I am running Asterisk 1.8.4.4 and want to use RLT (Release Link Trunking) over SS7 to our DMS-100 switch. All documentation that I have been able to find only shows PRI. Has anyone been able to use RLT over SS7? If so can you share your dialplan?
Maybe PRI has special connotations in Asterisk, but SS7 can be, and most often is, used over Primary Rate Interfaces.
Hi,
Can you let me know what you want to do first, so I can suggest you the better way with Nortel.
Regards,
Ketan
Here is a diagram of what we are trying to accomplish
User A makes a call to User B. User B makes a blind transfer to User C.
[code] User A ±--------------+ ±---------------+ ±--------------+
-
| | | | | |
| | | | | | |
±------->| |------->| |±------>|–+ |
| | | | | | |
| DMS-100 | | Asterisk | | |SIP Peer |
| | | | | | <----> User B
| | | | | | |
±-------+| |<-------| |<--------|–+ |
| | | | | | |
v | | | | | |
User C | | | | | |
±--------------+ ±---------------+ ±--------------+[/code]
Today, Asterisk will accept the blind transfer and the SIP Peer drops out.
[code] User A ±--------------+ ±---------------+ ±--------------+
-
| | | | | |
| | | | | | |
±------->| |±------>|–+ | | |
| | | | | | |
| DMS-100 | | |Asterisk | | SIP Peer |
| | | | | | <----> User B
| | | | | | |
±-------+| |<-------±–+ | | |
| | | | | | |
v | | | | | |
User C | | | | | |
±--------------+ ±---------------+ ±--------------+
[/code]
I would then like to have Asterisk release the call back to the DMS-100 via RLT. This would then release 2 dahdi channels.
[code]User A ±--------------+ ±---------------+ ±--------------+
-
| | | | | |
| | | | | | |
±------->|–+ | | | | |
| | | | | | |
| |DMS-100 | | Asterisk | | SIP Peer |
| | | | | | <----> UserB
| | | | | | |
±--------|–+ | | | | |
| | | | | | |
v | | | | | |
User C | | | | | |
±--------------+ ±---------------+ ±--------------+[/code]
BTW, Going down the PRI route on our DMS-100 is not an option.
Does RLT release the signalling path, or just the speech path? There is no way that Asterisk will release the signalling path, unless you use the Transfer application. The Transfer application is buggy on SIP and would think it is little used on Dahdi, if supported at all.
Doing a SIP blind transfer will not initiate an upstream blind transfer on a different technology. Asterisk will bridge the call itself.
I think the closest that Asterisk can get to pushing a blind transfer upstream is that you can configure it to recognize a refer-to address as non-local and actually generate a real INVITE from it.
RLT releases the signaling and the media; it’s a PSTN thing.
As it currently stands, you are correct, the RLT / 2BCT capabilities of Asterisk are limited to the PRI world.
Do you know if there any plans for RLT / 2BCT with SS7?
Howdy,
No commercial interests have driven us to investigate it.