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SPA301-G2 end to end call

Good day for everyone!

I have two endpoints CISCO SPA301-G2. We still have no any infrastructure (CUCM or Asterisk) for call controlling. But I heard that SIP-phones have feature to call each other without any IP-PBX typing ip-address on phone's keypad?

What feature is responsible for that? Does the SPA300 series have this feature?

3 Replies 3

Jaime Valencia
Cisco Employee
Cisco Employee

SPA endpoints are supported in the SMB area, might want to move your thread over there.

HTH

java

if this helps, please rate

I don't have direct experience with the SPA300 digital phone series. Others can provide more detailed knowledge about the phones.

Older Cisco Analog Terminal Adapters such as the SPA2102 and SPA3102 in their Administration Guide under Enable IP Dialing document how to manually type an sip uri on those ata's. The dialing technique is cumbersome. The SPA300 Admin Guide does mention Enable IP Dialing without detailing the manual dialing technique.

If you have a small number of other locations a much better way to do it than typing the ip address manually is to setup the sip uri as a phone Speed Dial and then simply dial the speed dial number. The phone can save eight Speed Dial entries. With unregistered ip dialing without a PBX call Control system you can encounter problems that need troubleshooting. There are configuration settings for making and answering unregistered calls. You may need port forwarding to solve network firewall problems.

Dan Lukes
VIP Alumni
VIP Alumni

Yes, SPA300 have the same feature set as SPA500 (but PoE) and almost same feature set as SPA gateways like SPA2102 or SPA112.

Well, I have no experience with IP dialing, but there are some generic advises. In appropriate ExtN tab you need to have:

  • Register: No (unless the line is registered to a PBX at the same time)
  • Enable IP Dialing: Yes
  • Make Call Without Reg: Yes
  • Ans Call Without Reg: Yes (if you wish to accept incoming calls as well)

If you needs to call 1234@127.11.22.33:5060, you can pick up the phone and dial "1234*127*11*22*33*5060# (source)

Note that "userid" must use numerical values, the "@" symbol is signified by the * key, and a,b,c, and d are all values between 0 and 255. If no port number is provided the default port is 5060 (source).

I assume there's working (read: suitable to carry SIP and RTP traffic) connectivity between originating phone and called peer. Namely, all NAT issues needs to be solved first.


You may try to use the Dial Plan trick described here to make dialing shortcuts.


Speed Dialing

According source, it is not possible to speed-dial to foreign server. For the purpose of previous sentence, the "foreign server" is the server not defined in a ExtN tab. So if you wish to speed dial phone 1234@ip.otherphone.name you should

  1. configure ip.otherphone.name parameters on ExtN tab
  2. use fnc=sd;ext=1234@$PROXY;vid=N

But there's chance it's not true - I didn't tried to define ext as 1234*127*11*22*33*5060, so try it by self. It may or may not work. All at all, Speed Dial definition needs to be bound to an ExtN with IP dialing enabled. Unless you specify vid=N the vid=1 is assumed.

Also note the "user" part of speed dial's ext= parameter is subject of Dial Plan processing.


Final generic note - sometime the port part of SIP URL: is optional and can be omitted assuming default 5060, but sometime is not optional and must be specified even if target port is the default. So if a configuration doesn't work with no port specified, try it again claiming port number explicitly.

You should be ready to capture SIP packets as well as syslog&debug messages from phone. It will help you to tweak the configuration according your wishes.

I'm unsure the direct IP "numbers" can be saved into local phone's personal directory.

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