Hello, I have a planet FXO 4 port gateway
I am using freepbx 3.0
I don´t know if I should add these analog ports using custom trunk or dadhi trunk
Thank you
Cesar
Is this an external device or a card in the system? If it is an external device you will probably connect to is using SIP. If it is a card in the system and is supported than DAHDI.
This is an external device
http://planet.com.tw/en/product/product.php?id=37038#appl
I will try to add a sip trunk,
Thank you Alan
I’ll let you know
Are you aware that you will need to login to the GUI of the VIP-480 and configure it too? I have never used this device so I can’t help you there.
Hello Alan
The status of the sip trunk is “unmonitored” (CLI sip show users)
In the status page I can see:
Ip Trunks online 1
Ip trunk registration 0
Yes. I am and I have done so
Thank you
Post your trunk config from FreePBX.
General Settings
Trunk Name?: vip480
Outbound CallerID?:
CID Options?: Allow any CID
Maximum Channels?: 50
Asterisk Trunk Dial Options? Override
Continue if Busy?: Check to always try next trunk On
Disable Trunk?: Disable Off
Dialed Number Manipulation Rules?
() + | . insert remove
Dial Rules Wizards?: Pick one
Outbound Dial Prefix?: 9
Export Dialplans as CSV?:
Outgoing Settings
Trunk Name?: myvip480
PEER Details?:
canreinvite=no
fromuser=58036366
host=192.168.2.50
port=5060
secret=9342009610
type=peer
username=58036366
allow=g723&g729
Incoming Settings
USER Context?:
USER Details?:
Registration
Register String?:
58036366:[email protected]
I have just added to peer details the qualify=yes string and now it says OK when I execute sip show peers (CLI)
I have checked the logs of asterisk and found this:
[2013-09-10 18:33:03] NOTICE[2002] chan_sip.c: – Registration for ‘[email protected]’ timed out, trying again (Attempt #66)
[2013-09-10 18:33:03] WARNING[2002] chan_sip.c: Remote host can’t match request REGISTER to call ‘0f5b649874bd730b5a25542f6e0f26a5@[::1]’. Giving up.
It’s on a fixed IP registration is not needed.
Also you should not need any authentication.
Lastly, do you have g.729 licenses installed? Why would you use a low bit rate CODEC on a LAN?
What should I delete from the “peer details” ?
I am not familiar with codec carachteristics, Which should I use?
Thank you
Cesar
Your CODEC should match, ulaw or alaw is fine (use region appropriate).
You should not need any authentication parameters, don’t overcomplicate.
I don’t know your gateway, key is you have to match it to Asterisk.
Check sample sip.conf for the version Asterisk you are running. It contains all the SIP peer variables and explanations that you can use in the FreePBX trunk module.
Remember FreePBX controls Asterisk, don’t modify the config files directly, it’s just a reference.
Ok for the codec
You say that I don’t need any authentication parameters because I am using a gateway in the same lan?
I deleted the register string as well as secret and user
In the status page it doesn’t show any more the trunk connections line
How do I know if the trunk is working?
Thank you
Cesar
For trunk status use the qualify operator?
Did you look at the docs ?
The status of the trunk is ok
I have checked the log file:
[2013-09-11 09:17:21] VERBOSE[10665][C-00000055] pbx.c: == Spawn extension (from-internal, 55660021, 8) exited non-zero on ‘SIP/710-00000059’
[2013-09-11 09:17:21] VERBOSE[10665][C-00000055] pbx.c: – Executing [h@from-internal:1] Hangup(“SIP/710-00000059”, “”) in new stack
[2013-09-11 09:17:21] VERBOSE[10665][C-00000055] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/710-00000059’
[2013-09-11 09:17:21] VERBOSE[10666][C-00000055] app_mixmonitor.c: == MixMonitor close filestream (mixed)
[2013-09-11 09:17:21] VERBOSE[10666][C-00000055] app_mixmonitor.c: == End MixMonitor Recording SIP/710-00000059