Mweb Talk SIP trunk with asterisk freepbx trunk configuration settings

[Zapo]

Good day all,

Im new to the broadband forum, I had an account that seemed to be deactivated and I couldn't find a way to message any admins or anything. So I just created a new one. (So for the admins, i donot know if they can merge my two accounts, pm me for more info please)

So my main issue,

I have a client who wants a voip pbx server,

I have installed and set up the asterisk with free pbx, I have struggled for about a week with trying to get my nexus sip account working, I still never got it working. ( Could never call outwards, well I could occasionaly)

I have now since moved over to a mweb mytalk sip trunk as nexus support simply said they dont give support with connecting the account to 3rd party devices.

Does anyone here know of the trunk sip settings to work with mweb sip trunk?

Outgoing:

Incoming:

RegisterString:

Currently I have just filled in the default parameters which freepbx provided when creating the trunk. it registers fine, but I can only make outgoing calls.

when I phone the mweb number (my voip number) from my phone it doesn't ring or anything it just hangs up, and there are no symptoms that the pbx received it. (ie its not my incoming route, as far as I know).

So if some one could provide me with some information and help relive me from my frustrated state I would greatly appreciate it.

I am sure that some here on the forums has successfully set up mweb talk on an asterisk freeepbx system.


Thanks

 

[Hummercellc]

Trunk Name: what ever you like
Outbound Caller ID: 27XXXXXXXXX (your sip number)

Outgoing settings:

Trunk Name: 27XXXXXXXXX (your sip number)

PEER Details:

username=27XXXXXXXXX
type=peer
qualify=yes
secret='sip password'
nat=yes
insecure=invite,port
host=196.28.95.12
fromuser=27XXXXXXXXX
fromdomain=196.28.95.12
dtmfmode=rfc2833
context=from-trunk
disallow=all
allow=g729 (change to alaw if you don't have g729 installed)
canreinvite=no

Incoming setting - Leave them all blank

Register string: 27XXXXXXXXX:YOUR-PASSWORD@196.28.95.12/27XXXXXXXXX

Remember to create a outbound route....

 

[Zapo]

Thanks for your response, I will give them a try.

May I ask why incoming must be blank? Does it not use them?

And do I need to make any changes to my global sip.config ?

 

[Hummercellc]

Thanks for your response, I will give them a try.

May I ask why incoming must be blank? Does it not use them?

And do I need to make any changes to my global sip.config ?

The incoming section causes more problems than it's worth, So instead you leave it blank and add the line
'context=from-trunk' to the outgoing section...

If you are using Asterisk 1.8 or later, the default config is fine.

Prior to 1.8 you only had to make sure you had the following:
You add this at the bottom under
Other SIP Settings

allowguest=no
alwaysauthreject=yes

 

[Zapo]

Thanks for you response,

I think I got it sorted.
I am using the latest version of freepbx. But I think I still added the allowguesy=no, alwaysauthreject=yes

However I am having an issue where the trunk registrations go offline?
They where connected then I came back and looked a while later and they where both offline.

What could be causing this?

I am running asterisk in a VM ontop of windows 7? I have some feeling that it is networking related some how. But the trunk lines are online but the registration isnt?

 

[Hummercellc]

Change your registration settings under the 'Asterisk SIP Settings'

Registration Settings

Registrations: 5 (registertimeout) 0 (registerattempts)
Registration Times: 60 (minexpiry) 120 (maxexpiry) 90 (defaultexpiry)

That should do the trick....

 

[rapidblue]

Trunk Name: what ever you like
Outbound Caller ID: 27XXXXXXXXX (your sip number)

Outgoing settings:

Trunk Name: 27XXXXXXXXX (your sip number)

PEER Details:

username=27XXXXXXXXX
type=peer
qualify=yes
secret='sip password'
nat=yes
insecure=invite,port
host=196.28.95.12
fromuser=27XXXXXXXXX
fromdomain=196.28.95.12
dtmfmode=rfc2833
context=from-trunk
disallow=all
allow=g729 (change to alaw if you don't have g729 installed)
canreinvite=no

Incoming setting - Leave them all blank

Register string: 27XXXXXXXXX:YOUR-PASSWORD@196.28.95.12/27XXXXXXXXX

Remember to create a outbound route....

Was also struggling to get incoming calls - this helped!! Thanks

 

[riazsabi]

I cant seem to get this to work. Same setup but using a Grandstream DP750 DECT setup

 

[dre84]

Any idea what the new settings will be on Freepbx 16? How to set up the trunk and in and out bound routes? i dont see any peer settings any more and it shows trunk online, but does not dail out or receive calls.

 

[dillydog]

You are most likely using the PJSIP Driver, you can change it back to ChanSIP or both and then setup the trunk as per above.

 

[dillydog]

Also note that Mweb/Ignite have 3 usable SBC, they all intermittently fail so it is good to know this. they are.

196.28.95.12 (sip.mweb.net)
197.84.140.140 (talk.isvoip.net) (This is in Cape Town)
197.80.140.140 (This is in JHB)
there is another DNS Name talk.mweb.net, this seems to toggle between the JHB and CPT systems.

 

[dre84]

ChanSip is no longer supported. There is not even an option for ChanSip any more.

 

[dillydog]

ChanSip is no longer supported. There is not even an option for ChanSip any more.

On FreePBX there is an option in the advanced settings to change the driver. But yes ChanSip has been grandfathered.