I bought a Grandstream GXP1625 and put it on the MESH network. The direct IP address of the phone the ARDEN mesh is 10.88.207.114.
I have no issues going through the menus to make a direct IP dial call. It works. It also works with a VOIP provider on a SIP account I have. I can dial out to normal phone numbers.
But here's the question: how do you edit the phone book entries to dial by direct IP?
I have tried all of the following settings:
x.x.x.x in work phone entry
x*x*x*x in work phone entry
x*x*x*x#5016 in work phone entry
I've tried this with and without the SIP account added.
I've tried adding a "fake" SIP account to dial out on MESH.
Has anyone with a Grandstream managed to get this to work?
---mark, KM6ZPO
I have no issues going through the menus to make a direct IP dial call. It works. It also works with a VOIP provider on a SIP account I have. I can dial out to normal phone numbers.
But here's the question: how do you edit the phone book entries to dial by direct IP?
I have tried all of the following settings:
x.x.x.x in work phone entry
x*x*x*x in work phone entry
x*x*x*x#5016 in work phone entry
I've tried this with and without the SIP account added.
I've tried adding a "fake" SIP account to dial out on MESH.
Has anyone with a Grandstream managed to get this to work?
---mark, KM6ZPO
I've found Grandstream VoIP devices among the easiest to use and configure. They tend to "just work" out of the box. Many phones do not allow IP-to-IP direct calling, but Grandstream does this by default. All of the models I've tested have worked very well and I'd certainly recommend them.
I have a GXP1625 on my desk, and I have direct IP Speed Dial configured on the single "programmable key" -- as well as having a complete Phonebook configured with direct IP calling. I've found the instructions graciously provided by Tim N8NQH to be invaluable -- and they work on the entire GXP16xx line of phones. Here is the URL for his Speed Dial instructions: http://tim-yvonne.com/ham/mesh/1630/speed-dial.htm
Also, here is a previous forum post by Tim N8NQH on these phones:
https://www.arednmesh.org/content/grandstrean-1600-series-voip-telephones
Hope this helps.
Another workaround and for about the same price as a GXP1630
would be RasPBX.
(http://www.raspberry-asterisk.org/)
After some configuring, you would be able to
dial by extension, any registered IP phone
dial by callsign, any IP phone on your network with a registered and published 'name' or IP address
( I use the last 4 characters of my network neighbor's callsign.
I can more easily remember somene's callsign over remembering their extension number. )
and more.
YMMV.
Chuck
Cheaper than a FreePBX Raspberry PI and less hassle, fewer points of failure was the option I went with: a Cisco SPA504G phone. Those are going for $25 shipped. They support IP dialing via Phonebook entry. (I know because I have another one.) I may build a PBX server later just for fun.
At this point I'm just putting together a proof of concept for a field deployment of a base station ARDEN Mesh AP with one phone and one or more remote nodes (via RF) with phones.
Do the same for account 2 if you have one.
The good news is that at least I have speed dial working so I can add one contact. Thanks for starting this thread!
Your dIal pattern looks correct. If it works for speed dial it should work in the contact entry as well. You may have to pick up or not pick up the receiver first.
More info for troubleshooting:
Port numbers must match up. By default, the IP dialing port is going to be 5060. If you configure your phone for a SIP service which requires a different port, you're not going to be able to receive a direct dial call.
If the two phones don't have matching codecs, the call may go through but you won't hear each other.
As an example, I use G722 on my Cisco SPA504G. This is compatible with the Grandstream GXP1625. The GS will call it PCMA.
From the Grandstream docs:
What voice codecs do Grandstream products support?
All the phones support similar codecs and are listed below. The BT100 does not support GSM and the GXV does not support G722. Each codec has its uniqueness for certain application.
HT Series - PCMU (G711u), PCMA (G711a), G729A/B, G723.1, G726-32, iLBCHT502 - PCMU (G711u), PCMA (G711a), G729A/B/E, G723.1, G726-40/24/16, iLBC
BT Series - PCMU (G711u), PCMA (G711a), G729A/B, G723.1, G722, G726-32, iLBC
BT200 supports PCMU (G711u), PCMA (G711a), G729A/B, G723.1 and GSM codecs.
GXP Series - PCMU (G711u), PCMA (G711a), G729A/B, G723.1, G722, iLBC
GXV3000 - PCMU (G711u), PCMA (G711a), G729A/B, G723.1, GSM
GXP2200 - PCMU (G711u), PCMA (G722a), G729A/B, G722
The points above about codec types and ports is all and necessary to know by a caller.
What may not be so obvious on multi-line phones is that each line has a different SIP port assigned to it. Usually, Line 1 (sometimes called the "default" line) is customarily assigned internally in the phone to port 5060. Subsequent lines require different ports. Thus, Line 2 will be on sip port 5062, Line 3 will be on port 5064, etc. The assignment of different ports to different line buttons is necessary for PBX's to be able to send calls to the appropriate line buttons when calls come in.
For direct dialling by ip address, when calling a multi-line phone, not only do you need to know the ip address of the phone, but on which line button the call should come into.
One other thing to remember about calling by direct ip address is that the phone must be configured to allow unauthenticated incoming calls. And, with multi-line phones, this is done on a per-line basis. This means that a caller must know in advance on which line button his call should arrive. And, there are phones out there that simply do not allow incoming calls that are not through a pbx. Thus, direct ip dialling to those phones will never work.
For all the individual details on how to make direct ip dialling work successfully, it might just be easier to put them behind a pbx which will then take care of knowing all the details on how to reach any kind of sip phone. In addition, it will almost always result in a shorter telephone number that one has to dial.
73, Mark, N2MH
I found a setting in Accounts (the main one I use for port 5060 that happens to be setup as an extension on my PBX) > Sip Settings > Security Settings
Make sure "Accept Incoming SIP from Proxy Only" is set to NO