ATA for outbound calls

[kenyon]

I recently setup Switchvox Free Edition so I could see if Switchvox would be a viable option for our company. I also picked up a couple of IP phones and a Linksys SPA3102, hoping to use it for both connecting an analog phone and connecting to a PSTN line to play with both incoming and outgoing calls.

I haven't been able to find how to set up Switchvox to use the SPA3102 for the PSTN line. I set it up as an extension, and if I call that extension I get dial tone and can then dial out, but that's not terribly helpful. I also have yet to find out how to get Switchvox to answer an incoming call. The settings I've looked at generally look like they would only apply to an installed Digium card.

[deckerp]

I would also like to see an ATA or VOIP Gateway type of device used. It seems like the Switchvox PBX always wants to authenticate to it as if it were a SIP Provider. In the case of AsteriskNOW you could configure it as a PEER.

How do I do this with Switchvox.

I'm using a Grandstream GXW4104 and Switchvox free edition.

[mikeshappell]

Under "System Setup", "VoIP Providers", choose to modify or add a new provider (depending on what you already have) and then select the "Click to show advanced options". There is an option called "Host Type" which will let you select "peer" or "provider".

[deckerp]

Thanks for the suggestion. I've tried both Peer and Provider but can't seem to make it work for either.

When setup as Provider I see the following in the Advanced Error Log:

1 01/08/2008 7:29 PM WARNING chan_sip.c: Host '192.168.123.3' does not implement 'REGISTER'
2 01/08/2008 7:29 PM NOTICE chan_sip.c: -- Registration for 'admin@sip_provider_101' timed out, trying again (Attempt #1)
3 01/08/2008 7:28 PM WARNING chan_sip.c: Host '192.168.123.3' does not implement 'REGISTER'
4 01/08/2008 7:28 PM NOTICE indications.c: Removed default indication country 'us'


When setup as Peer I don't see any new entries in the Advanced Error Log when I attempt a call, but when I do the Enable Outgoing Call Diagnostics Mode it seems to go to the proper rule assigned to the Grandstream.

- It looks like they both support Ulaw, Alaw, and GSM. I have all three checked on both ends.

- I've tried many combinations of DTMF mode, but I expected RFC2833 to work.

I did post a question for Grandstream and received the following reply:

Thanks for your email. We know of no compatibility issues with any SIP based system that utilizes the RFC 3261 international standard. I have been informed that our products are likely compatible with Switchvox, as Switchvox is a Digium / Asterisk product. Grandstream has completed in-depth interoperability testing with Digium / Asterisk, a Grandstream solution partner. You will find more information about interoperability on our website at http://www.grandstream.com/interoperability.html.

Any further suggestions would be appreciated.

[mikeshappell]

You might try to set Qualify Hosts to yes (as I think this sets insecure to very (worth a try). You might also look at Apply Incoming Call
Rules to Provider to no as well, but I would try the quality host thing first. Since I don't have one of these devices or experience with it, I can't be much more help.