Asterisk can with the sendtext cmd which is available in CVS-HEAD.
On 5/18/05, Chris Coulthurst <[EMAIL PROTECTED]> wrote:
>
>
>
> Can anyone explain the Polycom Text Messaging features built in to the IP
> 500/600? Can Asterisk (or something else) talk to it? I've seen vague
> references
Since the Polycom Instant Messaging features uses MSN Messenger, I
doubt it will work with Asterisk.
C F wrote:
Asterisk can with the sendtext cmd which is available in CVS-HEAD.
On 5/18/05, Chris Coulthurst <[EMAIL PROTECTED]> wrote:
Can anyone explain the Polycom Text Messaging features built
[EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Eric Wieling aka ManxPower
> Sent: Wednesday, 18 May 2005 8:41 PM
> To: C F; Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Polycom Instant Messaging
>
> Since the Polycom I
Eric Wieling aka ManxPower wrote:
Since the Polycom Instant Messaging features uses MSN Messenger, I doubt
it will work with Asterisk.
The Polycoms don't use MSN Messenger. Although I am not quite sure what
you mean... Anyways, they use SIMPLE.
--
Kristian Kielhofner
__
MSN Messenger does not support SIP, Windows Messenger does. There's a
difference between the two.
On 5/18/05, Eric Wieling aka ManxPower <[EMAIL PROTECTED]> wrote:
> Since the Polycom Instant Messaging features uses MSN Messenger, I
> doubt it will work with Asterisk.
>
> C F wrote:
>
> > Aster
The LCS 2005 client will also have SIP support.
Ryan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adrian A
Sent: Thursday, May 19, 2005 2:17 AM
To: Asterisk Users Mailing List - -Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom
erisk Users Mailing List - Non-Commercial Discussion; C F
|Subject: RE: [Asterisk-Users] Polycom Instant Messaging
|
|There is a really good article in this months months Von magazine on
|page 26 about why asterisk will need to adopt sip extensions for
|Microsoft messenger.
|
|I'd post here but
Sent: Thursday, 19 May 2005 2:45 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Polycom Instant Messaging
>
> If anyone has had success with IM with these phones in ANY
> configuration, I, as well as others Im sure woul
> If anyone has had success with IM with these phones in ANY
> configuration, I, as well as others Im sure would love to hear about how
> its done.
If this is all you want, then it can be done like this:
create a .call file that drops into a context that does the following
1. calls the polycom pho
Actually, it looks like I'm getting this problem on all my phones. When I was
testing my phones, most worked pretty well with an occasional complaint from
the Polycom.
I've moved them now to a different location and the ISP must have different
NAT translation going on that make it more difficult
: "Michael George" <[EMAIL PROTECTED]>
To:
Sent: Friday, May 27, 2005 11:26 PM
Subject: [Asterisk-Users] Polycom phones, UNREACHABLE
I'm having some trouble with Polycom Soundpoint phones. I have had good
luck
deploying them on a local network, but now I've tried put
oming unreachable?
> - Original Message -
> From: "Michael George" <[EMAIL PROTECTED]>
> To:
> Sent: Friday, May 27, 2005 11:26 PM
> Subject: [Asterisk-Users] Polycom phones, UNREACHABLE
>
>
> >I'm having some trouble with Polycom Soundpoi
Michael George wrote:
On Sat, May 28, 2005 at 11:10:30AM -0400, Steve Totaro wrote:
qualify = yes is what is causing the messages. You can assign a value
rather than yes. like 1000 or something or you can remove the qualify
statement alltogether. The message is just a warning. Eliminating
I'm having the same issues with the polycom phones, as well as with
Sipura ata's. I am also using on another natted network a sipura ata,
that I changed the settings on the sipura that might help, and it did
help, I havn't had an unreachable message since.
I'm not sure if on the second network the
I now have the problem solved (or so I think) even without the qualify
in sip.conf.
It happens to be that the problem was just that the firewall would not
allow the packets back in after a specific time. All I had to do was
create trigger rules on the firewall to allow the packets back in
based on
I just ran into an interesting problem. I have a Polycom IP500 that up
until today has been rock solid. We have started exploring video
conferencing and all of a sudden IP500 goes weird. If I initiate a call
to any other sip device, it works fine. But on an incoming call I get
no sound.
If I
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Monday, May 30, 2005 11:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom phones, UNREACHABLE
I now have the problem solved (or so I think) even
Does anyone know if there is any way to make the 'services' key do
anything?
How about a way to remap it?
Chris Coulthurst
[EMAIL PROTECTED]
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/ast
Hello, I just wiped out my old asterisk install and installed Asterisk
at Home. I was quickly able to get my Digium TDM422P working, 2 POTS
lines, 2 phones. I also got X-Lite working as a SIP extension. I then
tried to setup my Polycom IP 500, and this was not so easy...
Using AMP I created
Hi all,
I've been messing around with the g729 codec in
some phones I use and had made all phones use the codec for all calls for
testing purposes. The problem is when I attempt to dial out on my Polycom
IP 500 (test happens to be calling my cell phone) I can only hear sound coming
one wa
-Commercial Discussion
|Subject: [Asterisk-Users] Polycom and CallerID
|
|
|I'm having a problem with the callerID that the polycom IP600
|phones are
|displaying. I would like to modify the CIDName and leave CIDNumber as
|exactly what the phone call came in as(provided they aren't hiding
We have the following versions:
App Version: 1.4.1.0040 (SIP)
Bootrom: 2.6.1.0003
I also noticed that the polycom IP600 phones are Rev 3.
--johann
Chris Coulthurst wrote:
Which software pack to you have for the IP600? Sip.ld, bootrom, etc...
Chris Coulthurst
[EMAIL PROTECTED]
_
This works for me:
to display the following on the polycom phone:
From: Support-Group
x
<<--- the caller id number
you can use the following code in extension.conf:
exten => 301, 1, Dial(SIP/456&SIP/455&SIP/457, 30)
exten => 3
Hello,
if somebody is interested in Europe for 2 polycom
soundpoint ip 300 for testing with Asterisk contact me
out of list .
Regards
Harry
___
Appel audio GRATUIT partout dans le monde
Just so you know who you're dealing with:
-- Forwarded message --
From: harry gaillac <[EMAIL PROTECTED]>
Date: Jun 24, 2005 7:58 PM
Subject: Re: [Asterisk-Users] polycom soundpoint ip 300
To: Wilson Pickett
i piss on you Wilson Pickett
Harry from France
> &g
Have you considered playing with the timeouts?
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Pushor
Sent: Monday, June 27, 2005 4:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Polycom &
Subject: [Asterisk-Users] Polycom & VPN trouble
Hi All,
I am a remote office that is connected to my office via openvpn on UDP.
Voip has always worked well (after discovering g729). Initially I used a
softphone, then an analog set on a sipura 2000, then a polycom IP500 (I
still LOVE this p
I'm attempting to set up my SoundPoint 501 with my Asterisk server.
I've configured DHCP and TFTP and successfully updated both the BootRom and SIP
application. I've also created a custom cfg file for this phone's MAC address and
the settings seem to be taking just fine. I can see that the
Eric Rees wrote:
I am having a problem with Polycom auto-answer. I have the auto-answer
working between PhoneA and PhoneB, but when I try to use the intercom
between more then one phone I start having problems. PhoneA dials *3
which calls PhoneB, PhoneC, and PhoneD. All the phones ring, but only
ch 01, 2005 10:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom Auto-Answer
Eric Rees wrote:
> I am having a problem with Polycom auto-answer. I have the
auto-answer
> working between PhoneA and PhoneB, but when I try to use the interco
Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Polycom Auto-Answer
That's kind of what I thought, but I am trying to put together a phone
to multi-phone paging system. I all ready have and overhead paging
systems, but the powers-at-be want a phone paging s
B. J. Bomar wrote:
Take a look at
http://www.voip-info.org/wiki-Cisco+7940-7960+auto-answer+config. With a
few small modifications it should work like a champ on the Polycom phones.
B. J.
Polycom has a much better way to do auto-answer using SIP_INFO. My
sample configs have both a AutoAnswer and
Kristian Kielhofner wrote:
B. J. Bomar wrote:
Take a look at
http://www.voip-info.org/wiki-Cisco+7940-7960+auto-answer+config. With a
few small modifications it should work like a champ on the Polycom
phones.
B. J.
Polycom has a much better way to do auto-answer using SIP_INFO. My
sample conf
That worked great.
-Original Message-
From: B. J. Bomar [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 01, 2005 11:20 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Polycom Auto-Answer
Take a look at
http://www.voip-info.org/
Eric Wieling wrote:
Polycom has a much better way to do auto-answer using SIP_INFO. My
sample configs have both a AutoAnswer and a Ring_Answer (for intercom):
http://www.kriscompanies.com/modules.php?name=Downloads&d_op=viewdownload&cid=1
That's only for station to station paging, not all stat
Discussion
Subject: Re: [Asterisk-Users] Polycom Auto-Answer
B. J. Bomar wrote:
> Take a look at
> http://www.voip-info.org/wiki-Cisco+7940-7960+auto-answer+config. With a
> few small modifications it should work like a champ on the Polycom phones.
>
> B. J.
Polycom has a much better
Hey guys, I’m interested in the XML Support that the
Polycom phones have.
I want my techs to be able to view queue information via the
XML screen. Is this possible?
When I say queue information, I mean how many people are
waiting in the queues (3 queues combined), how long the wait
On Mar 7, 2005, at 5:52 PM, [EMAIL PROTECTED] wrote:
Hi, all
I have two questions regarding usage of Polycom SP300 with Asterisk.
No sure if it
is Astersisk or phone related, though.
1. When dialing an extension, one has to perss Dial or Send on the
phone after
number is entered. Is it possible
A quicker way to get to the missed calls list is to hit the down arrow button. Just exiting out just clears the display of missed calls and resets counter, the records are still in there. Look for the polycom remote reboot script and reboot the phones daily to clear the list for good so you d
Hello list:
I have a very odd problem. Seemingly randomly, my Polycom IP600 phones
will ring without a call being placed to it.
That is to say, a random phone will ring. Nothing shows up under Caller
ID. Even the buttons that light up to show an incoming call do not light
up. If you pick up the
, 8 March 2005 10:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Polycom SP300 questions
Hi, all
I have two questions regarding usage of Polycom SP300 with Asterisk. No sure if
it is Astersisk or phone related, though.
1. When dialing an extension
: [Asterisk-Users] Polycom SP300 questions
Hi, all
I have two questions regarding usage of Polycom SP300 with Asterisk.
No sure if it is Astersisk or phone related, though.
1. When dialing an extension, one has to perss Dial or Send on the
phone after number is entered. Is it possible to avoid this
I googled and googled but could not find anything regarding this problem.
I have Asterisk CVS-HEAD-03/19/05. been running Asterisk for over 6
months, no problems with my grandstreams. I'm fairly familiar with the
ins and outs of asterisk...
IP600 with latest sip 1.4.1 and bootrom from my FTP serv
I'm having a problem with my Polycom phones and hoping someone else has
experienced the same thing: Outbound calls are fine, and inbound calls
originating from another SIP phone are fine, but inbound calls to the
Polycom phone from an IAX channel sound like you're talking to a robot.
The perso
Title: polycom IP 500/600
Hi,
A few questions about polycom IP 500/600.
. how do I reset everything to factory default? The combination of 4,6,8,* seems only reset the network setup. Other settings, e.g. time server, SIP configure are still there.
. is a way to push ftp username/password v
http://www.voip-info.org/tiki-index.php?page=Polycom%20Phones
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: zaterdag 6 november 2004 19:48
To: Asterisk-a-users-list
Subject: [Asterisk-Users] Polycom 500 software?
Just unboxed two
Polycom ships out two different phones , ones with H323,and one with SIP
already loaded.
Thank you,
Steve Maroney
On Sat, 6 Nov 2004, Rich Adamson wrote:
> Just unboxed two brand new SoundPoint IP 500 phones. There was no software
> shipped with the units. Is the basic sip software (v1.3.1?) av
>
> On Sat, 6 Nov 2004, Rich Adamson wrote:
>
> > Just unboxed two brand new SoundPoint IP 500 phones. There was no
> software
> > shipped with the units. Is the basic sip software (v1.3.1?)
available
> > from somewhere this weekend?
> >
> > Tried digging around the Polycom site including registe
> >
> > Polycom ships out two different phones , ones with H323,and one with
> SIP
> > already loaded.
> >
> > Thank you,
> > Steve Maroney
>
> Correction, the polycom IP 500 ships without h.323 or SIP
> software (it only has a bootrom on it), and software is only
> distributed by polycom auth
> > > Polycom ships out two different phones , ones with H323,and one with
> > SIP
> > > already loaded.
> > >
> > > Thank you,
> > > Steve Maroney
> >
> > Correction, the polycom IP 500 ships without h.323 or SIP
> > software (it only has a bootrom on it), and software is only
> > distributed
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Peter Johnson
> Sent: Monday, November 08, 2004 1:02 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk
Rich Adamson wrote:
Polycom ships out two different phones , ones with H323,and one with
SIP
already loaded.
Thank you,
Steve Maroney
Correction, the polycom IP 500 ships without h.323 or SIP
software (it only has a bootrom on it), and software is only
distributed by polycom authorized VoIP partn
rs-
> > [EMAIL PROTECTED] On Behalf Of Peter Johnson
> > Sent: Monday, November 08, 2004 1:02 AM
> > To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> > Subject: RE: [Asterisk-Users] Polycom 500 software?
> >
> > > >
> > > &g
On Mon, 8 Nov 2004, Rich Adamson wrote:
> > That's not right.
> > New phones come loaded with the current relevant firmware.
> > Upgraded f/w is only available to/from certified resellers.
> > Or look on the wiki for where it is freely available.
>
> The two new 500's that were purchased from a P
: [Asterisk-Users] Polycom phone question
Does anybody know if the CS version of the Polycom handset will take the SIP
image. If I have read correctly, the CS version is for Cisco Call Manager,
and is Cisco "certified." Thanks in advance.
B. J.
__
I'm ordering some more phones - I have the Polycom IP 500's now and I
like them. I need some less expensive phones, and I'd like to stay
with all Polycoms for ease of administration. I've heard, though, that
the IP 300's don't support PoE even though their brochures say they do.
Has anybody
Received two new Polycom 500 phones. Dhcp and ftp configured properly to
load the various files including v2.5.0 bootrom.ld, etc. One of the phones
loaded all firmware and config files properly, registers with *, and is
usable.
The second phone loads bootrom.ld (from the same ftp server on the sam
Has anyone had an issue with the polycom's not discontinuing the mwi
chirp even after the message has been acknowledged?
--
James M. Milne
Nuvio Corporation
CCNA - CCNP - CIPTSS - CCIE
[EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi, I have was testing some of the different ring types with
my polycom 500, and the ALERT_INFO settings. Now when my phone receives a call
it won’t ring. All the other phones ring fine, and my phone wasn’t the
only one I rebooted with the changed sip.conf and impd.conf. I have reverted
bac
hi all,
can i hook up one of the polycom table top phones (soundpoint pro SE220
or 225, for example) to an FXO port on Asterisk? or do they require some
proprietary polycom PBX to work?
thanks in advance.
Flynn
___
Asterisk-Users mailing list
[EMAIL PR
Jeff Pyle [mailto:[EMAIL PROTECTED]
> Sent: Friday, 17 September 2004 11:27 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Polycom IP500
>
> I have two IP 500's on my Asterisk PBX. The IM features just kinda worked,
> wi
ECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff Pyle
Sent: Friday, September 17, 2004 5:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: FW: [Asterisk-Users] Polycom IP500
See, that's just the thing. I didn't. It just worked! I did some
limited packet traces
When I send the message from the phone, I use only the phones
extension. Or, I think I am. I set up each phone's "buddy list" with
the extension (only) of the other phone. When I send the message, I
just picked the buddy off the list.
Both the sip.conf and phone configs are pretty vanilla.
[20
I have a Polycom IP500 that will not dial out if it is
off-hook. From speakerphone it will work, I presume that it has to do with
caching of digits, but if the handset is picked up it will drop the dial
attempt after the 7th or 8th digit.
Has anyone else seen this and if so how do I cor
Anyone successfully set up one of the polycom soundpoint ip sidecars
with asterisk to monitor and allow transfer to monitored extensions?
How does it work? Any issues?
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing lis
I have a Polycom IP501 phone and have set it up to download the config from an
FTP server, it did this once and now is in an endless loop of trying to contact
the FTP server, failing, then rebooting.
When I watch the FTP server logs it looks like the phone starts a session, ends
it, starts it,
Carlos Chavez wrote:
Is there any way to increase the number of digits before the number is
diales automatically?
Yes,
I don't know about the 601s, but under the 301s and the 501s you can
edit the digit map via the web interface or the sip.cfg on your ftp server.
Doug
--
Ben Frank
On 2/9/06, Doug Lytle <[EMAIL PROTECTED]> wrote:
>
> Carlos Chavez wrote:
> > Is there any way to increase the number of digits before the number is
> > diales automatically?
> >
>
> Yes,
>
> I don't know about the 601s, but under the 301s and the 501s you can
> edit the digit map via the web
As an example, here is my custom digitmap:
dialplan.digitmap="9,911|9,411|0T|00T|1xx|9,011x.T|9,1[2-9]xx[2-9]xx|9,[2-9]xx|*7x|7x|*1xx|*8"
The | is used to separate different entries. The comma means that it'll
keep providing the dial tone after hitting 9. If you see a T after one
of t
you didn't mention which phone you have, but don't forget that to make
the services key, for example, behave the way you want, the key numbers
change between the 50x and th 601:
key.IP_500.31.function.prim=
key.IP_600.29.function.prim=
But I suspect that since your remapping happens, but you g
Guys.
I would like to hear some comments about people using polycoms 600 IP phones
and what their doing with their minibrowsers? Any inetresting apps that you
might want to share?
Thanks
AK
___
--Bandwidth and Colocation provided by Easynews.com --
A
On 2/23/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I have several Polycom 501 connected to asterisk. The phone has an
> ACD-login function that I'd like to use. But I can't find find much
> information about this.
>
> I've read a post on [EMAIL PROTECTED]
> (http://bugs.digium.com/v
ligt meddelande-
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För BJ Weschke
Skickat: den 23 februari 2006 13:44
Till: Asterisk Users Mailing List - Non-Commercial Discussion
Ämne: Re: [Asterisk-Users] Polycom 501 ACDlogin
On 2/23/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
>
ens...and I can't find any readmes' or
> manuals.
>
> Regards,
> Jan
>
>
> -Ursprungligt meddelande-
> Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För BJ Weschke
> Skickat: den 23 februari 2006 13:44
> Till: Asterisk Users Mailing List - Non-Comme
sprungligt meddelande-
> > Från: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] För
> BJ Weschke
> > Skickat: den 23 februari 2006 13:44
> > Till: Asterisk Users Mailing List - Non-Commercial
> Discussion
> > Ämne: Re: [Asterisk-Users] Polycom 501 ACDlogin
> >
&g
; Jan
> >
> >
> > -Ursprungligt meddelande-
> > Från: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] För
> BJ Weschke
> > Skickat: den 23 februari 2006 13:44
> > Till: Asterisk Users Mailing List - Non-Commercial
> Discussion
> > Ämne:
ng
Skickat: den 23 februari 2006 16:13
Till: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Ämne: RE: [Asterisk-Users] Polycom 501 ACDlogin
You don't need the Polycom ACD support in order to do ACD logins with Polycom
phones. Just dial an extension and call Age
> [test]
> type=friend
> secret=blahpoly
> insecure=yes
> host=dynamic
> qualify=500
> nat=no
> mailbox=testmailbox
> callerid=Yourname
> conext=local
> disallow=all
> allow=ulaw
> progressinband=no
>
> here is the local section of the dial plan.
> exten => 850,1,Goto(Mercury-Network,850,1)
> exte
What does the dialplan for the Polyocm 601 (the one the phone uses,
not Asterisk) look like?
You can see if it's a polycom or asterisk thing, by enabling sip
debug, and watch what is coming in from the Polycom. if nothing is
coming then it's the Polycom doing it.
On 2/23/06, Kevin Smith <[EMAIL PR
: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Douglas Garstang
|Sent: Thursday, February 23, 2006 9:13 AM
|To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
|Non-Commercial Discussion
|Subject: RE: [Asterisk-Users] Polycom 501 ACDlogin
|
|You don't need the Polycom ACD suppo
> Hey everyone, I haven't seen an issue quite like mine, so I am hoping
> anyone who used the Polycom 601's may have an idea.
>
> We are going to be switching our office over to Asterisk. All the phones
> are going to be 601's, I am going to set up a boot server, but for now I
> am just going
Hey guys,
Thanks for the suggestions. I did find the problem. Looking in the sip
debug, I was getting a 407 error, corrected that, then was getting a
404. Which lead me to look at my context and bam...typo, I had conext
instead of context. Corrected that and all is well. Thanks again.
Kevin
Does anybody know how to set polycom's default ring volume ? Everytime you
restart a polycom phone, ring defaults to a very low volume setting which is
kind of annoying having to set everytime you reboot.
Any hints?
___
--Bandwidth and Colocation provid
>|While I'm asking about the Polycom ip500, the answers for all phones
>|where mic/handset/headset levels are adjustable would be of
>interest to
>|many I'm sure.
>|
>|For the ip500, the default value for the handset seems to be
>|voice.gain.tx.analog.handset="3"
I have a number of IP600s and
Very good phone, strongly recommended.
Wojtek
- Original Message -
From: "Tim Connolly" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, March 09, 2006 5:05 PM
Subject: [Asterisk-Users] Polycom 4000 resul
Use the callerid rewite rule to prepend a 9 on the asterisk side
-Original Message-
From: "Bill Gibbs" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: 3/11/06 6:52 PM
Subject: [Asterisk-Users] Polycom - directory dia
: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] Polycom
- directory dial
This is not an Asterisk specific question but doesn’t
anyone know if you can automatically prepend a 9 on the call lists so clients
can return dial without having to repunch in the number
Do you mean, say number 444-555- calls in. You want to hit dial for
that number, from say the missed calls list, and have it on add a 9 in
front? If so just do this in extensions.conf
exten => _9NXXNXX,1,Dial(Zap/g1/${EXTEN} ;Takes calls with a 9
exten => _NXXNXX,1,Dial(Zap/g1/9${EX
Hi, all
Sorry for not exactly on-topic question
I got Polycom SP300 phones. Somehow they did not come with software. I will
call them on Monday, but in the meantime, I would like to get them going.
I need Polycom configuration template files (phone.cfg, sip.cfg and whatever
else they supply).
Chris Gamble wrote:
From their website, the key difference between the polycom 500 and 600 phones is the number of "lines" they support. What does this mean in terms of asterisk? Do I have to have a seperate extension for each of these lines or ?
Also, slightly off-topic, how does the 500 POE "
Just because the phone has the extra "lines" doesn't mean you are
required to use them. Each "line" can handle 2 calls.
The 600 has a working XML microbrowser, which the 50x does not.
The Polycom 501 (not sure if the 500 is the same) doesn't have a place
on the phone to plug power in. It gets
We use the 500's and they are great. It has 3 lines, which is plenty.
The resolution on the 600 screen is better, but unless you are using
graphics on the LCD, it isn't worth the extra $$. We have a couple of
300's and the screen is ugly. It is small and hard to read. I would
stick with the
I have a user that just got a broadband connection so she could have an
extension off our pbx. The service is DSL and uses a speedstream 5200
dsl router. I sent her a Polycom IP300. At first it would not access the
config files via ftp so I had tech support walk her through setting the
phone's
CVS Head from 07/07/2005
I'm trying to make an IP-501 auto answer a call.
exten => 301,1,SetVar(_ALERT_INFO="Ring_Ans")
exten => 301,2,SetVar(ALERT_INFO="Ring_Ans") # Tried both combinations
exten => 301,3,Dial(SIP/5001,15)
exten => 301,4,Hangup
Sip.cfg for Polycom phone
Ipmid.cfg
Asteri
you can also use answer as the ring type instead of ring-answer if you just
want it to pick up.
I would keep "Ring_Ans" the same throughout for simplicity
exten => 301,1,SetVar(_ALERT_INFO=Ring_Ans)
add an alert info type (say type 5)
then elsewhere add a corresponding ring type (say type
Hi, all
I have Polycom SP300 phones. My extension range is 1xx, so I added
corresponding entry to the digitmap.
By some reason this does not affect on-hook dialing. If I have phone
off-hook all is ok. dial extension 102 for example and it connects.
if phone is off-hooh, however, I have to pre
I've had one on my desk for a couple weeks now. What I've done:
Used DHCP to get the IP address / gateway / ntp server / dns server.
Not used DHCP to get the FTP server (rather than futz with my DHCP
server settings).
Manually set the FTP server IP/username/password on the phone (I didn't
want
chris gamble wrote:
I just received my first polycom 501, tried my best to follow all of the
documentation and configs on the wiki ( though some conflicted in which
case i tried both ways ), and at the end of the day can not connect my
phone to asterisk.
My Questsions: does the wiki information
Kristian Kielhofner wrote:
chris gamble wrote:
I just received my first polycom 501, tried my best to follow all of the
documentation and configs on the wiki ( though some conflicted in which
case i tried both ways ), and at the end of the day can not connect my
phone to asterisk.
My Questsio
Hi, all
I have Polycom SP300 phones.
Calls between those are ok and quality is great.
Then I have IAX2 soft phones (FireFly). Calls between those are OK too.
But when I have call b/w Polycom (SIP) and IAX, I have really bad echo at
Polycom phone side. IAX phone side is OK.
Any ideas?
Thanks
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