At 11:58 PM 10/18/2007, you wrote:
>I could write you a script to wash your car.
>You'd just need some kind of interface to do the
>mechanical part of the work.
I have a script to wash a car so you don't have to write
one: http://www.lazaino.com/application.html
Sorry, couldn't resist. Ira
I know of a call centre that bought a cheap projector for that purpose.
PaulH
On Thu, 2007-10-18 at 23:28 -0700, o o wrote:
> Has anyone used an LED wall display with asterisk? I have a customer
> who has an ancient telecorp system that drives an LED wall display. It
> shows the number of agent
As usual, we'll be jawing about any and all asterisk-related subjects
with the usual gang and any new people are always welcome, regardless
of your level of expertise. You can even come and ask questions, it's
guaranteed to be a more pleasant experience than it will be on IRC ;)
http://VoipUsersCo
Roger Schreiter wrote:
> What is a reliable way to read the real IP address of the origin
> of a SIP call?
Maybe SIPCHANINFO(peerip) or SIPCHANINFO(recvip)?
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and no
Rob Schall wrote:
> Here's what I'm looking to do
>
> exten => 10,1,Dial(SIP/1000&SIP/1001,15,wW)
> exten => 10,2,Voicemail(u1000)
>
>
> This should ring both phones and they should keep ringing for the
> alloted time before moving on. However, it appears that if one of the
> phones is Busy,
On Fri, 2007-10-19 at 07:22 -0700, bilal ghayyad wrote:
> My Asterisk version is 1.4 and I am trying to use the
> ResponseTimeOut() application to control the response
> time of the Background function, but when the running
> arrive for the ResponseTimeOut() then the call drop
> and my debuging say
Or your could use a touch screen with Flash Operator Panel. Just a
suggestion out of left field.
-Original Message-
From: Russell Brown [mailto:[EMAIL PROTECTED]
Sent: Friday, October 19, 2007 1:12 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Receptionists Phone sugge
I have a client that want to try the softphone with USB handsets route
to see if hardphones will even be needed. I always push for hardphones
(Polycom) so I am not sure about softphones or USB handsets.
This is going to be for a 300+ seat call center onsite and many offsite,
I plan on using Op
Hi List;
My Asterisk version is 1.4 and I am trying to use the
ResponseTimeOut() application to control the response
time of the Background function, but when the running
arrive for the ResponseTimeOut() then the call drop
and my debuging says:
No Application 'ResponseTimeout' for extension
(Test
for those of you who have not joined the conference call yet, I highly
recommend it. there is always several interesting tidbits that will
help you in your * implementations...
see you at 12:30p today!
daveC
randulo wrote:
As usual, we'll be jawing about any and all asterisk-related sub
C F wrote:
> How on earth does this prevent Glare? Or even reduce it?
>
I think he was providing his configuration in case there WAS a change he
could make to reduce it.
The only thing we could do was an option because our incoming lines were
arranged in a hunt group. We made sure that we di
Eric "ManxPower" Wieling wrote:
> The remote server is where your problem is.
>
Thanks for the reply but I can call the extension in question normally
and it works fine. The problem is that the IAX trunk appears to be
answering before it knows if the physical destination is available or
not. I ha
How on earth does this prevent Glare? Or even reduce it?
On 10/19/07, Gustavo Gonzalez <[EMAIL PROTECTED]> wrote:
> How I change my configuration to reduce this issue. I have this setting on
> my zapata.conf
>
> signalling=fxs_ks
> group=1
> callgroup=1
> pickupgroup=1
> channel=1
>
> signalling=f
Hi guys,
There is other way instead plain text to define passwords in sip.conf ?
In register, peers and extensions ?
Thanks.
--
Frederico Madeira
[EMAIL PROTECTED]
www.madeira.eng.br
___
--Bandwidth and Colocation Provided by http://www.api-digital.
It should be automatic. As long as you have a dial plan destination for
the H.323 endpoint, it does not matter what the transport and protocol
is. That's handled transparently by its various channels.
You will have to configure the SIP and H.323 settings for the channel
drivers, of course, but
The same way you do it with IAX2, pretty much.
http://www.voip-info.org/wiki-Asterisk+config+sip.conf
On Fri, 19 Oct 2007, bilal ghayyad wrote:
> Hi All;
>
> Alot of softswitches or PBX's does not accept to
> manipulate any SIP call without being registered
> firstly. So that means, I need aste
On 10/19/07, Per Jessen <[EMAIL PROTECTED]> wrote:
> Per Jessen wrote:
>
> > [EMAIL PROTECTED] wrote:
> >
> >> Did you set "NAT Keep Alive Enable: = Yes" for the line in question
> >> in the SPA's configuration?
> >>
> >
> > Uh, no, not specifically and I'm guessing it's not set by default?
>
> The
Eric "ManxPower" Wieling wrote:
> Voicemail will answer the line. 10 seconds is a pretty short timeout.
>
> What you need to do is copy the CLI output of your failed calls from
> BOTH servers and put them in this thread. Then we can SEE what Asterisk
> is ACTUALLY doing.
>
Thanks for making
Does anyone have this guide and be willing to share it with me?
Thank in advance?
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mai
We switched to T1(PRI) for high bandwidth & voice quality, echo
I am using TE212P(which is a dual span Echo Chancellor & hardware DTMF). I
have only one PRI connection from PSTN, but I implemented this 6 months agao
when there were no single span cards.
Sangoma just came with one in April, bu
We use:
http://www.ngnsky.com/product_info.php?cPath=21&products_id=50
when we have the remote extension blues.
It works quite well for us and the phone isn't that bad.
On 10/19/07, Vincent <[EMAIL PROTECTED]> wrote:
>
> Hi
>
> SIP is such a pain to use when NAT is involved that I'm willing t
Yehavi Bourvine +972-8-9489444 wrote:
> In any case, I'll try this week to upgrade to 1.4.6 version and then add
> IMAP
> support and inform what happens.
There have been _many_ IMAP related fixes sine 1.4.6. Please try the latest
version, 1.4.13, instead.
--
Russell Bryant
Senior Software E
Frederico,
Take a look at:
http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+md5secret
This is the only way I know of.
-- Alex
On Fri, 19 Oct 2007, Frederico Madeira wrote:
> Hi guys,
>
> There is other way instead plain text to define passwords in sip.conf ?
> In register, peers and
Hi All;
If I installed H.323 on asterisk, and the caller phone
was SIP endpoint while I need to route the call for a
destination via an H.323 trunk, so Asterisk will do
that SIP to H.323 translation automatically or I have
to do also a configuration to SIP to H.323
translation?
Regards
Bilal
___
Ade Vickers wrote:
> Is it possible in Asterisk 1.4.x to issue a dialplan command which will set
> a phantom SIP extension to "busy" for as long as a caller is in the "spam
> trap", & back to idle when they finally give up & hang up?
http://www.asterisk.org/node/48325
http://www.asterisk.org/node
Hello
I've never built an IVR before, so I was wondering if someone
could share some code from their extensions.conf that would perform
some of thoses steps:
1. When a call comes in from the TDM FXO port, answer
2. If no CID, play message "No CID available. Please type the number
where yo
I recently implemented a simple "spam trap" extension for telemarketers -
once identified as a telemarketer (usually they ask to speak to the person
in charge of recruiting/website/purchasing/etc.), I simply offer to put them
through to the person in question, & dump them on a special extension whi
Frederico Madeira wrote:
> Hi guys,
>
> There is other way instead plain text to define passwords in sip.conf ?
> In register, peers and extensions ?
>
> Thanks.
>
Depending on how your asterisk server is setup to run, if you chmod
/etc/asterisk as 750 and the files underneath as 640, then on
On 10/19/07, Michael J. Liberatore <[EMAIL PROTECTED]> wrote:
>
>
> In addition to my below question, i was wondering if anyone had a problem
> with installing zaptel on debian sarge. its a udev problem, make install
> thinks i am running udev, but when i fix the makefile to be like 1.4.4 which
>
On 10/17/07, Michael Graves <[EMAIL PROTECTED]> wrote:
> I'm in process of transitioning a number of offices to a hosted virtual
> pbx from Junction Networks. It's a combination of OpenSER and Asterisk.
> They have a nice click-to-call extension for Firefox, but I need the
> equivalent for IE so th
Hmm.. I think it should be cleaning it up post-call already. If not,
please open a bug on Mantis as that sounds like a bug.
On 10/19/07, Anthony Messina <[EMAIL PROTECTED]> wrote:
>
> Is there a variable for the filename that is created by the FollowMe
> application when "a" is specified as an op
On Fri, 19 Oct 2007 14:16:40 -0700, "Charles Alvis"
<[EMAIL PROTECTED]> wrote:
>http://www.ngnsky.com/product_info.php?cPath=21&products_id=50
Thanks. I'll check it out.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-
We tried with MS Exchange but couldn't get it to work (MS Exchange
doesn't support a master account).
CP
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olivier
Sent: Thursday, October 18, 2007 11:20 PM
To: Asterisk Users Mailing List - Non-Commer
I Just wanted to add something here,
Having separate VLAN does nothing in terms of QOS.
In fact having a computer feeding from phone make more sense because phone
will untag packets coming from PC.
and after that its all about your switch how to prioritize packets.
Unless there is a way in your swi
Well this is the bug I am having with the make install of 1.4.5.1:
http://bugs.digium.com/view.php?id=10156
Even though I got it to install ztcfg -vvv still says 1.4.4 also.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Fredrickson
Sent
Please see: http://spc.pifiu.com under "SPA Phone Admin guide"
On 10/19/07, Bruce Komito <[EMAIL PROTECTED]> wrote:
> Does anyone have this guide and be willing to share it with me?
>
> Thank in advance?
>
> Bruce Komito
> High Sierra Networks, Inc.
> www.servers-r-us.com
> (775) 236-5815
>
>
>
>
[EMAIL PROTECTED] wrote:
> On 10/19/07, Michael J. Liberatore <[EMAIL PROTECTED]> wrote:
>>
>> In addition to my below question, i was wondering if anyone had a problem
>> with installing zaptel on debian sarge. its a udev problem, make install
>> thinks i am running udev, but when i fix the makef
I hope 2 things need to be clear.
1) One call per line, needs to be set on the VoIP.
2)We user Polycom 501 for all Desktop & Polycom 601 for reception.
http://media.polycom.com/usa/en/products/voice/soundpoint_ip/601/demo/index.html
OK, what I mean by one call per line
-- Polycom of SIP Phone
Voicemail will answer the line. 10 seconds is a pretty short timeout.
What you need to do is copy the CLI output of your failed calls from
BOTH servers and put them in this thread. Then we can SEE what Asterisk
is ACTUALLY doing.
Alan Lord wrote:
> Eric "ManxPower" Wieling wrote:
>> Alan Lord
Hi All;
Alot of softswitches or PBX's does not accept to
manipulate any SIP call without being registered
firstly. So that means, I need asterisk to register
firstly then I can route my calls to that SIP trunk.
In IAX2, we use the register => , so what shall we do
in Asterisk? And how its format
Is there a variable for the filename that is created by the FollowMe
application when "a" is specified as an option to record the caller's name?
I'd like to clean up the recorded name files after the call is complete.
Thanks -Anthony
--
Anthony - http://messinet.com - http://messinet.com/~ame
Hi
SIP is such a pain to use when NAT is involved that I'm willing to buy
an IAX hardphone for someone who works remotely over the Net and needs
to get calls from our Asterisk server, itself behind a NAT.
Are there good, affordable IAX phones you would recommend?
Thank you.
___
In addition to my below question, i was wondering if anyone had a
problem with installing zaptel on debian sarge. its a udev problem,
make install thinks i am running udev, but when i fix the makefile to be
like 1.4.4 which works, when i load ztcfg it still says 1.4.4. so
something is not right..
It's just FOP which works well. Dependent on the quality of touch screen
obviously. I haven't spend any time with FOP using Touch screens myself but
I'm sure others here have. There was a thread a few days ago that got into
it a bit.
-Original Message-
From: Mike Clark [mailto:[EMAIL PR
Alan Lord wrote:
> Eric "ManxPower" Wieling wrote:
>> The remote server is where your problem is.
>>
>
> Thanks for the reply but I can call the extension in question normally
> and it works fine. The problem is that the IAX trunk appears to be
> answering before it knows if the physical destinati
Mojo with Horan & Company, LLC wrote:
> C F wrote:
>
>> How on earth does this prevent Glare? Or even reduce it?
>>
>>
> I think he was providing his configuration in case there WAS a change he
> could make to reduce it.
>
> The only thing we could do was an option because our incoming
Trunk and backport. IMHO that is the way to go.
Philipp Kempgen wrote:
> Philipp Kempgen wrote:
>
>> Steve Murphy wrote:
>>
>
>
>>> But, in 1.4, I really can't add a new CDR field and call it a 'bug fix'.
>>> It really is a 'new', 'enhanced' sort of thing. So, this kind of change
>>> wi
On 10/19/07, Kevin Smith <[EMAIL PROTECTED]> wrote:
>
>
> Robert McNaught wrote:
> > Hi,
> >
> > Has anyone had any great difficulties with QoS using the second
> > ethernet phone in these Polycom phones for desktop machines in a
> > converged network? I had heard that these can cause difficulties
Hi,
just noticed chan_mobile, which looks like it will do exactly as I need.
http://www.voip-info.org/wiki-Asterisk+Bluetooth+channels
However seems it is only for latest 1.4 but there is a mention of a
backport for 1.2
http://www.sigsegv.cx/sip-9.html
Anybody using this with something like 1.2.1
Hi Robert,
While I'm not sure how our network compares with yours, we run about
twenty 601 phones along with our office workstations (some stations are
without a phone). Each station with a phone is connected with the other
Ethernet port on the phone so we have one drop to each station. The
ph
Philipp Kempgen wrote:
> Steve Totaro wrote:
>
>> I am using Thunderbird 2.0.0.5. If using Outlook, I think the time is
>> correct.
>
> Does MS have a different attitude towards timezones? :)
Sorry. I forgot that they don't read RFCs. ;)
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 1
Steve Totaro wrote:
> I am using Thunderbird 2.0.0.5. If using Outlook, I think the time is
> correct.
Does MS have a different attitude towards timezones? :)
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and
Philipp Kempgen wrote:
> Steve Murphy wrote:
>> But, in 1.4, I really can't add a new CDR field and call it a 'bug fix'.
>> It really is a 'new', 'enhanced' sort of thing. So, this kind of change
>> will have to go into trunk at the moment.
>
> Sad but true.
> I guess it couldn't go in even if th
Steve Murphy wrote:
> On Wed, 2007-10-17 at 02:41 +0200, Philipp Kempgen wrote:
>> Steve Murphy wrote:
(Sorry for quoting so much but I need to keep the context.)
>>> For instance, if Zap/52 dials Zap/51,
>> 52 --- dials & talks ---> 51
>>
>>> who hookflashes and dials Zap/50,
>> 51 --- dials
Hello,
I tried a few months ago to use IMAP with Asterisk; I used either 1.4 or the
latest SVN at that time (sorry, don't remember).
After a day I had to remove it since Asterisk crashed, mostly in the IMAP
client code (the code of UW IMAP). My users wants IMAP back (they loved it) but
not in
On 10/19/07, Mike Clark <[EMAIL PROTECTED]> wrote:
>
> Do they play well with Vista?
Hah - I have no idea. We installed Vista on one laptop here when Dell
started shipping it. That lasted about 3 days and 10 support tickets
from the user. Then we reverted back to XP. Haven't touched Vista
sinc
The remote server is where your problem is.
Alan Lord wrote:
> Hello,
>
> This message is similar to one I posted before, but with a different
> subject line and I've revised the description to hopefully make it clearer.
>
> The basic problem is I am trying to dial 2 numbers simultaneously usin
All of the emails I get from the list have the correct time with the
exception of the typical list slowness.
All of your emails (and only your emails and spam) are approximately 11
or twelve hours in the future. The email I am responding to has the
correct day but the time reads 11:13 PM.
I am
Hello,
This message is similar to one I posted before, but with a different
subject line and I've revised the description to hopefully make it clearer.
The basic problem is I am trying to dial 2 numbers simultaneously using
the & construct. One device is a "locally attached" soft SIP phone. The
shadowym wrote:
> Or your could use a touch screen with Flash Operator Panel. Just a
> suggestion out of left field.
>
>
shadowym:
Do you have a specific setup w/touchscreen that you have deployed and
that works well?
Thanks,
Mike
___
--Bandwid
ResponseTimeout was deprecated in 1.2 and removed in 1.4. Was this
information not in the upgrade.txt file in 1.2 and 1.4?
bilal ghayyad wrote:
> Hi List;
>
> My Asterisk version is 1.4 and I am trying to use the
> ResponseTimeOut() application to control the response
> time of the Background f
bilal ghayyad wrote:
> My Asterisk version is 1.4 and I am trying to use the
> ResponseTimeOut() application to control the response
> time of the Background function, but when the running
> arrive for the ResponseTimeOut() then the call drop
> and my debuging says:
>
> No Application 'ResponseTi
[EMAIL PROTECTED] wrote:
> Hi,
>
> I'm running some Asterisk-machines, and on one of them i get this errors
> in the CLI, but i don't know what that means.
>
> Hardware:
> Digium 4-Port E1 Card with HWEC
> Intel Pentium D 3 GHz
> 2 GB RAM
> SATA Harddisk
> Supermicro Mainboard
>
> Software:
> l
On Wed, 2007-10-17 at 02:41 +0200, Philipp Kempgen wrote:
> Steve Murphy wrote:
>
> > It's not a bad idea, but I think the philosophy would be that whatever
> > turns CDR records into billing statements could/should/would decide
> > which to skip, and which to process, and not something in Asteris
Perssy Llamosas wrote:
> I doubt it.
>
> hxxp://boycottnovell.com/2007/10/02/opensuse-103-release/
>
I think that is the sort of thing the OP would classify as "religious
grounds".
/Per Jessen, Zürich
--
http://www.spamchek.com/ - your spam is our business.
__
Hi List;
I need to do IP Trunk between Asterisk and another
softswitch provider, the softswitch support SIP but
requires Asterisk to register for this IP Trunk (it
should appears as gatekeeper entity that does
registeration to another gatekeeper entity).
How can I configure this SIP trunk to do r
Any chance that your dtmf is not set up correctly ?
- Original Message -
From: Michael Munger
To: asterisk-users@lists.digium.com
Sent: Tuesday, October 16, 2007 10:30 PM
Subject: [asterisk-users] Background not listening?
This ridiculously simple IVR is not listening to di
On my SPA3102 on the Admin Advanced SIP page:
Subsitute VIA Addr: yes
Send Resp To Src Port: yes
I also set the RTP Port Min & RTP Port Max so that my NAT router could be set
up to forward RTP packets to this device.
This is quite a good posting about setting up Linksys devices to handle NAT
(
Hi,
incoming SIP calls have a channel name in the form of:
SIP/-
This is a way to get fetch the IP address of the remote side
of a SIP call - in most cases.
However, sometimes, instead of the IP address, there is a host
name in the channel name. I assume, this value in the channel name
is not th
Tzafrir Cohen wrote:
>
> By now there are quite a few x86_64 Asterisk users that complain if
> something breaks.
>
Been using it on a 64-bit P4 with debian 4.0/1 (amd64) for some time now
without a hitch.
___
--Bandwidth and Colocation Provided by htt
Per Jessen wrote:
> [EMAIL PROTECTED] wrote:
>
>> Did you set "NAT Keep Alive Enable: = Yes" for the line in question
>> in the SPA's configuration?
>>
>
> Uh, no, not specifically and I'm guessing it's not set by default?
The SPA921 config has a "NAT Keep Alive Intvl" which is set to 15 by
de
On 10/19/07, Tzafrir Cohen wrote:
> On Thu, Oct 18, 2007 at 11:24:24PM -0400, Baji Panchumarti wrote:
>
> > I hope you have better success than I did, my problem was
> > not so much with asterisk in particular but 64-bit in general.
> >
> > Examples of problems using CentOS 4.5 on x86_64
> >
On Thu, Oct 18, 2007 at 11:24:24PM -0400, Baji Panchumarti wrote:
> I hope you have better success than I did, my problem was
> not so much with asterisk in particular but 64-bit in general.
>
> Examples of problems using CentOS 4.5 on x86_64
>
> - many problems loading php5 & mysql from pack
On Fri, Oct 19, 2007 at 01:40:20AM +, Rony Ron wrote:
> Hello all,
> i would like to have references so i'm giving free help
> for any project (commercial or public).
One useful and obvious reference:
http://www.catb.org/~esr/faqs/smart-questions.html
--
Tzafrir Cohen
Does anyone have any suggestions for a decent receptionists phone?
Aastra? Grandstream?
Something with (potentially) lots of BLFs, large(ish) screen, headset
and most importantly the ability to transfer calls?
I've installed five Snom 370s that seemed ideal but my client is very
very unhappy as
[EMAIL PROTECTED] wrote:
> So your problem is:
>
> -- IAX2/alanb-3 answered SIP/101-081d1050
>
> Except the remote end didn't actually answer the call? The problem is
> your remote end... its answering the call. All the IAX hardphones I've
> seen don't seem to be the highest of quality honest
Doug wrote:
> At 20:40 10/18/2007, Rony Ron wrote:
>> Hello all,
>> i would like to have references so i'm giving free help
>> for any project (commercial or public).
>>
>> regards,
>
> Can you come over and wash my car?
I could write you a script to wash your car.
You'd just need some kind of in
There is a free dialer from http://www.snapanumber.com/
If I remember correctly, it will let you click on phone numbers in web pages.
--
--
Steven
http://www.glimasoutheast.org
"Michael Graves" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
> I'm in process of transitioning a nu
Eric "ManxPower" Wieling wrote:
> Alan Lord wrote:
>> Eric "ManxPower" Wieling wrote:
>>> The remote server is where your problem is.
>>>
>> Thanks for the reply but I can call the extension in question normally
>> and it works fine. The problem is that the IAX trunk appears to be
>> answering befo
Erik Anderson wrote:
> On 10/19/07, Steve Totaro <[EMAIL PROTECTED]> wrote:
>
>> Any advice on softphones, handsets, or practical experience with this
>> sort of deployment? It would be very nice if there was a central way of
>> provisioning the phones.
>>
>
> I've deployed several setups
for those of you who have not joined the conference call yet, I highly
recommend it. there is always several interesting tidbits that will
help you in your * implementations...
see you at 12:30p today!
daveC
randulo wrote:
As usual, we'll be jawing about any and all asterisk-related sub
How I change my configuration to reduce this issue. I have this setting on
my zapata.conf
signalling=fxs_ks
group=1
callgroup=1
pickupgroup=1
channel=1
signalling=fxs_ks
group=2
callgroup=1
pickupgroup=1
channel=2;
singalling=fxs_ks
group=3
callgroup=1
pickupgroup=1
channel=3;
singalling=fxs_k
On Fri, 2007-10-19 at 09:12 +0100, Russell Brown wrote:
> Does anyone have any suggestions for a decent receptionists phone?
> Aastra? Grandstream?
>
> Something with (potentially) lots of BLFs, large(ish) screen, headset
> and most importantly the ability to transfer calls?
Personally I'm happy
QueueMetrics is able to prepare a realtime screen meant for a video
projector or large LCD screen to display to show call-center stats in
real-time. We have quite a number of customers who used old linux boxes
connected to the right display that just start up, start firefox and go to
a spe
Hi,
I'm running some Asterisk-machines, and on one of them i get this errors
in the CLI, but i don't know what that means.
Hardware:
Digium 4-Port E1 Card with HWEC
Intel Pentium D 3 GHz
2 GB RAM
SATA Harddisk
Supermicro Mainboard
Software:
latest libpri/zaptel/asterisk of version 1.2
I tried
Russell Brown wrote:
>
> Does anyone have any suggestions for a decent receptionists phone?
> Aastra? Grandstream?
>
Linksys SPA94x/6x perhaps. I don't know if it has the transfer problem
or not.
/Per Jessen, Zürich
--
http://www.spamchek.com/ - your spam is our business.
_
Hi all,
Well, actually, I'm looking at asterisk from the development/SIP side of
things, not the cartoons. Or that's what I hope my project leader wants
me to do...
Best regards,
Matti Zemack, BBC R&D, Kingswood Warren, UK
http://www.bbc.co.uk/
This e-mail (and any attachments) is confidential a
[EMAIL PROTECTED] wrote:
> Did you set "NAT Keep Alive Enable: = Yes" for the line in question in
> the SPA's configuration?
>
Uh, no, not specifically and I'm guessing it's not set by default?
thanks.
/Per Jessen, Zürich
--
http://www.spamchek.com/ - your spam is our business.
o o wrote:
> Has anyone used an LED wall display with asterisk? I have a customer who has
> an ancient telecorp system that drives an LED wall display. It shows the
> number of agents signed in, calls in queue, hold time, etc. It also sounds an
> alarm if the hold time exceeds a set value. I'm l
I had a similar issue a while ago. Check your dial plan. Are you
forwarding to your cell phone's V-Mail as fallback? I had the issue
where I was getting callbacks from asterisk if one phone was on DnD and
the calll wasn't answered. Becarefull of your dial() commands and the
delays you use.
On 10/19/07, Steve Totaro <[EMAIL PROTECTED]> wrote:
>
> Any advice on softphones, handsets, or practical experience with this
> sort of deployment? It would be very nice if there was a central way of
> provisioning the phones.
I've deployed several setups internally using X-Lite and these headse
Kevin Smith wrote:
> Hi Robert,
>
> While I'm not sure how our network compares with yours, we run about
> twenty 601 phones along with our office workstations (some stations are
> without a phone). Each station with a phone is connected with the other
> Ethernet port on the phone so we have on
I think most phones somehow have this kind of behaviour : "transfer button
applies to ongoing call" and so on.
What happens if you don't press TRANSFER again (when display shows < Call A
> CallB) ?
Have you tried call parking ?
What if you used blind transfer instead ?
If receptionist is busy, ass
On Fri, Oct 19, 2007 at 11:15:35PM +0100, Alan Lord wrote:
> Frederico Madeira wrote:
> > Hi guys,
> >
> > There is other way instead plain text to define passwords in sip.conf ?
> > In register, peers and extensions ?
> >
> > Thanks.
> >
>
> Depending on how your asterisk server is setup to r
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