Hi,
> -Original Message-
> >- Sometimes a call won't go through, dialtone stays on after
> keypresses.
> >Strange. A powercycle of the iaxy usually helps
>
> Are you changing which phone is plugged into the iaxy before
> this happens? The changing of resistance levels I have
> noticed
MessageHi,
> -Original Message-
> From: Aaron Martin
> Sent: Tuesday, May 18, 2004 8:55 PM
> Does anyone have any recomendations for a free Windows softphone, SIP or
> IAX that supports the following features:
> * Message Waiting Indicator
> * Consultative Transfers
> * Speed Dials
DIAX
I'm having the exact same problems here - won't start with safe_asterisk.
I'm running a slightly dated CVS head (CVS-02/24/04-15:39:13) however I have
two machines running this date CVS, the other already has G.729 installed
and works fine - however it registered automatically with the voiceage
re
Hi,
> -Original Message-
> i have extensions in locations across a number of telco area codes.
> when someone in seattle picks up and dials 91234567, it would
> be nice to transform it to 92061234567. i would prefer not
> to have an extension context per area code. it would be cool
>
Contact digium about the new g729 codec mark was asking people to
test..
Thanks for the help Brian :)
--
jeremy bogan[ [EMAIL PROTECTED] ]
segment publishing - design.develop.host
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Try following the instructions inside the bristuff package to use the
zaptel in there.
bristuff uses stable asterisk and the patches seem to be very dependent
on the date stamp of CVS the package downloads (due to CVS changing alot
I guess).
On Wed, 2004-05-19 at 10:05, Michael Devenijn wrote:
I'm trying to make a call from an IAXPhone client - through the * PBX
to an 888 number using the IAXTel link. I'm using the basic conf files
for extensions and iax. I get successfully connected (at the
"Attempting native bridge" line of the output) and am then able to talk
both ways for 20 to 3
Contact digium about the new g729 codec mark was asking people to test..
ftp://ftp.digium.com/pub/asterisk/g729/beta
You must still have a key but its linked ot your mac addresses :)
bkw
- Original Message -
From: "Kris Stark" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday,
Just as a better example of how to use accountcode for this ... maybe we
could add areacode= to sip.conf and pull that into a var instead of wasting
the accountcode for this hrm...
[1234567]
username=whatever
secret=whatever
context=phones
accountcode=920
then in extensions.conf
[outbound]
Dear all
I read on the list back in 2003 that * does not support IXJ LineJACK
dialout yet
is this still the case?
Thanks
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On Tue, 2004-05-18 at 23:46, Eric Wieling wrote:
> On Tue, 2004-05-18 at 23:26, Randy Bush wrote:
> > i have extensions in locations across a number of telco area codes.
> > when someone in seattle picks up and dials 91234567, it would be
> > nice to transform it to 92061234567. i would prefer not
Are you using the safe_asterisk script to start up? G729 requires a
tty, which the script provides - at least so I've read... I can get
mine to segfault every time if I start up using just the asterisk
command, safe_asterisk works every time...
Nothing seems to want to work:
/usr/sbin/safe_asteri
On Tue, 2004-05-18 at 22:55, Jeremy Bogan wrote:
> > You must register the codec in order to be able to use it.
> >> May 12 19:27:42 WARNING[16384]: codec_g729b.c:511 load_module:
> >> Unable to
> >> initialize va stuff: -1
> >> Segmentation fault
> >> alberspilnx8:/bin # Ouch ... error while wri
On Mon, 2004-05-17 at 19:46, Jorge Verastegui wrote:
> When i make a call from Asterisk everything goes Ok,
> I do have a problem: when a call from the PSTN originates, the extension
> in Asterisk hangs up and I only hear silence in the PSTN for
> approximately 60 seconds.
I mean instead,
On Tue, 2004-05-18 at 23:26, Randy Bush wrote:
> i have extensions in locations across a number of telco area codes.
> when someone in seattle picks up and dials 91234567, it would be
> nice to transform it to 92061234567. i would prefer not to have
> an extension context per area code. it would
Randy Bush wrote:
i have extensions in locations across a number of telco area codes.
when someone in seattle picks up and dials 91234567, it would be
nice to transform it to 92061234567. i would prefer not to have
an extension context per area code. it would be cool to be able
to set a variable
You could be kinky and use accountcode= for that purpose
then dial(blah/${ACCOUNTCODE}${EXTEN})
bkw
- Original Message -
From: "Randy Bush" <[EMAIL PROTECTED]>
To: "splatters" <[EMAIL PROTECTED]>
Sent: Tuesday, May 18, 2004 10:26 PM
Subject: [Asterisk-Users] want to set a var in sip.conf
The funny thing is that in my experience VoIP actually works quite well over
the Internet. I am Danish but live in Malaysia, so I do quite a lot of VoIP
calls between those two locations. That can't possibly get any worse on the
public Internet. There are an average of 25 hops between Malaysia a
I'm not saying not to use them but firewalls and VPN are not very voip
friendly. VPN adds latency and jitter and firewalls play hell with RTP
ports.
bkw
- Original Message -
From: "Ronald R. McDaniel" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, May 18, 2004 10:27 PM
Subje
i have extensions in locations across a number of telco area codes.
when someone in seattle picks up and dials 91234567, it would be
nice to transform it to 92061234567. i would prefer not to have
an extension context per area code. it would be cool to be able
to set a variable in the sip.conf bi
ACL's are no way near as secure as firewalls and VPNs. ACLs only look at
IP address and ports. Spoof the IP address and find out the port and you
can get in. I am not saying that this would be an easy task, it would be
pretty difficult to do under most situations. Typically we use ACLs along
w
It's a very small delay my avg from houston to tampa is about 70 ms over the
tunnel and about 40 with out the tunnel on a good day. The thing that gets
you is the lack of QOS over the Net so get some good pipes. This is using a
vpn 3005 and a pix 506 with 168 bit encryptions on a nail vpn. If you
I understand that softphone are the answer in fact I deploy a ton of the Ip
comm version every week. I am under contract with the phones so I can't
sell them and there no easy way out of the contract.
As for 79XX's I have several office that have them working over a VPN backed
in to our main of
I personally think firewalls are a stopgap measure for the real problem. A
firewall and VPN are not a fool proof method of protection. Fix the real
problem instead of hiding it. I usually dont use a real firewall but ACLs
and other similar methods to lock down where/who can access a box. As for
Dear list..
I am trying to use * to answer a call coming in from the PSTN port of a
line jack
I am using mode=fxo in phone.conf but the line just rings and gins
in mode.=dialtone it works fine on the POTS port
any ideas what I am doing wrong?
Thanks
___
Title: Message
Does the Cisco softphone work with SIP? The fact sheet
only talks about SKINNY.
Paul
Mahler [EMAIL PROTECTED]
Signate, LLC665 Third
StreetSuite 100San Francisco,
CA 94107-1901 Asterisk Services and
Training
> You know, I'm not so sure this is limited to chan_capi. I have two
> asterisk boxes running, with one connected to my PSTN gateway (also
> using Asterisk). 1.0 stable works fine with my Cisco phone. CVS head
> works if I comment out the offending lines. Without commenting them out,
> the cisco p
Well - I would assume that most Asterisk instances run on Linux boxes, so
even if put directly on a public IP address it's quite possible to protect
the machine and do various VPN setup's (including IPSec). Speaking of
which - anybody got experience with VoIP and IPSec? I've never really used
IPS
Doug,
I don't believe that it would be a good idea to leave the Asterisk box
unprotected (without any firewall). This would leave you wide open for
people to access your internal system through the Asterisk box. We have
all been participating in a discussion about an article written by the
ingen
You must register the codec in order to be able to use it.
May 12 19:27:42 WARNING[16384]: codec_g729b.c:511 load_module:
Unable to
initialize va stuff: -1
Segmentation fault
alberspilnx8:/bin # Ouch ... error while writing audio data: : Broken
pipe
Ouch ... error while writing audio data: : Bro
You know, I'm not so sure this is limited to chan_capi. I have two
asterisk boxes running, with one connected to my PSTN gateway (also
using Asterisk). 1.0 stable works fine with my Cisco phone. CVS head
works if I comment out the offending lines. Without commenting them out,
the cisco phones
I have it working on an industrial single board pc. :)
David Hickman
TSG Computer Consulting - Auctions
314-865-4752 x2
On May 18, 2004, at 8:42 PM, Jacques Leisy wrote:
Anybody running * on a compact PCI platform?
I got a few CPCI boards on eBay including a T1 Natural Microsystems AG4000?
Any
Title: Message
humm
now that I think about I don't think it's free sorry my
mistake
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
listsSent: Tuesday, May 18, 2004 9:27 PMTo:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Free
So
Title: Message
what
about cisco's Ip comunicator? it's free so is the old cisco soft
phone.
If you
don't have access to it let me know
Doug BlockChief Information Officer of Efast
Funding713-983-4055 (Direct)888-338-3863 x 4055 (Toll
Free)713-983-4555 (Direct Fax)
-Original
On Tue, 2004-05-18 at 20:45, Vic Cross wrote:
> Is there a CVS-web of the * tree? I don't know how to drive CVS to give
> changelogs etc... Again, if there's a way for me to find out how/what to
> change, I can give it a go.
http://www.google.com/search?hl=en&lr=&ie=UTF-8&q=site%3Alists.digi
It's avaialble at:
http://www.carrieraccess.com/support/products/index.cfm/fuseaction/default_p
rod/cat_id/21.htm
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
VoIP Systems, Training & Consulting
> -Original Message-
> From: [EMAIL PR
Does anyone have any recomendations for a free
Windows softphone, SIP or IAX that supports the following features:
* Message Waiting Indicator
* Consultative Transfers
* Speed Dials
On Tue, 18 May 2004, brian k. west wrote:
> The problem isn't with asterisk chan_capi will have to be updated to deal
> with the changes.
Only someone with knowledge of the internals of * would know that the RTP
timestamps generated by * on an outgoing SIP leg would be affected by the
incoming c
Anybody running * on
a compact PCI platform?
I got a few CPCI
boards on eBay including a T1 Natural Microsystems AG4000?
Any hope to ever get
* running on that platform?
Linux Suse 9.0 is
running fine
Thanks
Jacques
has anyone done caller announce in MeetMe's before?
Dave P
>>> [EMAIL PROTECTED] 5/18/2004 5:50:49 PM >>>
With multiple parking lots you can give each person their own lot thus
exten
800 for everyone will connect them with just their call passing the lot
name
which you know for X customer.
bkw
Also on a side note if Kapejod isn't wanting keep chan_capi up to date then
someone needs to ask him if he will disclaim it so digium can include it and
help maintain it.
bkw
- Original Message -
From: "brian k. west" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, May 18, 200
> I'd love to fix the problem, but no-one is listening!
>
> I did what you said, captured Ethereal traces, found that timestamps do
> not increment, found BLATANT errors in rtp.c where a signed int is being
> used to hold return values from an unsigned int function... and had my
> bug report throw
A bit about my experience with the TDM-04 FXO. Only saw a few post on
this subject, thought I would contribute a little about my experience to
save others the hassle.
a. As an earlier poster noted, the driver for the FXO is in the wcfxs
module. Perhaps it should be renamed to something less con
I am completely new to * ( I know read the archives but this is a little
different case)
I am trying to setup a Sip system out side my security firewalls for home
users. I currently run a Cisco avvid solution internally but it's highly
firwalled. I am planning on building a pri out of my 3745 cisc
On Tue, 18 May 2004, brian k. west wrote:
> Lets look at this and FIX the problem instead of hacking it. What you need
> to do is install etherreal and capture a call and parse the timestamp info
> to see if they are slipping. Because they are perfect here.
>
> bkw
I'd love to fix the problem,
Yes, call Carrier Access support, number available on their website. They will
set you up with a support login and you can download the various manuals in pdf
format. I would send you one directly, but they are very large. 6-13 meg if I
remember correctly.
They also offer free classes on the Adit.
> At least the X100Ps should maintain a fair amount of their value.
You're kidding me, right??
-A.
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Lets look at this and FIX the problem instead of hacking it. What you need
to do is install etherreal and capture a call and parse the timestamp info
to see if they are slipping. Because they are perfect here.
bkw
- Original Message -
From: "Brian Cuthie" <[EMAIL PROTECTED]>
To: <[EMAI
> compared to? My P4/Xeon 2.8 does SLINR -> iLBC in 12ms so a 2.4ghz
> should take 14 (?)
Recompiling all of asterisk and not just the format_ilbc.o gives me 14ms.
Regards,
Andrew
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- I'm not a Linux user i'm trying to get in it ...
- Fedora core 1
- QuadBRI card bought from junghanns.net
- We want to use the card in TE mode to connect to the TELCO
- Downloaded BRISTUFF0.0.2(stable) latest from junghanss.net/asterisk
- followed the instructions on voip-info.org
- compiled ever
I am trying to find a manual for the Carrier Access Adit 600. Does anyone
know where I might be able to find one?
Thanks
-Jon
--
Jon J. Brandon [EMAIL PROTECTED] http://www.monsoonretail.com
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i am trying to make an automated outbound call
playing a message upon answer but i am having trouble triggering the
message except with a timer (not paying attention to call progress tones or
indications)
have the call working via php agi, just not sure
how to deal with the call progress/i
compared to? My P4/Xeon 2.8 does SLINR -> iLBC in 12ms so a 2.4ghz
should take 14 (?)
Andrew Kohlsmith wrote:
-fPIC -O3 -march=pentium4 -funroll-loops -fomit-frame-pointer -msse
-msse2 -mfpmath=sse
Made no difference whatsoever on my P4/Xeon 2.4GHz for iLBC (I don't use
speex).
-A.
___
[EMAIL PROTECTED] wrote:
>> I installed the latest CVS of everything, and we've been getting
>> random hangups.
Bruce Komito wrote:
> I, too, have a TDM400P with FXO cards and I am having the
> same problem.
After further investigation, I thought that I had a bad module in the
#1 position on my
That's the correct behavior. AGI commands are always executed in series.
Are you trying to dial out from a MeetMe conference?
Recently, someone posted this improved Meetme
http://www.areski.net/asterisk-meetme/about.php
The other option would be to execute the dial then use manager API to
redirect
William Suffill wrote:
check the caller id in your incoming extension before you pass to to a
end user. Reset $calleridname to unavaliable if no number is given
Thanks a good suggestion... How would I implement this??? Any docs/web
pages/examples you could point me to?
Thanks.
__
Yes, I've tried with SendDTMF, and it works, but if I do that, then * looses
control of the call. That is, the call is transfered to the new extensions
on the PBX but since * is not in the calll flow anymore, it doesn't know if
on the other end they have ansered or not.
- Original Message ---
Iain,
This is a known issue with the Cisco phone and Asterisk having to do
with a change made later in the cvs tree. Try 1.0 stable, or modify
rtp.c to comment out the two lines as follows:
/* Re-calculate last TS */
rtp->lastts = rtp->lastts + ms * 8;
//
On Tue, 2004-05-18 at 17:42, Isamar Maia wrote:
> > I don't know what to tell you, other than to echo the statement that
> > you'll probably be better served by installing a 4 FXO TDM400P card, even
> > though that's gonna cost you another US$400. You might try asking here on
> > the list if anybo
With multiple parking lots you can give each person their own lot thus exten
800 for everyone will connect them with just their call passing the lot name
which you know for X customer.
bkw
- Original Message -
From: "Andrew Kohlsmith" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tue
> I don't know what to tell you, other than to echo the statement that
> you'll probably be better served by installing a 4 FXO TDM400P card, even
> though that's gonna cost you another US$400. You might try asking here on
> the list if anybody wants to buy some X100P boards...
>
Sullivan,
Thank
I'm trying to produce some enhancements to one of the applications,
and am trying to use ast_request("zap", format, "pseudo") to create
a new channel on /dev/zap/pseudo, which I can then bind to a zaptel
conference and play a stream to it.
I've been using as inspiration the Radio Repeater app, app
> You could use app_valetparking to do this since it allows multiple parking
> lots you could use some some logic to do exactly that.
How do the multiple parking lots facilitate this feature? I agree
app_valetparking will do most of what he wants, but I don't understand how
the mulitple lot par
share what ever you come up wif please :)
bkw
- Original Message -
From: "Gavin Hollinger" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, May 18, 2004 4:06 PM
Subject: Re: [Asterisk-Users] Flexible Call Parking Solution
> Wow! how did I miss this! Will try today!
>
>
> ---
Serge Oleinikov wrote:
I was trying to replace the header. But looks like header contains some kind
of CRC
The format the rings are at are after what I found out uLaw compressed
8bit 8000hz mono
samples. But they also have a header infront of the file. I will play
arround with it later. Maybe
there
Wow! how did I miss this! Will try today!
- Original Message -
From: "brian" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, April 30, 2004 4:10 PM
Subject: [Asterisk-Users] Flexible Call Parking Solution
> These all work with sip native transfers and sip attended transfers i
> "Randy" == Randy Bush <[EMAIL PROTECTED]> writes:
Randy> how does a call to sip://[EMAIL PROTECTED] come in to asterisk so i
Randy> can route it?
I beleive it comes in to extension user in the default context:
[default]
exten => user,1,whatever...
You'll need to look in the variable ${SIP
Philipp,
Can you use wildcards on the caller id? Eg certain area codes get a
certain ring type others get a different ring type?
Also I have 2 pstn lines coming into asterisk, wonder if I can make one
of them (packet8) ring a certain tone and the other pstn (telstra) ring
a different tone.
(or ev
Abandon OSX. Make your system dual boot with Gentoo or Yellowdog and
compile on Linux. It's just easier. If you have an old PII I'd use that
instead.
TL
--
Todd Lieberman
[EMAIL PROTECTED]
http://tlsolutions.net
p. 215-495-0030
f. 215-495-0031
-Original Message-
From: [EMAIL PROTECTED
Hi!
> caller ID already tells you who is calling). However, I can put anything
> I want into the text boxes and nothing happens - I always get the
> "system ring tone".
No problem here, works just fine. If calls I get ringtone 2,
otherwise the default ringtone.
> o System Ring Tone
>
Where do I get app_valetparking and app_dial2?
They both sound like great apps!
Thanks
> Let me see if I can whip up something using app_valetparking and post it
> to
> the list that will do exactly this.
>
> bkw
>> I had to write my own Dial2 application to do this, which is a copy of
>> the
>
Hi
Running slackware 9.1 with compiled kernel patched to 2.6.6 running ok.
Having problems compiling zaphfc.c
First thing was the can't find "irq_vectors.h",
but "solved" by changing the #include "irq_vectors.h" to #include
"mach-default/irq_vectors.h" in /usr/src/linux/include/asm/ directory.
But
On Tue, 2004-05-18 at 15:45, Dan Fernandez wrote:
> Steve,
>
> Thanks for your respnose. The flash does seem to work. If I plug a phone on
> the x100p I can hear with the x100p flashes. I then get a dialtone. The
> problem is that when i try to dial again from that card, i get "cannot
> create zap
I, too, have a TDM400P with FXO cards and I am having the same problem.
However, as near as I can tell, the driver is already compiled with
AGGRESSIVE_SUPPRESSOR turned off. I was thinking the calls were dropped
from the sip side, but I haven't been able to confirm that. All
sip-to-sip calls, how
Thanks for the response.
Have you try the new TDM FXO cards? Does call progress work with those?
- Original Message -
From: "Vic Cross" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, May 13, 2004 5:46 AM
Subject: Re: [Asterisk-Users] problems with analog interface to PBX
Let me see if I can whip up something using app_valetparking and post it to
the list that will do exactly this.
bkw
- Original Message -
From: "Carlton J. O'Riley" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, May 18, 2004 1:47 PM
Subject: RE: [Asterisk-Users] call announce?
You could use app_valetparking to do this since it allows multiple parking
lots you could use some some logic to do exactly that.
bkw
- Original Message -
From: "Carlton J. O'Riley" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, May 18, 2004 1:47 PM
Subject: RE: [Asterisk-Use
Steve,
Thanks for your respnose. The flash does seem to work. If I plug a phone on
the x100p I can hear with the x100p flashes. I then get a dialtone. The
problem is that when i try to dial again from that card, i get "cannot
create zap channel". It seems that because the line is now off hook, the
if the dns has
_sip._tcp.my.dom. SRV 0 0 5060 asterisk.dom.ain.
_sip._udp.my.dom. SRV 0 0 5060 asterisk.dom.ain.
where asterisk.dom.ain. is an A RR of the asterisk pbx.
how does a call to sip://[EMAIL PROTECTED] come in to asterisk
so i can route it?
do i just put in sip.conf
Stephen R. Besch wrote:
Thomas Gallaway wrote:
Brian Capouch wrote:
Thomas Gallaway wrote:
My ringtones just work on all the grandstream's :-)
Do the "URLS" for the ringtones at the top show up as something
other than all zeroes?
I've fiddled with this until blue in the face, and the ring soun
Stephen R. Besch wrote:
Thomas Galloway wrote:
Stephen R. Besch wrote:
Duane wrote:
Grandstream v1.0.4.68 firmware
Am I missing something obvious about the new ringtone feature? The
4.68 firmware updates as usual from my TFTP server, the new version
shows up in the phone's web page, but the ring
Hello,
I have researched a few postings where users mentioned being able to
install Asterisk on Mac OS X Panther by adding some code after line 165 in
the Makefile and then compiling.
This has been unsuccessful for me.
I downloaded the asterisk-0.9.0.tar.gz tarball and am trying to install
from
I had to write my own Dial2 application to do this, which is a copy of the
app_dial.c source with this feature added. I didn't have it record the
incoming caller's name, but rather prompt the answering user as to whether
or not to accept the call. It would be trivial using extension logic to
have
I have about 5 snom 200 phones working fine with everything. Voicemail,
Transfers and all. Except I can't seem to use them to pickup parked calls
nor place a call on park. I also have sipura-2000 with analog phones that
are able to pickup parked calls and to park them. Most of them are on
firmware
On Tue, 2004-05-18 at 14:11, Kyle Hagan wrote:
> I use the Zultys 4x4 and it will allow me to announce before it
> transfers. But he GS BT-100 just transfers it right away.
Supervised aka consultative transfer is where you get to talk to the
person you are transfering to before you complete the t
check the caller id in your incoming extension before you pass to to a
end user. Reset $calleridname to unavaliable if no number is given
On Tue, 2004-05-18 at 15:18, Roger wrote:
> I have a question - if a user calls up w/ blocked caller id I get the
> following on my phone
>
> Incoming call fro
> -fPIC -O3 -march=pentium4 -funroll-loops -fomit-frame-pointer -msse
> -msse2 -mfpmath=sse
Made no difference whatsoever on my P4/Xeon 2.4GHz for iLBC (I don't use
speex).
-A.
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Thomas Gallaway wrote:
Brian Capouch wrote:
Thomas Gallaway wrote:
My ringtones just work on all the grandstream's :-)
Do the "URLS" for the ringtones at the top show up as something other
than all zeroes?
I've fiddled with this until blue in the face, and the ring sounds
just like the ring it
I've just noticed a strange behaviour with a MeetMe conference.
I have a pair of phones (GS BT102) on my desk, and dialled both of them
into a conference on speakerphone. If I spoke or made a sound, I heard
it replayed from both speakers together a split second later, as
expected.
I went away for
> Yep, you can get one of those (MVP810), refurbished for $2K. So for the
> 24 ports you need, that'll be $6K + a four-port hub.
> Or you could get 12 Sipura SPA-2000 for $100/each, a 16-Port Switch (get
> something with address-cache, like the LinkSys 4116) for $100, and a few
> power-strips. T
Thomas Galloway wrote:
Stephen R. Besch wrote:
Duane wrote:
Grandstream v1.0.4.68 firmware
Am I missing something obvious about the new ringtone feature? The
4.68 firmware updates as usual from my TFTP server, the new version
shows up in the phone's web page, but the ring tones, while present on
Thomas Gallaway wrote:
Jeremy McNamara wrote:
Stephen R. Besch wrote:
Duane wrote:
Grandstream v1.0.4.68 firmware
Am I missing something obvious about the new ringtone feature? The
4.68 firmware updates as usual from my TFTP server, the new version
shows up in the phone's web page, but the ring t
Hi, I had been using 4 X100P cards in my Asterisk box, but 2 of them
were sharing an interrupt. Therefore, periodically I would hear beeps
and clicks that I had assumed were a result of this. So, I ordered a
TDM400P with 4 FXO modules and installed it in the box last night.
Today, we've had noth
I have a question - if a user calls up w/ blocked caller id I get the
following on my phone
Incoming call from asterisk
This is the same on my Cisco 7940s and Polycom phones. For average
users this is not intuitive at all..
I'd like to configure this so if I deploy this at a customer site it
I use the Zultys 4x4 and it will allow me to announce before it
transfers. But he GS BT-100 just transfers it right away.
Kyle
brian k. west wrote:
No way to do that without writing your own custom application to do it.
bkw
- Original Message -
From: "Gavin Hollinger" <[EMAIL PROTECTED]
Jeremy McNamara wrote:
Stephen R. Besch wrote:
Duane wrote:
Grandstream v1.0.4.68 firmware
Am I missing something obvious about the new ringtone feature? The
4.68 firmware updates as usual from my TFTP server, the new version
shows up in the phone's web page, but the ring tones, while present on
Hey Brian,
See if you can do 7960 - SIP - * - IAX - * - SIP - GW. That will get you the
evidence you need to hear.
Quoting brian <[EMAIL PROTECTED]>:
> Strange I do 7960 => * => IAX all day long without one jitter or any bad
> audio. Now if both ends are NOT running the very latest(within the
Brian Capouch wrote:
Thomas Gallaway wrote:
My ringtones just work on all the grandstream's :-)
Do the "URLS" for the ringtones at the top show up as something other
than all zeroes?
I've fiddled with this until blue in the face, and the ring sounds
just like the ring it had before.
This is w
> I've just had the most appalling performance from * ever. Dialling:
>
> Cisco 7960 => asterisk => IAX
>
> produces sound drop outs so extreme that the call is useless. I noted this
> in an earlier post. Dialling:
>
> Cisco ATA186 => asterisk => IAX
>
> is fine.
>
> Frankly, I think this
Yep, you can get one of those (MVP810), refurbished for $2K. So for the
24 ports you need, that'll be $6K + a four-port hub.
Or you could get 12 Sipura SPA-2000 for $100/each, a 16-Port Switch (get
something with address-cache, like the LinkSys 4116) for $100, and a few
power-strips. Total is $1
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