Marco Mooijekind
2011-Dec-16 21:02 UTC
[asterisk-users] Dialing problem with Polycom phones after SIP update
Dear all, I'm using serveral Polycom 335 and 650 phones on Asterisk 1.8. All worked well. After applying the new Polycom UC 4.0.1 software update to the phones I notice the following: When dialing an extension, either on- or off hook, the phone immediately displays "SIP URL:..". This does not allow me to enter a regular numeric extension. The Polycom admin manual states that the phone displays the SIP URL input message if the phone is not registered. This is strange since i do see the phones registering themselves in the Asterisk verbose logging. Anyone experiencing this problem , any tips! Thanks in advance! Marco Mooijekind. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111216/259df874/attachment.htm>
Gord Urquhart
2011-Dec-16 21:24 UTC
[asterisk-users] Dialing problem with Polycom phones after SIP update
Does the phone show the line as registered? The little phone icon on the display should be solid for a registered line and just a outline for a unregistered line. Using wireshark to watch the SIP traffic is a easy way to ensure the REGISTER signally is complete. On Fri, Dec 16, 2011 at 1:02 PM, Marco Mooijekind < marco.mooijekind at gmail.com> wrote:> Dear all, > > I'm using serveral Polycom 335 and 650 phones on Asterisk 1.8. > All worked well. After applying the new Polycom UC 4.0.1 software update > to the phones I notice the following: > > When dialing an extension, either on- or off hook, the phone immediately > displays "SIP URL:..". > This does not allow me to enter a regular numeric extension. > The Polycom admin manual states that the phone displays the SIP URL input > message if the phone is not registered. > This is strange since i do see the phones registering themselves in the > Asterisk verbose logging. > > Anyone experiencing this problem , any tips! > > Thanks in advance! > > Marco Mooijekind. > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111216/b0d2cb6a/attachment.htm>