I really like the IP60x phones. Have started using the IP430, so far
after 20 or so they are fine.
But the IP30x and 50x I refuse to use.
The aastra 480i is also good.
The 9133i has promise.
I do not like the snoms - any.
Grandstream are so so
Budgetone is not bad for the price, but not en
sending them out. We did try the debug before ethereal but the tech at
VegaStream insisted we will need ethereal to troubleshoot this.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry
Jones
Sent: Tuesday, July 25, 2006 11:14 AM
To: Asterisk Users
How do you mean it does not recognize them? By the routing not
working properly?
Or by not outdialing properly?
No need for ethereal, just turn on sip debugging and it will display
the messages for you, just like * will.
On Jul 25, 2006, at 9:43 AM, Issac Simchayof wrote:
I am having a
Also on the poly, for the feature you are describing you be using
join not conference
And on all pbx I have seen poly implements conference just like they
historically have
On Jul 24, 2006, at 4:50 PM, Carlos Chavez wrote:
On Mon, 2006-07-24 at 16:55 -0400, Mike wrote:
My worst gripe
Asterisk does not yet support bridged calls
You can easily have a button labeled exec 1 ring on her phone at the
same time it rings the execs phone, and have one light if he is on
the phone
Also FOP works great
On Jul 23, 2006, at 3:42 PM, Mr. Jones wrote:
Thanks Sebastian -
You're righ
pass via dhcp
if you have control of their router, just redirect dhcp requests to
your boot server and control all from there.
On Jul 21, 2006, at 11:24 AM, Stephen Murphy wrote:
I have clients in a remote location and therefore do not have
access to their Polycom phone to input the ftp i
I set nat=yes & qualify=yes in sip.cfg for the phone, not on the
phone, and works well for me as far as calls go, have other issues
but calling in and out works fine.
On Jul 20, 2006, at 12:43 PM, Frank Cernese wrote:
I saw a similar question, but the solution didn't help me. I have
my
I would agree with you on just about everything.
Except the op had his fax connected via channel bank directly to *
and a pri on the other port - ie no packets involved here.
However - all faxing does involve the transfer of frames from one fax
to the other and that is was ecm handles. But
without ecm -
line errors will cause slight imperfections (dots) on transmitted image
with ecm -
retry, retry, retry, fail
On Jul 19, 2006, at 12:23 PM, Maxim Vexler wrote:
On 7/19/06, Lee Howard <[EMAIL PROTECTED]> wrote:
Jerry Jones wrote:
> Also if possible tur
Just a couple checks...
You are using G711u for the FXS - right?
Also if possible turn off ECM on the FAX machines
Otherwise I have never used Sangoma cars but this configuration works
very well with Digium cards, at least with asterisk, I do not use aah
On Jul 19, 2006, at 11:11 AM, Bruce
This will typically happen over internet connections. If the qualify
message is lost, or takes too long the * server will stop sending
calls. This is the normal function of qualify. I find that in most
cases it is a matter of the end user saturating his connection on his
end, assuming you a
If you at least setup your ftp server, and point the phones to it,
they will save a copy of their contact database so that will not be
lost.
Just edit and save an entry after server is ready and it will create
the file.
No too hard to use the web browser and look at each phone to get its
Is not immediate for use by FXS ports? If the line is ISDN then the
number would arrive in a setup message on the D-channel.
On Jul 14, 2006, at 8:34 AM, Giordano Grandis wrote:
I cannot use it, I have the immediate=yes in my zapata, the
extension will be always 's'
Thanks again for all
Hello all,
Does anyone know of a shipping channelised T3 card which works with
asterisk? Any experiences? Seem to remember that Digium had announced
one but do not see it on their site, and Sangoma looks to not support
M13. Looking to deliver multiple PRI to multiple sites aggregated
over
This works great...
exten =>,1,Macro(vmlogin,4)
[macro-vmlogin] ;Auto login
to VM
exten => s,1,SetVar(who=${CHANNEL})
exten => s,2,Cut(who=who,,1)
exten => s,3,noop(${who})
exten => s,4,Answer()
exten => s,5,Wait(1)
exten => s,6,VoicemailMain(s${who:
chan_zap.c: Updated conferencing on
94, with 0
conference users
Jul 3 17:49:21 VERBOSE[14197] logger.c: -- Hungup 'Zap/94-1'
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry
Jones
Sent: Monday, July 03, 2006 5:15 PM
To: Ast
Are you seeing any messages on the console? You should be seeing
something like "Starting simple switch"
We would need more info to help more.
On Jul 3, 2006, at 8:18 AM, Viktor Tatianin wrote:
Hello All
Please help me, I have next problem
When lift up handset at phone which connect to ch
You can always control via the dialplan from *
Just starting to play with the Aastra so not much knowledge on them yet.
On Jul 3, 2006, at 6:41 AM, Steve Davies wrote:
Hi,
Does anybody know whether it is possible to completely disable call
waiting on an Aastra 9112i?
The latest 1.4.x firmwa
If available you almost always will wish to use NI2, especially if
you like to see who is calling you
On Jun 30, 2006, at 2:01 PM, Aaron Paxson wrote:
I would work that out with your vendor, as the settings must be the
same on both sides.
If national won't work for you, ask them if th
I assume you are using g711a/u?
Anything else will have issues with dtmf inband
On Jun 29, 2006, at 2:52 PM, Michael George wrote:
On Thu, Jun 29, 2006 at 10:42:05AM -0700, Shane wrote:
Ther's probably a simple answer to this but I've searched
around and haven't located anything as yet. Is
do I see what the console is saying?
Jerry Jones wrote:
Do you have more than one call per line enabled on the Poly? Is it
the phone or asterisk returning the busy? What does the console say?
On Jun 27, 2006, at 5:29 PM, Mike Staver wrote:
I have one extension setup for each Polycom 501 I have
configuration files.
Cullin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry
Jones
Sent: Wednesday, June 28, 2006 10:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk 1.2.7.1 with Polycom 501 on
Assuming it is a dedicated private line p2p T1
Assuming that 23 calls at one time is sufficient
Install a T1 card in each server, plug the T1 in and set one end ofr
pri net, the other for pri cpe.
zaptel.conf and zapata.conf are the files you are looking for. Just
define the 23 cha
Do you have more than one call per line enabled on the Poly? Is it
the phone or asterisk returning the busy? What does the console say?
On Jun 27, 2006, at 5:29 PM, Mike Staver wrote:
I have one extension setup for each Polycom 501 I have, and when I
try to call out on a conference call, I
Couple thoughts/observations
Yes you WILL get this working:)
since your .cfg is just a snippet and not the complete lines, kinda
hard to know exactly.
Hopefully you have the complete file and are just sending the
relevant pieces to the list.
Also, if you enable sip debug, do you even see
I am running an IP601 on my desk and it is only monitoring up to 8.
If I add more, it drops the oldest and adds the new one.
running 1.6.6.0036
On Jun 19, 2006, at 11:40 AM, Kevin P. Fleming wrote:
- Douglas Garstang <[EMAIL PROTECTED]> wrote:
Polycom released their SIP software vers
Go to one of the numerous voip distribor sites and browse their
selections, there are several
Yes you can plug into you wiring, just don't forget to disconnect
your local telco line - bad things would happen
On Jun 12, 2006, at 5:19 PM, John Klimek wrote:
Ahh, thanks! That's what I
add an ATA with an fxs port or 2
On Jun 12, 2006, at 2:39 PM, John Klimek wrote:
Ok, I've done some more research and I don't think I want an FXO
box...
What I'd like to do is use BroadVoice (with their BYOD plan) and then
run Asterisk on my WRT54G router. I'd also like to use my regular ho
Unless you pruchae an exteral zone page adapter which accepts FXS
coonections, use an ata with fxo connection
I have never had to 'add' loop current, although I have not used all
fxo adapters. I can verify Adit and other channel bank fxo
connections do not require any voltage, the just give
use an fxo interface and 600ohm input on amp
On Jun 11, 2006, at 9:53 AM, Thomas Kenyon wrote:
Doug Lytle wrote:
Thomas Kenyon wrote:
I need to be able to connect an old PA system to an asterisk box,
which
basically works as a couple of amplifiers taking an analogue phone
signal and playin
If you have a PRI to a telco, they probably only have a single trunk
group with 23 channels in it for your connection.
Any calls to any of your numbers may come to you on any channel.
Channels are not dedicated to individual numbers.
In other words the first call may come in on channel 1, t
One end must be cpe, the other net
also one should provide the timing, the other recover the timing
On May 31, 2006, at 8:43 PM, Lacy Moore - Aspendora wrote:
What cards?
On 5/31/06, Bruce Reeves <[EMAIL PROTECTED]> wrote:
Both servers are runing Asterisk 1.2.7 and the T-1 is a cross over
Create a contact entry with their extension and enable buddy watch on it
It will then show up on an unused line key
On May 27, 2006, at 3:26 PM, Faris Raouf wrote:
I've somehow managed to battle may way through hinting issues with
type=peer type=friend and various other oddities and now have
I would like to suggest using any managed switch and hard setting the
ports to 100/full
I have found that the auto negotiation algorithm is generally to
blame on many switches.
As an example, connecting a cisco router to a netgear/dlink/3com/etc
will geneerate errors on the cisco interfac
http://www.plantronics.com/north_america/en_US/products/cat1150057/
cat5420035/prod5460010
Just read a positive review of this one
On May 24, 2006, at 3:53 PM, Kyle Sexton wrote:
Slight derail, does anyone have a good bluetooth headset that will
work with a cell phone and a PC at the same ti
From my copy of the manual
MP-1xx SIP User’s Manual 36 Document #: LTRT-65404
You can use the ‘Reset’ button to restore the MP-1xx networking
parameters to their factory
default values (described in Table 4-1) and to reset the username
and password.
Note that the MP-1xx returns to the soft
A big disadvantage for a hosted provider is creating manager accounts
for end users. for a single company installation it may not be a big
deal, but SIP TAPI looks much cleaner to me as a service provider.
Of course I have yet to get a good configuration for it working
properly.
On May 2
Uh - If the OP is trying to transfer an existing call, then he should
be using transfer not forward. You may not access the forward
function on a polycom with an active call. Forward will send
subsequent calls to your specified destination, not existing calls.
On May 19, 2006, at 8:53 AM,
I believe the hint priority must be in the same context as the phones
extension number, in this [local]
On May 12, 2006, at 6:58 AM, richard Coco wrote:
Hi all,
i am desperating, trying to configure an OptiPoint410
with the hint priority.
Here what i have...
OptiPoint410std-> exten 2001
I would guess either the DSL itself is bad or perhaps the dsl Modem.
perhaps calling Bellsouth would be helpful? Does other Internet
traffic get interrupted also?
On May 8, 2006, at 1:42 PM, Hadar Pedhazur wrote:
I haven't seen anything this strange, and it's 100% reproducible.
I'm hoping
Any CLASS MWI should also work. I do believe there is support at
least on the Digium cards, and I think the Vega also. I am just
installing a Vega now and will try to verify, but it might be awhile
before I can.
On May 7, 2006, at 3:46 PM, Garth Summey wrote:
I have 20 or so users with an
Edit your config files to enable persistance
Will remain across multiple calls, but not reboots
On May 4, 2006, at 2:51 PM, Jim Freeze wrote:
We are using the polycom 501 phones, and are having some challenges
with the volume setting. When a phone call comes in, the user ups the
volume for th
Attribute Values Default Interpretation call.rejectBusyOnDnd 0, 1 1 If set to 1, reject all incoming calls with the reason “busy” if do-not-disturb is enabled. Have not used, but looks like it may ignore the key if this is 0Let us know...On May 4, 2006, at 2:22 AM, <[EMAIL PROTECTED]> <[EMAIL PROTE
Perhaps a setgroup/checkgroup before your dial command?
On May 3, 2006, at 1:24 AM, Arne Morten Johansen wrote:
Yeah I do use ring groups at the moment. But the problem is that I
can’t control “the flow”.
Let’s take your example.
dial(SIP/dev1&SIP/dev2&SIP/dev3)
If I dial these 3 numb
What does you dial command look like?
On Apr 30, 2006, at 12:05 PM, Jim Lynch wrote:
I've installed [EMAIL PROTECTED] and gotten inbound calls going to an
extension, extension to extension calling works but I'm still
missing a few pieces. The most annoying one is that apparently
asterisk
You do not say how you have the two connected/
Are you connecting the * to stations via fxo or to lines via fxs on
the legacy?
On Apr 30, 2006, at 11:22 AM, Remco Barende wrote:
Hi list!
I managed to come reasonably far (farther than I thought I would)
but have two problems.
I still n
Yellow=ground - not used
Green = tip
Red = ring
connect green/red to rj pins 4/5
You could pick up a quarter mod line cord (mod to spade) and replace
the cord, or use a screw terminal block to connect to line.
Enjoy
On Apr 25, 2006, at 9:19 AM, Sean Cook wrote:
Ok... I am not a telephone
Not sure about your specific set, but some just place a load on the
line and disconnect the handset when you use the hold button. You
pick the handset up or press the button again it reconnects the
handset and removes the load. It does not actually send a flash to
place the call on hold, th
Pleaase read the archives or the wiki - you will shortly find you
need a wait in your dialplan
On Apr 19, 2006, at 8:10 AM, Jonathan k. Creasy wrote:
Below is a snipped debug on our PRI. We are getting number only for
the CallerID but the telco says they are sending us Name and
Number. We
increase your silence setting
On Apr 18, 2006, at 8:31 AM, Mike Garey wrote:
When someone hangs up before getting to the leave voicemail prompt,
asterisk still attempts to record a voicemail message, so I end up
getting a bunch of empty voicemails.. Is there any way to change this
behaviour, s
Are you seeing link on either end? Not sure if the GS shows or not.
On the Mac, open a terminal window and type ifconfig to see if the
port is active - ie has link
it should have a line similar to this if so
media: autoselect (100baseTX ) status: active
If this is correct then you h
Has anyone successfully implemented SIPTAPI with asterisk? It would
appear to require a true proxy. I assume it will need a seperate user
account to register and place calls, but I have been unable to get it
to attempt to register with asterisk.
If you have it working, example configuration
for voip devices.
On Apr 14, 2006, at 2:16 PM, John Novack wrote:
Jerry Jones wrote:
Yes it should all behave the way we are used to. However SIP IS
different. The exact behavior will be dependant upon the
individual hard phone.
Isn't that true only if it has a preprogr
Yes it should all behave the way we are used to. However SIP IS
different. The exact behavior will be dependant upon the individual
hard phone.
This of course is if using SIP which we do not know yet...
On Apr 14, 2006, at 1:43 PM, John Novack wrote:
Michael Collins wrote:
A few months
Use VLANs on Ploys all the time, but manually set also.
Of course switches and routers all need to be setup for the proper
vlan config also.
On Apr 12, 2006, at 12:18 PM, BJ Weschke wrote:
On 4/12/06, Rob Terhaar <[EMAIL PROTECTED]> wrote:
So has anyone had any experience working with the
I CAN VERIFY via aa dozen PRI from XO that yes indeed provide
incoming callerID on PRI. It arrives shortly after the setup message.
Hence the wait(1) required to display it.
Now if you are referring to sending caller name across PSTN - that
does NOT work since the terminating switch will do a
On Apr 10, 2006, at 12:35 PM, Steven wrote:
Thanks for the info.
I went to add the Wait(2), but am unsure where to do it.
My context is "from-pstn".
My [from-pstn] is:
[from-pstn]
exten => s,1,NoOp(${TIMESTAMP} PRI call in) ;I tried adding
this to see if "s" is used, but lothing was l
I had emaile poly support a couple months ago and they replied not yet.
On Apr 2, 2006, at 7:14 AM, Noah Miller wrote:
Is there any way to get a polycom 601 to do overlap dialing?
I can't find anything on the subject, which confirms my initial hunch:
I really doubt it. You could probably w
As a rule you are wasting your time trying to send calling name to
your telco. Unless your carrier is also the terminating providor for
everyone you call it will accomplish nothing.
Caller ID that the called party receives is by way of a lookup by the
terminating providor in a national data
Show channels?
On Mar 31, 2006, at 2:09 AM, Ronald Wiplinger wrote:
In the past I used SetGroup and CheckGroup to figure out if my
allowed providers lines are all used or not.
Since most of my provider have given me a single line anyway, I
wonder if there is a way to check if this (provider
Polycom with sidecars work? I like the idea of a
dedicated
FOP display but not sure why you would need it if you have a
Polycom with
sidecars.
-Original Message-
From: Jerry Jones [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 28, 2006 7:28 AM
To: Asterisk Users Mailing List - Non
We installed a snom with 3 sidecars. Kinda worked, but had so many
quirks they had us replace with a Polycom. All their other phones
were of the poly variety. We installed a dedicated lcd running FOP
for display. Receptionist was much happier.
One of the key problems was she like to set the
You need to create [new_context] in extensions.conf
then add the context=new_context to sip.conf so calls from from the
sip devices know which context to use. this can be in the general
section and apply to all sip devices, or add per device and each can
have its own context
just adding t
Yes they are analog
Yes you would use fxo to connect to asterisk.
No you do not have a T1.
On Mar 23, 2006, at 2:21 AM, mike webb wrote:
at my work we have a meridian 1. it has 6 lines from the outside
world coming in to it. if someone calls our number 870-238-2111
they will always reach o
Could always create named extensions and dial by name
On Mar 15, 2006, at 11:17 AM, C F wrote:
IIRC, it's something that is supported in the latest versions of SIP,
which Asterisk doesn't support yet.
On 3/15/06, Noah Miller <[EMAIL PROTECTED]> wrote:
Hi Giorgio -
we have an asterisk 1.2.1
Not used to Brazils signalling standards, but in the US all end user
signalling is via DTMF or dial pulse(rotary), so we know the Digium
cards support that.
Here FSK is only used for the CLASS family of features, ie callerID,
message waiting, etc.
Good Luck
On Mar 9, 2006, at 11:27 AM,
Best - T.38
Second - and way down the list - make all G711
On Mar 6, 2006, at 5:19 AM, [EMAIL PROTECTED] wrote:
Hi,
Thanks for your replies.
I am going to have many DID's and I have to provide each of them
this feature.
So I cannot solve this problem with a dedicated DID having G711. Is
yes
On Mar 1, 2006, at 6:02 AM, Arne Morten Johansen wrote:
Hi there.
Is it possible to have different sip users have the same CallerId
number
in sip.conf.
I need this because we got multiple companies on this Asterisk box.
Company A's internal numbers:
CID: User:
1000 - User 1
2000 -
Losing an audio packet here or there wouldn't normally be so bad
for fax. Normally I would expect the fax protocol, especially ECM
protocol, to be able to recover from it. However, Asterisk seems
to not work in an ideal fashion for this purpose. Whenever
Asterisk encounters a lost audi
On Feb 10, 2006, at 12:15 PM, Andrew Kohlsmith wrote:
I am starting to get the hang of this, I think. These are more
implementation questions; "is this a proper/good way of using/doing
this"
kind of questions.
The IP501 has three line appearances. I have learned that shared line
appearanc
On Feb 6, 2006, at 5:04 AM, <[EMAIL PROTECTED]>
<[EMAIL PROTECTED]> wrote:
Im curious. Does anyone have experienced echo-problems that later
where solved by buying a hardware-echo canceller such as the
Wildcard TE411P?
Yes. I turned off all echo can on the wildcards and bought external.
Also works fine for me, even with the default user/pw combo
On Jan 31, 2006, at 12:25 PM, Mojo with Horan & Company, LLC wrote:
vsftpd has always worked fine for me, but I did change the password
the polycom was expecting to send from the default one with capital
letters.
Moj
Ken D'Ambro
On Jan 31, 2006, at 10:03 AM, Michaël Gaudette wrote:
Hi,
I`ve been trying to figure out voicemail, but there is something
that is obviously escaping me. Using * 1.2.3, standard built with
asterisk-addons.
I have two voicemails, one is 702 and one is 705. Both in
different contexts, b
Just curious,
I have had issues with the number of monitored phones and getting out
of sync with reloads. Have you had similar issues? Which version of *?
On Jan 30, 2006, at 7:04 PM, Saul Diaz wrote:
Damon Estep wrote:
Anyone successfully set up one of the polycom soundpoint ip sidecars
We started out useing SPA2k but they were prone to stop talking to
the ethernet. OK after reboot for awhile but cannot keep going to
customer sites and rebooting things.
switched to spa2001 and somewhat better but they keept losing
registrations and then could not talk to them remotely. Again
On Jan 27, 2006, at 11:33 AM, Mimmus wrote:
Hi,
I'm trying to configure some Quality Of Service among an Asterisk
server
with RedHat3 and some IP phones on my LAN.
I read about 802.1p (level 2) QoS, using 3 bits of VLAN tag.
Two questions:
- do I need to use tagged links (trunks) end-to-end
I have many Poly installations and have not had this issue, EXCEPT -
the one time we permitted a customer to run their computer data
through the telephones.
We also have noticed a poor server config can cause this in testing.
Noticed when I had one person building * servers using Debian. Had
Where do you have to set the public IP? We use dhcp Poly behind
firewalls daily. Just set nat=yes in sip.conf
On Jan 23, 2006, at 3:26 PM, Bill Gibbs wrote:
I know the Polycoms work with NAT, but you have to specify the
public IP.
Is there anyway for it to discover the external IP autom
You need to have an extension defined for each number comig in. They
may be 4 digit if that is how your circuit is ordered. You then need
to create a dialplan to tell the call what to do.
Yes you could create a group in zapata to use for outdial
The pri will automatically allow up to 23 call
Don't know the max, but I have many more than 11 with never a problem.
On Jan 16, 2006, at 8:57 AM, Douglas Garstang wrote:
What's the maximum number of #include statments I can have in
extensions.conf?
I'm getting an error at the 11th one. I tried breaking twelve
#include's into 2 differ
set dtmf mode to inband and use g711
On Jan 11, 2006, at 12:21 PM, Andrew Berman wrote:
I am having an issue using a Polycom 501 and VoIP for outgoing
calls where if I call say my credit card company and try to follow
their PBX menu, the key presses never register with their PBX.
It's as
I have never had any echo on inside calls (acoustic) and have placed
many Poly in bare offices.
If on outside calls then you may need an acoustic echo can on your
line. I added a year ago and all complaints disappeared.
On Jan 11, 2006, at 12:16 PM, Jorge Mendoza wrote:
Carlos Chavez wrot
On Jan 11, 2006, at 11:12 AM, C F wrote:
You could use a 2 span T1 card from Digium and plug one span into a
channel bank, and have FXS ports on the CB for the fax machines. With
the latest firmwares from Digium these streams are bridged internaly
on the card, and don't even come on to the PCI
Any type of circuit available as an analog line is also available
over a T1. It just minimizes the amount of copper required to deliver
service.
You must look at you original order from your telephone company to
determine the type of circuits they are delivering. They may be POTS
1FB in w
No but shell scripts are pretty easy and will cleanup your file for you.
On Jan 5, 2006, at 1:07 PM, Douglas Garstang wrote:
Not everyone is a C programmer extraordinairre.
-Original Message-
From: Alyed Tzompa [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 05, 2006 11:59 AM
To: Doug
sounds like a digitmap issue.
We looked at using # originally, but interferred with too many IVR
type applications from people.
On Jan 5, 2006, at 12:00 PM, Mojo with Horan & Company, LLC wrote:
Because the Polycom softkey menus were so cumbersome, we chose to
use Asterisk's attended and
Of course most carriers these days charge extra per call path. So how
many simultaneous calls do you really need up?
On Dec 30, 2005, at 1:55 PM, trixter aka Bret McDanel wrote:
On Fri, 2005-12-30 at 21:38 +0200, Bogdan Moldovan wrote:
Depending on the forward type. You could put conditional o
On Dec 30, 2005, at 1:48 PM, trixter aka Bret McDanel wrote:
On Fri, 2005-12-30 at 14:06 -0500, Andrew Kohlsmith wrote:
On Friday 30 December 2005 13:23, trixter aka Bret McDanel wrote:
CLECs and ILECs largely are required to let you port your number
(there
are some potential issues that cna
PROTECTED]
Sent: Wednesday, December 28, 2005 3:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom check-sync
On 12/28/05, Jerry Jones <[EMAIL PROTECTED]> wrote:
Looking to implement a simple cli to reboot our Polycoms. The manual
says it w
Looking to implement a simple cli to reboot our Polycoms. The manual
says it will on response to a Notify with an event of check-sync.
Has anyone else used this? Any examples of a minimum packet to send?
tia
___
--Bandwidth and Colocation provided
Its all in the priority the phones assign responding to messages.
Sipura are also fast compared to Polycom.
this is not an accurate measure of network latency.
On Dec 27, 2005, at 2:11 PM, Dean Collins wrote:
I have a polycom 501, for some reason asterisk always shows the
round trip time t
Just to clarify, the frame slips are on the T1 circuit which is
delivering your PRI service.
If you are correctly adjusting your timing leaving to match the
recevieved timing from the CO there will be no slips. I am unaware of
any mechanism within an * server to report this. Of course T-BER
On Dec 21, 2005, at 7:53 AM, Rhonda Herron wrote:
Hi, Thanks for the reply...
The clicks are on every call and every few minutes or so, I guess
you could call it regular intermittance? :)
The only option for DTMF on my IAX phones are inband and outband-
neither of which will respond to a
Yes call progress tones (busy) can be generated by ATA devices. Not
sure if the 486 has a digitmap or not, may wish to checkit out if so.
I also believe it has an internal status webpage and syslog info you
could look at for clues.
On Dec 19, 2005, at 5:34 PM, Craig Bruenderman wrote:
I'v
This is a very big headache for me.
Alarms today normally use a protocol called contactID for
communications. These are very short dtmf tones and most devices have
a very hard time transmitting them. also any jitter etc causes them
to be unreadable. If anyone has a reliable method for trans
So how many are you trying to display? A hundred is pretty easily done.
On Dec 16, 2005, at 3:03 PM, Terry H. Gilsenan wrote:
Kerry Garrison wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Terry H.
Gilsenan
Sent: Friday, December 16, 2005 12
hehe
I just installed * with a T1 span to and Adit600 with 2fxs and 1fxo
The 8 fxo ports were for zone pageing
works great
should work with any fxo device and an existing page system
On Dec 9, 2005, at 11:34 AM, C F wrote:
Overhead paging is totally possible, there are several articles
avai
If the digium card is good then he has the proper cable config,
although his send may not be getting to the nortel. This is layer one
which must work before layer two, ie d channel.
What does the nortel say regarding the T1?
If this is good then your issue is with configuration not cableing.
Just in the process of figuring this out myself. i do have it working
on an IP601 with a sidecar. Here are my notes.
On the polycom
Create a contact directory entry for the extension you wish to
monitor. Yes the contact must match the exten= statement in your
dialplan. Note: It must reside
greetings all
I have been having intermittant issues with various flavors of ATA
not keeping their registrations over extended periods of time. This
has happened with both Sipura and Grnadstream ATA.
They all install and operate correctly for at least a week or so.
Then I notice screenful
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