Display hangup reason on the VVX 601

[gvegidy]

Hi,

I'm currently evaluating the VVX 601 before deploying a bigger number of phones. One issue that I found is this:

When the pbx hangs up a call because the called number is busy or invalid, I do not see any information about the hangup reason on the screen of the phone, I just hear a busy tone that is generated locally in the phone.

I tried submitting several different reasons from the pbx and I also tried adding an additional Q.850 reason header. But all this made no difference.

Here are sip traces of the replys I tried:

SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 172.17.6.52;rport=5060;received=172.17.6.52;branch=z9hG4bK4d88fb59CA8F1CEE
Call-ID: d8b133c4bd2b2303274da8a5250a682b
From: "42" <sip:42@voip-pbx.m.i2n>;tag=8B0CF535-464A0C6A
To: <sip:41@voip-pbx.m.i2n;user=phone>;tag=8692ef7f-6320-4265-8f90-68dafbeec99f
CSeq: 2 INVITE
Server: Asterisk PBX 16.4.1
Content-Length:  0

SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 172.17.6.52;rport=5060;received=172.17.6.52;branch=z9hG4bK5c3cf6b9AD13D14E
Call-ID: 62039ea3098b4c1d7448efee080a682b
From: "42" <sip:42@voip-pbx.m.i2n>;tag=3E49FC95-A5EDECCA
To: <sip:41@voip-pbx.m.i2n;user=phone>;tag=69dc0ab7-71aa-4540-af42-a6e36af3abbd
CSeq: 2 INVITE
Server: Asterisk PBX 16.4.1
Reason: Q.850;cause=17
Content-Length:  0

SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 172.17.6.52;rport=5060;received=172.17.6.52;branch=z9hG4bK4a849f4c1F54A2E9
Call-ID: 62e210e4cb552952ad4067f58e0a682b
From: "42" <sip:42@voip-pbx.m.i2n>;tag=58808C88-44B97EC5
To: <sip:88@voip-pbx.m.i2n;user=phone>;tag=1536c460-8795-4de9-81bf-8a66d7914a76
CSeq: 2 INVITE
Server: Asterisk PBX 16.4.1
Content-Length:  0

Phones from other vendors show a message like "User busy" or "Number unavailable" when receiving these packets.

How can I show such hangup reasons or causes to the user with the VVX 601?

Firmware version is 6.0.0.4796.

Thanks.

[SteffenBaierUK]

Hello @gvegidy ,

welcome to the Poly Community.

Not to my knowledge and this would require a feature request. Can the PABX not generate an Audible Feedback?

Jan 03, 2013 Question: How can I request a change aka feature request to the current Polycom SIP / UC Software?

Resolution: Please check => here <=

Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Best Regards

Steffen Baier

[gvegidy]

Thanks for your quick reply.

> Not to my knowledge and this would require a feature request.

I bought the phone through an authorized distributor (ALSO Germany) and not through a reseller. The MAC address of our test phone is 64:16:7F:0A:68:2B. Can you help me to look up a reseller to create the feature request?

> Can the PABX not generate an Audible Feedback?

The PBX can of course play back a recorded message. But this isn't as fast and precise as a message on the screen, so the user experience is not as good.

My users are used to ISDN phones and the good cause messages the ISDN network provides. The new SIP based system will always be compared to that and should provide the same level of user experience.

When reading the manual I saw that SIP warning headers are mentioned (option "voIpProt.SIP.header.warning.enable"). But I don't have experience with warning headers yet.

Would it be possible to use these warning headers to transmit the hangup reason to the phone and have the message from the warning header shown on the screen when the pbx disconnected the line?

I could probably patch the pbx to add such a header, but it is a bit of work and I'll only try it if you think that it can be a viable solution.

[SteffenBaierUK]

Hello @gvegidy ,

ALLNET GMBH sold the phone 07/12/2017 so it is out of warranty.

You can use this configuration:

voipProt.SIP.header.warning.enable="1"
voipProt.SIP.header.warning.codes.accept=""

The expected SIP message should look like this:

SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.23.3.201;branch=z9hG4bK28e7211C49CAC92
From: "207" <sip:service@172.23.3.201;user=phone>;tag=675FD13-71E2B364
To: "Someone" <sip:440@172.23.73.165>;tag=1
Call-ID: 1-54180@172.23.73.165
CSeq: 1 BYE
Contact: <sip:172.23.73.165:5060>
Warning: 399 172.23.73.165 "Hello World"
Content-Length: 0

Causing the phone to display:

Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Best Regards

Steffen Baier