I've search the world over but I haven't figured out a way to
have
valid/invalid options for entry when using the Read command...
I need to set a variable, but only want to allow certain values to be
valid options for that variable...
I hope I understand your question.
You can
There's actually a document included with the source code which will
take you through setting up an agent callback system. You can find it
in 'doc/queues-with-callback-members.txt'.
The 'AgentCallBackLogin' application has some issues, and since you
can
do the same thing with your
I am really grateful to all the experts on the mailing list who gave me
some very good advice on this problem which I experienced in China. I
think we have fixed the problem and the card is no longer reporting any
problems. We are able to dial out successfully and we will continue to
test.
Here
Check this one out... http://www.voip-info.org/wiki/view/Asterisk+non-root
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shaun Wingrin
Sent: Sunday, 17 August 2008 6:51 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
I am trying to look for a software (open source or proprietory) that could do
reporting on both queue and CDR in Asterisk 1.4.*
Could someone give me some suggestions?
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if after you tried both straight through crossover cables and
it still give you RED alarm. just tell them you can't get any
clocking signal. they'll probably send someone on site and test
the line.
Yes, I tried all sorts of cables and ended up getting the local contact
to complain to NETCOM.
Sounds like you're making progress. I would try the above span
definition without the crc4. That might do the trick.
Thanks Brad.
I already tried it without crc4 but it makes no difference.
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Ensure that in file indications.conf you have
[general]
country=cn ; not usa ! or if you are in Australia shortcut for Australia
Uros, that was a good reminder. However, I don't think it is related to this
problem.
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You don't need to install it. Just run kernel/xpp/utils/genzaptelconf
directly from the source directory.
Thanks Tzafrir.
My local contact is away today and so I could not get him to plug the
line to port 4. So, it is still in port 1.
Here is the output after running genzaptelconf.
#
I am trying to build a simple queue with several agents using
AgentCallBackLogin.
From what I read on the Internet and tried briefly, it seems to suggest that I
should be coding my own queue system for AgentCallBackLogin using AEL2 instead
of using the AgentCallBackLogin command because it is
I think it can't hurt to try a different release. Let me know how it
goes.
Thanks Igor.
I just upgraded zaptel to 1.4.11.
However, I am still seeing red in the alarm in zttool and the LED on
port 1 also shows red.
---
cat
Wow - that's nasty.
Almost like a broken card or MB. Ouch.
Should you call the supplier of the card and ask them about warranty?
PaulH
Thanks Paul.
The TE412P card is fine and the zaptel error in dmesg is fixed by
1.4.11.
However, the red alarms are still there.
Hi, just for (all of) you to know this is a known bug of zaptel
1.4.11, the firmware upload procedure is taking some time, operating
like a freeze during the process, so this message appears.
But this isn't a real problem, as it doesn't have any consequences
appart from the message.
The test for that is simple:
head -n 1 /proc/zaptel/*
Let's look at all four spans. Not just the first one.
Thanks Tzafrir.
# head -n 1 /proc/zaptel/*
== /proc/zaptel/1 ==
Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/ RED
== /proc/zaptel/2 ==
Span 2: TE4/0/2 T4XXP (PCI) Card
I tried the suggestion on
http://www.voip-info.org/wiki/view/Asterisk+MFC+R2
http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 but the red alarm is
still on.
An output of 'head -n 1 /proc/zaptel/*' after the fact might help us help
you. Did you re-run ztcfg after editing zaptel.conf ?
An output of 'head -n 1 /proc/zaptel/*' after the fact might help us
help
you. Did you re-run ztcfg after editing zaptel.conf ?
On a sidebar, let me suggest head -1q; it's neater.
Thanks Jay for your neater suggestion!
# head -n1 /proc/zaptel/*
== /proc/zaptel/1 ==
Span 1: TE4/0/1 T4XXP
Thanks Steve for your suggestions.
In China you will generally get either MFC/R2 or EuroISDN. MFC/R2 is
much more common.
This is exactly my current problem.
NETCOM in Shanghai just told my local contact it is an E1 and that's it.
I have no idea whether it is MFC/R2 or EuroISDN and so there
If you're interested, our working circuit in Chengdu with a China
Telecom PRI is configured as:
/etc/zaptel.conf:
span=1,0,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
loadzone = cn
defaultzone=cn
/etc/asterisk/zapata.conf (just including the pertinent lines):
switchtype=euroisdn
i've installed several Asterisk systems in Shanghai Beijing.
Thanks Edwin.
The remote site is in Shanghai and NETCOM is the telco.
Do you know if their E1 line is MFC/R2 or EuroISDN?
red alarm usually means there's no clocking signal.
check all your cables (crossover vs straight through)
Yes, to try port 4 again.
Thanks Tzafrir.
Why do I have to plug it into port 4?
Backup the existing zaptel.conf and run genzaptelconf (no need to
unload
/ reload any modules). What is the output of 'head -n 1
/proc/zaptel/*'
after that?
I could not find genzaptelconf probably because I
You don't need to install it. Just run kernel/xpp/utils/genzaptelconf
directly from the source directory.
Yes mate - I was just 1 sec away from reinstalling zaptel.
Why do I have to plug it into port 4?
Do I have to plug the line into port 4 instead of port 1?
This time, I am trying to remotely install Asterisk in China.
I was told that an E1 line has been installed and so I plug it into port
1 of a TE412P.
On the box, first of all, I just installed Zaptel 1.4.10.1.
# service zaptel restart
Unloading zaptel hardware drivers:ERROR: Module zaptel is in
My best guess from looking at that is that its a driver bug. The last
thing that happens before the lockup seems to be an ioctl call to the
device.
Hope it helps,
Igor H.
Thanks Igor.
Does it mean that I should install a later release of zaptel?
Are you sure that they're plugged into port 1 and not port 4? It is a
rather
common mistake to believe that the port numbers start at the bottom of
the card and not at the top.
Thanks Tilghman.
I checked with the guys in the remote office and he is certain that he
has plugged the E1 line
exten=7000,n,GotoIfTime(9:00-17:00,mon-fri,*,*?rcl_off_opn)
7000 is the extension of main menu
Where do I put the reference to open hours menu in the statement
above.
exten=7000,n,GotoIfTime(9:00-17:00,mon-fri,*,*?rcl_off_opn)
[...code for office close...]
exten = 7000,n(rcl_off_opn),
Do you do contract work?
Thanks for making my day :-)
I am sure there are lots of much more experienced Asterisk people out
there who will respond to your email.
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What do I have to code in the main menu to do the following.
If between the hours of 9am - 5pm go to open hours
All other hours go to after hours
You can do something like:
exten = main switch
no,n,GotoIfTime(9:00-17:00,mon-fri,*,*?rcl_off_opn)
___
You should probably avoid giving incoming access to outgoing..
Thanks Paul.
[incoming]
...
include = internal
include = outgoing
The thing is if I don't have this include = outgoing in [incoming], I
will not be able to dial out at all.
Any thoughts?
With an ISDN10/20/30/etc, I would just put all the lines into an
'incoming' context - and make sure that incoming context doesn't have
any includes (unless you really need them...)
Can someone please have a look at below to see if this would be the best
and secure practice of using context in
i'm using 64-bit Ubuntu Server Edition 8.04
I just use GMT+0, but i'm on Singapore whcih should be at GMT+8, but
if
i use GMT+8 the system does not give the correct time.
You should actually be using Asia/Singapure rather than guess.
i'm not using ntp, coz when i do i also don't get
http://bugs.digium.com/12426
There's also a fix there that I don't fully understand (and I'm not
sure that that fix does not cause damage, so don't just apply it).
I am installing Zaptel 1.4.10.1 and I encountered the same problem.
As Jerry said, compile problem goes away if I uncheck xpp in
For posterity, always make sure that some junior admin hasn't used a home
router/gateway as an emergency hub stuffed underneath somebody's desk. Those
pesky extra DHCP servers don't play nice with others.
Just a suggestion, will defining a special VLAN just for the Polycom phone be
able to
Just want to know if anyone has used instant messaging using Polycom and
Asterisk.
From Google, I did not really see IM being mentioned at all. It appears
no one is interested to implement it in Asterisk. Or I guess people
would rather use Jabber or other IM messengers.
Tilghman, you are spot on!
As it turns out, this parameter is not documented at all in
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+voicemail.c
onf or anywhere on Internet.
Also, in that voicemail wiki, there seems to be a lot of parameters
that
wasn't explained at all.
I
I was thinking about dividing my users into different groups (contexts)
in voicemail.conf so that I could use voicemail show users for
[context] to manage them easier.
However, I found out that I should not do that because if I am using
[macro-stdexten] in extensions.conf, I will need to
As a result, I just go back to put all users in [default] in
voicemail.conf.
Am I missing anything?
What do those contexts mean in your setup (beside being arbitrary
groups)?
I just want to group the mailboxes by say department rather than putting
them all under [default].
So, I could
I haven't been following the conversation, but why don't you use
searchcontexts=yes in voicemail.conf? As long as you don't specify a
particular context when calling Voicemail, it will look through all
contexts
until it finds a matching mailbox.
Tilghman, you are spot on!
As it turns out,
Anyone have recommendations for wireless headsets that work well with
Polycom phones and Asterisk?
I am waiting for delivery one of the following types:
Plantronics CS351 Monaural SupraPlus Wireless Professional Headset
System Voice Tube
You should probably clean it up and put it up on the wiki. I don't
think
anyone has put up a step-by-step like you did before.
There might be much easier additions/modifications done to it, and it
will
be available to everybody.
Done. No problem - glad to be of service to the open-source
Lee, John (Sydney) would like to recall the message, [asterisk-users] Newbie
Asterisk: Install Asterisk as non-root.
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You should probably clean it up and put it up on the wiki. I don't
think
anyone has put up a step-by-step like you did before.
There might be much easier additions/modifications done to it, and it
will
be available to everybody.
Done. No problem - glad to be of service to the open-source
First of all, thanks Philipp, Alan, Tzafrir and James for your valuable
comments. I have listed below the exact list of commands to run for
reinstalling asterisk 1.4.* as non-root on a Redhat / Fedora distro.
Hope others can benefit.
I have the following comments/questions though:
1) #What
I was following the instruction on
http://www.voip-info.org/wiki-Asterisk+non-root to re-install my
Asterisk as non-root when I had the following questions/issues:
1) Use your system's preferred method of adding a new user. Examples:
Red Hat: adduser -c Asterisk PBX -d /var/lib/asterisk -u
I was able to disable the DND button (no 9) on IP60x by putting the
following line in sip.cfg.
keys key.scrolling.timeout=1
key.IP_600.9.function.prim=Null/
However, I could not do so for the Services Button (no 29) on IP600 (or
Applications button on IP01)
keys
In The future of Telephony, it says ... We should also note for
security's sake you should always make sure that your [incoming] context
never allows outbound dialing. (If by chance it did, people could dial
into your system and make outbound toll calls that would be charged to
you!)
The book
The only things I set in relation to echo cancellation is in zapata.conf
where I put echocancel=yes
Ouch...any idea what echo cancellation your system is using?
PaulH
On Thu, 2008-05-08 at 14:55 +1000, Lee, John (Sydney) wrote:
the relaxdmtf (or similar) option in zaptel can make
I have this simple queue for the reception set up such that the console
queue has only one agent.
I checked the number in the queue and if there is someone there, I play
back a busy please be patient message and then join the call to the
queue.
If there is no one in the queue, the caller will go
Queue(console,r)
would do what you want, but so you would need to have two entry points
to
queue.
Thanks Atis. Your suggestion did magic!
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To
dmesg | grep -i zap
Should give you a version, and an echo cancellation technology.
Thanks Paul.
# dmesg | grep -i zap
Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.4.6
Zaptel Echo Canceller: MG2
Zaptel Transcoder support loaded
Besides the Background() app mentioned, you might like the WaitExten()
app
Thanks guys for your response.
I have had much success with Read() as below so that whenever I press a
key before the sound file finishes playing, it will read the digit and
move to the next line.
exten =
the relaxdmtf (or similar) option in zaptel can make this work a bit
better...but it's a try at your own risk option!
PaulH
Thanks Paul.
I have further findings into the problem.
While the message is being played, if I press a key during the pause
or break between words, then the key will
If you have access to the console you can do many things.
For instance, you can originate test calls.
Tzafrir, thanks for your response and sorry for not being specific.
You raised a very good point about accessing the console.
For example, I made a major change to some of the config files
I will be installing Asterisk in a few offices which I don't have any
colleagues over there to help me.
Let's suppose I installed Asterisk in such a site. I tested it to my
satisfaction and I went back to my home office.
One day, a customer called me to say that he had a problem calling out
or
Check the number of calls waiting in the queue, then play the message
if
more than 0
example code (written in the TBird IDE)
Exten = 100,1,Answer()
Exten = 100,n,Set(NumWaiting=${QUEUE_WAITING_COUNT(${QUEUENAME})})
Exten = 100,n,GotoIf($[${NumWaiting} = 0]?JoinQueue)
Exten =
I am trying to find out if Polycom (I am using IP601) can display the
speed dial list using last name first instead of first name first.
Currently, the speed dial list displays first name first.
Thanks.
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Just want to know if anyone has used instant messaging using Polycom and
Asterisk.
From Google, I did not really see IM being mentioned at all. It appears
no one is interested to implement it in Asterisk. Or I guess people
would rather use Jabber or other IM messengers.
DND does not do anything for me BLF-wise either (shame). Simply
picking up
the handset won't do, at that point the phone is giving you a dialtone
but
nothing is sent to the server. You actually have dial out. Try
actually
calling somebody, the state should change to InUse.
Thanks Mike and
I am working on Polycom IP601 console with expansion module.
I want to put on the BLF (busy lamp field) feature on all the
contact/speed dial names I put on the console but I could not get it to
work.
*CLI core show version
Asterisk 1.4.13 built by root @ hostname on a i686 running Linux on
If I understand correctly (and that's my experience) the BLF will only
light
up when the phone is ringing/on a call. Asterisk doesn't support all
those
fancy status that you can select from the phone.
Mike, thanks for your response.
I think my test is worse than that. I pressed DND on one
I am exploring the contacts directory in Polycom and I am wondering if a
big -directory.xml on the boot server will eat up the memory
and crash the Polycom phone once downloaded onto the phone.
The asterisk directory extension is good but because users cannot see
the names I thought to
When I downloaded the sip and bootrom from Polycom website, I noticed a
file called SoundPointIPWelcome.wav. However, I have no idea where and
when it was used. I played the wav file but I have never heard the
phone using this wav file before. Does anyone know what it is used for?
It's played at the completion of the boot process. It's always been
very quiet on the models I've worked with.
Thanks Erik. I can probably replace it with my beloved Mozart Symphony
no 40 :-)
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Any suggestion for a headset (cord and cordless) for IP601?
Any good (and economical) ones from Polycom or Platronics?
Thanks.
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I had this problem before...the following appeared in a previous post...
For some reasons, the * and 1 must be pressed pretty quickly together on
the Polycom phone before it can be transmitted successfully to Asterisk.
Does anyone know if that can be tuned?
Sure... go to features.conf, and change
All Polycom phones use the same firmware and bootroms - one reason why
the sip.ld is so damn large for them.
Thanks Rob.
Alleluia! Rob, I will take your word for it - it solves all my worries
in deploying different models to the same environment like IP5XX and
IP6XX.
I have a question about DHCP and boot server supporting more than 1
model of Polycom phones.
According to Polycom standards, Polycom phone boots up to get a DHCP
address and at the same time getting a boot server string (with username
and password) to logon to boot server to download SIP, bootROM
I assume span 2 is set ti T1...
Thanks James. I will check.
Also you should be using a crossover if your going from the card
direct
to the channel bank. Remember an RJ48 crossover and an RH45 crossover
are not the same.. It you are using an RJ48 crossover and your span 2
is
T1 then try
Have you tried setting the card as being T1 instead of E1 for the port
connected to the channel bank?
Thanks Paul.
I was thinking about the same thing as I was leaving work today.
I will try to set the jumper just on port 2 and let you know.
___
--
I was thinking about the same thing as I was leaving work today.
I will try to set the jumper just on port 2 and let you know.
Yes, that fixed the problem.
Thanks James and Paul.
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Good to here,
I know the time off set US - AU is terrible when you need support.
I have continued to configure the analogue phone by just adding new
extensions (just like any VOIP phone) to extensions.conf as follows:
exten = 5162,1,SetMusicOnHold(cpwr)
exten = 5162,n,Dial(Zap/32,20)
exten =
Any luck with the channel bank?
Thanks for the reminder Paul but so far no luck.
I have been getting:
1) *** Initialising: Trying to frame D4 / ESF on the channel bank
2) Red flashing light on port 2 of the TE412P card
I have checked a few things here and there but I think I must have
missed
I was trying to find out how I could put in a greeting when a caller
***first*** joins the queue.
I searched high and low but could only find (in queues.conf):
. announce, which is announcement to the agent
. announce-frequency which is announcement of queue position
.
I would think you'll need to do a Playback() of this message before
the
caller enters the queue, as I'm not aware of such an option provided
by
app_queue.
Exten=100,1,Answer()
Exten=100,n,Playback(greetings-earthling)
Exten=100,n,Queue(xyzqueue)
Exten=100,n,Hangup
Thanks Mark for your
What kind of information are you looking for? configuration or? If you
look in our manuals our cards and the Digium cards configure the same
in zaptel and zapata.
Hi James, I have purchased a CB24-FXS-UNIV and am using a TE412P card
with a Asterisk 1.4 box.
One port of the card is connected to
I am planning to roll out Asterisk to some offices and I have been
thinking about how to disaster proof the box.
For the production box, of course, it will have RAID 1 disk drives and
2GB memory at least.
a) If the office burns down, there is nothing much I could do.
b) If it is software error,
I am working on a menu to accept input from a caller like as follows:
Exten = 100,1,Answer()
Exten = 100,n,Playback(LONG-MESSAGE)
Exten = 100,n,Read(OPTION,,2)
...
When I tested it, I noticed if I start pressing a key before the
Playback() is finished, the input is not buffered (simply ignored)
I am trying to build a simple queue for the receptionist phone.
In other words, there is only 1 agent and that is the receptionist
phone.
I just defined a few lines in queues.conf
[console]
strategy = ringall
member = SIP/4000 ;4000 is the console extension
In extensions.conf, it is:
I might of got my wires crossed here, but I'm looking for a way to
disable
musiconhold for individual users.
Good question Adrian.
I never thought about that but I googled a bit and here seems to be the
answer:
http://lists.digium.com/pipermail/asterisk-users/2007-August/193721.html
So you either need to go a Goto(context,4000,1) or to drop it to the
queue
with Queue(console) etc.
I have chosen to use Goto(context,4000,1) from a programmer's
perspective although queue(console) works just as good.
Thanks guys.
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I am using Polycom IP600 phone. If I call a phone which has DND (do not
disturb) enabled, the message to the caller will be The person on
extension ... is on the phone, please leave a message
Is there a way to pick the person ... not available message instead?
I am writing an extension to accept speed dial nos.
However, I forgot that these speed dials are the same for all offices
and thus would ideally be shared by offices which will host their own
Asterisk box.
I read from a few postings that this database cannot be replicated to
other Asterisk box.
I
Does 'show features' display the correct information?
PaulH
Thanks Paul
*CLI show features
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# #
Attended Transfer
One Touch Monitor
On our system i got:
Zap/1-1 answered SIP/106-091a2750
-- User hit '*1' to record call. filename: wav|
auto-1205385048-106-0434225491|m
Our dialplan looks like:
_0X' = 1. Dial(zap/g1/${EXTEN}||Ww)
(from show dialplan)
PaulH
Thanks Paul.
I think the problem is *1 is
I think the problem is *1 is being ignored or cannot be transmitted
successfully to Asterisk.
Finally I resolved the problem.
For some reasons, the * and 1 must be pressed pretty quickly
together on the Polycom phone before it can be transmitted successfully
to Asterisk.
I think I cannot deny
I think if you install sox you will get soxmix.
Thanks Paul
I did yum install sox and Asterisk will automatically combine .in.wav
and .out.wav together into .wav
That is excellent!
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Special dialplans for reception are entirely up to you. The only
reason
reception phones have different dialplans to normal extensions is that
often people want the receptionist's phone to behave a little
differently.
Thanks Rob.
I talked to the receptionist this afternoon. She said it
You could organise a system where people set their DND on the Asterisk
server, then use devstate to generate a flashing lightbut it would
be a bit of fiddling around, and if people used their DND button on
the
phone it would not be displayed
We looked at this once, and decided
I thought it was quite easy to implement but I cannot get one-touch
recording to work. Here are the changes what I did:
I restarted Asterisk after the change (because reload does not work for
changes in features.conf).
I press *1 on the Polycom IP600 phone to record a conversation but no
new wav
I thought it was quite easy to implement but I cannot get one-touch
recording to work. Here are the changes what I did:
I restarted Asterisk after the change (because reload does not work for
changes in features.conf).
I press *1 on the Polycom IP600 phone to record a conversation but no
new wav
I was reading a Polycom brochure and it appears that there is really no
special receptionist console and the console is basically a IP601. Is
this correct?
The only difference is to purchase an expansion module in order to have
more shortcut keys for the girls.
So, apart from the hardware, as
Yes, use your first solution, but precede it with a call to the Read()
application for the user to enter their conference number. This will
put
it into a channel variable, e.g. ${CONF}, which you can then put in
place
of the hard coded number.
Thanks Tony for your advice.
Below is a working
I have been told to use Rhino Channel Bank but I am yet to set it up and
I appreciate if someone can show me some doco of using Rhino on an E1/T1
with TE410.
Thanks.
I've been asked to provide a system for 200 extensions, most of which
will
be existing analogue POTS handsets, not IP handsets.
We had a similar issue where the connector was not pushed in hard
enough.
I know that sounds like a joke, but it isn't!
PaulH
Thanks Paul - it also happened to my phone!
Thanks so much.
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I was successful to control the max users (10) if I hardcode the
conference room number (in this case 101) as follows:
exten = 8600,1,Playback(conf-thereare)
exten = 8600,2,MeetMeCount(101)
exten = 8600,3,Playback(conf-peopleinconf)
exten = 8600,4,MeetMeCount(101,CONFCOUNT)
exten =
I just want to confirm couple of things.
a) If I change an entry in sip.cfg (voice.volumne.persist for example)
on the boot server, the only way I could effect the change on the phone
is to reboot it.
b) What about if I set prov.polling.enabled, will the night time poll
effect the change (even
As far as I recall it can be done from the config file only. Here is
the
relevant line from sip.cfg:
device device.set=1 device.auth.localAdminPassword.set=1
device.auth.loc
alAdminPassword=YOUR-PASSWORD-HERE /
What sip release are you referring to?
I am looking at sip 1.6.x and sip.cfg
Polycoms need to reboot if you do much more than pick up the handset
and dial a number. A change of config of any scale certainly qualifies
here.
===That is a bit disappointing!
If you alter the sip.cfg file on your TFTP/FTP/HTTP server, the
Polycoms should pick up the fact that config file
Without portfast, you're looking at about 30
seconds for STP to negotiate whenever the port bounces, during which
time higher layer protocols are unavailable. This may interfere with
CDP and DHCP, if you're using those.
I am using DHCP and I could briefly recall my PC hanging for a short
while
especially when coupled with the fact that they take so long to
reboot.
It reminds me of the good old days of Microsoft Win95 and NT.
I must admit, I'm surprised that they don't handle the config file
changing for them on the server better - I had thought they were better
than that.
Yes, I just
I have been testing with Polycom IP600 phones for a month or so.
I found out that there are frequent problems with the handset.
The problem is I can hear the other end but the other end cannot hear
me.
I have already downloaded the latest bootROM 3.1.3 and sip 2.1.2
However, there are no
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