Hello,
I would like to ask for a feature that should have been available, but does not seem to be supported.
Accepting inbound calls from and sending outbound calls to SIP URIs.
Now that VoIP calls via the SIP protocol are becoming the norm, support fro accepting direct calls via SIP URI is sorely lacking.
Essentially, we should be able to select SIP URI as a Destination and enter name@domain.com for Inbound Routes, and IVRs.
Is there a reason this has not been implemented yet?
Dia,
What is the purpose to select SIP URI as a Destination for inbound routes and IVRs?
I am assuming you are talking about routing the incoming calls to external SIP server, if so,
I think you can follow below steps:
1. Create a SIP trunk, point to the external SIP server;
2. Create an outbound route and select the SIP trunk in step 1;
3. Setup the inbound route destination as outbound route create in step 2;
I am not sure if I understand your request properly, if not, please correct me.
Dear Oscar,
As you will certainly know, the most basic communication between SIP UAs is via direct calls, without a SIP Proxy between them. I am fully aware that when the SIP protocol was designed NAT was not on the horizon.
Most B2BUAs are primarily used to hide the internal network topology, protect IP phones from attacks, collect CDRs, and facilitate calls between extensions, and they are very successful at it.
The idea of being able to send a call directly to a SIP URI, should not require the setup of a trunk, since there is no registration involved, no credentials required for an outgoing call. Just a field to enter the SIP URI.
To accept inbound calls via SIP URIs, an Inbound Route should to be setup, fro directing inbound calls to a destination other than the default /.
The ability to accept and send direct SIP calls, is built-in to Asterisk, there is nothing else to install, so there should be an option to enable it.
Some will argue that enabling direct SIP calls can create security issues.The security issues are there on a daily basis, whether direct SIP calls are enabled or disabled. Whoever doesn't want them, can just turn off that option.
Let me know if more information is required.
Dia, There is an option in the S PBX, see below:
Not sure if this would help.
If you would need route the incoming calls direct to S PBX, I think with this option would help.
But if you would like to route to S PBX first then route to other SIP server, I would suggest you to customize asterisk dial plan. Folow below guide:
Hey Oscar,
That might be the proper option to follow.
Can I reach you via email to ask a specific question?
Yes, my email address: oscar.colka@gmail.com
So what was the solution? We need to to enable SIP URI dial in to S100 too. How can we do that?
Unified Communications
