Re: [Asterisk-Users] Polycom IP500 Quickstart page or files?
On 9/22/05, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: It's best for you to set up an ftp server instead of a tftp server, but I don't think you'll enjoy setting up a soundpoint phone without either of them. The Polycom Phones page in the wiki was pretty much all I needed to set mine up: http://www.voip-info.org/tiki-index.php?page=Polycom+Phones More specifically, the link most of the way down: http://www.krisk.org/asterisk/pcom/ was the starting point for my whole configuration. thx for the info - naturally I planned to consult the wiki but I asked thinking maybe someone had set up a simple quickstart page somewhere. (called instant gratification :) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500 Quickstart page or files?
Good luck! Soundpoint phones, in my opinion, are worth every second spent setting them up. I was able to get it the ip500 working with asterisk on three lines and it's a beautiful phone for the $200 or so I paid for it. I'll bet the remaining ip500's will be available fairly cheap too, with the 501 out now. Can you tell me (maybe a stupid question) is it possible to write out the current config files? I guess not, but that would be a handy way to start, now that the phone is working. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500 Quickstart page or files?
At 12:50 9/23/2005, you wrote: Good luck! Soundpoint phones, in my opinion, are worth every second spent setting them up. I was able to get it the ip500 working with asterisk on three lines and it's a beautiful phone for the $200 or so I paid for it. I'll bet the remaining ip500's will be available fairly cheap too, with the 501 out now. Can you tell me (maybe a stupid question) is it possible to write out the current config files? I guess not, but that would be a handy way to start, now that the phone is working. You could use FireFox to save, or print out the config ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500 Quickstart page or files?
It's best for you to set up an ftp server instead of a tftp server, but I don't think you'll enjoy setting up a soundpoint phone without either of them. The Polycom Phones page in the wiki was pretty much all I needed to set mine up: http://www.voip-info.org/tiki-index.php?page=Polycom+Phones More specifically, the link most of the way down: http://www.krisk.org/asterisk/pcom/ was the starting point for my whole configuration. Good luck! Soundpoint phones, in my opinion, are worth every second spent setting them up. Mojo Wilson Pickett wrote: Hi, I just got my ip500 back after months of waiting. Is there an easy way to get it hooked up to asterisk without [t]ftp servers and all that or is there a quickstart page somewhere? tia r ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500 Quickstart page or files?
Huh? You can easily configure an IP500 via a web browser. Just point the URL to the IP addr of the telephone. -- Tom Hayden On 9/22/05, Wilson Pickett [EMAIL PROTECTED] wrote: Hi, I just got my ip500 back after months of waiting. Is there an easy way to get it hooked up to asterisk without [t]ftp servers and all that or is there a quickstart page somewhere? tia r ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500 Quickstart page or files?
but you do not get all of the features via a web browser to customize.On 9/22/05, Tom Hayden [EMAIL PROTECTED] wrote:Huh? You can easily configure an IP500 via a web browser. Just pointthe URL to the IP addr of the telephone. --Tom HaydenOn 9/22/05, Wilson Pickett [EMAIL PROTECTED] wrote: Hi, I just got my ip500 back after months of waiting. Is there an easy way to get it hooked up to asterisk without [t]ftp servers and all that or is there a quickstart page somewhere? tia r ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users--Tom___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP TelephonyPhone: 518-631-2855 x205Fax: 518-631-2856 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500 Quickstart page or files?
After dealing with a Poly 301 I rather use the FTP server and config files, even for a single phone, download the manual and stuff from freedomphones.net/polycom Tom Vile wrote: but you do not get all of the features via a web browser to customize. On 9/22/05, *Tom Hayden* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Huh? You can easily configure an IP500 via a web browser. Just point the URL to the IP addr of the telephone. -- Tom Hayden On 9/22/05, Wilson Pickett [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, I just got my ip500 back after months of waiting. Is there an easy way to get it hooked up to asterisk without [t]ftp servers and all that or is there a quickstart page somewhere? tia r ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500 Mass Configurations
Cody Lerum wrote: Can I just pull unchanged lines out? No. Many of the 'defaults' are only defaults because they are in the sample configuration files, and if you upload new files that don't have the defaults, the features will not work the same way (or at all). I have personally seen Call Waiting indications work wrongly without the defaults, for example. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500 / Registration Question?
sip show register will display the sip registrations the server has performed to other peers, not other peers to it this is also true for iax not sure why you split the registrations into 2 instead of using friend, friend works fine for me and I have not heard of any issues of using it for hardphones On Aug 12, 2005, at 11:21 AM, Kenny Kant wrote: Hello again, I have a bunch of Polycom IP500 Phones with Boot 2.6.2 and SIP 1.4.1. I have defined seperate user and peer settings for my extensions as per posts I have seen in here. I can access voicemail...etc and the phone seem work fine. Question: when I do sip show registration there is nothing listed and/or sip show subscriptions nothing is there. But when I do sip show peers I see a list of my phones, same for sip show users. Shouldnt I see my phones as registered or something similar under these two sections? I have them set to register, and like I said they are working fine. Any help? Thanks, Kenny __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500 - Phone TIme
Paul Hales wrote: It now works - but only in the latest (1.5+) firmware releases. Where are the 1.5 releases? I see only 1.4.1 on all the Polycom sites. Regards, Richard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP500 Forward problem codec issue
I'm using ulaw, but seeing this problem as well.
Are you using CVS? I would swear it didn't do this to me in earlier tests, but
it is doing it now. I will try to track down the specific change tonight ...
My solution for now is to Answer() the call before dialing out. I changed all
of my outbound dialing rules from:
[trunklocal]
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
To:
[trunklocal]
exten = _9NXX,1,Answer
exten = _9NXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
This seems to fix it, and I haven't identified any side effects.
I need to do this anyway to workaround an early-media problem I have.
Does it work for you after this change?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Herrick
Sent: Saturday, April 30, 2005 8:49 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Polycom IP500 Forward problem codec issue
Polycom IP500 Forward problem codec issue
All,
Im running the Polycom IP500 phones at several sites. My * server is
at a collocation site and I have complete control of the T1s running to the
remote sites with the IP500 phones. Connectivity to the PSTN is
through a Cisco 2600 with a PRI card. During initial testing I ran
G711/ulaw but have added G729 licenses to the system.
Problem: When the forwarding function on the Polycom phones is enabled the
forward/transfer does work but the caller does not hear any ringing.
During the time that the caller should hear ringing the * console produces
pages of errors.
snip
..
Apr 30 08:41:03 NOTICE[2813]: channel.c:1314 ast_read: Dropping incompatible
voice frame on Local/[EMAIL PROTECTED],2 of format g729 since our native format
has changed to ulaw Apr 30 08:41:03 NOTICE[2813]: channel.c:1314 ast_read:
Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format g729
since our native format has changed to ulaw ..
/snip
I have tested this with the phones behind a PIX firewall with NAT, behind a PIX
firewall without NAT, and without a firewall at all. Nat is not the problem.
In the SIP.conf canreinvite=no so all traffic should be passing through the *
server.
The problem seems to be in the translation of the G729 packets from the
phone to the G711 packets to the router. Only during the forwarding
process is this a problem.
Here is a snip from the console when it worked.
(Note: it worked because I was ringing two phones with this line in my
extensions.conf (exten = --6081,1,Dial(SIP/--6081SIP/--6091,20)
=SNIP
-- Executing Goto(SIP/---..241.35-40400490, TPN|--6081|1) in new
stack
-- Goto (TPN,--6081,1)
-- Executing Dial(SIP/---.---.241.35-40400490,
SIP/--6081SIP/--6091|20) in new stack
-- Called --6081
-- Called --6091
-- Got SIP response 302 Moved Temporarily back from --.92.27
-- Now forwarding SIP/---.---.---.35-40400490 to 'Local/[EMAIL PROTECTED]'
(thanks toSIP/--6091-6268)
-- Executing Dial(Local/[EMAIL PROTECTED],2,
SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/--6081-e558 is ringing
-- SIP/---.---.241.35-f522 is making progress passing it to
Local/[EMAIL PROTECTED],2
-- Local/[EMAIL PROTECTED],1 is making progress passing it to
SIP/---.---.241.35-40400490
-- SIP/---.---.241.35-f522 answered Local/[EMAIL PROTECTED],2
-- Local/[EMAIL PROTECTED],1 answered SIP/---.---.---.35-40400490
== Spawn extension (TPN, --6081, 1) exited non-zero on 'Local/[EMAIL
PROTECTED],2ZOMBIE'
-- Attempting native bridge of SIP/---.---.241.35-40400490 and
SIP/---.---.241.35-f522
==/SNIP
Now here is the console output with a single phone defined in the
extensions.conf (exten = --6081,1,Dial(SIP/--6091,20)
*SNIP
Asterisk-A*CLI
-- Executing Goto(SIP/---.---.241.35-40418730, Charity|--3263|1) in new
stack
-- Goto (Charity,---263,1)
-- Executing Dial(SIP/---.---.241.35-40418730, SIP/--3263|18) in new
stack
-- Called --3263
-- Got SIP response 302 Moved Temporarily back from ---.---.243.5
-- Now forwarding SIP/---.---.241.35-40418730 to 'Local/[EMAIL PROTECTED]'
(thanks to SIP/--3263-f670)
-- Executing Dial(Local/[EMAIL PROTECTED],2,
SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/---.---.241.35-36ca is making progress passing it to
Local/[EMAIL PROTECTED],2
-- Local/[EMAIL PROTECTED],1 is making progress passing it to
SIP/---.---.241.35-40418730 Apr 29 11:30:03 NOTICE[2197]: channel.c:1314
ast_read: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of
format
g729 since our native format has changed to ulaw pages of the same error
Apr 29 11:19:18 NOTICE[2185]: channel.c:1314 ast_read: Dropping incompatible
voice frame on Local/[EMAIL PROTECTED],2 of format
g729 since our native format has changed to ulaw
-- SIP/---.---.241.35-4e1f answered Local/[EMAIL PROTECTED],2
-- Local/[EMAIL PROTECTED],1 answered
Re: [Asterisk-Users] Polycom IP500 Forward problem codec issue
On May 2, 2005 10:31 am, Charlie Watts wrote:
I'm using ulaw, but seeing this problem as well.
Are you using CVS? I would swear it didn't do this to me in earlier tests,
but it is doing it now. I will try to track down the specific change
tonight ...
My solution for now is to Answer() the call before dialing out. I changed
all of my outbound dialing rules from:
Same problem encountered here. My solution is to answer and play a sec of
silence before the dial proceeds - if i don't answer both parties are
connected but can't hear each other.
joe
[trunklocal]
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
To:
[trunklocal]
exten = _9NXX,1,Answer
exten = _9NXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
This seems to fix it, and I haven't identified any side effects.
I need to do this anyway to workaround an early-media problem I have.
Does it work for you after this change?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott Herrick
Sent: Saturday, April 30, 2005 8:49 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Polycom IP500 Forward problem codec issue
Polycom IP500 Forward problem codec issue
All,
Im running the Polycom IP500 phones at several sites. My * server is
at a collocation site and I have complete control of the T1s running to
the remote sites with the IP500 phones. Connectivity to the PSTN is
through a Cisco 2600 with a PRI card. During initial testing I ran
G711/ulaw but have added G729 licenses to the system.
Problem: When the forwarding function on the Polycom phones is enabled the
forward/transfer does work but the caller does not hear any ringing. During
the time that the caller should hear ringing the * console produces pages
of errors. snip
..
Apr 30 08:41:03 NOTICE[2813]: channel.c:1314 ast_read: Dropping
incompatible voice frame on Local/[EMAIL PROTECTED],2 of format g729
since our native format has changed to ulaw Apr 30 08:41:03 NOTICE[2813]:
channel.c:1314 ast_read: Dropping incompatible voice frame on
Local/[EMAIL PROTECTED],2 of format g729 since our native format has
changed to ulaw .. /snip
I have tested this with the phones behind a PIX firewall with NAT, behind a
PIX firewall without NAT, and without a firewall at all. Nat is not the
problem.
In the SIP.conf canreinvite=no so all traffic should be passing through the
* server.
The problem seems to be in the translation of the G729 packets from the
phone to the G711 packets to the router. Only during the forwarding
process is this a problem.
Here is a snip from the console when it worked.
(Note: it worked because I was ringing two phones with this line in my
extensions.conf (exten =
--6081,1,Dial(SIP/--6081SIP/--6091,20)
=SNIP
-- Executing Goto(SIP/---..241.35-40400490, TPN|--6081|1) in
new stack -- Goto (TPN,--6081,1)
-- Executing Dial(SIP/---.---.241.35-40400490,
SIP/--6081SIP/--6091|20) in new stack
-- Called --6081
-- Called --6091
-- Got SIP response 302 Moved Temporarily back from --.92.27
-- Now forwarding SIP/---.---.---.35-40400490 to 'Local/[EMAIL PROTECTED]'
(thanks toSIP/--6091-6268) -- Executing
Dial(Local/[EMAIL PROTECTED],2,
SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/--6081-e558 is ringing
-- SIP/---.---.241.35-f522 is making progress passing it to
Local/[EMAIL PROTECTED],2
-- Local/[EMAIL PROTECTED],1 is making progress passing it to
SIP/---.---.241.35-40400490 -- SIP/---.---.241.35-f522 answered
Local/[EMAIL PROTECTED],2 -- Local/[EMAIL PROTECTED],1 answered
SIP/---.---.---.35-40400490 == Spawn extension (TPN, --6081, 1) exited
non-zero on 'Local/[EMAIL PROTECTED],2ZOMBIE' -- Attempting native
bridge of SIP/---.---.241.35-40400490 and
SIP/---.---.241.35-f522
==/SNIP
Now here is the console output with a single phone defined in the
extensions.conf (exten = --6081,1,Dial(SIP/--6091,20)
*SNIP
Asterisk-A*CLI
-- Executing Goto(SIP/---.---.241.35-40418730, Charity|--3263|1) in
new stack -- Goto (Charity,---263,1)
-- Executing Dial(SIP/---.---.241.35-40418730, SIP/--3263|18) in
new stack -- Called --3263
-- Got SIP response 302 Moved Temporarily back from ---.---.243.5
-- Now forwarding SIP/---.---.241.35-40418730 to
'Local/[EMAIL PROTECTED]' (thanks to SIP/--3263-f670) -- Executing
Dial(Local/[EMAIL PROTECTED],2,
SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/---.---.241.35-36ca is making progress passing it to
Local/[EMAIL PROTECTED],2
-- Local/[EMAIL PROTECTED],1 is making progress passing it to
SIP/---.---.241.35-40418730 Apr 29 11:30:03 NOTICE[2197]: channel.c:1314
ast_read: Dropping incompatible voice frame on
Local/[EMAIL PROTECTED],2 of format g729 since our native format has
changed to ulaw pages of the same error Apr 29
Re: [Asterisk-Users] Polycom IP500 Forward problem codec issue
Joe and Charlie,
YES, that fixed the problem. I did move the whole network to G729 but
it was never a codec problem.
I'm not running CVS, it's 1.0.3 at the moment.
Thanks
Scott H
Joe Baptista wrote:
On May 2, 2005 10:31 am, Charlie Watts wrote:
I'm using ulaw, but seeing this problem as well.
Are you using CVS? I would swear it didn't do this to me in earlier tests,
but it is doing it now. I will try to track down the specific change
tonight ...
My solution for now is to Answer() the call before dialing out. I changed
all of my outbound dialing rules from:
Same problem encountered here. My solution is to answer and play a sec of
silence before the dial proceeds - if i don't answer both parties are
connected but can't hear each other.
joe
[trunklocal]
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
To:
[trunklocal]
exten = _9NXX,1,Answer
exten = _9NXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
This seems to fix it, and I haven't identified any side effects.
I need to do this anyway to workaround an early-media problem I have.
Does it work for you after this change?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott Herrick
Sent: Saturday, April 30, 2005 8:49 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Polycom IP500 Forward problem codec issue
Polycom IP500 Forward problem codec issue
All,
Im running the Polycom IP500 phones at several sites. My * server is
at a collocation site and I have complete control of the T1s running to
the remote sites with the IP500 phones. Connectivity to the PSTN is
through a Cisco 2600 with a PRI card. During initial testing I ran
G711/ulaw but have added G729 licenses to the system.
Problem: When the forwarding function on the Polycom phones is enabled the
forward/transfer does work but the caller does not hear any ringing. During
the time that the caller should hear ringing the * console produces pages
of errors. snip
..
Apr 30 08:41:03 NOTICE[2813]: channel.c:1314 ast_read: Dropping
incompatible voice frame on Local/[EMAIL PROTECTED],2 of format g729
since our native format has changed to ulaw Apr 30 08:41:03 NOTICE[2813]:
channel.c:1314 ast_read: Dropping incompatible voice frame on
Local/[EMAIL PROTECTED],2 of format g729 since our native format has
changed to ulaw .. /snip
I have tested this with the phones behind a PIX firewall with NAT, behind a
PIX firewall without NAT, and without a firewall at all. Nat is not the
problem.
In the SIP.conf canreinvite=no so all traffic should be passing through the
* server.
The problem seems to be in the translation of the G729 packets from the
phone to the G711 packets to the router. Only during the forwarding
process is this a problem.
Here is a snip from the console when it worked.
(Note: it worked because I was ringing two phones with this line in my
extensions.conf (exten =
--6081,1,Dial(SIP/--6081SIP/--6091,20)
=SNIP
-- Executing Goto(SIP/---..241.35-40400490, TPN|--6081|1) in
new stack -- Goto (TPN,--6081,1)
-- Executing Dial(SIP/---.---.241.35-40400490,
SIP/--6081SIP/--6091|20) in new stack
-- Called --6081
-- Called --6091
-- Got SIP response 302 Moved Temporarily back from --.92.27
-- Now forwarding SIP/---.---.---.35-40400490 to 'Local/[EMAIL PROTECTED]'
(thanks toSIP/--6091-6268) -- Executing
Dial(Local/[EMAIL PROTECTED],2,
SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/--6081-e558 is ringing
-- SIP/---.---.241.35-f522 is making progress passing it to
Local/[EMAIL PROTECTED],2
-- Local/[EMAIL PROTECTED],1 is making progress passing it to
SIP/---.---.241.35-40400490 -- SIP/---.---.241.35-f522 answered
Local/[EMAIL PROTECTED],2 -- Local/[EMAIL PROTECTED],1 answered
SIP/---.---.---.35-40400490 == Spawn extension (TPN, --6081, 1) exited
non-zero on 'Local/[EMAIL PROTECTED],2ZOMBIE' -- Attempting native
bridge of SIP/---.---.241.35-40400490 and
SIP/---.---.241.35-f522
==/SNIP
Now here is the console output with a single phone defined in the
extensions.conf (exten = --6081,1,Dial(SIP/--6091,20)
*SNIP
Asterisk-A*CLI
-- Executing Goto(SIP/---.---.241.35-40418730, Charity|--3263|1) in
new stack -- Goto (Charity,---263,1)
-- Executing Dial(SIP/---.---.241.35-40418730, SIP/--3263|18) in
new stack -- Called --3263
-- Got SIP response 302 Moved Temporarily back from ---.---.243.5
-- Now forwarding SIP/---.---.241.35-40418730 to
'Local/[EMAIL PROTECTED]' (thanks to SIP/--3263-f670) -- Executing
Dial(Local/[EMAIL PROTECTED],2,
SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/---.---.241.35-36ca is making progress passing it to
Local/[EMAIL PROTECTED],2
-- Local/[EMAIL PROTECTED],1 is making progress passing it to
SIP/---.---.241.35-40418730 Apr 29 11:30:03 NOTICE[2197]: channel.c:1314
ast_read: Dropping incompatible voice frame on
Local/[EMAIL PROTECTED],2 of format g729
RE: [Asterisk-Users] Polycom IP500 - Phone TIme
It now works - but only in the latest (1.5+) firmware releases. Later, PaulH -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Scobie Sent: Friday, 29 April 2005 6:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 - Phone TIme Paul Hales wrote: And my dreamthat one day Polycom phones will support Australian Daylight savings... But it's only a dream. Unless I am missing something, you don't need to dream about it - set it in ipmid.cfg. Look at the Sip Admim PDF for an explanation of: tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=4 tcpIpApp.sntp.daylightSavings.start.date=1 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=10 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=1 Regards, Richard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP500 - Phone TIme
And my dreamthat one day Polycom phones will support Australian Daylight savings... But it's only a dream. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rick Baranowski Sent: Friday, 29 April 2005 3:03 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom IP500 - Phone TIme We set ours through the web interface on the phone Here is what we use for Phoenix. tcpIpApp.sntp.daylightSavings.enable=0 tcpIpApp.sntp.gmtOffset=-25200 tcpIpApp.sntp.address=207.46.130.100 Rick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Morin Sent: Thursday, April 28, 2005 8:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 - Phone TIme -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Morin Sent: Thursday, April 28, 2005 11:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 - Phone TIme Wiley Siler wrote: Does anyoe know where I can set the timezone in the configuration files? I am in Phoenix, AZ which has a GMT offset of -7 hours but when I enter this into the gmt fields in ipmid.cfg nothing seems to happen. Here are the fields... tcpIpApp.sntp.address= tcpIpApp.sntp.gmtOffset= I never set the timezone in the Polycom config file. I set it in the DHCPd config file. /etc/dhcpd.conf: option ntp-servers 172.17.2.1; option time-offset -21600; Subquestion to this ( although I much prefer setting the offset in the ipmid.cfg file myself ): How do you specify a negative offset when you are using the dhcp server that comes with windows server? Sean You need to use the hex value. Go to http://www.cisco.com/warp/public/109/calculate_hexadecimal_dhcp.html and at the bottom of the page there is a chart with the offsets and the hex value. Dan BUT, the hex values on the cisco site have periods in them...don't include those, just the 8 characters. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500 - Phone TIme
Paul Hales wrote: And my dreamthat one day Polycom phones will support Australian Daylight savings... But it's only a dream. Unless I am missing something, you don't need to dream about it - set it in ipmid.cfg. Look at the Sip Admim PDF for an explanation of: tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=4 tcpIpApp.sntp.daylightSavings.start.date=1 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=10 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=1 Regards, Richard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500 - Phone TIme
Wiley Siler wrote: Does anyoe know where I can set the timezone in the configuration files? I am in Phoenix, AZ which has a GMT offset of -7 hours but when I enter this into the gmt fields in ipmid.cfg nothing seems to happen. Here are the fields... tcpIpApp.sntp.address= tcpIpApp.sntp.gmtOffset= I never set the timezone in the Polycom config file. I set it in the DHCPd config file. /etc/dhcpd.conf: option ntp-servers 172.17.2.1; option time-offset -21600; ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500 - Phone TIme
I've only programmed my IP500 through the web interface. On the web interface, I set the GMT Offset to -6 (for Central). It works for me. BTW, I'm using pool.ntg.org as my sntp server. Now what doesn't work, is that when I change something in the web interface and the phone reboots, the time goes back to last time I booted. The only way to reset it is to power cycle the phone. The SNTP Resync Period doesn't seem to work. Doug At 01:11 PM 4/28/2005, you wrote: Does anyoe know where I can set the timezone in the configuration files? I am in Phoenix, AZ which has a GMT offset of -7 hours but when I enter this into the gmt fields in ipmid.cfg nothing seems to happen. Here are the fields... tcpIpApp.sntp.address= tcpIpApp.sntp.gmtOffset= Are these the correct ones? Thanks, Wiley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP500 - Phone TIme
H my phones are static IP. Sooo... Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling aka ManxPower Sent: Thursday, April 28, 2005 11:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 - Phone TIme Wiley Siler wrote: Does anyoe know where I can set the timezone in the configuration files? I am in Phoenix, AZ which has a GMT offset of -7 hours but when I enter this into the gmt fields in ipmid.cfg nothing seems to happen. Here are the fields... tcpIpApp.sntp.address= tcpIpApp.sntp.gmtOffset= I never set the timezone in the Polycom config file. I set it in the DHCPd config file. /etc/dhcpd.conf: option ntp-servers 172.17.2.1; option time-offset -21600; ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP500 - Phone TIme
Problem found Polycom field below is in seconds not in hours like in the web interface... So for my -7 hour offset have to multiply 3600 (num of secs in 1 hour) times the number of hours (7) Value = -25200 That gives me Mountain time accurately. For anyone else who needs time on their Polycom set... There you go! Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Thursday, April 28, 2005 11:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 - Phone TIme H my phones are static IP. Sooo... Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling aka ManxPower Sent: Thursday, April 28, 2005 11:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 - Phone TIme Wiley Siler wrote: Does anyoe know where I can set the timezone in the configuration files? I am in Phoenix, AZ which has a GMT offset of -7 hours but when I enter this into the gmt fields in ipmid.cfg nothing seems to happen. Here are the fields... tcpIpApp.sntp.address= tcpIpApp.sntp.gmtOffset= I never set the timezone in the Polycom config file. I set it in the DHCPd config file. /etc/dhcpd.conf: option ntp-servers 172.17.2.1; option time-offset -21600; ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP500 - Phone TIme
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Thursday, April 28, 2005 12:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 - Phone TIme H my phones are static IP. Sooo... Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling aka ManxPower Sent: Thursday, April 28, 2005 11:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 - Phone TIme Wiley Siler wrote: Does anyoe know where I can set the timezone in the configuration files? I am in Phoenix, AZ which has a GMT offset of -7 hours but when I enter this into the gmt fields in ipmid.cfg nothing seems to happen. Here are the fields... tcpIpApp.sntp.address= tcpIpApp.sntp.gmtOffset= I never set the timezone in the Polycom config file. I set it in the DHCPd config file. /etc/dhcpd.conf: option ntp-servers 172.17.2.1; option time-offset -21600; If I remember correctly, when setting the offset in the cfg files it needs to be specified in seconds, not hours - for instance in ipmid.cfg: SNTP tcpIpApp.sntp.resyncPeriod=86400 tcpIpApp.sntp.address=131.107.1.10 tcpIpApp.sntp.gmtOffset=-25200. Marty ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500 - Phone TIme
Eric Wieling aka ManxPower wrote: Wiley Siler wrote: Does anyoe know where I can set the timezone in the configuration files? I am in Phoenix, AZ which has a GMT offset of -7 hours but when I enter this into the gmt fields in ipmid.cfg nothing seems to happen. Here are the fields... tcpIpApp.sntp.address= tcpIpApp.sntp.gmtOffset= I never set the timezone in the Polycom config file. I set it in the DHCPd config file. /etc/dhcpd.conf: option ntp-servers 172.17.2.1; option time-offset -21600; Subquestion to this ( although I much prefer setting the offset in the ipmid.cfg file myself ): How do you specify a negative offset when you are using the dhcp server that comes with windows server? Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP500 - Phone TIme
Actually,I just setup the server to update its tiem and DHCP gets set automatically on my domain controller/DHCP Server. Then I set the IP statically in my phones and tell them the SNTP server is my domain controller. http://www.google.com/search?hl=enlr=q=set+time+net+windowsbtnG=Searc h On phones that I have used DHCP with, they all get the time correctly since the server is correct. Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Kennedy Sent: Thursday, April 28, 2005 1:10 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Polycom IP500 - Phone TIme Eric Wieling aka ManxPower wrote: Wiley Siler wrote: Does anyoe know where I can set the timezone in the configuration files? I am in Phoenix, AZ which has a GMT offset of -7 hours but when I enter this into the gmt fields in ipmid.cfg nothing seems to happen. Here are the fields... tcpIpApp.sntp.address= tcpIpApp.sntp.gmtOffset= I never set the timezone in the Polycom config file. I set it in the DHCPd config file. /etc/dhcpd.conf: option ntp-servers 172.17.2.1; option time-offset -21600; Subquestion to this ( although I much prefer setting the offset in the ipmid.cfg file myself ): How do you specify a negative offset when you are using the dhcp server that comes with windows server? Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP500 - Phone TIme
Wiley Siler wrote: Does anyoe know where I can set the timezone in the configuration files? I am in Phoenix, AZ which has a GMT offset of -7 hours but when I enter this into the gmt fields in ipmid.cfg nothing seems to happen. Here are the fields... tcpIpApp.sntp.address= tcpIpApp.sntp.gmtOffset= I never set the timezone in the Polycom config file. I set it in the DHCPd config file. /etc/dhcpd.conf: option ntp-servers 172.17.2.1; option time-offset -21600; Subquestion to this ( although I much prefer setting the offset in the ipmid.cfg file myself ): How do you specify a negative offset when you are using the dhcp server that comes with windows server? Sean You need to use the hex value. Go to http://www.cisco.com/warp/public/109/calculate_hexadecimal_dhcp.html and at the bottom of the page there is a chart with the offsets and the hex value. Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP500 - Phone TIme
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Morin Sent: Thursday, April 28, 2005 11:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 - Phone TIme Wiley Siler wrote: Does anyoe know where I can set the timezone in the configuration files? I am in Phoenix, AZ which has a GMT offset of -7 hours but when I enter this into the gmt fields in ipmid.cfg nothing seems to happen. Here are the fields... tcpIpApp.sntp.address= tcpIpApp.sntp.gmtOffset= I never set the timezone in the Polycom config file. I set it in the DHCPd config file. /etc/dhcpd.conf: option ntp-servers 172.17.2.1; option time-offset -21600; Subquestion to this ( although I much prefer setting the offset in the ipmid.cfg file myself ): How do you specify a negative offset when you are using the dhcp server that comes with windows server? Sean You need to use the hex value. Go to http://www.cisco.com/warp/public/109/calculate_hexadecimal_dhcp.html and at the bottom of the page there is a chart with the offsets and the hex value. Dan BUT, the hex values on the cisco site have periods in them...don't include those, just the 8 characters. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP500 - Phone TIme
We set ours through the web interface on the phone Here is what we use for Phoenix. tcpIpApp.sntp.daylightSavings.enable=0 tcpIpApp.sntp.gmtOffset=-25200 tcpIpApp.sntp.address=207.46.130.100 Rick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Morin Sent: Thursday, April 28, 2005 8:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 - Phone TIme -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Morin Sent: Thursday, April 28, 2005 11:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 - Phone TIme Wiley Siler wrote: Does anyoe know where I can set the timezone in the configuration files? I am in Phoenix, AZ which has a GMT offset of -7 hours but when I enter this into the gmt fields in ipmid.cfg nothing seems to happen. Here are the fields... tcpIpApp.sntp.address= tcpIpApp.sntp.gmtOffset= I never set the timezone in the Polycom config file. I set it in the DHCPd config file. /etc/dhcpd.conf: option ntp-servers 172.17.2.1; option time-offset -21600; Subquestion to this ( although I much prefer setting the offset in the ipmid.cfg file myself ): How do you specify a negative offset when you are using the dhcp server that comes with windows server? Sean You need to use the hex value. Go to http://www.cisco.com/warp/public/109/calculate_hexadecimal_dhcp.html and at the bottom of the page there is a chart with the offsets and the hex value. Dan BUT, the hex values on the cisco site have periods in them...don't include those, just the 8 characters. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom ip500 (Not-Registered)
Dan Morin wrote: I just got a few Polycom IP500s and Ive been following the info in the wiki trying to configure them. From what I can tell, they seem to be setup correctly (wellthey dont work so obviously not) however, when they try to register with Asterisk, the following error shows up in the Logs: Apr 25 15:22:05 NOTICE[1718]: Registration from '' failed for '192.168.0.222' Apr 25 15:22:20 DEBUG[1718]: Auto destroying call '[EMAIL PROTECTED]' Where 192.168.0.222 is the IP of the phone. The two single quotes seem to indicate that no credentials are being passed to * (?). If anyone has any experience with these, please let me know. I can post the configs if that would help. Thanks in advance. Dan Please do. Specifically, sip.conf ( or whatever your sip configuration file for the phones is ), the individual phone settings and your sip.conf file from asterisk ( relevant parts only ). Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom ip500 (Not-Registered)
Make sure the address is the same as the userid, IE reg.X.auth.userId=username and reg.X.address=username -Original Message- From: [EMAIL PROTECTED] on behalf of Sean Kennedy Sent: Mon 4/25/2005 9:11 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Polycom ip500 (Not-Registered) Dan Morin wrote: I just got a few Polycom IP500s and I've been following the info in the wiki trying to configure them. From what I can tell, they seem to be setup correctly (well.they don't work so obviously not.) however, when they try to register with Asterisk, the following error shows up in the Logs: Apr 25 15:22:05 NOTICE[1718]: Registration from '' failed for '192.168.0.222' Apr 25 15:22:20 DEBUG[1718]: Auto destroying call '[EMAIL PROTECTED]' Where 192.168.0.222 is the IP of the phone. The two single quotes seem to indicate that no credentials are being passed to * (?). If anyone has any experience with these, please let me know. I can post the configs if that would help. Thanks in advance. Dan Please do. Specifically, sip.conf ( or whatever your sip configuration file for the phones is ), the individual phone settings and your sip.conf file from asterisk ( relevant parts only ). Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users winmail.dat___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP500 phones do not update time fromtime server
Here is that Part: -- TCP_IP netMon tcpIpApp.netMon.enabled=1 tcpIpApp.netMon.period=30/ SNTP tcpIpApp.sntp.resyncPeriod=86400 tcpIpApp.sntp.address=10.12.14.33 tcpIpApp.sntp.gmtOffset=-25200 tcpIpApp.sntp.daylightSavings.enable=1 tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 tcpIpApp.sntp.daylightSavings.start.month=4 tcpIpApp.sntp.daylightSavings.start.date=1 tcpIpApp.sntp.daylightSavings.start.time=2 tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 tcpIpApp.sntp.daylightSavings.stop.month=10 tcpIpApp.sntp.daylightSavings.stop.date=1 tcpIpApp.sntp.daylightSavings.stop.time=2 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1 tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=1/ port RTP tcpIpApp.port.rtp.filterByIp=1 tcpIpApp.port.rtp.filterByPort=0 tcpIpApp.port.rtp.forceSend= tcpIpApp.port.rtp.mediaPortRangeStart=/ /port /TCP_IP Seshu Kanuri, Seshu (Company IT) wrote: Does anyone know how Polycom 500s will be able to update their time. My setup for a time sync with Public domain Time servers is not successful. Seshu Can you look for the sntp entry in your ipmid.cfg file and post it in it's entirety? Sean NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500 phones do not update time from time server
Kanuri, Seshu (Company IT) wrote: Does anyone know how Polycom 500s will be able to update their time. My setup for a time sync with Public domain Time servers is not successful. Seshu NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users We have to ip300 phones and the time server update works just fine. Make sure you have a gateway and dns defined. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500 phones do not update time from time server
Kanuri, Seshu (Company IT) wrote: Does anyone know how Polycom 500s will be able to update their time. My setup for a time sync with Public domain Time servers is not successful. Seshu Can you look for the sntp entry in your ipmid.cfg file and post it in it's entirety? Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500 phones do not update time from time server
If the time is off by exactly x hours, check the *timezone* in ipmid.cfg. On 4/14/05, Kanuri, Seshu (Company IT) [EMAIL PROTECTED] wrote: Does anyone know how Polycom 500s will be able to update their time. My setup for a time sync with Public domain Time servers is not successful. Seshu NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500 phones do not update time from time server
Kanuri, Seshu (Company IT) wrote: Does anyone know how Polycom 500s will be able to update their time. My setup for a time sync with Public domain Time servers is not successful. Seshu We had a user with a Sonic Wall Firewall who needed to set the snpt server to the IP address of his firewall in order to get the time to update. Not sure of the fw version. -rb ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500 phones do not update time from time server
Kanuri, Seshu (Company IT) wrote: Does anyone know how Polycom 500s will be able to update their time. My setup for a time sync with Public domain Time servers is not successful. We set the NTP server and timezone using ISC DHCPd. option ntp-servers 172.16.7.1; option time-offset -21600; -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP500
That's what I'm about to try, I keep getting pulled off of this project to go do other things. Thanks for the input. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrei (MPI) Sent: Thursday, January 06, 2005 5:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Tim Jackson wrote: Copied your sip.conf and changed the settings and I'm getting the exact same error. I'm also running 1.3.4 of the SIP app for the IP500. Someone has already pointed out that you might have ran into a network problem. What's the network setup between phone and the server? Asterisk CVS-v1-0-01/06/05-00:11:36 built by [EMAIL PROTECTED] on a i686 I was unable to use Asterisk from latest CVS, I am using version from 12/02 CVS. I was getting authorization failed in CLI, and phone could not make calls with CVS-latest Asterisk. Might be something similar in your setup? Just copy /usr/src/asterisk from old server and try make install.. Please, someone, comment on latest changes in CVS for SIP configurations? Might enforced md5 passwords etc? Or anything like that? context=noawnser A typo, right? Andrei ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP500
How about your zapata.conf and zaptel.conf files? Were they updated for the new card? W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Jackson Sent: Thursday, January 06, 2005 12:09 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Polycom IP500 Earlier tonight I moved our * box from an old Compaq w/ 3 X100P clone cards to a 1U IBM server with a TDM04B card. I finally got the card working in the server, but I'm having issues with these Polycom IP500s now. Using the exact same config from the old server I'm getting weird errors. Dial a number on the phone and it gives you dialtone but no user interaction (if that makes sense) then after about 35-40 seconds it displays Line used remotely and hangs up. Inbound calls ring, but you can't answer them, registration seems to be ok, but I'm at a loss. sip.conf: [101] type=friend callerid=Tim Jackson - Home 101 secret=itsasekret username=101 host=dynamic dtmfmode=rfc2833 nat=yes canreinvite=no context=default allow=all OR [101] type=peer callerid=Tim Jackson 101 secret=itsasekret host=dynamic dtmfmode=rfc2833 nat=yes mailbox=101 canreinvite=no context=default disallow=all allow=ulaw app log from phone: 0106005724|res |4|00|[ResFinderC]: Failed to download file BigLogo.bmp, errno 0xdd. 0106005724|net |*|00|Initial log entry. Current logging level 4 key 0106005724||*|00|Initial log entry. Current logging level 4 ssps 0106005724||*|00|Application, comp. 1: Label=PolyDSP Orion Mem2 FS1, Version=1.1.2.0002 17-May-04 15:28 0106005724|ssps |*|00|Application, comp. 1: P/N=3150-11580-112. 0106005724|pps |*|00|Initial log entry. Current logging level 4 sip 0106005724||*|00|Initial log entry. Current logging level 4 cfg 0106005724||4|00|Edit|Parse error 4 with local cfg /ffs0/local/0004f2010524-phone_cfg.zzz 0106005724|app1 |*|00|Initial log entry. Current logging level 4 cfg 0106005724||4|00|Edit|Parse error 4 with local cfg /ffs0/local/0004f2010524-phone_cfg.zzz 0106005724|cfg |5|00|Edit|Simple|error 0x0 when locking (param get) 0106005726|cfg |5|00|Edit|Simple|error 0x0 when locking (param get) 0106005726|so |*|00|[SoNcasC]: App-Ctx (Tim Jackson) [0-101] 0106005727|cfg |5|00|Edit|Simple|error 0x0 when locking (setting lcl.ml.lang=) 0106005727|cfg |5|00|Edit|Simple|error 0x0 when locking (param get) 0106005727|slog |*|00|Initial log entry. Current logging level 4 0106005927|cfg |4|00|Edit|Parse error 4 with local cfg /ffs0/local/0004f2010524-phone_cfg.zzz debug sip peer 101 Sip read: REGISTER sip:192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6 From: Tim Jackson sip:[EMAIL PROTECTED];tag=36767043-B9FDB2DA To: sip:[EMAIL PROTECTED] CSeq: 1 REGISTER Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060;methods=INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Max-Forwards: 70 Expires: 3600 Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.9.202.2 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6;received=192.9.202.2;rpo rt=5060 From: Tim Jackson sip:[EMAIL PROTECTED];tag=36767043-B9FDB2DA To: sip:[EMAIL PROTECTED];tag=as024fe72d Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.9.202.2:5060 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6;received=192.9.202.2;rpo rt=5060 From: Tim Jackson sip:[EMAIL PROTECTED];tag=36767043-B9FDB2DA To: sip:[EMAIL PROTECTED];tag=as024fe72d Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=angelinacounty.net, nonce=243b35d1 Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms asterisk*CLI Sip read: REGISTER sip:192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKf79496d429FB42CD From: Tim Jackson sip:[EMAIL PROTECTED];tag=36767043-B9FDB2DA To: sip:[EMAIL PROTECTED] CSeq: 2 REGISTER Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060;methods=INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Authorization: Digest username=101, realm=angelinacounty.net, nonce=243b35d1, uri=sip:192.9.200.9:5060, response=11f3478d812d35993018150f29fb5e81, algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 192.9.202.2 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKf79496d429FB42CD;received=192.9.202.2;rpo
RE: [Asterisk-Users] Polycom IP500
They were updated, to reflect the new card. And I can call in perfectly. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Thursday, January 06, 2005 2:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 How about your zapata.conf and zaptel.conf files? Were they updated for the new card? W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Jackson Sent: Thursday, January 06, 2005 12:09 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Polycom IP500 Earlier tonight I moved our * box from an old Compaq w/ 3 X100P clone cards to a 1U IBM server with a TDM04B card. I finally got the card working in the server, but I'm having issues with these Polycom IP500s now. Using the exact same config from the old server I'm getting weird errors. Dial a number on the phone and it gives you dialtone but no user interaction (if that makes sense) then after about 35-40 seconds it displays Line used remotely and hangs up. Inbound calls ring, but you can't answer them, registration seems to be ok, but I'm at a loss. sip.conf: [101] type=friend callerid=Tim Jackson - Home 101 secret=itsasekret username=101 host=dynamic dtmfmode=rfc2833 nat=yes canreinvite=no context=default allow=all OR [101] type=peer callerid=Tim Jackson 101 secret=itsasekret host=dynamic dtmfmode=rfc2833 nat=yes mailbox=101 canreinvite=no context=default disallow=all allow=ulaw app log from phone: 0106005724|res |4|00|[ResFinderC]: Failed to download file BigLogo.bmp, errno 0xdd. 0106005724|net |*|00|Initial log entry. Current logging level 4 key 0106005724||*|00|Initial log entry. Current logging level 4 ssps 0106005724||*|00|Application, comp. 1: Label=PolyDSP Orion Mem2 FS1, Version=1.1.2.0002 17-May-04 15:28 0106005724|ssps |*|00|Application, comp. 1: P/N=3150-11580-112. 0106005724|pps |*|00|Initial log entry. Current logging level 4 sip 0106005724||*|00|Initial log entry. Current logging level 4 cfg 0106005724||4|00|Edit|Parse error 4 with local cfg /ffs0/local/0004f2010524-phone_cfg.zzz 0106005724|app1 |*|00|Initial log entry. Current logging level 4 cfg 0106005724||4|00|Edit|Parse error 4 with local cfg /ffs0/local/0004f2010524-phone_cfg.zzz 0106005724|cfg |5|00|Edit|Simple|error 0x0 when locking (param get) 0106005726|cfg |5|00|Edit|Simple|error 0x0 when locking (param get) 0106005726|so |*|00|[SoNcasC]: App-Ctx (Tim Jackson) [0-101] 0106005727|cfg |5|00|Edit|Simple|error 0x0 when locking (setting lcl.ml.lang=) 0106005727|cfg |5|00|Edit|Simple|error 0x0 when locking (param get) 0106005727|slog |*|00|Initial log entry. Current logging level 4 0106005927|cfg |4|00|Edit|Parse error 4 with local cfg /ffs0/local/0004f2010524-phone_cfg.zzz debug sip peer 101 Sip read: REGISTER sip:192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6 From: Tim Jackson sip:[EMAIL PROTECTED];tag=36767043-B9FDB2DA To: sip:[EMAIL PROTECTED] CSeq: 1 REGISTER Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060;methods=INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Max-Forwards: 70 Expires: 3600 Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.9.202.2 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6;received=192.9.202.2;rpo rt=5060 From: Tim Jackson sip:[EMAIL PROTECTED];tag=36767043-B9FDB2DA To: sip:[EMAIL PROTECTED];tag=as024fe72d Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.9.202.2:5060 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6;received=192.9.202.2;rpo rt=5060 From: Tim Jackson sip:[EMAIL PROTECTED];tag=36767043-B9FDB2DA To: sip:[EMAIL PROTECTED];tag=as024fe72d Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=angelinacounty.net, nonce=243b35d1 Content-Length: 0 to 192.9.202.2:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms asterisk*CLI Sip read: REGISTER sip:192.9.200.9:5060 SIP/2.0 Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bKf79496d429FB42CD From: Tim Jackson sip:[EMAIL PROTECTED];tag=36767043-B9FDB2DA To: sip:[EMAIL PROTECTED] CSeq: 2 REGISTER Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060;methods=INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Authorization: Digest username=101, realm=angelinacounty.net, nonce=243b35d1, uri=sip:192.9.200.9:5060
RE: [Asterisk-Users] Polycom IP500
FROM MY SIP.CONF
[1000]
type=friend
host=dynamic
context=local
allow=ulaw
secret=YESITIS
callerid=Front Desk 1000
[EMAIL PROTECTED]
dtmfmode=rfc2833
nat=0
FROM MY EXTENSION.CONF
[local]
include = mainmenu
include = parkedcalls
include = trunklocal
include = trunktollfree
include = trunkld
include = trunkint
include = sip
YOURS
sip.conf:
[101]
type=friend
callerid=Tim Jackson - Home 101
secret=itsasekret
username=101
host=dynamic
dtmfmode=rfc2833
nat=yes
canreinvite=no
context=default
allow=all
May as well just set allow=ulaw unless you are eally using something
else.
Does your extensions.conf have a context default which is set up with
something like...
[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten = _9XXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9XXX,2,Congestion
exten = _9480NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9480NXX,2,Congestion
exten = _9602NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9602NXX,2,Congestion
exten = _9623NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9623NXX,2,Congestion
Where TRUNK is passed in from a global?
MINE GLOBALS
;Trunk Info
TRUNK=ZAP/g1 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
On a guess, it seems like your context for incoming could be correct and
your context for out may be wrong.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim
Jackson
Sent: Thursday, January 06, 2005 7:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Polycom IP500
They were updated, to reflect the new card. And I can call in perfectly.
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Thursday, January 06, 2005 2:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Polycom IP500
How about your zapata.conf and zaptel.conf files? Were they updated for
the new card?
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim
Jackson
Sent: Thursday, January 06, 2005 12:09 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Polycom IP500
Earlier tonight I moved our * box from an old Compaq w/ 3 X100P clone
cards to a 1U IBM server with a TDM04B card. I finally got the card
working in the server, but I'm having issues with these Polycom IP500s
now. Using the exact same config from the old server I'm getting weird
errors. Dial a number on the phone and it gives you dialtone but no user
interaction (if that makes sense) then after about 35-40 seconds it
displays Line used remotely and hangs up. Inbound calls ring, but you
can't answer them, registration seems to be ok, but I'm at a loss.
sip.conf:
[101]
type=friend
callerid=Tim Jackson - Home 101
secret=itsasekret
username=101
host=dynamic
dtmfmode=rfc2833
nat=yes
canreinvite=no
context=default
allow=all
OR
[101]
type=peer
callerid=Tim Jackson 101
secret=itsasekret
host=dynamic
dtmfmode=rfc2833
nat=yes
mailbox=101
canreinvite=no
context=default
disallow=all
allow=ulaw
app log from phone:
0106005724|res |4|00|[ResFinderC]: Failed to download file BigLogo.bmp,
errno 0xdd.
0106005724|net |*|00|Initial log entry. Current logging level 4 key
0106005724||*|00|Initial log entry. Current logging level 4 ssps
0106005724||*|00|Application, comp. 1: Label=PolyDSP Orion Mem2
FS1, Version=1.1.2.0002 17-May-04 15:28
0106005724|ssps |*|00|Application, comp. 1: P/N=3150-11580-112.
0106005724|pps |*|00|Initial log entry. Current logging level 4 sip
0106005724||*|00|Initial log entry. Current logging level 4 cfg
0106005724||4|00|Edit|Parse error 4 with local cfg
/ffs0/local/0004f2010524-phone_cfg.zzz
0106005724|app1 |*|00|Initial log entry. Current logging level 4 cfg
0106005724||4|00|Edit|Parse error 4 with local cfg
/ffs0/local/0004f2010524-phone_cfg.zzz
0106005724|cfg |5|00|Edit|Simple|error 0x0 when locking (param get)
0106005726|cfg |5|00|Edit|Simple|error 0x0 when locking (param get)
0106005726|so |*|00|[SoNcasC]: App-Ctx (Tim Jackson) [0-101]
0106005727|cfg |5|00|Edit|Simple|error 0x0 when locking (setting
lcl.ml.lang=)
0106005727|cfg |5|00|Edit|Simple|error 0x0 when locking (param get)
0106005727|slog |*|00|Initial log entry. Current logging level 4
0106005927|cfg |4|00|Edit|Parse error 4 with local cfg
/ffs0/local/0004f2010524-phone_cfg.zzz
debug sip peer 101
Sip read:
REGISTER sip:192.9.200.9:5060 SIP/2.0
Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6
From: Tim Jackson sip:[EMAIL PROTECTED];tag=36767043-B9FDB2DA
To: sip:[EMAIL PROTECTED]
CSeq: 1 REGISTER
Call-ID: [EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]:5060;methods=INVITE, ACK, BYE, CANCEL,
OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4
Max-Forwards: 70
Expires: 3600
Content-Length: 0
11 headers
Re: [Asterisk-Users] Polycom IP500 - problems with multiple simultaneous calls
Noah Miller wrote: I guess the phone just doesn't register as busy when there is only one call on a line. It has to have two calls on a line appearance to register as busy. Has anyone figured out how to disable this hold feature and just have the second call go to the second line, the third call to the third line, etc? This is call waiting; if you can find a way to disable call waiting in the Polycom config, it will work the way you desire. I've looked for it before (briefly) and couldn't find it, but if you can figure out how to do it please let us all know :-) There are ways using SetGroup and CheckGroup to accomplish this in the dialplan, but it's better to let the phone handle it (because if it works, you can go back to registering three line appearances to the same SIP username and it will work properly). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500
Tim, For what it's worth, from my working sip.conf for Polycoms: [2010] type=friend username=usr2010 callerid=MyName 2010 secret=nobodyknowswhatitis host=dynamic dtmfmode=inband context=admin defaultip=192.168.1.10 progressinband=no Notes: dtmfmode=inband and progressinband=no - that seems to be recommended from * sample sip.conf file for Polycoms. defaultip= setting helped with network issues, not only with Polycoms, with Cisco 7940 as well. Also in main sip.conf: [general] ... disallow=all ; Allow all codecs allow=ulaw,alaw maxexpirey=7200 defaultexpirey=3600 canreinvite=no Also, if you are not behind NAT, why nat=yes? And if NAT is in use, what is your network infrastructure? Also, what is Polycom's SIP firmware version? (mine is 1.3.4 from October 2004). And of course: what is Asterisk and zaptel version? What is your zapata.conf (just curious)? Andrei Tim Jackson wrote: Earlier tonight I moved our * box from an old Compaq w/ 3 X100P clone cards to a 1U IBM server with a TDM04B card. I finally got the card working in the server, but I'm having issues with these Polycom IP500s now. Using the exact same config from the old server I'm getting weird errors. Dial a number on the phone and it gives you dialtone but no user interaction (if that makes sense) then after about 35-40 seconds it displays Line used remotely and hangs up. Inbound calls ring, but you can't answer them, registration seems to be ok, but I'm at a loss. sip.conf: [101] type=friend callerid=Tim Jackson - Home 101 secret=itsasekret username=101 host=dynamic dtmfmode=rfc2833 nat=yes canreinvite=no context=default allow=all ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500
snip Inbound calls ring, but you can't answer them, registration seems to be ok, but I'm at a loss. /snip Had this same behavior with IP500s last week. For us, the solution was NAT related. Our server is on the LAN with the phones, but I was doing 1-1 NAT of an IP on the outside to the server. I changed to doing port-forwarding of the IAX2 port (4869 or whatever it is) from the firewall, and all was well. HTH, -Ron ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP500
This isn't a dialplan issue, it's a SIP issue. The same dialplan and
sip.conf are working perfectly with the other server.
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Thursday, January 06, 2005 9:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Polycom IP500
FROM MY SIP.CONF
[1000]
type=friend
host=dynamic
context=local
allow=ulaw
secret=YESITIS
callerid=Front Desk 1000
[EMAIL PROTECTED]
dtmfmode=rfc2833
nat=0
FROM MY EXTENSION.CONF
[local]
include = mainmenu
include = parkedcalls
include = trunklocal
include = trunktollfree
include = trunkld
include = trunkint
include = sip
YOURS
sip.conf:
[101]
type=friend
callerid=Tim Jackson - Home 101
secret=itsasekret
username=101
host=dynamic
dtmfmode=rfc2833
nat=yes
canreinvite=no
context=default
allow=all
May as well just set allow=ulaw unless you are eally using something
else.
Does your extensions.conf have a context default which is set up with
something like...
[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten = _9XXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9XXX,2,Congestion
exten = _9480NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9480NXX,2,Congestion
exten = _9602NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9602NXX,2,Congestion
exten = _9623NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9623NXX,2,Congestion
Where TRUNK is passed in from a global?
MINE GLOBALS
;Trunk Info
TRUNK=ZAP/g1 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
On a guess, it seems like your context for incoming could be correct and
your context for out may be wrong.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim
Jackson
Sent: Thursday, January 06, 2005 7:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Polycom IP500
They were updated, to reflect the new card. And I can call in perfectly.
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
Siler
Sent: Thursday, January 06, 2005 2:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Polycom IP500
How about your zapata.conf and zaptel.conf files? Were they updated for
the new card?
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim
Jackson
Sent: Thursday, January 06, 2005 12:09 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Polycom IP500
Earlier tonight I moved our * box from an old Compaq w/ 3 X100P clone
cards to a 1U IBM server with a TDM04B card. I finally got the card
working in the server, but I'm having issues with these Polycom IP500s
now. Using the exact same config from the old server I'm getting weird
errors. Dial a number on the phone and it gives you dialtone but no user
interaction (if that makes sense) then after about 35-40 seconds it
displays Line used remotely and hangs up. Inbound calls ring, but you
can't answer them, registration seems to be ok, but I'm at a loss.
sip.conf:
[101]
type=friend
callerid=Tim Jackson - Home 101
secret=itsasekret
username=101
host=dynamic
dtmfmode=rfc2833
nat=yes
canreinvite=no
context=default
allow=all
OR
[101]
type=peer
callerid=Tim Jackson 101
secret=itsasekret
host=dynamic
dtmfmode=rfc2833
nat=yes
mailbox=101
canreinvite=no
context=default
disallow=all
allow=ulaw
app log from phone:
0106005724|res |4|00|[ResFinderC]: Failed to download file BigLogo.bmp,
errno 0xdd.
0106005724|net |*|00|Initial log entry. Current logging level 4 key
0106005724||*|00|Initial log entry. Current logging level 4 ssps
0106005724||*|00|Application, comp. 1: Label=PolyDSP Orion Mem2
FS1, Version=1.1.2.0002 17-May-04 15:28
0106005724|ssps |*|00|Application, comp. 1: P/N=3150-11580-112.
0106005724|pps |*|00|Initial log entry. Current logging level 4 sip
0106005724||*|00|Initial log entry. Current logging level 4 cfg
0106005724||4|00|Edit|Parse error 4 with local cfg
/ffs0/local/0004f2010524-phone_cfg.zzz
0106005724|app1 |*|00|Initial log entry. Current logging level 4 cfg
0106005724||4|00|Edit|Parse error 4 with local cfg
/ffs0/local/0004f2010524-phone_cfg.zzz
0106005724|cfg |5|00|Edit|Simple|error 0x0 when locking (param get)
0106005726|cfg |5|00|Edit|Simple|error 0x0 when locking (param get)
0106005726|so |*|00|[SoNcasC]: App-Ctx (Tim Jackson) [0-101]
0106005727|cfg |5|00|Edit|Simple|error 0x0 when locking (setting
lcl.ml.lang=)
0106005727|cfg |5|00|Edit|Simple|error 0x0 when locking (param get)
0106005727|slog |*|00|Initial log entry. Current logging level 4
0106005927|cfg |4|00|Edit|Parse error 4 with local cfg
/ffs0/local/0004f2010524-phone_cfg.zzz
debug sip peer 101
Sip read:
REGISTER sip:192.9.200.9:5060 SIP/2.0
Via: SIP/2.0/UDP 192.9.202.2:5060;branch=z9hG4bK7ad588653F72B1C6
RE: [Asterisk-Users] Polycom IP500
Copied your sip.conf and changed the settings and I'm getting the exact same error. I'm also running 1.3.4 of the SIP app for the IP500. Asterisk CVS-v1-0-01/06/05-00:11:36 built by [EMAIL PROTECTED] on a i686 running Linux [channels] echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=2 txgain=2 usecallerid=yes context=inbound-pots signalling=fxs_ks callerid=Unknown Caller group = 1 channel = 1-2 echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=2 txgain=2 usecallerid=yes context=noawnser signalling=fxs_ks callerid=Unknown caller group = 1 channel = 3-4 -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrei (MPI) Sent: Thursday, January 06, 2005 11:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Tim, For what it's worth, from my working sip.conf for Polycoms: [2010] type=friend username=usr2010 callerid=MyName 2010 secret=nobodyknowswhatitis host=dynamic dtmfmode=inband context=admin defaultip=192.168.1.10 progressinband=no Notes: dtmfmode=inband and progressinband=no - that seems to be recommended from * sample sip.conf file for Polycoms. defaultip= setting helped with network issues, not only with Polycoms, with Cisco 7940 as well. Also in main sip.conf: [general] ... disallow=all ; Allow all codecs allow=ulaw,alaw maxexpirey=7200 defaultexpirey=3600 canreinvite=no Also, if you are not behind NAT, why nat=yes? And if NAT is in use, what is your network infrastructure? Also, what is Polycom's SIP firmware version? (mine is 1.3.4 from October 2004). And of course: what is Asterisk and zaptel version? What is your zapata.conf (just curious)? Andrei Tim Jackson wrote: Earlier tonight I moved our * box from an old Compaq w/ 3 X100P clone cards to a 1U IBM server with a TDM04B card. I finally got the card working in the server, but I'm having issues with these Polycom IP500s now. Using the exact same config from the old server I'm getting weird errors. Dial a number on the phone and it gives you dialtone but no user interaction (if that makes sense) then after about 35-40 seconds it displays Line used remotely and hangs up. Inbound calls ring, but you can't answer them, registration seems to be ok, but I'm at a loss. sip.conf: [101] type=friend callerid=Tim Jackson - Home 101 secret=itsasekret username=101 host=dynamic dtmfmode=rfc2833 nat=yes canreinvite=no context=default allow=all ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500
Tim Jackson wrote: Copied your sip.conf and changed the settings and I'm getting the exact same error. I'm also running 1.3.4 of the SIP app for the IP500. Someone has already pointed out that you might have ran into a network problem. What's the network setup between phone and the server? Asterisk CVS-v1-0-01/06/05-00:11:36 built by [EMAIL PROTECTED] on a i686 I was unable to use Asterisk from latest CVS, I am using version from 12/02 CVS. I was getting authorization failed in CLI, and phone could not make calls with CVS-latest Asterisk. Might be something similar in your setup? Just copy /usr/src/asterisk from old server and try make install.. Please, someone, comment on latest changes in CVS for SIP configurations? Might enforced md5 passwords etc? Or anything like that? context=noawnser A typo, right? Andrei ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP500
Check the polycom config for the phone for msg.mwi.1.subscribe. If you have you mailbox listed in there remove it, this field needs to be blank. Instead include your mailbox in your sip.conf for the polycom. Reboot your phone and line 1 should work again. The subscribe seems to be tying up line 1 and keeping you from receiving or making calls. -- Jeremy Apple [EMAIL PROTECTED] VCCH, Inc. On Thu, 2005-01-06 at 15:17 -0600, Tim Jackson wrote: Copied your sip.conf and changed the settings and I'm getting the exact same error. I'm also running 1.3.4 of the SIP app for the IP500. Asterisk CVS-v1-0-01/06/05-00:11:36 built by [EMAIL PROTECTED] on a i686 running Linux [channels] echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=2 txgain=2 usecallerid=yes context=inbound-pots signalling=fxs_ks callerid=Unknown Caller group = 1 channel = 1-2 echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=2 txgain=2 usecallerid=yes context=noawnser signalling=fxs_ks callerid=Unknown caller group = 1 channel = 3-4 -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrei (MPI) Sent: Thursday, January 06, 2005 11:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Tim, For what it's worth, from my working sip.conf for Polycoms: [2010] type=friend username=usr2010 callerid=MyName 2010 secret=nobodyknowswhatitis host=dynamic dtmfmode=inband context=admin defaultip=192.168.1.10 progressinband=no Notes: dtmfmode=inband and progressinband=no - that seems to be recommended from * sample sip.conf file for Polycoms. defaultip= setting helped with network issues, not only with Polycoms, with Cisco 7940 as well. Also in main sip.conf: [general] ... disallow=all ; Allow all codecs allow=ulaw,alaw maxexpirey=7200 defaultexpirey=3600 canreinvite=no Also, if you are not behind NAT, why nat=yes? And if NAT is in use, what is your network infrastructure? Also, what is Polycom's SIP firmware version? (mine is 1.3.4 from October 2004). And of course: what is Asterisk and zaptel version? What is your zapata.conf (just curious)? Andrei Tim Jackson wrote: Earlier tonight I moved our * box from an old Compaq w/ 3 X100P clone cards to a 1U IBM server with a TDM04B card. I finally got the card working in the server, but I'm having issues with these Polycom IP500s now. Using the exact same config from the old server I'm getting weird errors. Dial a number on the phone and it gives you dialtone but no user interaction (if that makes sense) then after about 35-40 seconds it displays Line used remotely and hangs up. Inbound calls ring, but you can't answer them, registration seems to be ok, but I'm at a loss. sip.conf: [101] type=friend callerid=Tim Jackson - Home 101 secret=itsasekret username=101 host=dynamic dtmfmode=rfc2833 nat=yes canreinvite=no context=default allow=all ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP500 - problems with multiplesimultaneous calls
I have these very phones and took me a while to figure this out myself. The phone considers each line registration to be a line with a second line. So, call line while someone is on a call and another instance will appear below. That means you only need one registered instance for the phones to get two incoming calls. If however you want to have a second registered extension rung if the first is busy, you have to configure that in Asterisk and the phone so that it moves to the next Dial command on a busy from the phone. However, to get a busy from the phone, you have to configure the phone not to have the second ringing instance for the single registration. Confused? I was... So... Your phone display looks some6thing like this... LINE 1 LINE 2 BLAH In order to make line 1 toll into line 2, you need to make the phone SIP back that it is busy when Line 1 is engaged. Default out of the box is to show the incoming call on line below the current call on line one. This will be confitgured in the mac-phone.cfg file for your phone. Probably in this section... divert divert.1.contact= divert.1.autoOnSpecificCaller=1 divert.2.contact= divert.2.autoOnSpecificCaller=1 divert.3.contact= divert.3.autoOnSpecificCaller=1 divert.4.contact= divert.4.autoOnSpecificCaller=1 divert.5.contact= divert.5.autoOnSpecificCaller=1 divert.6.contact= divert.6.autoOnSpecificCaller=1 fwd divert.fwd.1.enabled=0 divert.fwd.2.enabled=0 divert.fwd.3.enabled=0 divert.fwd.4.enabled=0 divert.fwd.5.enabled=0 divert.fwd.6.enabled=0/ busy divert.busy.1.enabled=0 divert.busy.1.contact= divert.busy.2.enabled=0 divert.busy.2.contact= divert.busy.3.enabled=0 divert.busy.3.contact= divert.busy.4.enabled=0 divert.busy.4.contact= divert.busy.5.enabled=0 divert.busy.5.contact= divert.busy.6.enabled=0 divert.busy.6.contact=/ noanswer divert.noanswer.1.enabled=0 divert.noanswer.1.timeout=60 divert.noanswer.1.contact= divert.noanswer.2.enabled=0 divert.noanswer.2.timeout=60 divert.noanswer.2.contact= divert.noanswer.3.enabled=0 divert.noanswer.3.timeout=60 divert.noanswer.3.contact= divert.noanswer.4.enabled=1 divert.noanswer.4.timeout=60 divert.noanswer.4.contact= divert.noanswer.5.enabled=0 divert.noanswer.5.timeout=60 divert.noanswer.5.contact= divert.noanswer.6.enabled=0 divert.noanswer.6.timeout=60 divert.noanswer.6.contact=/ dnd divert.dnd.1.enabled=0 divert.dnd.1.contact= divert.dnd.2.enabled=0 divert.dnd.2.contact= divert.dnd.3.enabled=0 divert.dnd.3.contact= divert.dnd.4.enabled=0 divert.dnd.4.contact= divert.dnd.5.enabled=0 divert.dnd.5.contact= divert.dnd.6.enabled=0 divert.dnd.6.contact=/ /divert Note the No Answer and Busy Fwd parameters. These will be yor starting point and should be docuented in the Polycom Admin PDF. The * side should just be an extension definition that moves from one Line to the Next if busy. The bad news is it seems like some of the features for these phones just don't work so I cannot guarantee that what I sent will work. I finally gave up on this for lack of tiem and just registered a single Line and let the phone ring in a second line to the extension. Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: Wednesday, January 05, 2005 5:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Polycom IP500 - problems with multiplesimultaneous calls Hi All - I've got a load of Polycom phones, and for the most part, I think they're great, but one thing that is bugging the heck out of me (and my users) is the on-hold feature. When you're on a call, and another one comes in, it doesn't ring the second line appearance on the phone, even though I have it registered separately, and I've tried to make my dialplan go to the second appearance/registration. Instead, the second call rings on the first line, and allows you to put the first call on hold, and take the second call. To do so, though, you have to press the little down arrow and then press Answer. When the third call comes in, it will ring the 2nd line. I find this to be non-intuitive, but I can get used to it. My receptionists, however, are finding it REALLY painful. I'd just like to make the first call go to line appearance 1, the second simultaneous call to go to line appearance 2, etc. Maybe somebody figured out a neat dialplan thing to get this done. My config that doesn't do what I want looks like this: ; The first line appearance is registered to 18, the second to 1802, and the third to 1803 exten = 18,1,Dial(SIP/18,20) exten = 18,2,Voicemail(u18) exten = 18,102,Goto(1802,1) exten = 1802,1,Dial(SIP/1802,20) exten = 1802,2,Voicemail(b18) exten = 1802,102,Goto(1803,1) exten = 1803,1,Dial(SIP/1803,20) exten = 1803,2,Voicemail(b18) exten = 1803,102,Voicemail(b18) exten = 1803,103,Hangup I guess the phone just doesn't register as busy when there is
RE: [Asterisk-Users] Polycom IP500
http://www.freedomphones.net/polycom/files/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Sent: Sunday, December 05, 2004 4:14 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Polycom IP500 Does anyone have a location to download the latest Polycom firmware etc? Other than the extranet site, because I am not a reseller, there fore I have no login. [minirant] And shouldn't end users be granted access to this kind of thing anyway? Geeze [/minirant] Thanks, Chris Cherry -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.289 / Virus Database: 265.4.5 - Release Date: 12/3/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500
Does anyone have a location to download the latest Polycom firmware etc? Other than the extranet site, because I am not a reseller, there fore I have no login. [minirant] And shouldn't end users be granted access to this kind of thing anyway? Geeze [/minirant] Found the same thing. IP phones are great, corporate policy sucks big time. The only way to get firmware from Polycom (their policy, not mine) is to become a certified reseller. Check the asterisk archives as someone has posted a site several times with current firmware. Its probably in the wiki as well. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500
--On Monday, December 06, 2004 7:11 AM -0600 Rich Adamson [EMAIL PROTECTED] wrote: The only way to get firmware from Polycom (their policy, not mine) is to become a certified reseller. Check the asterisk archives as someone has posted a site several times with current firmware. Its probably in the wiki as well. There is a reseller (1av8r2fly) who advertises on eBay as OurPhoneBooth He seems very Asterisk friendly. I plan to buy a Polycom phone from him if he is an authorized reseller and will support the phone. Has anyone had dealings with this vendor? (Maybe he's even on the list. Please chime in.) /edg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500
Im working on getting 1.3.4.. Will post. On Thu, 02 Dec 2004 13:15:29 -0500, Andrei (MPI) [EMAIL PROTECTED] wrote: I would sniff UDP packets with tcpdump and see what is going on: separately in 10.24.102 and 10.24.100 segment. Do you have any general NAT settings in sip.conf, like externip and local network etc? How about RTP port range? It would be easier to understand your setup if you post your sip.conf, as well as phone's MAC.cfg file (which may be too big for this newsgroup - email is fine). Also, asterisk sip registration and error messages would give us some more info. Andrei Tim Jackson wrote: I've already added nat=yes. Nothing fancy/special on the routing between these. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrei (MPI) Sent: Thursday, December 02, 2004 10:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Hello Tim, You are saying that: phone is on 10.24.102.0/24 and Asterisk resides on 10.24.100.0/24. Honestly, I see at least one hop forwarding here and possible network issues right away. At one moment, not in a NAT environment, but having phone IP 172.16.100.xxx and Linux server ip 172.16.1.xxx - I had to add nat=yes in sip.conf in order to get the phone to work. Sincerely, Andrei Tim Jackson wrote: Theres no NAT going on here. Just 1 router in between, phones reside on 10.24.102.0/24 Asterisk resides on 10.24.100.0/24, so this isn't a NAT problem. Any other ideas? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrei (MPI) Sent: Wednesday, December 01, 2004 11:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Tim, You may see description of new 1.3.4 firmware at polycom.com (check - http://www.polycom.com/common/pw_cmp_updateDocKeywords/0,1687,3641,00.p d f ) released in October. Though, it was proven over time that troubles with a SIP phone like not hearing one side or the other is NAT related problems. You may want to investigate firewall setup. I am not saying it is not phone related, but the phone would be the last one to blame. Also, may I express my feelfings about Cisco and Polycom - not allowing direct firmware download for their phones - that sucks big time. I will get the firmware this way or the other. They just force me to waste my time again and again contacting their dealers and searching the internet. That should just enrage customers, in my opinion. Are they so big, they do not even care? Sincerely, Andrei Tim Jackson wrote: Any idea if 1.34 fixes the problems with the phones being up for long periods of time and weird call problems (I cannot hear remote caller, but they can hear me) ? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Wednesday, December 01, 2004 6:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom IP500 Any idea if 1.34 makes Daylight Savings work for us people in Australia? PaulH -Original Message- From: Andrei (MPI) [mailto:[EMAIL PROTECTED] Sent: Thursday, 2 December 2004 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Hi Chris, First of all, you need to configure ftp or tftp and watch syslog closely - what the phone is looking for at boot time. You would need to put config files into (t)ftp directory, named according to MAC address of you phone. XML and Web is really weird - they do not even share same config data. For example, I had to change address of SNTP server (clock) and it still was showing as 'clock' on Web admin page for the phone. I will email you the config files I got from a good fellow from this list not so long ago... and those config files do really help! Also, I suggest that you upgrade your SIP firmware if you have not done it yet. I got the Polycom 500 with firmware which was very old and incapable to work with asterisk. Mine is 1.3.1 now (http://www.freedomphones.net/polycom/files/). If anyone has SIP firmware 1.3.4 - please make it downloadable somewhere on the internet? Thank you, Andrei Chris Cherry wrote: Hey All, First Time Writing. I'm trying to set up my IP500 phones to register SIP with *. I input all the (I assume) correct data in to the fields on the Web Interface. And I get no notification that the phone is even attempting to register, no failed messages etc. I have read that the Web interface
Re: [Asterisk-Users] Polycom IP500
I got 1.3.4 already. No major changes, works smoothly so far. Matthew Marlowe wrote: Im working on getting 1.3.4.. Will post. On Thu, 02 Dec 2004 13:15:29 -0500, Andrei (MPI) [EMAIL PROTECTED] wrote: I would sniff UDP packets with tcpdump and see what is going on: separately in 10.24.102 and 10.24.100 segment. Do you have any general NAT settings in sip.conf, like externip and local network etc? How about RTP port range? It would be easier to understand your setup if you post your sip.conf, as well as phone's MAC.cfg file (which may be too big for this newsgroup - email is fine). Also, asterisk sip registration and error messages would give us some more info. Andrei Tim Jackson wrote: I've already added nat=yes. Nothing fancy/special on the routing between these. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrei (MPI) Sent: Thursday, December 02, 2004 10:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Hello Tim, You are saying that: phone is on 10.24.102.0/24 and Asterisk resides on 10.24.100.0/24. Honestly, I see at least one hop forwarding here and possible network issues right away. At one moment, not in a NAT environment, but having phone IP 172.16.100.xxx and Linux server ip 172.16.1.xxx - I had to add nat=yes in sip.conf in order to get the phone to work. Sincerely, Andrei Tim Jackson wrote: Theres no NAT going on here. Just 1 router in between, phones reside on 10.24.102.0/24 Asterisk resides on 10.24.100.0/24, so this isn't a NAT problem. Any other ideas? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrei (MPI) Sent: Wednesday, December 01, 2004 11:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Tim, You may see description of new 1.3.4 firmware at polycom.com (check - http://www.polycom.com/common/pw_cmp_updateDocKeywords/0,1687,3641,00.p d f ) released in October. Though, it was proven over time that troubles with a SIP phone like not hearing one side or the other is NAT related problems. You may want to investigate firewall setup. I am not saying it is not phone related, but the phone would be the last one to blame. Also, may I express my feelfings about Cisco and Polycom - not allowing direct firmware download for their phones - that sucks big time. I will get the firmware this way or the other. They just force me to waste my time again and again contacting their dealers and searching the internet. That should just enrage customers, in my opinion. Are they so big, they do not even care? Sincerely, Andrei Tim Jackson wrote: Any idea if 1.34 fixes the problems with the phones being up for long periods of time and weird call problems (I cannot hear remote caller, but they can hear me) ? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Wednesday, December 01, 2004 6:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom IP500 Any idea if 1.34 makes Daylight Savings work for us people in Australia? PaulH -Original Message- From: Andrei (MPI) [mailto:[EMAIL PROTECTED] Sent: Thursday, 2 December 2004 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Hi Chris, First of all, you need to configure ftp or tftp and watch syslog closely - what the phone is looking for at boot time. You would need to put config files into (t)ftp directory, named according to MAC address of you phone. XML and Web is really weird - they do not even share same config data. For example, I had to change address of SNTP server (clock) and it still was showing as 'clock' on Web admin page for the phone. I will email you the config files I got from a good fellow from this list not so long ago... and those config files do really help! Also, I suggest that you upgrade your SIP firmware if you have not done it yet. I got the Polycom 500 with firmware which was very old and incapable to work with asterisk. Mine is 1.3.1 now (http://www.freedomphones.net/polycom/files/). If anyone has SIP firmware 1.3.4 - please make it downloadable somewhere on the internet? Thank you, Andrei Chris Cherry wrote: Hey All, First Time Writing. I'm trying to set up my IP500 phones to register SIP with *. I input all the (I assume) correct data in to the fields on the Web Interface. And I get no notification that the phone is even attempting to register, no failed messages etc. I have read
Re: [Asterisk-Users] Polycom IP500
You may see description of new 1.3.4 firmware at polycom.com (check - http://www.polycom.com/common/pw_cmp_updateDocKeywords/0,1687,3641,00.pdf ) released in October. Though, it was proven over time that troubles with a SIP phone like not hearing one side or the other is NAT related problems. You may want to investigate firewall setup. I am not saying it is not phone related, but the phone would be the last one to blame. Also, may I express my feelfings about Cisco and Polycom - not allowing direct firmware download for their phones - that sucks big time. I will get the firmware this way or the other. They just force me to waste my time again and again contacting their dealers and searching the internet. That should just enrage customers, in my opinion. Are they so big, they do not even care? I'd certainly agree with you. Just received two new 500's, one of which would not load bootrom.ld for anything. Polycom's support line, when questioned about downloading code, indicated the stuff was only available through certified resellers in order to control 'quality'. Thought that was really funny since the 500's were ordered from a reseller with SIP image, and the reseller never even bothered to include a CD or url. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500
Rich Adamson wrote: ... Thought that was really funny since the 500's were ordered from a reseller with SIP image, and the reseller never even bothered to include a CD or url. ... Yes, that's a fact. You were lucky he knew what exactly he was selling. I could tell more horror stories as I was contacting Polycom again and again looking for a reseller that would sell 10+ 600s and Polycom would give me contacts of their dealers who would not even sure what IP telephony is, could not find even product codes etc... Finally some person from Polycom said to me Our guy who is responsible for IP telephones is on vacation this week, please leave him voicemail, he will call you next week. He never called back, even next month Though, enough about that. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP500
Theres no NAT going on here. Just 1 router in between, phones reside on 10.24.102.0/24 Asterisk resides on 10.24.100.0/24, so this isn't a NAT problem. Any other ideas? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrei (MPI) Sent: Wednesday, December 01, 2004 11:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Tim, You may see description of new 1.3.4 firmware at polycom.com (check - http://www.polycom.com/common/pw_cmp_updateDocKeywords/0,1687,3641,00.pd f ) released in October. Though, it was proven over time that troubles with a SIP phone like not hearing one side or the other is NAT related problems. You may want to investigate firewall setup. I am not saying it is not phone related, but the phone would be the last one to blame. Also, may I express my feelfings about Cisco and Polycom - not allowing direct firmware download for their phones - that sucks big time. I will get the firmware this way or the other. They just force me to waste my time again and again contacting their dealers and searching the internet. That should just enrage customers, in my opinion. Are they so big, they do not even care? Sincerely, Andrei Tim Jackson wrote: Any idea if 1.34 fixes the problems with the phones being up for long periods of time and weird call problems (I cannot hear remote caller, but they can hear me) ? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Wednesday, December 01, 2004 6:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom IP500 Any idea if 1.34 makes Daylight Savings work for us people in Australia? PaulH -Original Message- From: Andrei (MPI) [mailto:[EMAIL PROTECTED] Sent: Thursday, 2 December 2004 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Hi Chris, First of all, you need to configure ftp or tftp and watch syslog closely - what the phone is looking for at boot time. You would need to put config files into (t)ftp directory, named according to MAC address of you phone. XML and Web is really weird - they do not even share same config data. For example, I had to change address of SNTP server (clock) and it still was showing as 'clock' on Web admin page for the phone. I will email you the config files I got from a good fellow from this list not so long ago... and those config files do really help! Also, I suggest that you upgrade your SIP firmware if you have not done it yet. I got the Polycom 500 with firmware which was very old and incapable to work with asterisk. Mine is 1.3.1 now (http://www.freedomphones.net/polycom/files/). If anyone has SIP firmware 1.3.4 - please make it downloadable somewhere on the internet? Thank you, Andrei Chris Cherry wrote: Hey All, First Time Writing. I'm trying to set up my IP500 phones to register SIP with *. I input all the (I assume) correct data in to the fields on the Web Interface. And I get no notification that the phone is even attempting to register, no failed messages etc. I have read that the Web interface is crap and the XML config files is the way to go. Does anyone have a basic config file that doesn't change any defaults? I couldn't seem to find one. Extra Info: Server is 192.168.0.3 Phone name/ext I want to be 301 Thanks, Chris Cherry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit
RE: [Asterisk-Users] Polycom IP500
Tim, I've experienced routing problems before where I could dial a phone on a different subnet and I could hear them, but they couldn't hear me. Is it possible that the phone learns the route, then loses it later? I ended up setting the default gateway on all of my phones to a router that knows about all of my subnets, instead of my internet gateway. Ty -Original Message- From: Tim Jackson [mailto:[EMAIL PROTECTED] Sent: Thursday, December 02, 2004 9:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 Theres no NAT going on here. Just 1 router in between, phones reside on 10.24.102.0/24 Asterisk resides on 10.24.100.0/24, so this isn't a NAT problem. Any other ideas? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrei (MPI) Sent: Wednesday, December 01, 2004 11:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Tim, You may see description of new 1.3.4 firmware at polycom.com (check - http://www.polycom.com/common/pw_cmp_updateDocKeywords/0,1687,3641,00.pd f ) released in October. Though, it was proven over time that troubles with a SIP phone like not hearing one side or the other is NAT related problems. You may want to investigate firewall setup. I am not saying it is not phone related, but the phone would be the last one to blame. Also, may I express my feelfings about Cisco and Polycom - not allowing direct firmware download for their phones - that sucks big time. I will get the firmware this way or the other. They just force me to waste my time again and again contacting their dealers and searching the internet. That should just enrage customers, in my opinion. Are they so big, they do not even care? Sincerely, Andrei Tim Jackson wrote: Any idea if 1.34 fixes the problems with the phones being up for long periods of time and weird call problems (I cannot hear remote caller, but they can hear me) ? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Wednesday, December 01, 2004 6:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom IP500 Any idea if 1.34 makes Daylight Savings work for us people in Australia? PaulH -Original Message- From: Andrei (MPI) [mailto:[EMAIL PROTECTED] Sent: Thursday, 2 December 2004 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Hi Chris, First of all, you need to configure ftp or tftp and watch syslog closely - what the phone is looking for at boot time. You would need to put config files into (t)ftp directory, named according to MAC address of you phone. XML and Web is really weird - they do not even share same config data. For example, I had to change address of SNTP server (clock) and it still was showing as 'clock' on Web admin page for the phone. I will email you the config files I got from a good fellow from this list not so long ago... and those config files do really help! Also, I suggest that you upgrade your SIP firmware if you have not done it yet. I got the Polycom 500 with firmware which was very old and incapable to work with asterisk. Mine is 1.3.1 now (http://www.freedomphones.net/polycom/files/). If anyone has SIP firmware 1.3.4 - please make it downloadable somewhere on the internet? Thank you, Andrei Chris Cherry wrote: Hey All, First Time Writing. I'm trying to set up my IP500 phones to register SIP with *. I input all the (I assume) correct data in to the fields on the Web Interface. And I get no notification that the phone is even attempting to register, no failed messages etc. I have read that the Web interface is crap and the XML config files is the way to go. Does anyone have a basic config file that doesn't change any defaults? I couldn't seem to find one. Extra Info: Server is 192.168.0.3 Phone name/ext I want to be 301 Thanks, Chris Cherry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited
RE: [Asterisk-Users] Polycom IP500
Routing here isn't an issue. The router that is their gateway has all internal routes learned via OSPF. Connectivity to the * box is fine (all 100mbit, the interface the phones are on is a dot1q sub-interface though). I'm 100% confident that it's not an routing/nat problem (no NAT taking place). But I've given up. No real answers on the polycom issue, I've just taken the phones and the * box down. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ty Purcell Sent: Thursday, December 02, 2004 9:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 Tim, I've experienced routing problems before where I could dial a phone on a different subnet and I could hear them, but they couldn't hear me. Is it possible that the phone learns the route, then loses it later? I ended up setting the default gateway on all of my phones to a router that knows about all of my subnets, instead of my internet gateway. Ty -Original Message- From: Tim Jackson [mailto:[EMAIL PROTECTED] Sent: Thursday, December 02, 2004 9:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 Theres no NAT going on here. Just 1 router in between, phones reside on 10.24.102.0/24 Asterisk resides on 10.24.100.0/24, so this isn't a NAT problem. Any other ideas? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrei (MPI) Sent: Wednesday, December 01, 2004 11:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Tim, You may see description of new 1.3.4 firmware at polycom.com (check - http://www.polycom.com/common/pw_cmp_updateDocKeywords/0,1687,3641,00.pd f ) released in October. Though, it was proven over time that troubles with a SIP phone like not hearing one side or the other is NAT related problems. You may want to investigate firewall setup. I am not saying it is not phone related, but the phone would be the last one to blame. Also, may I express my feelfings about Cisco and Polycom - not allowing direct firmware download for their phones - that sucks big time. I will get the firmware this way or the other. They just force me to waste my time again and again contacting their dealers and searching the internet. That should just enrage customers, in my opinion. Are they so big, they do not even care? Sincerely, Andrei Tim Jackson wrote: Any idea if 1.34 fixes the problems with the phones being up for long periods of time and weird call problems (I cannot hear remote caller, but they can hear me) ? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Wednesday, December 01, 2004 6:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom IP500 Any idea if 1.34 makes Daylight Savings work for us people in Australia? PaulH -Original Message- From: Andrei (MPI) [mailto:[EMAIL PROTECTED] Sent: Thursday, 2 December 2004 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Hi Chris, First of all, you need to configure ftp or tftp and watch syslog closely - what the phone is looking for at boot time. You would need to put config files into (t)ftp directory, named according to MAC address of you phone. XML and Web is really weird - they do not even share same config data. For example, I had to change address of SNTP server (clock) and it still was showing as 'clock' on Web admin page for the phone. I will email you the config files I got from a good fellow from this list not so long ago... and those config files do really help! Also, I suggest that you upgrade your SIP firmware if you have not done it yet. I got the Polycom 500 with firmware which was very old and incapable to work with asterisk. Mine is 1.3.1 now (http://www.freedomphones.net/polycom/files/). If anyone has SIP firmware 1.3.4 - please make it downloadable somewhere on the internet? Thank you, Andrei Chris Cherry wrote: Hey All, First Time Writing. I'm trying to set up my IP500 phones to register SIP with *. I input all the (I assume) correct data in to the fields on the Web Interface. And I get no notification that the phone is even attempting to register, no failed messages etc. I have read that the Web interface is crap and the XML config files is the way to go. Does anyone have a basic config file that doesn't change any defaults? I couldn't seem to find one. Extra Info: Server is 192.168.0.3 Phone name/ext I want to be 301 Thanks, Chris Cherry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman
Re: [Asterisk-Users] Polycom IP500
Hello Tim, You are saying that: phone is on 10.24.102.0/24 and Asterisk resides on 10.24.100.0/24. Honestly, I see at least one hop forwarding here and possible network issues right away. At one moment, not in a NAT environment, but having phone IP 172.16.100.xxx and Linux server ip 172.16.1.xxx - I had to add nat=yes in sip.conf in order to get the phone to work. Sincerely, Andrei Tim Jackson wrote: Theres no NAT going on here. Just 1 router in between, phones reside on 10.24.102.0/24 Asterisk resides on 10.24.100.0/24, so this isn't a NAT problem. Any other ideas? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrei (MPI) Sent: Wednesday, December 01, 2004 11:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Tim, You may see description of new 1.3.4 firmware at polycom.com (check - http://www.polycom.com/common/pw_cmp_updateDocKeywords/0,1687,3641,00.pd f ) released in October. Though, it was proven over time that troubles with a SIP phone like not hearing one side or the other is NAT related problems. You may want to investigate firewall setup. I am not saying it is not phone related, but the phone would be the last one to blame. Also, may I express my feelfings about Cisco and Polycom - not allowing direct firmware download for their phones - that sucks big time. I will get the firmware this way or the other. They just force me to waste my time again and again contacting their dealers and searching the internet. That should just enrage customers, in my opinion. Are they so big, they do not even care? Sincerely, Andrei Tim Jackson wrote: Any idea if 1.34 fixes the problems with the phones being up for long periods of time and weird call problems (I cannot hear remote caller, but they can hear me) ? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Wednesday, December 01, 2004 6:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom IP500 Any idea if 1.34 makes Daylight Savings work for us people in Australia? PaulH -Original Message- From: Andrei (MPI) [mailto:[EMAIL PROTECTED] Sent: Thursday, 2 December 2004 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Hi Chris, First of all, you need to configure ftp or tftp and watch syslog closely - what the phone is looking for at boot time. You would need to put config files into (t)ftp directory, named according to MAC address of you phone. XML and Web is really weird - they do not even share same config data. For example, I had to change address of SNTP server (clock) and it still was showing as 'clock' on Web admin page for the phone. I will email you the config files I got from a good fellow from this list not so long ago... and those config files do really help! Also, I suggest that you upgrade your SIP firmware if you have not done it yet. I got the Polycom 500 with firmware which was very old and incapable to work with asterisk. Mine is 1.3.1 now (http://www.freedomphones.net/polycom/files/). If anyone has SIP firmware 1.3.4 - please make it downloadable somewhere on the internet? Thank you, Andrei Chris Cherry wrote: Hey All, First Time Writing. I'm trying to set up my IP500 phones to register SIP with *. I input all the (I assume) correct data in to the fields on the Web Interface. And I get no notification that the phone is even attempting to register, no failed messages etc. I have read that the Web interface is crap and the XML config files is the way to go. Does anyone have a basic config file that doesn't change any defaults? I couldn't seem to find one. Extra Info: Server is 192.168.0.3 Phone name/ext I want to be 301 Thanks, Chris Cherry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase
RE: [Asterisk-Users] Polycom IP500
I've already added nat=yes. Nothing fancy/special on the routing between these. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrei (MPI) Sent: Thursday, December 02, 2004 10:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Hello Tim, You are saying that: phone is on 10.24.102.0/24 and Asterisk resides on 10.24.100.0/24. Honestly, I see at least one hop forwarding here and possible network issues right away. At one moment, not in a NAT environment, but having phone IP 172.16.100.xxx and Linux server ip 172.16.1.xxx - I had to add nat=yes in sip.conf in order to get the phone to work. Sincerely, Andrei Tim Jackson wrote: Theres no NAT going on here. Just 1 router in between, phones reside on 10.24.102.0/24 Asterisk resides on 10.24.100.0/24, so this isn't a NAT problem. Any other ideas? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrei (MPI) Sent: Wednesday, December 01, 2004 11:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Tim, You may see description of new 1.3.4 firmware at polycom.com (check - http://www.polycom.com/common/pw_cmp_updateDocKeywords/0,1687,3641,00.p d f ) released in October. Though, it was proven over time that troubles with a SIP phone like not hearing one side or the other is NAT related problems. You may want to investigate firewall setup. I am not saying it is not phone related, but the phone would be the last one to blame. Also, may I express my feelfings about Cisco and Polycom - not allowing direct firmware download for their phones - that sucks big time. I will get the firmware this way or the other. They just force me to waste my time again and again contacting their dealers and searching the internet. That should just enrage customers, in my opinion. Are they so big, they do not even care? Sincerely, Andrei Tim Jackson wrote: Any idea if 1.34 fixes the problems with the phones being up for long periods of time and weird call problems (I cannot hear remote caller, but they can hear me) ? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Wednesday, December 01, 2004 6:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom IP500 Any idea if 1.34 makes Daylight Savings work for us people in Australia? PaulH -Original Message- From: Andrei (MPI) [mailto:[EMAIL PROTECTED] Sent: Thursday, 2 December 2004 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Hi Chris, First of all, you need to configure ftp or tftp and watch syslog closely - what the phone is looking for at boot time. You would need to put config files into (t)ftp directory, named according to MAC address of you phone. XML and Web is really weird - they do not even share same config data. For example, I had to change address of SNTP server (clock) and it still was showing as 'clock' on Web admin page for the phone. I will email you the config files I got from a good fellow from this list not so long ago... and those config files do really help! Also, I suggest that you upgrade your SIP firmware if you have not done it yet. I got the Polycom 500 with firmware which was very old and incapable to work with asterisk. Mine is 1.3.1 now (http://www.freedomphones.net/polycom/files/). If anyone has SIP firmware 1.3.4 - please make it downloadable somewhere on the internet? Thank you, Andrei Chris Cherry wrote: Hey All, First Time Writing. I'm trying to set up my IP500 phones to register SIP with *. I input all the (I assume) correct data in to the fields on the Web Interface. And I get no notification that the phone is even attempting to register, no failed messages etc. I have read that the Web interface is crap and the XML config files is the way to go. Does anyone have a basic config file that doesn't change any defaults? I couldn't seem to find one. Extra Info: Server is 192.168.0.3 Phone name/ext I want to be 301 Thanks, Chris Cherry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies
Re: [Asterisk-Users] Polycom IP500
I would sniff UDP packets with tcpdump and see what is going on: separately in 10.24.102 and 10.24.100 segment. Do you have any general NAT settings in sip.conf, like externip and local network etc? How about RTP port range? It would be easier to understand your setup if you post your sip.conf, as well as phone's MAC.cfg file (which may be too big for this newsgroup - email is fine). Also, asterisk sip registration and error messages would give us some more info. Andrei Tim Jackson wrote: I've already added nat=yes. Nothing fancy/special on the routing between these. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrei (MPI) Sent: Thursday, December 02, 2004 10:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Hello Tim, You are saying that: phone is on 10.24.102.0/24 and Asterisk resides on 10.24.100.0/24. Honestly, I see at least one hop forwarding here and possible network issues right away. At one moment, not in a NAT environment, but having phone IP 172.16.100.xxx and Linux server ip 172.16.1.xxx - I had to add nat=yes in sip.conf in order to get the phone to work. Sincerely, Andrei Tim Jackson wrote: Theres no NAT going on here. Just 1 router in between, phones reside on 10.24.102.0/24 Asterisk resides on 10.24.100.0/24, so this isn't a NAT problem. Any other ideas? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrei (MPI) Sent: Wednesday, December 01, 2004 11:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Tim, You may see description of new 1.3.4 firmware at polycom.com (check - http://www.polycom.com/common/pw_cmp_updateDocKeywords/0,1687,3641,00.p d f ) released in October. Though, it was proven over time that troubles with a SIP phone like not hearing one side or the other is NAT related problems. You may want to investigate firewall setup. I am not saying it is not phone related, but the phone would be the last one to blame. Also, may I express my feelfings about Cisco and Polycom - not allowing direct firmware download for their phones - that sucks big time. I will get the firmware this way or the other. They just force me to waste my time again and again contacting their dealers and searching the internet. That should just enrage customers, in my opinion. Are they so big, they do not even care? Sincerely, Andrei Tim Jackson wrote: Any idea if 1.34 fixes the problems with the phones being up for long periods of time and weird call problems (I cannot hear remote caller, but they can hear me) ? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Wednesday, December 01, 2004 6:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom IP500 Any idea if 1.34 makes Daylight Savings work for us people in Australia? PaulH -Original Message- From: Andrei (MPI) [mailto:[EMAIL PROTECTED] Sent: Thursday, 2 December 2004 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Hi Chris, First of all, you need to configure ftp or tftp and watch syslog closely - what the phone is looking for at boot time. You would need to put config files into (t)ftp directory, named according to MAC address of you phone. XML and Web is really weird - they do not even share same config data. For example, I had to change address of SNTP server (clock) and it still was showing as 'clock' on Web admin page for the phone. I will email you the config files I got from a good fellow from this list not so long ago... and those config files do really help! Also, I suggest that you upgrade your SIP firmware if you have not done it yet. I got the Polycom 500 with firmware which was very old and incapable to work with asterisk. Mine is 1.3.1 now (http://www.freedomphones.net/polycom/files/). If anyone has SIP firmware 1.3.4 - please make it downloadable somewhere on the internet? Thank you, Andrei Chris Cherry wrote: Hey All, First Time Writing. I'm trying to set up my IP500 phones to register SIP with *. I input all the (I assume) correct data in to the fields on the Web Interface. And I get no notification that the phone is even attempting to register, no failed messages etc. I have read that the Web interface is crap and the XML config files is the way to go. Does anyone have a basic config file that doesn't change any defaults? I couldn't seem to find one. Extra Info: Server is 192.168.0.3 Phone name/ext I
Re: [Asterisk-Users] Polycom IP500
Hi Chris, First of all, you need to configure ftp or tftp and watch syslog closely - what the phone is looking for at boot time. You would need to put config files into (t)ftp directory, named according to MAC address of you phone. XML and Web is really weird - they do not even share same config data. For example, I had to change address of SNTP server (clock) and it still was showing as 'clock' on Web admin page for the phone. I will email you the config files I got from a good fellow from this list not so long ago... and those config files do really help! Also, I suggest that you upgrade your SIP firmware if you have not done it yet. I got the Polycom 500 with firmware which was very old and incapable to work with asterisk. Mine is 1.3.1 now (http://www.freedomphones.net/polycom/files/). If anyone has SIP firmware 1.3.4 - please make it downloadable somewhere on the internet? Thank you, Andrei Chris Cherry wrote: Hey All, First Time Writing. I'm trying to set up my IP500 phones to register SIP with *. I input all the (I assume) correct data in to the fields on the Web Interface. And I get no notification that the phone is even attempting to register, no failed messages etc. I have read that the Web interface is crap and the XML config files is the way to go. Does anyone have a basic config file that doesnt change any defaults? I couldn't seem to find one. Extra Info: Server is 192.168.0.3 Phone name/ext I want to be 301 Thanks, Chris Cherry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP500
Any idea if 1.34 makes Daylight Savings work for us people in Australia? PaulH -Original Message- From: Andrei (MPI) [mailto:[EMAIL PROTECTED] Sent: Thursday, 2 December 2004 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Hi Chris, First of all, you need to configure ftp or tftp and watch syslog closely - what the phone is looking for at boot time. You would need to put config files into (t)ftp directory, named according to MAC address of you phone. XML and Web is really weird - they do not even share same config data. For example, I had to change address of SNTP server (clock) and it still was showing as 'clock' on Web admin page for the phone. I will email you the config files I got from a good fellow from this list not so long ago... and those config files do really help! Also, I suggest that you upgrade your SIP firmware if you have not done it yet. I got the Polycom 500 with firmware which was very old and incapable to work with asterisk. Mine is 1.3.1 now (http://www.freedomphones.net/polycom/files/). If anyone has SIP firmware 1.3.4 - please make it downloadable somewhere on the internet? Thank you, Andrei Chris Cherry wrote: Hey All, First Time Writing. I'm trying to set up my IP500 phones to register SIP with *. I input all the (I assume) correct data in to the fields on the Web Interface. And I get no notification that the phone is even attempting to register, no failed messages etc. I have read that the Web interface is crap and the XML config files is the way to go. Does anyone have a basic config file that doesn't change any defaults? I couldn't seem to find one. Extra Info: Server is 192.168.0.3 Phone name/ext I want to be 301 Thanks, Chris Cherry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP500
Any idea if 1.34 fixes the problems with the phones being up for long periods of time and weird call problems (I cannot hear remote caller, but they can hear me) ? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Wednesday, December 01, 2004 6:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom IP500 Any idea if 1.34 makes Daylight Savings work for us people in Australia? PaulH -Original Message- From: Andrei (MPI) [mailto:[EMAIL PROTECTED] Sent: Thursday, 2 December 2004 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Hi Chris, First of all, you need to configure ftp or tftp and watch syslog closely - what the phone is looking for at boot time. You would need to put config files into (t)ftp directory, named according to MAC address of you phone. XML and Web is really weird - they do not even share same config data. For example, I had to change address of SNTP server (clock) and it still was showing as 'clock' on Web admin page for the phone. I will email you the config files I got from a good fellow from this list not so long ago... and those config files do really help! Also, I suggest that you upgrade your SIP firmware if you have not done it yet. I got the Polycom 500 with firmware which was very old and incapable to work with asterisk. Mine is 1.3.1 now (http://www.freedomphones.net/polycom/files/). If anyone has SIP firmware 1.3.4 - please make it downloadable somewhere on the internet? Thank you, Andrei Chris Cherry wrote: Hey All, First Time Writing. I'm trying to set up my IP500 phones to register SIP with *. I input all the (I assume) correct data in to the fields on the Web Interface. And I get no notification that the phone is even attempting to register, no failed messages etc. I have read that the Web interface is crap and the XML config files is the way to go. Does anyone have a basic config file that doesn't change any defaults? I couldn't seem to find one. Extra Info: Server is 192.168.0.3 Phone name/ext I want to be 301 Thanks, Chris Cherry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500
Tim, You may see description of new 1.3.4 firmware at polycom.com (check - http://www.polycom.com/common/pw_cmp_updateDocKeywords/0,1687,3641,00.pdf ) released in October. Though, it was proven over time that troubles with a SIP phone like not hearing one side or the other is NAT related problems. You may want to investigate firewall setup. I am not saying it is not phone related, but the phone would be the last one to blame. Also, may I express my feelfings about Cisco and Polycom - not allowing direct firmware download for their phones - that sucks big time. I will get the firmware this way or the other. They just force me to waste my time again and again contacting their dealers and searching the internet. That should just enrage customers, in my opinion. Are they so big, they do not even care? Sincerely, Andrei Tim Jackson wrote: Any idea if 1.34 fixes the problems with the phones being up for long periods of time and weird call problems (I cannot hear remote caller, but they can hear me) ? -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Wednesday, December 01, 2004 6:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom IP500 Any idea if 1.34 makes Daylight Savings work for us people in Australia? PaulH -Original Message- From: Andrei (MPI) [mailto:[EMAIL PROTECTED] Sent: Thursday, 2 December 2004 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 Hi Chris, First of all, you need to configure ftp or tftp and watch syslog closely - what the phone is looking for at boot time. You would need to put config files into (t)ftp directory, named according to MAC address of you phone. XML and Web is really weird - they do not even share same config data. For example, I had to change address of SNTP server (clock) and it still was showing as 'clock' on Web admin page for the phone. I will email you the config files I got from a good fellow from this list not so long ago... and those config files do really help! Also, I suggest that you upgrade your SIP firmware if you have not done it yet. I got the Polycom 500 with firmware which was very old and incapable to work with asterisk. Mine is 1.3.1 now (http://www.freedomphones.net/polycom/files/). If anyone has SIP firmware 1.3.4 - please make it downloadable somewhere on the internet? Thank you, Andrei Chris Cherry wrote: Hey All, First Time Writing. I'm trying to set up my IP500 phones to register SIP with *. I input all the (I assume) correct data in to the fields on the Web Interface. And I get no notification that the phone is even attempting to register, no failed messages etc. I have read that the Web interface is crap and the XML config files is the way to go. Does anyone have a basic config file that doesn't change any defaults? I couldn't seem to find one. Extra Info: Server is 192.168.0.3 Phone name/ext I want to be 301 Thanks, Chris Cherry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP500 problem updating bootrom
Nevermind, I had to upgrade to 1.4.2 first then 1.5 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Marlowe Sent: Tuesday, September 21, 2004 1:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Polycom IP500 problem updating bootrom I've had an IP300 for a while now and it's been working fine. I just got an IP500 and when it connects to the FTP server it downloads the new bootrom and says error loading. The bootrom is fine and works on the 300... In addition, I downloaded a new copy to be sure and it still doesn't work. Can anyone give me some advice? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500
I have two IP 500's on my Asterisk PBX. The IM features just kinda worked, without any extra configuration. Messenging and presense. I've heard of folks trying to interface this functionality with MSN Messenger and such, without much success. Can't help you much there. - Jeff On Fri, 17 Sep 2004 11:12:03 +1000, Paul Hales [EMAIL PROTECTED] wrote: The Polycom phones have an instant messaging function - any idea what is required to make it work? PaulH Adairs ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP500 vs Cisco 7940
The Polycom IP500s do support customized ringtones and can use a customized ALERT_INFO for all of them. One thing that is worth noting in this comparison is that the IP500 doesn't support the XHTML microbrowser that the IP600 does. Since they both use the same SIP application I am hoping they enable this in future but as of now it doesn't work. I actually had 30 of these before I found this out but would still recommend these over any phone in the price range. Jody N. Rudolph Heartland Communications Internet Services, Inc 1301 Boadway Paducah, KY 42001 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Laird Sent: Thursday, September 09, 2004 1:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 vs Cisco 7940 On Sep 9, 2004, at 9:53 AM, Matt G wrote: I've been asked to determine which phones our organization should go with. And I've narrowed it down to the Polycom IP500 or the Cisco 7940. From my travels through google, it's hard to find a definitive comparison of the two phones. So I thought I would ask the people that have probably used both. From what I can tell, the only major benefit the Cisco has over the Polycom is * 24 ring tones * XML support * Help Button * Larger Screen (is this true? 2x24 vs the 160x80 on the polycom) The screen on the Cisco isn't very big, either--192x96 or so, if I remember correctly. I'm running 6.3 on my 7940, and I haven't seen the ability to do anything interesting with ringtones. In theory, you can feed new tones to it, but you can't use them for ALERT_INFO-driven distinctive ringing. The XML support is okay, but rumors suggest that the newest Polycom firmware supports something very close to XHTML, which would be a lot more powerful then Cisco's sparsely-documented XML dialect. Another question that came up while discussing the Cisco phones was if the 24 ring tones are 'assignable' (ie, user calls in with callerid saying 'sales' and it rings a certain way, if they call in with callerid saying 'tech support' it rings something else). I couldn't find any information on this on google, so if anyone has the answer to this that would be great. I don't think it can do that. You can set ALERT_INFO in Asterisk to Bellcore-drX, where X is 1..5, and the phone will ring slightly differently, but that may or may not be good enough for your purposes. Other than that, the polycom seems to have all the features we want, and according to the wiki works quite well with asterisk and has many features enabled that seem pretty interesting (MWI, etc). The Cisco's on the other hand seem less straightforward to configure and not as much talk on the wiki, nor support. MWI works just fine on the 7940, so I'm not sure that I'd count that as an advantage for the Polycom. I haven't seen a Polycom in person, but I haven't heard anything bad about them. My 7940 works well, and I wouldn't hesitate to recommend it, but for the money, the Polycom is quite likely a better phone. I didn't find the 7940 to be particularly difficult to configure, *EXCEPT* for the initial installation of the SIP firmware. It's a multi-step upgrade, because you can't directly upgrade from the SCCP image that it ships with to a modern SIP image. Once you get past that, it isn't too bad, particularly if you have multiple phones. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500 vs Cisco 7940
what is the price range in us dollars? - Original Message - From: Jody N. Rudolph [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, September 09, 2004 11:32 AM Subject: RE: [Asterisk-Users] Polycom IP500 vs Cisco 7940 The Polycom IP500s do support customized ringtones and can use a customized ALERT_INFO for all of them. One thing that is worth noting in this comparison is that the IP500 doesn't support the XHTML microbrowser that the IP600 does. Since they both use the same SIP application I am hoping they enable this in future but as of now it doesn't work. I actually had 30 of these before I found this out but would still recommend these over any phone in the price range. Jody N. Rudolph Heartland Communications Internet Services, Inc 1301 Boadway Paducah, KY 42001 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Laird Sent: Thursday, September 09, 2004 1:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 vs Cisco 7940 On Sep 9, 2004, at 9:53 AM, Matt G wrote: I've been asked to determine which phones our organization should go with. And I've narrowed it down to the Polycom IP500 or the Cisco 7940. From my travels through google, it's hard to find a definitive comparison of the two phones. So I thought I would ask the people that have probably used both. From what I can tell, the only major benefit the Cisco has over the Polycom is * 24 ring tones * XML support * Help Button * Larger Screen (is this true? 2x24 vs the 160x80 on the polycom) The screen on the Cisco isn't very big, either--192x96 or so, if I remember correctly. I'm running 6.3 on my 7940, and I haven't seen the ability to do anything interesting with ringtones. In theory, you can feed new tones to it, but you can't use them for ALERT_INFO-driven distinctive ringing. The XML support is okay, but rumors suggest that the newest Polycom firmware supports something very close to XHTML, which would be a lot more powerful then Cisco's sparsely-documented XML dialect. Another question that came up while discussing the Cisco phones was if the 24 ring tones are 'assignable' (ie, user calls in with callerid saying 'sales' and it rings a certain way, if they call in with callerid saying 'tech support' it rings something else). I couldn't find any information on this on google, so if anyone has the answer to this that would be great. I don't think it can do that. You can set ALERT_INFO in Asterisk to Bellcore-drX, where X is 1..5, and the phone will ring slightly differently, but that may or may not be good enough for your purposes. Other than that, the polycom seems to have all the features we want, and according to the wiki works quite well with asterisk and has many features enabled that seem pretty interesting (MWI, etc). The Cisco's on the other hand seem less straightforward to configure and not as much talk on the wiki, nor support. MWI works just fine on the 7940, so I'm not sure that I'd count that as an advantage for the Polycom. I haven't seen a Polycom in person, but I haven't heard anything bad about them. My 7940 works well, and I wouldn't hesitate to recommend it, but for the money, the Polycom is quite likely a better phone. I didn't find the 7940 to be particularly difficult to configure, *EXCEPT* for the initial installation of the SIP firmware. It's a multi-step upgrade, because you can't directly upgrade from the SCCP image that it ships with to a modern SIP image. Once you get past that, it isn't too bad, particularly if you have multiple phones. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1033 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP500 vs Cisco 7940
In July I bought one from CDW for $280.75. Ty -Original Message- From: hank smith [mailto:[EMAIL PROTECTED] Sent: Thursday, September 09, 2004 2:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 vs Cisco 7940 what is the price range in us dollars? - Original Message - From: Jody N. Rudolph [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, September 09, 2004 11:32 AM Subject: RE: [Asterisk-Users] Polycom IP500 vs Cisco 7940 The Polycom IP500s do support customized ringtones and can use a customized ALERT_INFO for all of them. One thing that is worth noting in this comparison is that the IP500 doesn't support the XHTML microbrowser that the IP600 does. Since they both use the same SIP application I am hoping they enable this in future but as of now it doesn't work. I actually had 30 of these before I found this out but would still recommend these over any phone in the price range. Jody N. Rudolph Heartland Communications Internet Services, Inc 1301 Boadway Paducah, KY 42001 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Laird Sent: Thursday, September 09, 2004 1:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 vs Cisco 7940 On Sep 9, 2004, at 9:53 AM, Matt G wrote: I've been asked to determine which phones our organization should go with. And I've narrowed it down to the Polycom IP500 or the Cisco 7940. From my travels through google, it's hard to find a definitive comparison of the two phones. So I thought I would ask the people that have probably used both. From what I can tell, the only major benefit the Cisco has over the Polycom is * 24 ring tones * XML support * Help Button * Larger Screen (is this true? 2x24 vs the 160x80 on the polycom) The screen on the Cisco isn't very big, either--192x96 or so, if I remember correctly. I'm running 6.3 on my 7940, and I haven't seen the ability to do anything interesting with ringtones. In theory, you can feed new tones to it, but you can't use them for ALERT_INFO-driven distinctive ringing. The XML support is okay, but rumors suggest that the newest Polycom firmware supports something very close to XHTML, which would be a lot more powerful then Cisco's sparsely-documented XML dialect. Another question that came up while discussing the Cisco phones was if the 24 ring tones are 'assignable' (ie, user calls in with callerid saying 'sales' and it rings a certain way, if they call in with callerid saying 'tech support' it rings something else). I couldn't find any information on this on google, so if anyone has the answer to this that would be great. I don't think it can do that. You can set ALERT_INFO in Asterisk to Bellcore-drX, where X is 1..5, and the phone will ring slightly differently, but that may or may not be good enough for your purposes. Other than that, the polycom seems to have all the features we want, and according to the wiki works quite well with asterisk and has many features enabled that seem pretty interesting (MWI, etc). The Cisco's on the other hand seem less straightforward to configure and not as much talk on the wiki, nor support. MWI works just fine on the 7940, so I'm not sure that I'd count that as an advantage for the Polycom. I haven't seen a Polycom in person, but I haven't heard anything bad about them. My 7940 works well, and I wouldn't hesitate to recommend it, but for the money, the Polycom is quite likely a better phone. I didn't find the 7940 to be particularly difficult to configure, *EXCEPT* for the initial installation of the SIP firmware. It's a multi-step upgrade, because you can't directly upgrade from the SCCP image that it ships with to a modern SIP image. Once you get past that, it isn't too bad, particularly if you have multiple phones. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1033 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP500 vs Cisco 7940
A local vendor here carries IP500s for sub $200. Right now they are out of stock, but he has more coming in. If you want his contact info msg me off list. -Tim -Original Message- From: Ty Purcell [mailto:[EMAIL PROTECTED] Sent: Thursday, September 09, 2004 2:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom IP500 vs Cisco 7940 In July I bought one from CDW for $280.75. Ty -Original Message- From: hank smith [mailto:[EMAIL PROTECTED] Sent: Thursday, September 09, 2004 2:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 vs Cisco 7940 what is the price range in us dollars? - Original Message - From: Jody N. Rudolph [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, September 09, 2004 11:32 AM Subject: RE: [Asterisk-Users] Polycom IP500 vs Cisco 7940 The Polycom IP500s do support customized ringtones and can use a customized ALERT_INFO for all of them. One thing that is worth noting in this comparison is that the IP500 doesn't support the XHTML microbrowser that the IP600 does. Since they both use the same SIP application I am hoping they enable this in future but as of now it doesn't work. I actually had 30 of these before I found this out but would still recommend these over any phone in the price range. Jody N. Rudolph Heartland Communications Internet Services, Inc 1301 Boadway Paducah, KY 42001 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Laird Sent: Thursday, September 09, 2004 1:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 vs Cisco 7940 On Sep 9, 2004, at 9:53 AM, Matt G wrote: I've been asked to determine which phones our organization should go with. And I've narrowed it down to the Polycom IP500 or the Cisco 7940. From my travels through google, it's hard to find a definitive comparison of the two phones. So I thought I would ask the people that have probably used both. From what I can tell, the only major benefit the Cisco has over the Polycom is * 24 ring tones * XML support * Help Button * Larger Screen (is this true? 2x24 vs the 160x80 on the polycom) The screen on the Cisco isn't very big, either--192x96 or so, if I remember correctly. I'm running 6.3 on my 7940, and I haven't seen the ability to do anything interesting with ringtones. In theory, you can feed new tones to it, but you can't use them for ALERT_INFO-driven distinctive ringing. The XML support is okay, but rumors suggest that the newest Polycom firmware supports something very close to XHTML, which would be a lot more powerful then Cisco's sparsely-documented XML dialect. Another question that came up while discussing the Cisco phones was if the 24 ring tones are 'assignable' (ie, user calls in with callerid saying 'sales' and it rings a certain way, if they call in with callerid saying 'tech support' it rings something else). I couldn't find any information on this on google, so if anyone has the answer to this that would be great. I don't think it can do that. You can set ALERT_INFO in Asterisk to Bellcore-drX, where X is 1..5, and the phone will ring slightly differently, but that may or may not be good enough for your purposes. Other than that, the polycom seems to have all the features we want, and according to the wiki works quite well with asterisk and has many features enabled that seem pretty interesting (MWI, etc). The Cisco's on the other hand seem less straightforward to configure and not as much talk on the wiki, nor support. MWI works just fine on the 7940, so I'm not sure that I'd count that as an advantage for the Polycom. I haven't seen a Polycom in person, but I haven't heard anything bad about them. My 7940 works well, and I wouldn't hesitate to recommend it, but for the money, the Polycom is quite likely a better phone. I didn't find the 7940 to be particularly difficult to configure, *EXCEPT* for the initial installation of the SIP firmware. It's a multi-step upgrade, because you can't directly upgrade from the SCCP image that it ships with to a modern SIP image. Once you get past that, it isn't too bad, particularly if you have multiple phones. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP500 vs Cisco 7940
Hank, IP500's are 199$ USD at voipsupply.com and Cisco 7940 are 295.99$ USD at voipsupply.com Thanks, Matt hank smith wrote: what is the price range in us dollars? - Original Message - From: Jody N. Rudolph [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, September 09, 2004 11:32 AM Subject: RE: [Asterisk-Users] Polycom IP500 vs Cisco 7940 The Polycom IP500s do support customized ringtones and can use a customized ALERT_INFO for all of them. One thing that is worth noting in this comparison is that the IP500 doesn't support the XHTML microbrowser that the IP600 does. Since they both use the same SIP application I am hoping they enable this in future but as of now it doesn't work. I actually had 30 of these before I found this out but would still recommend these over any phone in the price range. Jody N. Rudolph Heartland Communications Internet Services, Inc 1301 Boadway Paducah, KY 42001 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Laird Sent: Thursday, September 09, 2004 1:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 vs Cisco 7940 On Sep 9, 2004, at 9:53 AM, Matt G wrote: I've been asked to determine which phones our organization should go with. And I've narrowed it down to the Polycom IP500 or the Cisco 7940. From my travels through google, it's hard to find a definitive comparison of the two phones. So I thought I would ask the people that have probably used both. From what I can tell, the only major benefit the Cisco has over the Polycom is * 24 ring tones * XML support * Help Button * Larger Screen (is this true? 2x24 vs the 160x80 on the polycom) The screen on the Cisco isn't very big, either--192x96 or so, if I remember correctly. I'm running 6.3 on my 7940, and I haven't seen the ability to do anything interesting with ringtones. In theory, you can feed new tones to it, but you can't use them for ALERT_INFO-driven distinctive ringing. The XML support is okay, but rumors suggest that the newest Polycom firmware supports something very close to XHTML, which would be a lot more powerful then Cisco's sparsely-documented XML dialect. Another question that came up while discussing the Cisco phones was if the 24 ring tones are 'assignable' (ie, user calls in with callerid saying 'sales' and it rings a certain way, if they call in with callerid saying 'tech support' it rings something else). I couldn't find any information on this on google, so if anyone has the answer to this that would be great. I don't think it can do that. You can set ALERT_INFO in Asterisk to Bellcore-drX, where X is 1..5, and the phone will ring slightly differently, but that may or may not be good enough for your purposes. Other than that, the polycom seems to have all the features we want, and according to the wiki works quite well with asterisk and has many features enabled that seem pretty interesting (MWI, etc). The Cisco's on the other hand seem less straightforward to configure and not as much talk on the wiki, nor support. MWI works just fine on the 7940, so I'm not sure that I'd count that as an advantage for the Polycom. I haven't seen a Polycom in person, but I haven't heard anything bad about them. My 7940 works well, and I wouldn't hesitate to recommend it, but for the money, the Polycom is quite likely a better phone. I didn't find the 7940 to be particularly difficult to configure, *EXCEPT* for the initial installation of the SIP firmware. It's a multi-step upgrade, because you can't directly upgrade from the SCCP image that it ships with to a modern SIP image. Once you get past that, it isn't too bad, particularly if you have multiple phones. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 1033 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
