External Phones - Tunnel, Ringback

[grodinj]

Does anyone know of any SIP phones that will work with 3CX off site without having to use the tunnel software? I am a little annoyed at having to turn on the computer if I need to use the phone. I am using Polycom 601's.

Another annoyance; I have a couple of external phones in my PBX system that use the SIP Proxy Manager ("Tunnel"). When they try to call extensions inside the office, the ringback (the tone that indicates the other phone is ringing) and disconnect (when one person hangs up, the other end automatically ends the call) work FINE! However, when the external phones call and outside number, there is no ringback and disconnect does not work. The call-e will hang up but the caller's phone stays connected and the timer keeps tickin'. At least the server shows the call as disconnected and not eating up minutes with my telco.

I have a Sangoma A101D gateway and have the tones generation set to "always". Not sure what else it could be.

Thanks,
Justin

 

[eagle2]

Hi Justin,

all your complains are related to NAT/firewall issues. The 3CX tunnel is one of the possible ways to overcome it.
Generally any SIP phone supporting STUN should work as external extension, unless remote router is not blocking it or NATing incorrectly.

If you mean site-to-site connectivity I would recommend using a kind of VPN tunnels between them using routers, this has also security impact. The topic has been discussed many times, I would also recommend to consider MikroTik (RB750) as inexpensive, powerful MPLS routers. You may establish a pseudo-wire type of tunnel, implement QoS, etc.

Regards

 

[snowsam]

While I was able to use polycoms off site with my old * system, you are correct that they don't work offsite without the tunnel on 3cx. I've also tried Yealink phones (mid-summer 2011) and there was some sort of bug that was preventing them from working correctly. I did not chase it too hard. There have been firmware upgrades since then and I have not tried again.

Instead I've successfully used Grandstream (handytone 286, GXP1400/1405) as well as Linksys / Sipura SPA-2102(or similar model) remotely without using the tunnel.

I hope that helps!
Sam

 

[telewebtech]

Justin,

It's definitely a networking issue and most SIP phones come with STUN settings. If you are looking for recommendations, Telewebtech has also had reliable success with Grandstream handsets (http://www.Grandstream.com) for remote extensions, particularly the GXP 1450, GXP 2100, and GXP 2000.

As for quality with remote extensions when not using the tunnel, I have heard marked improvements in several cases when the PBX hands off the call to the devices (make PBX delivers audio false for those extensions).

Hope this helps!

 

[leejor]

when the external phones call and outside number, there is no ringback and disconnect does not work. The call-e will hang up but the caller's phone stays connected and the timer keeps tickin'. At least the server shows the call as disconnected and not eating up minutes with my telco.

I have a Sangoma A101D gateway and have the tones generation set to "always". Not sure what else it could be.

I'm assuming that it is a setting in your gateway. Audio should be cut through once digits have been sent, there may be an initial ring from the gateway but any further audio should be from the line. As far as the disconnect, again, a setting in the gateway, as this is a T1/E1 gateway, it should respond from a disconnect "bit" from the far end and relay the SIP message to 3CX that the called party has hung up

 

[grodinj]

all your complains are related to NAT/firewall issues. The 3CX tunnel is one of the possible ways to overcome it.
Generally any SIP phone supporting STUN should work as external extension, unless remote router is not blocking it or NATing incorrectly.

Thanks for the input. Looks like Polycom Soundpoint phones do not have or support STUN. I have been banging my head against a wall for 2 days with NAT configurations and no luck. I'll just go get one of the recommended STUN supported SIP phones.

 

[grodinj]

I'm assuming that it is a setting in your gateway. Audio should be cut through once digits have been sent, there may be an initial ring from the gateway but any further audio should be from the line. As far as the disconnect, again, a setting in the gateway, as this is a T1/E1 gateway, it should respond from a disconnect "bit" from the far end and relay the SIP message to 3CX that the called party has hung up

Just for the heck of it, I restored the Polycom 501 to factory defaults and re-provisioned using the 3CX generated [MAC]_phone.cfg file and that fixed the disconnect issue. Still no ringback though (only when dialing outside lines). According to the Sangoma engineers recommendations, these are my gateway settings:
inbandProgressTonesGeneration="ALWAYS"
inbandProgressTonesIndicator="PROGRESS"