I have a Cisco IP Phone model SPA504G. It's on the list of recommended IP phones: https://www.arednmesh.org/content/ip-phones-used-mesh
It works, it can receive phone calls at AREDN IP 10.3.51.101
Problem is, I can't figure out how to dial an AREDN IP address with the menus / keypad on the phone.
I've dug through the phone's web interface as well. There's some kind of phone book in there, but it doesn't seem to work.
Anyway, if anyone has a web resource or just knows the correct key sequence to dial by IP direct to another AREDN IP phone (no PBX, no SIP), I would greatly appreciate the help.
---mark, KM6ZPO
It works, it can receive phone calls at AREDN IP 10.3.51.101
Problem is, I can't figure out how to dial an AREDN IP address with the menus / keypad on the phone.
I've dug through the phone's web interface as well. There's some kind of phone book in there, but it doesn't seem to work.
Anyway, if anyone has a web resource or just knows the correct key sequence to dial by IP direct to another AREDN IP phone (no PBX, no SIP), I would greatly appreciate the help.
---mark, KM6ZPO
Go into the Personal Directory settings.
Add an entry like this:
n=A;p=10.88.207.114
Where n = the name of the directory entry
P= the IP address.
You can do this via the dial pad, but it's a real pain entering IP addresses that way.
Then from the keypad, select the Directory, select the entry and dial. Works great!
Denis KD1HA
---mark, KM6ZPO
From what I am reading here, IP to IP calling is workable without a PBX?
I have my SPA-941 connected and my MikroTik (lite) assigned an IP address. And I have another phone enroute to another member (with an active node) that lives across town.
If this does work. My next shipment will go to our state EMCOM liaison.
After that, maybe a PBx.
Bruce, N1JGB
I'm ready to give up and just get a set of grandstreams because it shouldn't be this hard.
After viewing a utube video and reading Mark's instructions,
I think the Cisco's Personal Directory should look like the attachment.
What does your Personal Directory look like?
73, Chuck
Is there a SPA504G setting that allows/disallows inbound SIP calls?
[2026-08-05 11:00:53] VERBOSE[28713][C-00000018] pbx.c: Executing [939492@from-internal:1] Wait("SIP/1074-00000026", "1") in new stack
[2026-08-05 11:00:54] VERBOSE[28713][C-00000018] pbx.c: Executing [939492@from-internal:2] Dial("SIP/1074-00000026", "SIP/10.167.115.250,30") in new stack
[2026-08-05 11:00:54] VERBOSE[28713][C-00000018] netsock2.c: Using SIP VIDEO TOS bits 136
[2026-08-05 11:00:54] VERBOSE[28713][C-00000018] netsock2.c: Using SIP VIDEO CoS mark 6
[2026-08-05 11:00:54] VERBOSE[28713][C-00000018] netsock2.c: Using SIP RTP TOS bits 184
[2026-08-05 11:00:54] VERBOSE[28713][C-00000018] netsock2.c: Using SIP RTP CoS mark 5
[2026-08-05 11:00:54] VERBOSE[28713][C-00000018] app_dial.c: Called SIP/10.167.115.250
[2026-08-05 11:00:54] VERBOSE[28713][C-00000018] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[2026-08-05 11:00:54] VERBOSE[28713][C-00000018] pbx.c: Executing [939492@from-internal:3] Goto("SIP/1074-00000026", "Utilities,Sorry,1") in new stack
[2026-08-05 11:00:54] VERBOSE[28713][C-00000018] pbx_builtins.c: Goto (Utilities,Sorry,1)
[2026-08-05 11:00:54] VERBOSE[28713][C-00000018] pbx.c: Executing [Sorry@Utilities:1] Wait("SIP/1074-00000026", "1") in new stack
Does the "(1:0/0/1)" indicate my PBX does not like the number dialed or
the SPA504G at the other end does not like the number I dialed or
?
73, Chuck
Thank you.
I saw a post from Brian WD9HXC requesting a tunnel.
I sent him client credentials.
Once linked I noticed he advertised a phone, so I set up a SIP dial-by-callsign on my PBX.
It didn't work and, after a few days of trying things,
he adjusted his end and it now works.
I just encourage Brian to post what he did here.
;-)
3s, Chuck
The SPA504G doesn’t have a specific “allow inbound SIP calls” toggle. But inbound IP‑to‑IP SIP only works if several settings are correct:
Line Enable = yes
SIP Port = 5060 and SIP Transport = UDP
NAT Mapping/Keep Alive = disabled
Make Call Without Reg = yes (this is the key one)
The PBX log “(1:0/0/1)” means the PBX sent the SIP INVITE but the SPA504G never responded — which is exactly what happens when one of the above settings blocks direct SIP.
Once those are set, the phone will ring on inbound IP calls.
So lets say the user ID is "your call", then to get an incoming call what needs to be dialed is this:
"your call"@xx.xx.xx.xx if you do not fill in the user ID field then only the IP needs to be dialed.
You can probably do this without changing you call plan.
I am using the 504G on my shack node and I am using MicroSIP on my h2 node. I can call my Shack node by dialing just the IP. I have to use :siph2@xxx.xxx.xxx.xxx (x being your IP address) in order to call the h2 node.
If direct dialing is possible, what is the correct way to enter the 10‑period sequence?
I am not sure if you can do just one extension and not affect the others. I may play around with that and see. Otherwise what you will need to do is put the number in your personal directory. Then you can dial it trough that without any issues. The 504G is not the most user friendly when direct dialing, but you can do it. By changing your call plan so the phone sees the * as a period instead of an asterisk. But again it may affect you retrieving messages if you need that for retrieving them.
Hope that answers some of the questions regarding direct dialing.