Incoming Calls - 2 PSTN 2 VOIP Line Configuration Question

[cnguyen]

I'm having problems with multiple incoming phone calls working properly...
I performed test calls into the office main # from 4 different cell phones. I was able to get 2 calls in ( ring the group, then placed on hold). The 3rd call into the office just kept on ringing, same with 4th call.

Environment Setup:
3CX Version: 3.1 Beta 3.0.2328.0
4 Aastra 55i Phones (Extensions 100 -> 103)
Grandstream GXW-4104

PSTN1: 617-777-1111 (main number / Incoming Only)
PSTN2: 617-777-1112 (Incoming / Out Going)
VOIP1: 617-777-1113 (Incoming / Out Going)
VOIP2: 617-777-1114 (Incoming / Out Going)

Ring Group: Ext 888 (all 4 phones are part of the group); open office hrs.
Digital Receptionist: Ext 777; only responds during closed hours.

All calls are coming in on the main #.

Some questions I have are:

1. Do I need to enable "Bind to Meda Server" for each extension created?

2. On the Grandstream/ Channels page/ Phone Number Settings Section: I only have 2 entries (since I only have 2 pstn lines); do i have to add entries for VOIP 1 and 2 Internal SIP User ID and Authenticate ID ?

3. Since I only want the main number to receive incoming calls, how does 3cx handle this?; So that incoming calls are received and distributed to available lines (1pstn and 2voip) so that the main # is not tied up...; or how does this work?

4. Is it possible to configure the L1, L2, L3, L4 buttons on the Aastra phones to correspond directly to the 4 available lines (617-777-1111 through 617-777-1113, respectively)?... ie Some one calls in and Ext 100 answers on L2, places call on hold ; Person at extension 101 could press L2 and answer the call on hold.

This would be an ideal setup for most of my client offices...

Thanks in advance for any replies!

 

[DaKhalli]

Re: Incoming Calls - 2 PSTN 2 VOIP Line Configuration Questi

I'm having problems with multiple incoming phone calls working properly...
I performed test calls into the office main # from 4 different cell phones. I was able to get 2 calls in ( ring the group, then placed on hold). The 3rd call into the office just kept on ringing, same with 4th call.

Environment Setup:
3CX Version: 3.1 Beta 3.0.2328.0
4 Aastra 55i Phones (Extensions 100 -> 103)
Grandstream GXW-4104

PSTN1: 617-777-1111 (main number / Incoming Only)
PSTN2: 617-777-1112 (Incoming / Out Going)
VOIP1: 617-777-1113 (Incoming / Out Going)
VOIP2: 617-777-1114 (Incoming / Out Going)

Ring Group: Ext 888 (all 4 phones are part of the group); open office hrs.
Digital Receptionist: Ext 777; only responds during closed hours.

All calls are coming in on the main #.

Some questions I have are:

1. Do I need to enable "Bind to Meda Server" for each extension created?

Only if audio does not work i thought, but maybe it will help.

2. On the Grandstream/ Channels page/ Phone Number Settings Section: I only have 2 entries (since I only have 2 pstn lines); do i have to add entries for VOIP 1 and 2 Internal SIP User ID and Authenticate ID ?

to register the phone with the sip server you have to put in the sip data.

3. Since I only want the main number to receive incoming calls, how does 3cx handle this?; So that incoming calls are received and distributed to available lines (1pstn and 2voip) so that the main # is not tied up...; or how does this work?

i would say forward to an extension, if that extension is busy forward to another extension i would say, but maybe theres a better solution.
4. Is it possible to configure the L1, L2, L3, L4 buttons on the Aastra phones to correspond directly to the 4 available lines (617-777-1111 through 617-777-1113, respectively)?... ie Some one calls in and Ext 100 answers on L2, places call on hold ; Person at extension 101 could press L2 and answer the call on hold.

I think its part of the call queue feature, but havent played with it yet.

This would be an ideal setup for most of my client offices...

Thanks in advance for any replies!

hope this information will help you a bit and is correct, please let anyone bang my headagainst a wall if not

 

[SY]

Read manual.

Each RTP connection requires 2 UDP ports. Configure "Ports to use for external calls" on General Settings page (and open it on your firewall if necessary) to receive calls as many as you want

 

[cnguyen]

Thanks, I over looked that in the manual....

But will this help if the main number for the office is a PSTN line connected to the Grandstream? I'm trying to understand how the calls come in and are distributed between PSTN & VOIP...

1. Call comes in on main # (pstn 1) connected to grandstream, then forward to ring group.
2. User 1 at extension 100 picks up call. line 1 pstn is occupied.
3. Call 2 comes in on main #, ring group rings all phones.
4. User 2 at extension 101 picks up call. line 2 pstn is occupied.
5. Call 3 comes in on main #, Caller hears the phone ring, but ring group never rings and 3cx server does not recognize or see the call.

I'm starting to think that this is not possible.??? I'm trying different things and retesting the scenario above...

I tried to enable "bind to server" for all extensions, but that resulted in the calls being dropped after 15secs of being picked up and placed on hold...

20:00:31.187 MediaServerReporting::RTPSender [MS004000] Call(7) Ext.101: (ACTIVE): Can't send RTP stream to 0.0.0.0:3000 destination unreachable
20:00:22.218 CallLegImpl:nConnected [CM103001] Call(7): Created audio channel for Ext.101 (192.168.0.57:3000) with Media Server (192.168.0.20:7026)
20:00:22.218 StratInOut:nConnected [CM104005] Call(7): Setup completed for call from Ln:10001@GrandStream to Ext.101
20:00:22.218 CallLegImpl:nConnected [CM103001] Call(7): Created audio channel for Ln:10001@GrandStream (192.168.0.21:5020) with Media Server (192.168.0.20:7024)
20:00:19.031 CallConf:nProvisional [CM103003] Call(7): Ext.101 is ringing
20:00:18.812 CallConf:nIncoming [CM103002] Call(7): Incoming call from 617548XXXX (Ln:10001@GrandStream) to sip:[email protected]

Is this configuration (Incoming calls on PSTN line -> distributed to PSTN or VOIP through ring group) similar to what anyone else is setting up???? or do all the lines have to be PSTN for incoming and use VOIP for outgoing only?

Thanks for all help so far.

 

[cnguyen]

I thought this over after reading through other posts...

The way I had it setup initially was correct... and the line forwarding for incoming calls through the main number should be handle by the carrier (veizon, for PSTN 1 and 2) and for VOIP Lines , I was able to enable call forward if busy, by logging onto the broadvoice account to enable it.

So basically...

Call comes in -> PSTN1 (if busy forward to) -> PSTN2 (if busy forward to) -> VOIP 1 (if busy forward to) VOIP 2 (if busy forward to) -> PSTN 1

If some one could tell me if this is correct, that would be greatly appreciated!