Sip trunk connection to moldcell

[asd_rojd]

Hello
we need to migrate from asterisk to 3cx(version 15.5.15502.6). Problem with configuration of General sip trunk

old settings from asterisk are
Outbound CallerID:+37378111111
Trunk Name:moldcell

type=peer
qualify=yes
nat=yes
insecure=invite,port
host=MSS.MOLDCELL.MD
fromdomain=user1.sip.moldcell.md
disallow=all
canreinvite=no
autodomain=no
allow=alaw
Can some body help?

 

[eddv123]

Yes, on 3CX simply set this up as a Generic SIP trunk. All that should be required is to migrate the settings across. Since 3CX is GUI based and not CLI then it should be a lot easier.

I dont see any password field in your configuration here so can only assume it is an IP authenticated SIP trunk connection ?

 

[eddv123]

Also I see you are using NAT, so have a router/firewall in front of your system (which I would always recommend). Here is some information on how 3CX wants your firewall/router configured:

https://www.3cx.com/blog/voip-howto/firewall-nat-pat-stun/
https://www.3cx.com/docs/manual/firewall-router-configuration/

There are more specific guides for more popular router/firewall brands if you Google search also.

 

[JohnS_3CX]

If you are you new to 3CX then the following guide will give you all the information you need to setup your trunk. As eddv123 mentioned, if the provider is not listed, select the “Generic”option in the country drop down menu and then choose between “Generic VoIP Provider,” or “Generic SIP Trunk,” (But please note if using a generic provider we are not be able to guarantee that 3CX works with this VoIP Provider).

https://www.3cx.com/docs/manual/sip-trunks/

 

[asd_rojd]

Firewall already configured (we want to migrate) all is ok.
problem where exact in which filed i can put these settings in 3XC.
I understood that in trunk in "Outbound Parameters" but problem where exact (field) and what i need to put.

 

[leejor]

I understood that in trunk in "Outbound Parameters" but problem where exact (field) and what i need to put.

In most cases, the Outbound Parameters are best left at the defaults unless there is an issue. They are most often changed when outbound caller ID has problems.

https://www.3cx.com/blog/docs/voip-provider-configuration/

 

[asd_rojd]

after changing some outbound parameters
a compared INVITE's from asterisk and 3cx
green probably the same


how i can change rest parameters on 3cx to receive the same INVITE's ?

 

[leejor]

Some has to do with the audio ports and Codecs. Before you start changing things too much...what is not working?

 

[YiannisH_3CX]

Try disabling SRTP under the 3CX trunk settings / Options and see if that works. Everything else look pretty much the same.

 

[asd_rojd]

All is configured, but now to call on short numbers(rescue services and etc) my provider ask me to receive invite with section user=phone in invite


INVITE sip:[email protected];user=phone SIP/2.0
From: <sip:[email protected];user=phone>;tag=11639
To: sip:[email protected];user=phone

now it works ok with

INVITE sip:[email protected] SIP/2.0
From: <sip:[email protected]>;tag=11639
To: sip:[email protected]

is any way to add section user=phone in invite?

 

[YiannisH_3CX]

Currently this is not possible so i would recommend using a supported provider that is tested and fully compatible with 3CX. You will save a lot of manual configuration and guessing work and you will be running a supported configuration.