with the new intercom feature, is there something that needs to be configured on the receiving end? i have followed the config guide to get the button configured on our admins phones, but when the button is pressed it just rings on the boss' phone like a speed dial button would. what needs to be done to get the call to just answer automatically when the IC button is pressed?
thanks in advance!
Hello uhlaw3500,
like with any other feature I always recommend to check the Admin Guide first.
Page 149 covers how to Configure Intercom Calls.
In addition the community's VoIP FAQ contains this post here:
Oct 25, 2011 Question: How can I change my Ringtone or Ring in a special manner for a certain incoming call?
Resolution: Please check => here <=
The updated Post from the 15/10/2014 details how to do this and troubleshoot it in addition.
Please ensure you always check the community FAQ and/or utilize the community search before posting any new topics or follow up post’s.
Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.
Best Regards
Steffen Baier
Polycom Global Services
If official support is required please check how to phone or open a case here
----------------right, as stated in my original post i followed the admin guy when i did the configuration. and it is not working. which is why i posted here. any help would be appreciated.
thanks.
Hello uhlaw3500,
The FAQ post contains detailed information on how to set this up and how to troubleshoot this.
It is pretty basic and requires configuration changes on both the calling and called phone.
Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.
Best Regards
Steffen Baier
Polycom Global Services
If official support is required please check how to phone or open a case here
----------------Bummer. I am not able to get this to work either. It must not function with how asterisk handles alertInfo.
INVITE sip:1001@X.X.X.X:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.60.155:5060;branch=z9hG4bKce9afe85B4434F3B From: "Test" <sip:1000@X.X.X.X>;tag=B74A4DFD-6DA03463 To: <sip:1001@X.X.X.X;user=phone> CSeq: 1 INVITE Call-ID: 980cfccd-24a7d953-dadfcad5@192.168.60.155 Contact: <sip:1000@192.168.60.155:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomVVX-VVX_600-UA/5.2.0.8330 Accept-Language: en Supported: replaces,100rel Allow-Events: conference,talk,hold Alert-Info: intercom-derp Max-Forwards: 70 Content-Type: application/sdp Content-Length: 510 ===================================== INVITE sip:1001@X.X.X.X:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.60.155:5060;branch=z9hG4bKb1c5a8abD1235A9D From: "Test" <sip:1000@X.X.X.X>;tag=B74A4DFD-6DA03463 To: <sip:1001@X.X.X.X;user=phone> CSeq: 2 INVITE Call-ID: 980cfccd-24a7d953-dadfcad5@192.168.60.155 Contact: <sip:1000@192.168.60.155:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomVVX-VVX_600-UA/5.2.0.8330 Accept-Language: en Supported: replaces,100rel Allow-Events: conference,talk,hold Alert-Info: intercom-derp Authorization: Digest username="1000", realm="asterisk", nonce="4aba0dd7", uri="sip:1001@X.X.X.X:5060;user=phone", response="0c715edde28ad696f20a34c407ee338d", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 510 =============================================== INVITE sip:1001@192.168.60.99:5060 SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK24f2fc2d;rport Max-Forwards: 70 From: "Test" <sip:1000@X.X.X.X>;tag=as499168f5 To: <sip:1001@192.168.60.99:5060> Contact: <sip:1000@X.X.X.X:5060> Call-ID: 74ab6c4029f62842188d759d687c5fde@X.X.X.X:5060 CSeq: 102 INVITE User-Agent: FPBX-12.0.6(11.13.1) Date: Tue, 04 Nov 2014 22:37:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer P-Asserted-Identity: "Test" <sip:1000@X.X.X.X> Content-Type: application/sdp Content-Length: 318
I can see the alertinfo injected into the SIP Header by the polycom phone, but as soon as it hits the server and it invites the other extension, it strips the alertinfo, which only rings the phone instead of auto answering.
Hello TylerCMS,
welcome back to the Polycom Community.
The new Intercom features is especially for call servers that do not support the Info Header.
For Asterisk you simply follow the examples => here <= The phone does not to create the Alert-Info as the Asterisk server already does.
Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.
Best Regards
Steffen Baier
Polycom Global Services
If official support is required please check how to phone or open a case here
----------------Hi,
Yes, I have been using a custom softkey for many years, but I figured I would try the new built in intercom function since it allows us to use the BLF buttons to select who we want to intercom without having to type in an extension manually.
I will revert back to my EFK solution. Thanks for the clarification