UPDATE: This is old configuration you can try the new configuration in the post above
The Portech MV-372 gave me a lot of headache configuring it right. The configuration for Asterisk specified in documentations wasn't an option for me, working as extension, two dial stages while it works but come on, dial the extension, wait for signal, dial the number...
On the net i found some non-working configs so i had to spend 4 days trying to make it work as a Trunk and not as an extension.
Ok lets move on with configuration ;)
First we will configure the Portech MV-372 i believe this configuration will also work with Portech MV-370.
Login to your portech
- Route
- Mobile To Lan Settings:
Item | CID | URL
0 | * | 100
notes: in URL you put your extension of your asterisk you want call from mobile to go. It can be extension, ringgroup, ...
- Lan To Mobile Settings:
Item | URL | Call Num
0 | * | #
notes: URL will match the IP address that will allow to dial through portech, since my server is behind NAT i allowed all IP, haven't try to specify IP. Call num # will receive the number dialed from ip(soft)phone.
- Mobile To Lan Settings:
- SIP Settings
- Service Domain
Mobile 1 (Realm 1)
Display Name: Sim1
User Name: 1001
Register Name: 1001
Register Password: xxxxxx (choose a password)
Domain Server: 192.168.x.x (you asterisk IP)
Proxy Server: 192.168.x.x
Mobile 2 (Realm 1)
Display Name: Sim2
User Name: 1002
Register Name: 1002
Register Password: xxxxxx (choose a password)
Domain Server: 192.168.x.x (you asterisk IP)
Proxy Server: 192.168.x.x
notes: Username and registername you can change it to your needs just have a note of them since you will be entering those in asterisk trunks.
- Port Settings
Just make sure SIP Port for Mobile 1 is 5060 and
SIP Port for Mobile 2 is 5062
- Service Domain
Dont forget to save changes (should reboot after saving)
Now lets move to Asterisk.
Login to your FreePBX/Trixbox and add SIP Trunk
Outbound Called ID: xxxxxxx (put you number here)
Maximum Channels: 1
go to Outgoing settings
Trunk Name: SIM1 (i have called it that way)
PEER Details:
host=192.168.x.x (your Portech IP address)
type=peer
Incoming Settings:
USER Context: 1001 (important must match username/registername at Sip settings of Portech)
USER Details:
type=friend
secret=xxxxxx (match SIP Settings password from Portech)
username=1001 (match SIP Settings from Portech)
qualify=yes
nat=yes
canreinvite=no
context=from-internal
host=192.168.x.x (Portech IP)
And then just click Submit Changes (don't forget the Orange bar on top after you make changes in your server)
Add another SIP Trunk for SIM2
Outbound Called ID: xxxxxxx (put you number here)
Maximum Channels: 1
go to Outgoing settings
Trunk Name: SIM2
PEER Details:
host=192.168.x.x (your Portech IP address)
type=peer
port=5062 (important - this is for Mobile 2, remmber Port Settings on Portech?)
Incoming Settings:
USER Context: 1002 (important must match username/registername at Sip settings of Portech)
USER Details:
type=friend
secret=xxxxxx (match SIP Settings password from Portech)
username=1002 (match SIP Settings from Portech)
qualify=yes
nat=yes
canreinvite=no
context=from-internal
host=192.168.x.x (Portech IP)
port=5062
Again apply changes.
We're almost done. Now to make this work we have to create Outboud Route, so click Outbound Routes
Put Route name as you wish, i have called it Portech_1 (since i will add another and will make it Portech_2)
Dial Patterns: i have put 049XXXXXX because i want only mobile numbers from the same provider to go through this trunk (through Portech) i mean i want to cut the costs right? But because you don't have my gun on your head go ahead and do whatever it suits you.
Trunk Sequence: i added SIP/SIM1 and SIP/SIM2
You can sepparate Trunks from OutRoutes if you have sim cards from two different providers, just create another Outbound Route remove one Trunk from trunk sequence of the first route that we created and add it to this new one.
Only thing you left to do now is click Submit Changes then Apply Configuration Changes and pray for this to work.
Hope it will work for you.
9 comments:
Hi,
after 2 days of senseless seaching, i found your "manual" and... it works. Thank you!
Two questions: 1. Can you confirm the same behaviour, that the status of the portech under "SIP Settings"-"Service Domain" is still "Not registered" even if the Trixbox said "Trunk online"?!
2. Have you also a solution for the incomming calls, means Inbound Routes?!
Thanks in advance...
Ian
like the previous comment - says Not registered even though trunk is available in FOP - and I still can't make or recieve call on this!
Hi - just to say that my last comment was madness!! I had no signal on my SIM - which I now have after switching providers - outgoing calls work a treat :-)
Did you ever accomplish getting inbound calls diverted to trixbox?
Hi,
Please try the new configuration that i have posted and come back if you have problems.
I apologize for writing this late but i didn't configure my blog to email me when i have comments lol, but now its fixed.
Thanks
hi guys, i am trying to terminate call from my voip switch to mv 372, but it didnt work.
cause i couldnt configure it properly i guess.
my internet connection is nat firewalled, so i cant get register the device(mv372) with my voip switch (server)
does any one have any idea how to configure it properly with voip switch ??
it gives me lot of pain cause since last 3 weeks i am trying , but failed.i will appreciate anybodys help regarding this.
my email id is : mdhb35776@yahoo.com
and turuntaz@hotmail.com
thanks
Sorry delowar but i have never worked with Voipswitch so i can't help you there. If you had asterisk/freepbx i would be glad to help.
Thank you!
I spent two hours without success to link a pbxnsip to a Portch MV370 due to wrong routing table.
Your post gave me the right way to make it works.
RBB
hii i had followed your manual to configure mv372 with asterisk free pbx.but i want to know how to check whether it is configured or not.can any one help me
I just found your blog and want to say thank you.
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