity.net
context=outbound
insecure=port,invite
allow=all
Please add the following configuration to your /etc/asterisk/extensions.conf
Add the following to the bottom of your extensions.conf
Code:
[outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/$\{EXTEN}@outbound)
exten => _011.,1,Dial(SIP/$\{EXTEN}@outbound)
; e911 must be enabled. see DIDs > NPANXXNXXX > Action > e911
exten => _911,1,Dial(SIP/911@outbound)
[inbound]
exten => mynumberishere,1,Answer
...I looked in these files wasnt sure whether i should manually add these settings or not, and there is no mention of usernames or passwords in that configuration offered by vitelity. So I used the GUI in PBXIAF to add an extension 702, trunk, inbound, outbound route. I'm not sure the settings I added there were correct. I'm also not sure the settings on the vitelity portal are correct either. I do have a subaccount created on vitelity and it says it is online, I tried it out with YATE and it connects and dials out to my verizon cellphone, however I get no audio either way. I have opened up a ton of those ports as it says in a pdf i found on pbxiaf website that said to open 5060-5082 and 10000:20000. In other places it says to not open any ports at all to the device, so I'm confused on what I'm supposed to do. I would welcome any help from someone with experience on vitelity or pbxiaf for raspberry or pbx in general.
Thank you all in advance, hope to hear from you soon!
Edit ----
I added this information straight into the files it asks for and saved/restarted but none of the info is pulled up in the gui.
At the top of your /etc/asterisk/sip.conf right after the general settings add:
register =>myusername:[email protected]:5060
Near the bottom add:
Code:
[inbound]
type=friend
dtmfmode=auto
host=inbound29.vitelity.net
context=inbound
username=myusername
secret=mypass
allow=all
insecure=port,invite
canreinvite=no
[outbound]
type=friend
dtmfmode=auto
host=outbound.vitelity.net
username=myusername
fromuser=myusername
secret=mypass
trustrpid=yes
sendrpid=yes
allow=all
canreinvite=no