[asterisk-users] Dialing problem with Polycom phones after SIP update

[Eric Wieling]

Polycom (r) UC Software: Configuration File Conversion Utility\

On the page http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip560.html

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Justin Sherrill
Sent: Tuesday, December 20, 2011 8:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dialing problem with Polycom phones after SIP update

Out of curiosity, what is "the Polycom script"?

I obviously haven't moved from 3.2.x firmware yet.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric Wieling
Sent: Friday, December 16, 2011 4:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dialing problem with Polycom phones after SIP update

Did you run your old configurations thru the Polycom script to convert them to work with 3.3+?

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Marco Mooijekind
Sent: Friday, December 16, 2011 4:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dialing problem with Polycom phones after SIP update

Hello Gord,

the line icon is solid black, which should indicate the lines are registered. 

Marco.



On Fri, Dec 16, 2011 at 10:24 PM, Gord Urquhart <gordurq at gmail.com> wrote:


	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.
		
		
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