Peer to Peer SIP Telephony, Will It Work? Part three

In a previous post, I kinda lost my luster with SIP and WebRTC is just another technology riped with more failures if compared to dial tone.

In the original experiment, I had tested if I could have SIP telephones work peer to peer without much intervention at say an IP PBX or what. In reality, SIP doesn’t work that well in that form, despite Avaya and Aastra selling similar systems over a decade ago.

The test subjects were

  • A Cisco 2801 Router, using Cisco Call Manager Express 8.5 (functioning more for the Voice Register service)_
  • An Avaya 9650 (intended to be a collection to The Museum)
  • Several Mitel IP phones from a friend of mine in Montana (5330s and 5224 self purchased over 5 years ago. Set to SIP from MiNet because I don’t have a functioning ICP phone system)
  • Aastra 480i IP phones (purchased in 2016 to replace some Cisco phones that was a lost cause)

I finally got the Voice Register (for the purposes of this post, these phones act like POTS phones that plug into an Ethernet switch). I also was able to get FXO to finally work with my consumer Xfinity 1FH trunk.

On the Cisco side, the number goes into the voice port and connects to the Private Line Automatic Ringdown to a hunt-group number of 400, set in the voice-register hunt prompt. Members are SIP extensions as set up in voice-register-dn and voice-register-pool (hardware or phone) tags. Inbound calls work fine, as well as outbound calls as well. This never worked previously. Caller ID works fine and all sets can interpret it.

The Problem?

If the trunk has some bad grounding, and starts to do “ghost calling” (previously discussed on The Museum), Aastra and Mitel IP sets will ring “off the hook”. This is because they can handle Direct IP Calling. If you use a cloud IP provider, this could be used nefariously; but given the clear LAN nature of this setup, this could wake you up in the middle of the night. Direct IP calling does not properly disconnect what would be an unanswered call. It will ring until somone picks up and hears the pusedo dial tone.

(Sidenote: In late 2016, I woke up my mother when one of the 480s was ringing off the hook. Ether I misdialed, the network was clogged up, or what. Because we have a 390 (identical just with a analog connection), it was believed that someone was calling late. Configurations must be done very carefully both on the call controller and the device itself.

When making a call from the Avaya 9650 to the Mitel, similar effect, however calling a similar Mitel or Aastra set, I hung up, the call ended on the remote end. When that bad grounding or even perhaps an unlisted number comes into play, the router’s IP address is the “number” and this results into the said issues. If you disconnect the trunk, the Avaya realizes the call is gone, but the Mitels and Aastras won’t let go, and will continue to ring and ring and ring because it thinks there is still a command from the SIP router to tell the phones a call is still incoming.

Despite restricting direct IP calling, these desk sized shenanigans will do whatever it thinks it can do.

The Solution?

The recommended path if you want to sunset Cisco’s proprietary SCCP/Skinny protocol, is to get a 6900 series or the newer 8800 or 7800 series IP Phones. I may get a couple 6900s because a) it can support SIP and b) from my understanding these phones functioned closer to the very original 7900s, ones with  79×0 model numbers. Anything newer is GUI and uses Java, and uses a weird script to provision the VOIP sets. With newer non GUI models;  I can manually set the address to the SIP router, than having to be teased when I toggle down the screens on ether my 7961 or 7970 where it could be a setting I can change but I don’t see the softkey when I press Settings to do so.

What about that Definity?

My Definity just happened to come to me in 2015 from a follower at my other site. I had no intentions to grow it, except for a 4 other proprietary multi wired sets. I have received boatloads of 6400 series sets, some more than I had requested. Given the Definity is also the home of the AUDIX voice mail service (of which I am still trying to undo some major screwups dating a couple years ago), that it would still be the primary PBX in the house. The challenges to getting the Definity to work on residential broadband service has been well known even in Facebook groups. Caller ID is also a hit or miss and can be expensive to do to get line boards. The value added features of additional PC port, caller ID, higher audio quality to a handful of users along with off building extensions, like the garage, doghouse, etc, and other added functionality is the reason why I decided to return to VOIP to at least 4 extensions in the house. Since that thing is running, and taking care of it well, why not use it?


One thought on “Peer to Peer SIP Telephony, Will It Work? Part three

  1. Steven June 26, 2019 / 10:54 pm

    Reblogged this on and commented:

    DevNotes: Your SIP Router is Calling you. Please try catching it! How various SIP phones work differently and could annoy you!

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