Hello, I've tried to transfer a IAX call to a number configured on a
traditional
PBX, but it doesn't work. I have a traditional PBX connected with a zap channel
to Asterisk in the following way:
IAX/SIP client -- Asterisk (FXO) -- (FXS) traditional PBX --- OFFICE
Phones
Asterisk is connected
Hi list,
I tried to use DISA in order to get the line when I call with my mobile
phone but the system doesn't recognise my DTMF tones when I call to a SIP
trunk.
Everything is working Ok if I use a ZAP Trunks.
I tried to google to find a solution but I wasn't able to find any.
Hi guys, I have recently installed a Asterisk Server with CDR Call Detail
Records. I have installed it over a Asterisk 1.2 , but now It do not run
. I have installed it with the following procedure:
# yum install ncurses
#yum install openh323-devel
# yum install mysql-server
# yum install
Hi,
I've been looking for a good SIP application for Windows Mobile for ages.
I found speaQ, but it has the same problem as any other softphone for
Windows Mobile.
You see, it uses the speaker to output the conversation instead of the
phone speaker,
you know the one that is used when you
Hi everyone!
I'm having some issue trying to place calls with asterisk connected to
an E1 R2 from Telmex Argentina. The other E1 port is connected to a
Meridian which also uses R2 protocol. Calls sometimes fail with
different error messages such as: Unicall protocol error 32773, 32772,
32769.
I had the same issue. I needed to install #yum install mysql-devel.
Once I did this the addons compiled the file fine.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pablo
Almido
Sent: Wednesday, January 17, 2007 9:43 AM
To:
I don't think that the first priority (exten = _44XX,1,Answer) is ok,
have you tried without it?
Try not answering and post what happens.
On 1/17/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hello, I've tried to transfer a IAX call to a number configured on a
traditional
PBX, but it doesn't
Long story short...
Snom's ...
Retrieve button... works when MWI is *NOT* lit but does *NOT* work when
it is lit.
Any advice
Useragent: snom360/6.5.2
Function: F_RETRIEVE
[EMAIL PROTECTED] ~]# asterisk -rx show version
Asterisk 1.2.13 built by root @ pbx on a i686 running Linux on
I'm using asterisk 1.2.9.1 and mysql 3.23, asterisk add-ons 1.2.3.
All seems to work normally with realtime voicemail, reads vmbox
parameters from the db fine. When I try to change the password,
asterisk operates normally, enter new password ok, re-enter new
password ok, password has been
Subject: Re: [asterisk-users] windows mobile 5 softphone for square screen
devices
Anton Krall a écrit :
Guys, anybody has seen or is using some kind of softphone on any square
screen device with WM5? Ive tried sjlabs one and xten for pocket pc and
they
do work on Wm5 but they are designed
Snom's ...
Retrieve button... works when MWI is *NOT* lit but does *NOT*
work when
it is lit.
Any advice
Do you have an asterisk extension in your dialplan? See
http://www.voip-info.org/wiki-Asterisk+phone+snom, especially the part
about:
Making the MWI work with ASTERISK
Asterisk
I have not confirmed this independently, but I believe this is fixed
if you disable the Show message light when a call is missed feature
in the phone config. Alternatively, try pressing X to clear the
missed call indication before pressing Retrieve
Might work... Might not :)
Steve
On 1/17/07,
On 1/17/07, JR Richardson [EMAIL PROTECTED] wrote:
I'm using asterisk 1.2.9.1 and mysql 3.23, asterisk add-ons 1.2.3.
All seems to work normally with realtime voicemail, reads vmbox
parameters from the db fine. When I try to change the password,
asterisk operates normally, enter new
Steve Davies wrote:
I have not confirmed this independently, but I believe this is fixed
if you disable the Show message light when a call is missed feature
in the phone config. Alternatively, try pressing X to clear the
missed call indication before pressing Retrieve
Might work... Might not :)
On 1/18/07, JR Richardson [EMAIL PROTECTED] wrote:
I'm using asterisk 1.2.9.1 and mysql 3.23, asterisk add-ons 1.2.3.
All seems to work normally with realtime voicemail, reads vmbox
parameters from the db fine. When I try to change the password,
asterisk operates normally, enter new
Hi,
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?
Robert Jenkins.
Is AbsoluteTimeout designed to work with canreinvite=yes?
If not, are the any other options for disconnecting a call after a
predefined duration when using canreinvite=yes?
Thanks!
David
___
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David Thomas wrote:
Is AbsoluteTimeout designed to work with canreinvite=yes?
If not, are the any other options for disconnecting a call after a
predefined duration when using canreinvite=yes?
Reinvites only reinvite the MEDIA, not the SIGNALING. It should work,
but I've not tested it
Dear Friends and Supporters!
I try to install the Asterisk 1.4, and I needs to activate the hint to for
the call pickup feature. However, the hint is enabled and I can see the
status of the extensions by run command show hints. It show the phones are
Idle. However, it would NOT be able to
exten = asterisk,1,VoicemailMain(${CALLERIDNUM})
I use that one myself. Does the Snom attempt to dial asterisk when
you hit Retrieve? What error do you get? Sure it's in the right
context (I screw that up ALL the time)...
Stefano
___
--Bandwidth
Hi Guys
I'm conecting 2 astersk servers using this arquitecture
(Ext softphone)==sip==(asterisk 1)iax2 trunk(asterisk 2)
===alaw==(pstn)
If i call from the Ext to the asterisk 2 the sound is perfect, but
if i call from Ext to the pstn, i can hear perfect but they tell me
that
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
What kind of pstn interface are you using? For analog interfaces try
adjusting txgain in zapata.conf.
Yelson Vivas wrote:
Hi Guys
I'm conecting 2 astersk servers using this arquitecture
(Ext softphone)==sip==(asterisk 1)iax2
Richard Soderblom wrote:
Hi.
Has anyone had any success in implementing a callback or ringback
function in Asterisk?
I've had a look at the callback-voicemail example on voip-info.org
http://www.voip-info.org/wiki/view/Asterisk+tips+callback
However it won't quite work for me.
I need it for
I have a client that has 5 Snom 320s. 4 work great, one does not. I upgrade
the firmware to the latest (6.5.2) and the problem goes away, but then comes
back a couple days later.
There is a slight packet loss on the phone (about 1%), though there is no
packet loss on any of the other phones.
I have solved the problem, I have already install mysql-devel and then
# cd asterisk-addons-1.4.0
# make distclean
# ./configure
# make
# make install
# make samples
My Call Detail Records is running.
2007/1/17, Savoy, Kevin - Williston, ND [EMAIL PROTECTED]:
I had the same issue. I
I have the same problem. Please reply to the list if you figure it out.
I'll do the same.
_
From: Pablo Almido [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 17, 2007 9:43 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk 1.4 and CDR
Hi guys, I have
On 1/17/07, JR Richardson [EMAIL PROTECTED] wrote:
I'm using asterisk 1.2.9.1 and mysql 3.23, asterisk add-ons 1.2.3.
All seems to work normally with realtime voicemail, reads vmbox
parameters from the db fine. When I try to change the password,
asterisk operates normally, enter new
Had this happen when I put a SNOM 360 On the Lan over Fiber. The
Fiber transivers where stuck to 100tx and it was botching things. I
put a 10base hub between the fiber and the phone and it worked.
Disable the auto network config and I think that you can set the
unplug to ignore and another
Hi. I had successful confiured my Asterix PBX, but now I need to connect it
to a Nortel Meridian phone plant, I got a Digium T100P on the machine I got
the Asterisk and I already connect it with the phone plant, I need to know
what configuration do I have to do so the ip extensions can make calls
On the voip-info.org wiki there are good tips to get snoms to play nice on
lans. I personally have experienced wierdness using particular switches
(cheap ones).
also note that snom now has an auto-update subscription URL in their support
wiki, if you use the URL it makes updating a 4.X to 6.X
I use that one myself. Does the Snom attempt to dial asterisk when
you hit Retrieve? What error do you get? Sure it's in the right
context (I screw that up ALL the time)...
Sometimes it's asterisk, sometimes it's unknown sometimes, it's
Unknown so:
exten =
On Wednesday 17 January 2007 3:35 pm, Gustavo Andrés Salazar Giraldo wrote:
Hi. I had successful confiured my Asterix PBX, but now I need to connect it
to a Nortel Meridian phone plant, I got a Digium T100P on the machine I got
the Asterisk and I already connect it with the phone plant, I need
Hmm, the use of autopause in queues.conf introduces a new issue. When a
queue member is on a call, the queue continues to try to send calls to
the member's interface. Getting the 'Busy Here' response from the SIP
device causes the caller to continue holding.
The new issue is that autopause
Hi folks,
When my Sipura 2k is registered to Asterisk, I get some peculiar error
messages repeated in the logs every 30 seconds. I've put a snippet up in
http://pastebin.co.uk/9067 for you to see. I don't have any complicated
setups. Just 2 sip.conf entries for the Sipura, and 2 more for
On Wednesday 17 January 2007 3:43 pm, Colin Anderson wrote:
Sometimes it's asterisk, sometimes it's unknown sometimes, it's
Unknown so:
exten = asterisk,1,VoicemailMain(${CALLERIDNUM})
exten = Unknown,1,VoicemailMain(${CALLERIDNUM})
exten = unknown,1,VoicemailMain(${CALLERIDNUM})
This
On 1/17/07, Victor Perez [EMAIL PROTECTED] wrote:
Tried that, it didn't work but maybe I didn't configure it right. Anyways
how can I route all outgoing calls from that specific extension to use that
trunk?
Put that extension in a different context.
Yes, i am using E1/PRI
Thanks, for your help.
On 1/17/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Wednesday 17 January 2007 3:35 pm, Gustavo Andrés Salazar Giraldo
wrote:
Hi. I had successful confiured my Asterix PBX, but now I need to connect
it
to a Nortel Meridian phone plant, I 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
Browsing through the developers documentation and 1.4 source, I see
references to STUN in the code and documentation.
Does 1.4 have support for STUN, if so how is it configured?
Regards,
David
___
--Bandwidth and Colocation provided by Easynews.com --
Well Guys.. I just bought the X-PDA one and indeed it has 240x240 support,
costs around 29 USD and very well worth it, works great on an iPAQ 6945 via
wireless and using my BT headset all sound goes to the headset and not the
speaker, which is great and solves the eternal problem of having to
James Fromm wrote:
Hmm, the use of autopause in queues.conf introduces a new issue. When a
queue member is on a call, the queue continues to try to send calls to
the member's interface. Getting the 'Busy Here' response from the SIP
device causes the caller to continue holding.
The new
Hi All,
I have a (legacy) Praxton PBX, it has a PRI T1 input card and 64
analog extensions through 4 amphenol connectors.
We receive 12 voice channels (other 12 are idle) and have 100 DIDs.
No caller ID thru PRI though.
The Praxton box is amazing in terms of configuration and flexibility
DoH! I missed that ringinuse. Thanks!
Julian Lyndon-Smith wrote:
James Fromm wrote:
Hmm, the use of autopause in queues.conf introduces a new issue. When
a queue member is on a call, the queue continues to try to send calls
to the member's interface. Getting the 'Busy Here' response from
Okay. That doesn't help. What forces * to look at the DB rather than
waiting on a registration ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Thomas
Sent: Monday, January 15, 2007 9:09 AM
To: Asterisk Users Mailing List - Non-Commercial
On 1/17/07, Tim Connolly [EMAIL PROTECTED] wrote:
Okay. That doesn't help. What forces * to look at the DB rather than
waiting on a registration ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Thomas
Sent: Monday, January 15, 2007 9:09 AM
To:
On 1/17/07, David Thomas [EMAIL PROTECTED] wrote:
On 1/17/07, Tim Connolly [EMAIL PROTECTED] wrote:
Okay. That doesn't help. What forces * to look at the DB rather than
waiting on a registration ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Hello,
The IT folks at a hospital in New Zealand have approached us about
deploying Asterisk, but they would like to talk to people at other
hospitals that have already done this.
If anyone works at a hospital that has deployed Asterisk, or deployed
Asterisk at a hospital would you please get in
I guess I'm missing something else. 'ringinuse = no' doesn't change
anything. While on a call, the queue still sends another call and
proceeds to set the member paused after receiving 'Busy Here' back from
the SIP device.
My queues.conf is:
[general]
persistentmembers = no
I aaume you are calling in on a PSTN line? If so what fxo are you using
with Asterisk.
Doug
On Wed, 17 Jan 2007, Giuffredi wrote:
Hi list,
I tried to use DISA in order to get the line when I call with my mobile
phone but the system doesn't recognise my DTMF tones when I call to a SIP
On 1/18/07, JR Richardson [EMAIL PROTECTED] wrote:
I use the same database for the sip, iax, exten and vm, different
tables. When a sip device registers, asterisk writes to the database
with updates to the sip table ipaddress, port and regseconds, so I
don't think there is a write permissions
Sometimes timers need to be adjusted on the mfcr2 source code.
Sometimes is missconfiguration. Anyway, may be this document can help
you out to debug the problem:
http://moy.ivsol.net/unicall/mfcr2-asterisk-unicall-0.2-english.pdf
Kind Regards
On 1/17/07, Facundo Ameal [EMAIL PROTECTED] wrote:
Hi Stefan,
Thanks for your answer, but it's a error of me in cut, the goto are good:
[Cal-In]
exten = _81120,1,Goto(C-Internal,100,1)
exten = _81121,1,Goto(C-Internal,200,1)
[C-Internal]
exten = 100,1,Ringing
exten = 100,2,Wait,1
exten = 100,3,Answer
exten =
I had been wondering the same thing, I haven't really found any useful
information.
I use:
exten = 123,1,Dial(SIP/123)
exten = 123,2,Voicemail(u123)
exten = 123,102,Voicemail(b123)
If you set DND on the SIP phone usually it sends 486 busy here and
jumps to 102. If you reject the call it usually
Enclosed bellow is the fragment from extenstions.conf which does two things:
*41 - Does the ring-back staff.
*42 - Calls back the last one who called you.
Regards, __Yehavi:
; regular local extensions:
; The flow is: If not available or no answer send to mailbox if
Hello friends,
I am trying to install asterisk 1.4.0 . I am configuring it as follows:-
./configure --prefix=/home/vivek/downloads/install/asterisk/
But still while running 'make install', it tries to install it in
/var/lib/asterisk/ and stops because of failing permissions.
I have provided
Is there any reason why you could not do this?
exten = 123,1,Dial(SIP/123)
exten = 123,n,Goto(s-${DIALSTATUS},1)
exten = 123,n,HangUp
exten = s-NOANSWER,1,Voicemail(u123)
exten = s-NOANSWER,2,Hangup
exten = s-BUSY,1,Voicemail(b123)
exten = s-BUSY,2,Hangup
Or you could have
Hi everyone,
Is it possible to call an asterisk function out an AGI script? How do I do
this?
Thank you,
Thomas
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Hi all,
Are there any issues to be concerned about when calls come in from PSTN
to a PRI card and are forwarded back out the same PRI card? Anything
different have to be enabled in zaptel.conf or zapata.conf or the
Sangoma configs to make this work? What about using .call files that
join two
Hi all,
Is there a simple way to disable external outgoing calls on the basis of
calling extension ?
In other words I would like to have two different groups of internal
extensions, one enabled to place calls on the external PSTN,
the other only enabled to place internal calls or receiving
Hi Adam,
I have the same problem.
Are you sure is an echo canceller problem?
Following advices from this list I discovered that I had an IRQ shared.
Untill now I didn't try the new setup but I really hope that this was the
problem.
If you manage to solve the problem in
I'm a voip service provider and i'm setting up a asterisk box to
register around 100 lines from my central softswitch. This asterisk
box will be placed inside a customer and has a digium card to be
interconected with customer's pabx.
My problem is that when asterisk send register message, my
My *guess* is that the semicolon is being interpreted as a commend
marker, so you might need to escape it with a '\'.
I had problems with this, however, when using it via the [EMAIL PROTECTED]
management interface, because the '\' is stripped on display and so lost
if you view and save a working
Ok, this works pretty fine!
Thank you very much!
On 17/01/07, Steve Langstaff [EMAIL PROTECTED] wrote:
My *guess* is that the semicolon is being interpreted as a commend
marker, so you might need to escape it with a '\'.
I had problems with this, however, when using it via the [EMAIL
Hello, I intend to buy a FXO/FXS device from Linksys. I'm thinking
about SPA3102.
What you guys think about it. Is ok, is working with asterisk, can i use it
like voip peer. Thank you for your advice.
Jonson.
___
--Bandwidth and Colocation provided by
should have sent this to the list, Gordon how are you getting on with BT?
- Original Message -
From: Gordon Henderson [EMAIL PROTECTED]
To: Wireless [EMAIL PROTECTED]
Sent: Friday, January 12, 2007 10:45 PM
Subject: Re: [asterisk-users] Echo...
On Fri, 12 Jan 2007, Wireless wrote:
On 1/17/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
All,
Just came across the prompt #3 from inside the top menu of VM in latest
stable. Allison does not announce the prompt, but if you know it is there,
you can press 3 successfully follow the prompts from there to send your
message to
Okay, i'll move my discuss to asterisk-users.
Thank you.
On 1/17/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Wed, Jan 17, 2007 at 04:39:03PM +0800, 黄宗宁 wrote:
Jonson Player wrote:
Hello, I intend to buy a FXO/FXS device from Linksys.
I'm thinking about SPA3102. What you guys thik
Hi guys,
I got also problems with SPA 941 and 942, pour sound (a kind of click noise)
that when i set volume sound lower almost can't notice, but still exists.
I also notice on SIP to SIP calls , echo that could only be justified by
Handsets hardware quality. When i make calls using Xlite with
I like the idea of Virtual PBX, but I don't like python language.
Are there other implementations ?
I'd like some java or php thing.
On 1/16/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Jan 16, 2007 at 10:18:05AM +0500, Kate Kretz wrote:
Dear Sirs,
let me repost my question again,
Freepbx GUI let's you create different administrators with different
permissions!
On 1/17/07, Kate Kretz [EMAIL PROTECTED] wrote:
I like the idea of Virtual PBX, but I don't like python language.
Are there other implementations ?
I'd like some java or php thing.
On 1/16/07, Tzafrir Cohen
On Wed, Jan 17, 2007 at 11:47:35AM +, Marco Mouta wrote:
Freepbx GUI let's you create different administrators with different
permissions!
But can you separate the permissions by context/domain?
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL
Cant remember the url but I googled it. Xten also without luck.. the main
problem is the 240x240 screen...
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of mitcheloc
|Sent: Wednesday, January 17, 2007 1:48 AM
|To: Asterisk Users
My mistake Tzafrir, you are right!
On 1/17/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Wed, Jan 17, 2007 at 11:47:35AM +, Marco Mouta wrote:
Freepbx GUI let's you create different administrators with different
permissions!
But can you separate the permissions by context/domain?
--
Anton Krall a écrit :
Guys, anybody has seen or is using some kind of softphone on any square
screen device with WM5? Ive tried sjlabs one and xten for pocket pc and they
do work on Wm5 but they are designed for standard screens, anybody using
anything on square ones?
We are using PPCIAX.
--
Same result, more FXO interfaces.
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Al
Envoyé : 17 janvier 2007 00:39
À : asterisk-users@lists.digium.com
Objet : [asterisk-users] TDM404B VS TDM2401B
Hi List,
any good comparison between TDM404B and TDM2401B
i'm not very
housi mueller wrote:
The dnsmgr in Aterisk 1.4.0 seems not to work. I enabled DNS lookups
in dnsmgr.conf but after reloading the conf files * never refreshes DNS
lookups. Any ideas how to debug this issue?
The DNS manager is not used very much in Asterisk 1.4 at all; don't
expect it to
kjcsb wrote:
I had something vaguely similar. Asterisk was replying on the wrong
interface/network card. Might be worth checking.
Asterisk does not choose (or have any control over) which interface is
used for packet transmission. That is the responsibility of your
operating system's IP stack.
Hello Chris,
we have a number of clients who deployed very large CCs over the 200 agent
range.
Your idea #1 is pretty sound and I believe that's what most people are
doing. I would like to add a couple of points of attention:
- having hundreds of agents on a box means a lot of synchronous
Hi
I have two small question, if you can help me ;=)
Problems with Answer+Music
my extension:
[Cal-In]
exten = _81120,1,Goto(C-Internal,100,1)
exten = _81121,1,Goto(C-Internal,200,1)
[C-Phibee]
exten = 100,1,Ringing
exten = 100,2,Wait,1
exten = 100,3,Answer
exten =
Hi all,
some body told me that you can make asterisk to register itself to another
asterisk server. i just want to know whether it really can be done or not. i
have googled a lot but no answeres.
--
Regards
Rizwan Hisham
Software Engineer
___
You can by using a register entry in either SIP or IAX connections. Try
searching voip-info.org for the details of sip.conf and iax.conf and also
connectiong 2 asterisk servers.
On 1/17/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hi all,
some body told me that you can make asterisk to register
Hi,
Am 17.01.2007 um 15:07 schrieb Noc Phibee:
Problems with Answer+Music
my extension:
[Cal-In]
exten = _81120,1,Goto(C-Internal,100,1)
exten = _81121,1,Goto(C-Internal,200,1)
[C-Phibee]
exten = 100,1,Ringing
exten = 100,2,Wait,1
exten = 100,3,Answer
exten =
I realize I cannot use inband audio for phones (voicemail and internal ivr,
password for external trunks and other thing not working)
So I put everywhere rfc2833.
Doing this, anyway, make any EXTERNAL IVR NOT working.
I see a lot of posts about this, but no solution, becouse using inband
audio
Greetings,
I have never done any agi programming, but my first thought is maybe you need
a wait statement after answering?
John Beaman
Telecom Specialist
Voice Telecommunications Services Department.
Good Samaritan National Campus
605-362-3331
[EMAIL PROTECTED] 1/15/2007 10:53:51 AM
I
On 1/18/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hi all,
some body told me that you can make asterisk to register itself to another
asterisk server. i just want to know whether it really can be done or not. i
have googled a lot but no answeres.
--
Regards
Rizwan Hisham
Software Engineer
Tried that, it didn't work but maybe I didn't configure it right. Anyways
how can I route all outgoing calls from that specific extension to use that
trunk?
Thanks.
On 1/16/07, Tim Panton [EMAIL PROTECTED] wrote:
On 16 Jan 2007, at 19:56, Victor Perez wrote:
I have Teliax trunk set to ulaw
Hi.
Has anyone had any success in implementing a callback or ringback
function in Asterisk?
I've had a look at the callback-voicemail example on voip-info.org
http://www.voip-info.org/wiki/view/Asterisk+tips+callback
However it won't quite work for me.
I need it for local SIP users which most
In a couple different locations I have some clients that are having
intermittent problems.
All of my other customers aren't complaining of issues. Whenever I conduct a
test, everything is fine. No call quality issues to speak of.
What can I do to log\monitor these clients so I can
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