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-boun...@lists.digium.com [mailto:asterisk-users-boun...@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-boun...@lists.digium.com [mailto:asterisk-users-boun...@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 gord...@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.mooijek...@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 -- _ -- 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 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Dialing problem with Polycom phones after SIP update
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-boun...@lists.digium.com [mailto:asterisk-users-boun...@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-boun...@lists.digium.com [mailto:asterisk-users-boun...@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-boun...@lists.digium.com [mailto:asterisk-users-boun...@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 gord...@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.mooijek...@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 -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Dialing problem with Polycom phones after SIP update
Eric Wieling wrote: Polycom (r) UC Software: Configuration File Conversion Utility\ On the pagehttp://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip560.html And for those of us without Windows, this utility appears to work fine under wine. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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
[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. -- _ -- 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
Re: [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.mooijek...@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 -- _ -- 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
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 gord...@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.mooijek...@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 -- _ -- 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 -- _ -- 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
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-boun...@lists.digium.com [mailto:asterisk-users-boun...@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 gord...@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.mooijek...@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 -- _ -- 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 -- _ -- 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
