I found the solution to having calls from SugarCRM auto-answer on the
extension I route it to. I am documenting here for users who may want to
do the same thing.
Setup
I have a second extension on my phone (500) which the calls are routed
to, but there is no reason this wold not work on the
)
-Original Message-
From: [EMAIL PROTECTED] on behalf of Stephen Bosch
Sent: Fri 3/9/2007 4:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Polycom call parking feature and Asterisk callparking
Hi:
I want to make parking calls easier for my
Stephen Bosch wrote:
Chris Mason (Lists) wrote:
I am using SugarCRM together with the asterisk plugin, which allows me
to click a number, SugarCRM calls my extension then places the call when
I pickup.
I would like to have that extension auto-answer. I set it up as line 3
on my phone so
Bill Gibbs wrote:
Using the Park button actually requires more work than just doing an
attended transfer to the park extension
Does it? How does it work exactly?
What I would expect:
Press the Park button; hear the announcement; hang up
The way it works now:
Press the Transfer button;
Stephen Bosch wrote:
Bill Gibbs wrote:
Using the Park button actually requires more work than just doing an
attended transfer to the park extension
Does it? How does it work exactly?
What I would expect:
Press the Park button; hear the announcement; hang up
The way it works now:
Press the
Eric ManxPower Wieling wrote:
Stephen Bosch wrote:
Bill Gibbs wrote:
Using the Park button actually requires more work than just doing an
attended transfer to the park extension
Does it? How does it work exactly?
What I would expect:
Press the Park button; hear the announcement; hang up
:[EMAIL PROTECTED] On Behalf Of Stephen
Bosch
Sent: Saturday, March 10, 2007 12:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom call parking feature and
Asteriskcallparking
Bill Gibbs wrote:
Using the Park button actually requires more work
I have the polycom auto-answering calls to the extension I am using by
using this in the dialplan
exten = 830,1,SIPAddHeader(Alert-Info: RANR)
exten = 830,n,Dial(SIP/830,25,t)
However, I want the feature for the SugarCRM contact feature, which uses
the manager interface to place the call to
Hi:
I want to make parking calls easier for my hard-working users. Is there
a way to make the Polycom call park feature work with Asterisk?
In case it just works out of the box, I haven't tried it yet; but the
call park feature isn't enabled on the Polycom phones by default.
-Stephen-
Chris Mason (Lists) wrote:
I am using SugarCRM together with the asterisk plugin, which allows me
to click a number, SugarCRM calls my extension then places the call when
I pickup.
I would like to have that extension auto-answer. I set it up as line 3
on my phone so normal calls do not get
Chris Mason (Lists) wrote:
I am using SugarCRM together with the asterisk plugin, which allows me
to click a number, SugarCRM calls my extension then places the call when
I pickup.
I would like to have that extension auto-answer. I set it up as line 3
on my phone so normal calls do not get
The dialplan looks OK, depending of course on the numbers you trying to
dial. If you want the phone to wait for a given timeout period after the
digits are entered add a T immediately after the specific dialplan
rule. (ie: xx[2-9]xxT). I'm assuming from your rules you need to
dial a 9
»Steven Ringwald« wrote:
Any Polycom gurus out there? If so, I have a few config file questions.
First off, does anyone have the daylight savings time rules written
for this Sunday's big change?
Secondly, if there any way in the config file to tell the phone not to
display the number of
DST rules can be found by searching the sip.cfgg file for SNTP.
You will find a cluster of time parameters, including the month and
day upon which to change DST.
Thanks,
Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
iaxtel:
I am using SugarCRM together with the asterisk plugin, which allows me
to click a number, SugarCRM calls my extension then places the call when
I pickup.
I would like to have that extension auto-answer. I set it up as line 3
on my phone so normal calls do not get auto-answered. However, I have
Here is some of my actual Polycom config files. The only thing that has
been changed is the hostname of the PBX. We assign the SIP user ID as
the MAC of the phone with a -a -b -c, etc appended to it for each of the
line appearances. These config files do not have the DST stuff added yet.
»Steven Ringwald« wrote:
Any Polycom gurus out there? If so, I have a few config file questions.
First off, does anyone have the daylight savings time rules written for
this Sunday's big change?
Secondly, if there any way in the config file to tell the phone not to
display the number of
Doug Lytle wrote:
»Steven Ringwald« wrote:
Any Polycom gurus out there? If so, I have a few config file questions.
First off, does anyone have the daylight savings time rules written
for this Sunday's big change?
Secondly, if there any way in the config file to tell the phone not
to
Dave Fullerton wrote on 3/6/07 9:33 AM:
Polycom's 2.1.0 firmware has the new DST settings as the default. This
is what they use for the SNTP element:
SNTP tcpIpApp.sntp.resyncPeriod=86400 tcpIpApp.sntp.address=
tcpIpApp.sntp.address.overrideDHCP=0 tcpIpApp.sntp.gmtOffset=
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Miller
Sent: Tuesday, March 06, 2007 7:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Questions
Dave Fullerton wrote on 3/6/07 9:33 AM
Any Polycom gurus out there? If so, I have a few config file questions.
First off, does anyone have the daylight savings time rules written for
this Sunday's big change?
Secondly, if there any way in the config file to tell the phone not to
display the number of missed calls? I don't mind it
Jason Walker wrote:
I have users in my dialplan that go from SIP to Cell
When they are at their desk and they hit reject call, it goes to the
next thing in the dialplan, thus transferring to their cell. Not what
they want. Is it possible to change the reject button to make it go to
voice
I have users in my dialplan that go from SIP to Cell
When they are at their desk and they hit reject call, it goes to the
next thing in the dialplan, thus transferring to their cell. Not what
they want. Is it possible to change the reject button to make it go to
voice mail or a new ext?
Hi Guys,
A while back (several months ago) I was having issues with wmy Polycom's and
Asterisk. I was told to use a specific set of firmware and sip version. I am
unable to find that email. Anyone know which ones work well with Asterisk ? (I
believe it was 2.x )
Thanks,
Dovid B wrote:
Hi Guys,
A while back (several months ago) I was having issues with wmy
Polycom's and Asterisk. I was told to use a specific set of firmware
and sip version. I am unable to find that email. Anyone know which
ones work well with Asterisk ? (I believe it was 2.x )
I have yet to
- Original Message -
From: Doug Lytle [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, February 27, 2007 7:05 PM
Subject: Re: [asterisk-users] Polycom Firmware
Dovid B wrote:
Hi Guys,
A while back (several
On Tue, 2007-02-27 at 19:14 +0200, Dovid B wrote:
Doug is this for the sip version or firmware ? As far as I know once you go
beyond a certain firmware version with polycom you cant go back.
Dovid
We used bootrom version 2.6.1.
And yes, once you go to version 3.x, you cannot go back.
Found
Dovid B wrote:
- Original Message - From: Doug Lytle [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, February 27, 2007 7:05 PM
Subject: Re: [asterisk-users] Polycom Firmware
Dovid B wrote:
Hi Guys
Dovid B wrote:
Doug is this for the sip version or firmware ? As far as I know once
you go beyond a certain firmware version with polycom you cant go back.
Sip 1.5.2
Bootrom 3.1.3
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Sip 1.5.2
Bootrom 3.1.3
Anyone know any good reasons NOT to use the latest? I believe Bootrom
3.2.2 B and Firmware (SIP) 2.1.0. It's also the stuff the new IP 650's
are using.
-Kenneth
___
--Bandwidth and Colocation provided by Easynews.com --
I know this is not a Polycom support forum, but I also know there are a lot
of you with a great deal of Polycom experience.
Is there anyway to remove the Attended Transfer but keep the Blind
transfer? Or better yet, just swap the two soft buttons locations?
I know you can remap the Hard buttons,
In later 1.6.x firmwares there is a config option for allow transfer on
proceeding that basically allows you to do a blind transfer by just
hitting the transfer key again rather than having to select Blind.
Shawn Kelley wrote:
I know this is not a Polycom support forum, but I also know there
-users] Polycom SIP 501 Transfer Question
In later 1.6.x firmwares there is a config option for allow transfer on
proceeding that basically allows you to do a blind transfer by just
hitting the transfer key again rather than having to select Blind.
Shawn Kelley wrote:
I know this is not a Polycom
I have been given the task of getting a Polycom IP 601 running SIP
v2.0.1.0291 to register with our SER proxy and be able to interact with
our Asterisk server for voice mail. The Asterisk server currently sends
unsolicited NOTIFY messages to turn on/off the message waiting light.
Most of
Hi Friends,
This is Chandra from India. I have installed and configured Asterisk in our
company. I want to provide Polycom IP 501 model phones to our employees. I am
unable to find the dealer for these phones in India. Where can I buy these
phones in India? If anybody knows, please tell me the
If you already havent seen this:
http://dir.indiamart.com/impcat/video-telephone.html
cheerz
- Ben.
Crazy Boy wrote:
Hi Friends,
This is Chandra from India. I have installed and configured Asterisk
in our company. I want to provide Polycom IP 501 model phones to our
employees. I am unable
Jason,
Email me off-list and I will ship you a pack of usable configs.
Thanks,
Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
iaxtel: 700:248:2637 x:1500
http://www.sheltonjohns.com
On Jan 26, 2007, at 3:48 PM,
From what I know this log show everything working Perfect.
Then it goes to the Welcome screen then after a long time of processing,
it errors out with a 0x1 error
Any Ideas?
1005195711|so |4|00|-- Initial log entry --
1005195711|so |4|00|+++ Note that bootrom log
Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Polycom Provistioning Issue
From what I know this log show everything working Perfect.
Then it goes to the Welcome screen then after a long time of processing,
it errors out with a 0x1 error
Any Ideas?
1005195711|so |4
Looks like the network time server isn't provisioned.
--
Bill
1005195752|app1 |4|00|Could not load time from 0.0.0.0(0.0.0.0).
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options
Fixed that issue but it does not change the error
0126204105|cfg |3|00|Image sip.ld has not changed
0126204105|copy |3|00|Download of 'sip.ld' succeeded on attempt 1 (addr
1 of 1)
0126204105|cfg |3|00|Downloaded application image is identical to
current version
0126204105|cfg |3|00|Phone
]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Polycom Provistioning Issue
Fixed that issue but it does not change the error
0126204105|cfg |3|00|Image sip.ld has not changed
0126204105|copy |3|00|Download of 'sip.ld' succeeded on attempt 1 (addr
1 of 1
PROTECTED] On Behalf Of Jason
Walker
Sent: Friday, January 26, 2007 12:10
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Polycom Provistioning Issue
From what I know this log show everything working Perfect.
Then it goes to the Welcome screen then after a long time
13:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Provistioning Issue
?xml version=1.0 standalone=yes?
!-- Default Master SIP Configuration File--
!-- Edit and rename this file to Ethernet-address.cfg for each
phone.--
!-- $Revision: 1.14
.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Robert Jenkins
Sent: 16 January 2007 20:44
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Polycom IP601 - some hints working,
not others?
Hi,
I've got
I've just realised - the directory entries that have working Buddy watch are
the first in sequence when the extensions are sorted into NAME order, which
the phones do when saving their directory files.
Looks like it could be a watch limit in that version of the firmware?
Could it be that it's
I have a soundpoint 501 phone that has locked up twice now. You can make
a call but when a call is sent to it, it responds with sip busy
messages. You get the same message when the phone is in do not disturb.
I reset to defaults the first time and it worked for a week or so and
then stopped. The
Hi,
I've got an Asterisk setup including a TDM2400 for analog trunks
extensions plus two IP501s an IP601 (all firmware 1.6.7 as supplied).
The initial buddy / hint setup was fairly straightforward, but I have a
strange problem in that some extensions don't show any status indication.
Asterisk
-users] Polycom IP601 - some hints working, not
others?
Hi,
I've got an Asterisk setup including a TDM2400 for analog trunks
extensions plus two IP501s an IP601 (all firmware 1.6.7 as
supplied).
The initial buddy / hint setup was fairly straightforward, but I have
a
strange problem
[mailto:[EMAIL PROTECTED]
Sent: 16 January 2007 22:46
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [asterisk-users] Polycom IP601 - some hints
working, not others?
Are all of the sip phones in the same context?
-Original Message
: [asterisk-users] Polycom IP601 - some hints working, not
others?
Hi,
Yes, there are just the three Polycoms (200 - 202), the rest of the
system
is analog. The Polycoms always 'see' each other, the problem is with
them
seeing some Zap channels.
Although the 501s don't have the display of the 601
Al wrote:
I'm facing a weird issue, polycom phones work fine in the main office,
in remote office it says,
Registration from 'sip:[EMAIL PROTECTED]' failed for '70.59.21.112' -
Wrong password
the odd thing is Linksys phone works without any issue!!
Just a guess, put nat=yes in the sip.conf
I have never seen a registration failure solved with nat=yes.
Doug Lytle wrote:
Al wrote:
I'm facing a weird issue, polycom phones work fine in the main office,
in remote office it says,
Registration from 'sip:[EMAIL PROTECTED]' failed for '70.59.21.112' -
Wrong password
the odd thing is
Are you using tftp or ftp provisioning? If so, check your server
declaration in sip.cfg in your polycom configs directory.
Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
iaxtel: 700:248:2637 x:1500
Eric, I did, what's happening is that the unauthorized message from
the registering Polycom never reaches the Polycom because asterisk
doesn't know it's natted. Therfore the Polycom never supplies the
creditentials.
On 1/14/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
I have never seen a
Eric ManxPower Wieling wrote:
I have never seen a registration failure solved with nat=yes.
Just a guess, put nat=yes in the sip.conf for that phone and see if
it helps.
Hence the, Just a guess.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little
Hello list,
I was wondering if any of you guys have had any luck with polycom in remote
offices,
I'm facing a weird issue, polycom phones work fine in the main office, in
remote office it says,
Registration from 'sip:[EMAIL PROTECTED]' failed for '70.59.21.112' - Wrong
password
the odd thing
Does anybody happen to know the input power specs for the Polycom IP 500
and IP 600? We've mixed up our power supplies and we've got a whole box
of them and can't figure out which go to the Polycoms. I would rather
not kill the phones by trying random ones
501 - 12V, 1A and a power/data cable
601 - 24V, 0.5A
650 - 24V, 0.5A
- Dave
On Jan 3, 2007, at 11:48 AM, Peder @ NetworkOblivion wrote:
Does anybody happen to know the input power specs for the Polycom
IP 500 and IP 600? We've mixed up our power supplies and we've got
a whole box of them
The 501 is 12VDC, and the 601 is 24VDC, as I recall. There was a post a
few months ago that said that plugging the 24VDC into a IP501 will fry
the phone.
Peder @ NetworkOblivion wrote:
Does anybody happen to know the input power specs for the Polycom IP 500
and IP 600? We've mixed up our
FWIW, our Polycom IP601 phones use a transformer with output: 24VDC
500mA (center contact is positive).
A Polycom reseller (or Polycom sales) could probably give you
information on these other two models.
Alvin
Peder @ NetworkOblivion wrote:
Does anybody happen to know the input power specs
The IP600 is 12v!!! I fried a 600 when I used power adapter from 601.
On 1/3/07, Alvin Austin [EMAIL PROTECTED] wrote:
FWIW, our Polycom IP601 phones use a transformer with output: 24VDC
500mA (center contact is positive).
A Polycom reseller (or Polycom sales) could probably give you
Peter,
I have 600's that are 12V 1.5A, + in the center. This differs from some
of the other answers, maybe those differences are regional (although
that would seem rather silly).
HTH
B
Peder @ NetworkOblivion wrote:
Does anybody happen to know the input power specs for the Polycom IP 500
I haven't read trough the thread well enough. The 600 is 12V 1.5A
indeed. Too bad they don't all have the same voltage.
LST wrote:
The IP600 is 12v!!! I fried a 600 when I used power adapter from 601.
On 1/3/07, *Alvin Austin* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
FWIW,
Older models, 500 and 600, are 12V, newer 601s are 24v
--
Chris Mason
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED]
--
This message has been scanned for viruses and
dangerous content by
Good morning,
I have a Polycom 601 with two side cars. I created a list of contacts in XML
and it shows up on the side cars exaclty how I set it up in the
-directory.xml file (in the order that I wanted it etc.). However
when I hit the directories button and then contact directory I
I don't think that's possible. We have the same issue.
-Original Message-
From: Dovid B [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 27, 2006 8:04 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Polycom 601 Contacts List
Good morning,
I have a Polycom 601
, December 27, 2006 10:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Polycom 601 Contacts List
I don't think that's possible. We have the same issue.
-Original Message-
From: Dovid B [mailto:[EMAIL PROTECTED]
Sent
.
-Original Message-
From: Jonathan k. Creasy [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 27, 2006 10:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Polycom 601 Contacts List
There is an index in the configuration file which I believe
Hi -
We'll still need to see more of your dialplan. By your description,
it looks like the call is failing because the Dial() times out.
Take two... My calls are NOT FAILING. Never have so let me restate...
Call comes in receptionist answers. For some ungodly reason this client
does not want
(FYI client did not want VM... Don't ask...)
[general]
static=yes
writeprotect=no
[incoming]
exten = s,1,NoOP(${EXTEN})
exten = s,2,Goto(main-aa,s,1)
exten = 13015550835,1,Goto(main-aa,s,1)
exten = 3015550835,1,Goto(main-aa,s,1)
exten = 5550835,1,Goto(main-aa,s,1)
exten =
Change step 2 on your internal extensions to do whatever you want to do
(change the ringer, callID, whatever) then go to main-aa,s,1. Or, change
step 2 to go someplace else, at somplace else, do whatever you want to do,
and then go to main-aa,s,1. The second method is easier to change if, later
Hey all... Scenario
(INTERNAL)
1 Call comes in to receptionist and gets transferred to someone
2 No one picks up that transfer
3 Call goes back to receptionist
Now when the call goes back to the receptionist, how can I change either
the ringer, the callerID or both?
* TIA
Hi -
(INTERNAL)
1 Call comes in to receptionist and gets transferred to someone
2 No one picks up that transfer
3 Call goes back to receptionist
Now when the call goes back to the receptionist, how can I change either
the ringer, the callerID or both?
We'll need to see a little more info to
Noah Miller wrote:
Hi -
(INTERNAL)
1 Call comes in to receptionist and gets transferred to someone
2 No one picks up that transfer
3 Call goes back to receptionist
Now when the call goes back to the receptionist, how can I change either
the ringer, the callerID or both?
If you're looking
Hi -
(INTERNAL)
1 Call comes in to receptionist and gets transferred to someone
2 No one picks up that transfer
3 Call goes back to receptionist
Now when the call goes back to the receptionist, how can I change either
the ringer, the callerID or both?
If you're looking for the
Noah Miller wrote:
Hi -
We'll still need to see more of your dialplan. By your description,
it looks like the call is failing because the Dial() times out.
Take two... My calls are NOT FAILING. Never have so let me restate...
Call comes in receptionist answers. For some ungodly reason this
Has anyone ever gotten the Polycom Status feature, accessible via the 'MyStat'
soft-key to work? When you change the status in this way, the phone does not
send any communication to Asterisk, and it seems to have no effect in incoming
calls. So... what's it for?
Doug
On 12/13/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Has anyone ever gotten the Polycom Status feature, accessible via the
'MyStat' soft-key to work? When you change the status in this way, the phone
does not send any communication to Asterisk, and it seems to have no effect
in incoming
13, 2006 9:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom MyStat
On 12/13/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Has anyone ever gotten the Polycom Status feature, accessible via the 'MyStat'
soft-key to work? When you change
On 12/13/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Has anyone ever gotten the Polycom Status feature, accessible via the
'MyStat' soft-key to work? When you change the status in this way, the phone
does not send any communication to Asterisk, and it seems to have no effect
in incoming
Is anyone else having trouble getting a Polycom IP4000 (running SIP
1.6.7 and BootROM 3.1.3) to download its configuration files from a
vsftpd 2.0.1 server? We have 100+ IP501s that manage this without
problems, but the IP4000 keeps timing out.
We have opened a case with Polycom, but they are
Do you have the latest firmware files from polycom and sample
configurations? Can you get the phone to accept those? Any reason why you
are using FTP? Http has worked without a hitch. What does your logs say?
On 12/13/06, Anthony Rodgers [EMAIL PROTECTED] wrote:
Is anyone else having trouble
On 12/13/06, LST [EMAIL PROTECTED] wrote:
I think that is strictly a Polycom to Polycom thing (Buddywatch). I do
not believe it affects Asterisk (i.e. Busy = DND). With that being said,
I don't think it works very well even with all Polycom phones. I can
change my status to Busy and look
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom MyStat
On 12/13/06, LST [EMAIL PROTECTED] wrote:
I think that is strictly a Polycom to Polycom thing (Buddywatch). I do not
believe it affects Asterisk (i.e. Busy = DND). With that being said, I
Anyone else have problems with soft buttons not being responsive at all? 2
of the 4 soft buttons do not respond, no matter how hard you push. It is an
IP500. Well over 1 year old.
___
--Bandwidth and Colocation provided by Easynews.com --
I know this is not asterisk specific but we all know this group is used
for Polycom issues as well...
I have hints working ok on Asterisk. However the Polycom phone will
only show the buddies key if there is not a call. This defeats the
purpose of using the buddies to see if you can transfer
How did you do this ?
- Original Message -
From: Jerry Jones [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, November 30, 2006 3:12 PM
Subject: Re: [asterisk-users] Polycom 601 Second Incoming Call
you can
you can change the configs to have multiple beeps, and adjust the
timing of them, but when we tried the problem then is the beep is not
added to the incoming audio, but replaces it, so you lose the far end
speaking, went back to default.
On Nov 29, 2006, at 3:34 PM, Dovid B wrote:
Hi
On Wed, Nov 29, 2006 at 11:34:41PM +0200, Dovid B said:
I have a Polycom 601 that when the user is on the phone they only hear
one beep and the CID of the second incoming call is not shown. Is
there a way to have the CID show up for the second call ? And a way to
configure the phone to beep
Walt Reed wrote:
On Wed, Nov 29, 2006 at 11:34:41PM +0200, Dovid B said:
I have a Polycom 601 that when the user is on the phone they only hear
one beep and the CID of the second incoming call is not shown. Is
there a way to have the CID show up for the second call ? And a way to
configure
Hi List,
I have a Polycom 601 that when the user is on the phone they only hear one beep
and the CID of the second incoming call is not shown. Is there a way to have
the CID show up for the second call ? And a way to configure the phone to beep
more often if there is another call coming in. The
Hi,
I've recently got some Polycom 501 601 phones.
I have buddy watch working showing the status of users listed in the
directory.
I would like to also have the status of the trunks (ZAP via TDM2400E SIP)
on the IP601 Sidecar display, but I cannot so far find any info on this?
Thanks,
Have you tried setting up a hint for a ZAP channel?
exten = foo,hint,ZAP/bar
Then make a directory entry for foo in your Polycom directory for foo -
just as you would if the hint was for a SIP channel.
CP
On Nov 14, 2006, at 4:26 AM, Robert Jenkins wrote:
Hi,
I've recently got some
Hi List,
I have a Polycom 601 with one side car. It has one
sip account on it with 6 line displays. The receptionist is having a problem
where if she has 2-3 calls on the phone and another comes in when she tries to
get it, the phone just wont grab the call and it just rings. The phone is
Hello,
Any recommendations on Polycom Soundpoint IP601 dealers in the Toronto /
London ON areas?
Thanks,
Alvin
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I am having an issue with doing FTP auto provisioning of
Polycom 501s when they are behind a NAT. If I put the phone on the same
subnet as the provision server it loads the configs and changes fine but as
soon as I put in behind a NAT it comes up with cannot contact boot server. I
have
ate: Mon, 6 Nov 2006 19:19:48
-0600 Subject: [asterisk-users] Polycom autoprovision behind a NAT
I am having an
issue with doing FTP auto provisioning of Polycom 501’s when they are
behind a NAT. If I put the phone on the same subnet as the provision server
it loads the configs and chan
Try to ask this on asterisk commercialdiscussions list. Or check www.voipdepot.ca.
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as to what is going on so I will try passive and see if that helps.
Thanks!
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Russ Beaupre
Sent: Monday, November 06, 2006
7:45 PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
Polycom
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