I have a legacy Rolm / Siemens PBX system within my campus. This legacy system is connected to an Asterisk server via a qsig PRI trunk. Within my legacy system, we have named our lines wth the pattern "User Name" <> "User Department". We have a set of Polycom Soundpoint IP670 phones and a couple of VVX1500D media phones.
Whenever a call is made from the legacy system to one of the Polycom phones, the call does not go through. When I call an extension that terminates in my Asterisk box or a Cisco phone that we have for testing, the call goes through fine. The problem only appears to happen with the Polycom phones. Specifically, when a legacy phone calls an extension on a Polycom phone, the legacy phone does not return a ringing tone or show any call progress. The line on the legacy phone remains silent until the call times out on the Asterisk box and is directed to voicemail. Through this whole process, the Polycom phone never shows any sign of an incoming call. The screen doesn't change and the phone doesn't ring. What we have found is that the <> in the middle of the legacy phone's display name seems to be related to the problem. Any line with a <> in the name can not place a call to a Polycom phone. Any line without a <> in the middle of the name can call the Polycom without a problem. We've tested this behavior on multiple of my legacy lines and on multiple versions of Polycom phone firmware. With no other variables changing, adding <> to the name kept the calls from going through and removing <> from the name allowed the calls to go through normally.
Does anyone know if there is a way to get a Polycom phone to accept an incoming calling name with the symbols <> in the name? I have several thousand phone lines with that symbol pattern in the name, so I want to try to see if there is anything that can be done to correct the problem at the Polycom phone before I even consider having someone go through the legacy lines and remove the characters. One other related question... If these characters are not going to be able to work with the Polycom firmware, are there any other characters that we should avoid using in display names?
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welcome to the Polycom Community.
Without checking myself I am not aware of the restrictions within the SIP RFC on allowed characters in an Invite.
You mentioned you tested several Polycom Software Versions but failed to provide the Version Numbers of the one's you tested.
I can only refer you to your Polycom Reseller to work with them to get a ticket raised with us to check if your call scenario is violating the RFC or if something is broken in our builds.
If you are unable to do so via your reseller you can raise a case with our local support via PPI (payPerIncident) and work with the team on your issue.
Above all depends on you having a wireshark trace taken on a spanned / mirrored port where the Polycom Phone is connected showing the incoming SIP messaging to the Phone.
Polycom Global Services
just as a follow up I had a look internally at the latest upcoming UCS 4.0.x release Notes and could see the following fix:
69469 The display name with special character < or > causes phones to respond with a bad request.
Looking through this voip-69469 case notes seems similar to what you originally described in above message.
I am unable to 100% confirm this without actually seeing a Wireshark Trace but it seems the RFC 3261 allows "<" and ">" are legal characters in the FROM field within an invite.
As suggested above please work with your Polycom Reseller so you can liaise with Polycom Support directly and ensure you provide a wireshark trace, app & boot log and the configuration you have used to provision the phone and mention voip-69469.
Polycom Global Services