Am 09.04.2010 13:10, schrieb Vieri:
> Please listen to the following sound file:
I've experienced similar (well, vaguely similar) distortion on a
"horstbox pro" when echo cancellation is switched on for the zap
channels (ISDN).
Turning it off resulted in no distortion at all, but then i experien
call an IAX User directly and he puts me on hold I hear the [signal]
sound files and not the [default] ones, but why?
Any pointers please ?
Thanks - Oliver
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Michiel van Baak schrieb:
> Or use the provided chan_skinny
I tested chan_skinny some time ago, I remember it didn't work as
expected, just can't remember what it was. Perhaps I should also test it
again.
Thanks!
-- o
--
_
-
> Olivier schrieb:
> Thanks for the suggestion anyway, I'm going to test this just out of
> curiosity :-)
And that's what i get in the CLI:
> Got SIP response 501 "Not Implemented" back from XXX.XXX.XXX.XXX
Well, I guess I should really give chan_sccp another shot ...
Thanks anyway!
-- o
--
Olivier schrieb:
> 2010/2/14 Oliver Nittka
>
>> - dialplan app SendText() only works on a connected channel, AFAIK, and
>> I'm not sure if the 7940 with SIP firmware honours it at all.
>>
>> Have you tried to call the phone with auto-answer ?
>
I'm
Hello all,
this may be slightly offtopic :-)
I have some Cisco 7940 phones with SIP firmware, connected to an
Asterisk 1.2.18-BRIstuffed-0.3.0-PRE-1y-g (HorstBox Pro with custom
extensions.conf).
On some of the phones, two lines are configured, one for business, one
for private calls.
When forwa
Magnus Benngård schrieb:
> But when I dial 0317998985, the phone rings but no "reaction" fron ABCTI
> so
> I do understand that I have missed something... :( but what?
Your setup looks OK to me.
One thing: did you configure a hint in extensions.conf?
Should be in the context where your extension
Hallo,
ABCTI (an open-source CTI client for Asterisk) has moved to beta stage.
Find it on:
http://abcti.sourceforge.net
For the first time, we now have windows installers that actually work ;-)
We would appreciate any feedback you can give.
Regards,
-- o
_
Hoggins! schrieb:
> Hello,
>
> Well, I promise I will try it, as I've never been able to find such a
> software compatible with Asterisk > 1.4.
> Maybe I'm blind too.
That's exactly why i started this project. To my knowledge, there's no
free and easy to setup CTI client for Asterisk. Well, i hav
Tzafrir Cohen schrieb:
> What is the required configuration in Asterisk?
Well, good you mention it, i forgot to explain that in the README!
you just need to configure a user on the Asterisk server in manager.conf
like this:
[oly]
secret = XXX
permit = 192.168.178.0/255.255.255.0
read = call,or
Hallo everybody,
today i've released the very first (and very alpha :-) version of ABCTI,
a CTI solution
for Asterisk written in python, using pyGTK and py-Asterisk. It talks
directly to the
Manager API, so no additional software is required on the Asterisk server.
You can find it on Sourceforge
.conf
[channels]
language=de
group=1
signalling=pri_net
switchtype=euroisdn
context=default
;rxgain=-4.0
;txgain=-4.0
channel => 1-15,17-31
pridialplan=unknown
echocancel=yes
What can I do?
Thanks, Oliver
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Christian Victor wrote:
> 2009/3/30 Peer Oliver Schmidt <mailto:po...@theinternet.de>>
>
> The Horst-Box Professional has a lot of problems in the ADSL area
> (like stopping transfers after a dozen or so megabytes for example),
> and I have had lots of need
ons, and always made sure to follow
the instructions to the letter, with regards to configuration reset.
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aster
& ast_strlen_zero(o->pdu))) {
Try reverting that line, and see if that helps with your problem. And
maybe someone with a better understanding of C can take a look at the
above problem.
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PGP Key ID: 0x83E1C2EA
d: Failed to
> scan service '/var/spool/asterisk/outgoing/astup.call'
Do you by chance use bristuff?
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Apparently, there is a SIP(diversionheader) field that fixes the problem
below, but I cannot find any docs or examples of how to use it in my
dialplan. Any help would be appreciated. We have a Cisco CallManager
where users forward their numbers, so PSTN->PSTN calls get this error...
-Greg
<--
Apparently, there is a SIP(diversionheader) field that fixes the problem
below, but I cannot find any docs or examples of how to use it in my
dialplan. Any help would be appreciated. We have a Cisco CallManager
where users forward their numbers, so PSTN->PSTN calls get this error...
-Greg
<--
On Mon, 2008-03-31 at 23:07 +0100, Razza wrote:
> On 31/03/2008, J. Oquendo <[EMAIL PROTECTED]> wrote:
> YMMV Change to reflect your firmware (e.g. P003-07-4-xx)
>
> 8< SNIP >8
>
> I removed the following lines:
> P003-07-4-00
> P003-07-4-00
>
> And tried both of these
On Mon, 2008-03-24 at 19:52 +0100, Philipp Kempgen wrote:
> mark morreny schrieb:
>
> > I am working on deploying voip for my company and would like to seek some
> > advice on the number of E1 lines we need to rent.
>
> E1 is not VoIP. :-)
It is if provisioned for 30 channels of concattenated da
On Feb 7, 2008, at 2:07 PM, Tzafrir Cohen <[EMAIL PROTECTED]>
wrote:
> On Thu, Feb 07, 2008 at 07:53:12PM +, Ben Willcox wrote:
>> Olivier wrote:
>>> At the opposite, I think it could be useful for an Asterisk server
>>> to
>>> act as XMPP User Activity provider (ie update XEP-0108 field
ct for
7960/1 and 7971. When I get back home, I will login to the asterisk
servers and tell you what IPs the registration requests have in them.
> ----
> From : Greg Oliver <[EMAIL PROTECTED]>
> To : Asterisk Users Mailing List - Non-
On Feb 2, 2008, at 2:11 PM, John Von Essen <[EMAIL PROTECTED]> wrote:
> I posted an email a few days regarding a problem with hearing the
> voicemail greeting on my sip phones.
>
> It turns out to be a phone/stun/linksys issue - not an asterisk issue.
> Which brings up a couple of questions
You need $dnis.
On Jan 30, 2008, at 11:08 PM, "Prashant Sharma"
<[EMAIL PROTECTED]> wrote:
Hi,
I am new to asterisk configuration.
I want to get called number in features.conf.
I am defining a feature in features.conf and that feature got
executed on pressing a particular DTMF key sequenc
Cisco routers with DSPs as ip2ip gw will do it if you want to spend a
few bucks
On Jan 29, 2008, at 2:36 PM, "Khaled Chehab" <[EMAIL PROTECTED]>
wrote:
> Dears
>
> Any one knows a standalone voip transcoder software name,not an ip
> pbx.
> What I want is to transcode the incoming sip c
On Mon, 2007-12-10 at 17:58 -0800, Robert McNaught wrote:
> Hi
>
> Does anyone have any recommendations of an SMS gateway which you can
> just sign up for on a pay-as-you-go basis for testing, for use with
> Asterisk?
>
> Thanks
>
> Robert McNaught
>
In and Out Bound SMS from *, or just * -> S
On Thu, 2007-12-06 at 10:32 -0500, John Bittner wrote:
> The fix for this is not to use the normal Cisco IOS. Must use 12.4T
> version. It is a Cisco bug.
I would suggest jumping to greater than 12.4.11T as they introduced all
kinds of DTMF fixes there as well..
> -Original Message-
> On
On Thu, 2007-11-15 at 07:31 +0100, Olivier wrote:
>
>
> 2007/11/14, Greg Oliver <[EMAIL PROTECTED]>:
> On Tue, 2007-11-13 at 08:44 -0500, Anciso, Roy wrote:
> > Hello List,
> >
> > Does anyone have access to the soft key c
On Thu, 2007-11-15 at 05:34 +0100, Patrick wrote:
> On Wed, 2007-11-14 at 09:06 -0500, Anciso, Roy wrote:
> > The Cisco Documentation states that you can modify standard and
> > nonstandard softkey templates. They may not be xml files. I just
> > assumed they were xml since that is what is used t
On Tue, 2007-11-13 at 08:44 -0500, Anciso, Roy wrote:
> Hello List,
>
> Does anyone have access to the soft key configuration files for the
> Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and
> didn’t find much up there.
>
> Thanks
>
Softkeys running both SCCP and SIP firmwa
gards
Peer Oliver Schmidt
PGP Key ID: 0x83E1C2EA
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You first have to uncompress them. Even if both audio
streams use the same codec, they are compressed thus have to be
uncompressed for the mixing of the audio to happen.
Better?
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i4hylafax.
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e 40, with the 80 about nearly
twice the price of the 60.
Hope that helps.
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On Thu, 2007-06-28 at 14:52 +0200, Olivier wrote:
>
> 2007/6/27, Greg Oliver <[EMAIL PROTECTED]>:
> On Tue, 2007-06-26 at 21:45 +0200, Olivier wrote:
> > Hi,
> >
> > Has anyone met any success, installing localized (ie
>
On Wed, 2007-06-27 at 14:32 -0600, Stephen Bosch wrote:
> Hi, folks:
>
> Thoughts? Who here has used BRI in North America? And when you did, what
> interface hardware did you use?
>
> -Stephen-
>
>
I grew up on BRI when the internet first started taking off here. All
terminated into Asce
On Tue, 2007-06-26 at 21:45 +0200, Olivier wrote:
> Hi,
>
> Has anyone met any success, installing localized (ie non-english)
> menus within SIP firmware enabled Cisco 7941 ?
>
> Those phones seem to be trying to download localized menus from Cisco
> Call Manager but as they are managed by an Ast
On Fri, 2007-06-01 at 21:28 +0200, Pavel Jezek wrote:
> we are using 7941 with sip v8.2(2)SR3, it working quite well ;-)
>
>
> Eric Lubow wrote:
> > All,
> >
> >I am having a lot of trouble with the Cisco 7961G phones. I have
> > managed to get them up and running with Asterisk to the point
On Fri, 2007-06-01 at 10:18 +0530, Vamsi Pottangi wrote:
> Hi Greg,
>
> Narrowed the problem ot be that of codec mismatch ;-) Damn
> CCM, doesn't provide proper debugs.
>
> I have another query with CCM and Asterisk integration. In CCM cluster
> Phones register to 1st CCM and they fallback to 2n
On Wed, 2007-05-30 at 18:03 -0500, Eric "ManxPower" Wieling wrote:
> David Boyd wrote:
> > Does that mean that even when dynamic dns entries exist and the time
> to
> > live is set to 15 minutes asterisk will continue to try using the
> old
> > expired results?
I can also say that my experience i
On Wed, 2007-05-23 at 19:53 +0530, Vamsi Pottangi wrote:
> Hi,
>
> I was able to work out SIP trunk between Asterisk and CCM 4.x without
> any issues. Whereas SIP trunk in CCM 5.x is not working with Asterisk.
> Asterisk is sending OPTIONS messages to CCM 5.x for which CCM is not
> replying. For t
On Wed, 2007-05-16 at 23:19 +0100, Robert Lister wrote:
> I was wondering if it is possible (in 1.2.x) to get the SIP response code
> back after doing Dial().
>
> Dial() seems to treat most call-setup problems as dialstatus CONGESTION, and
> some are NOANSWER, but I want to know the SIP response
On Wed, 2007-05-16 at 13:57 -0500, Bruce Reeves wrote:
> How sad, cnet misspelled Polycom and Cisco didn't make the cut.
Yeah, Cisco and MSoft are on BAD terms since the inking of the deal with
Nortel.. MSoft got mad when they moved from Windows Server to Linux for
their CallManager platform, and
On Sun, 2007-05-13 at 20:54 +0300, Dovid B wrote:
> I am actually getting DTMF over SIP when people call in to a clients system
> that is running a2billing. They are using RFC2833.
>
If you are using a Cisco router anywhere in the loop, there is a known
bug that causes rfc2833 and inband signall
On Fri, 2007-05-11 at 18:44 -0400, Jon Pounder wrote:
> > On 5/11/07, Alex Balashov <[EMAIL PROTECTED]> wrote:
> >> On Fri, 11 May 2007, C F said something to this effect:
> >>
> >> > Not according to Verizon (in my area anyhow), We tried it and it
> >> didn't
> >> > work. The verizon technician in
ompile asterisk each time such an issue comes
up. So it would be great, if it was possible to only compile the ilbc
codec, place it in the folders with the other codecs and have the
distribution take care of the rest. Any suggestions on how to do that?
Thanks ag
lbc to ulaw? How do I do that?
Thank you very much again!
Oliver
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When calling from one phone to the other I get the following message:
chan_sip.c:2841 sip_call: No audio format found to offer. Cancelling
call to phone2
Thank you very much again!
Oliver
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or:
"Unable to find a codec translation path from ilbc to ulaw"
Setup SIP-phone:
disallow=all
allow=ilbc
Setup PSTN-Gateway:
disallow=all
allow=ulaw
I've googled for overn an houre. But no luck. So I'd really apreciate
any help!
Thanks!
Oliver
ds
Peer Oliver Schmidt
PGP Key ID: 0x83E1C2EA
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oblem.
Thanks again, and have a good night.
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= 0x1fff03ff
Apr 3 17:23:17 server42 kernel: [263330.900720] CIPmask2
= 0x0
Apr 3 17:23:17 server42 kernel: [263330.900726] CallingPartyNumber
= default
Apr 3 17:23:17 server42 kernel: [263330.900733]
CallingPartySubaddress = default
Apr 3 17:23:17 server42 kerne
kcapi: put [0x1] id#1
LISTEN_REQ len=26
Apr 3 15:06:52 server42 kernel: [255150.690252] kcapi: appl 1 down
At 15:06:35 the loading stopped.
Is this helpful, or do you need a higher verbosity?
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[size=32]
I/O ports at dcc0 [size=32]
[EMAIL PROTECTED]:~# capiinit status
1 fcpci running fcpci-dcc0-10A1 3.11-07 0xdcc0 10
Driver from ubuntu edgy
> A debug log (capi trace) from the driver or kernelcapi helps to see
> what messages are wrong/missing.
What is the best
ch, but it did not work. It waits quite a long time
before the chan-capi error message comes up, according to the time
stamp it is about 12 seconds. It is kind of strange, that the whole
startup process for asterisk usually takes only about 4-5 seconds.
D
/var/log/asterisk/messages
[Mar 31 16:17:03] ERROR[3850] chan_capi.c: Unable to listen on contr1
(error=0x100f)
Is this helpful, or do you need more information?
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0, i.e.
just asterisk -c chan-capi does not find the controller. Starting
asterisk with verbosity turned up, most of the time the controller is
found.
Might this be a problem of missing CPU power (PII-400)?
Any and all help is greatly appreciated.
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Please contact me on [EMAIL PROTECTED]
Thanks,
Oliver Vermeulen
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On Fri, 2006-10-13 at 13:08 -0500, Jessee J Holmes wrote:
> Actually, come to think of it, I don't know who will support it. Does
> Asterisk support G.722? From what I know it doesn't, is it included in
> the 1.4 beta? Will they support it? If Asterisk doesn't support it,
> then the phone won't do
On Fri, 2006-10-13 at 11:53 -0500, Tim Connolly wrote:
>
> Does anyone know what triggers the 7970 to update its config? I
> was able to get it to update to SIP, but the config I used initially
> won't go away. I am making small changes to the SEPxxx.cnf.xml file and
> rebooting the phone,
On Tue, 2006-10-10 at 15:16 -0700, Alyed Tzompa wrote:
> What I want is to transfer some calls to a Cisco extension, so think I don't
> need to do the upgrade to CM5.
>
> I'm I right?
>
> Alyed
Yes - you are right. On your CCM, go to a phone and check the CSS of
the device and the partition o
When you do a factory reset on a 41/61/70/71, it actually deletes ALL of
the firmware except the bootloader from the phone. You would have to
have all of the 70s firmware files that come with them in order to boot
them. The term70.default.loads tells the phone what version of software
to tftp. D
On Fri, 2006-09-29 at 20:26 -0700, Dan Austin wrote:
> Greg wrote:
> > 4.1.3 supports SIP trunks - I would HIGHLY recommend you move to that.
> > Anything over 4.0 supports SIP trunking.
>
> While it is true that CCM 4.0 and up supports SIP trunking, it is not
> all rainbows an butterflies. The 4
4.1.3 supports SIP trunks - I would HIGHLY recommend you move to that.
Anything over 4.0 supports SIP trunking.
-Greg
On Thu, 2006-09-28 at 19:32 +0200, Yusuf wrote:
> Hi,
>
> I recently had to hook up to Cisco Call Manager 4.1.3, and it only
> supports H323. SO I used ooh323, and a strange thi
On Thu, 2006-09-28 at 07:54 -0500, Tom wrote:
> At 05:39 AM 9/28/2006, you wrote:
> >Any pros / cons on getting one over the other ? I was wondering what
> >the main differences were.
>
> New phones (7941) support 802.3af POE. Old phones only Cisco special
> POE. New phones don't work with old
found any phone with bigger buttons. And each time the phone
rings, it lights up a dark room.
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out some old Compaqs had weird problems, which went away by going to a
new PC. Old Dell is working fine.
HTH
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quite some time. Do the bristuff
thing, and install the CHAN_CAPI-CM driver afterwards.
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JAMAICA DID'S - 1-876
NOW ACTIVE ON www.didx.org
Oliver Vermeulen
World Venture Group Telecom
Corporate Address:
147 New Haven Point Lane
West Palm Beach , FL , Miami
USA DID: +1 (305)722-1457
BE DID: +(32)9-395-5620
UK DID: +(44)870-478-8896
SIP : [EMAIL PROTECTED] <mailt
JAMAICA DID'S -
1-876
NOW ACTIVE ON www.didx.org
Cheers,
Oliver
VermeulenWorld Venture Group
Telecom
Office: +(40)21-569-4700Office2:
+(40)31-860-0030Fax:
+(40)31-860-0031USA DID: +1 (305)722-1457BE DID: +(32)9-395-5620UK
DID: +(44)870-478-8896msn:
[EMAIL PROTECTED
Hi
List,
We have new DID's
aviable for the following countrys :
- Romania Bucharest
40-21+
- Jamaica
1-876+
Go to www.didx.org
Or contact me of the
list.
Cheers,
Oliver
VermeulenWorld Venture Group
Telecom
Corporate
Address:Str Avionului
Nr 35/bl16J/3Buch
Dose anybody know how to put skype behind a usa proxy ?
Thanks,
O
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Hi
List,
I'm looking for a
Asterisk radius module ... Anybody has one ?
Thanks,
Oliver
Oliver
VermeulenWorld Venture Group
Telecom
Corporate
Address:Str Avionului
Nr 35/bl16J/3Bucharest, 014333 RomaniaOffice: +(40)21-569-4700Office2:
+(40)31-860-0030Fax:
+(40)3
erence. Unity is
> currently acting as our IVR. Would that make any difference?
>
> Thanks.
>
> On 5/24/06, Greg Oliver <[EMAIL PROTECTED]> wrote:
> On Wed, 2006-05-24 at 07:30 -0700, Gary Richardson wrote:
> > On the route pattern configuration
on
> > Subject: Re: [Asterisk-Users] CallerID
> >
> > You should set the presentation flags to private.
> > http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+CallingPres
> >
> > On 5/23/06, Greg Oliver <[EMAIL PROTECTED]> wrote:
> > > I am
ager.
>
> On the gateway configuration page, there is a 'Calling Party
> Selection' box. Changing the values in that drop down does not have
> any affect on the callerid.
>
> Thanks.
>
> On 5/23/06, Greg Oliver <[EMAIL PROTECTED]> wrote:
>
On Tue, 2006-05-23 at 10:46 -0700, Gary Richardson wrote:
> Hey guys,
>
> When a call comes in via the PSTN to our Call Manager 3.2 and is
> forwarded (via unity and H323), the caller id is set to our Unity
> Voicemail instead of the caller id from the PSTN. We're using the
> oh323 channel in this
On Tue, 2006-05-23 at 06:27 -0400, Steve Totaro wrote:
> Greg Oliver wrote:
> > I am trying to set CIDNum to nothing, but my outgoing PRI controlled by
> > another PBX seems to fill in something when asterisk does not.. If I
> > set a number either in the sip channel f
On Tue, 2006-05-23 at 09:32 +0100, Mark Ackroyd wrote:
> Here in the UK on pri, setting the callerid to 0, withholds it.
>
> > I am trying to set CIDNum to nothing, but my outgoing PRI controlled by
> > another PBX seems to fill in something when asterisk does not.. If I
> > set a number either i
I am trying to set CIDNum to nothing, but my outgoing PRI controlled by
another PBX seems to fill in something when asterisk does not.. If I
set a number either in the sip channel for the phone, or from
extensions.con, it is realized.. If I try to leave them blank, or even
Not Defined, the main n
On Mon, 2006-05-22 at 12:16 -0400, Greg Boehnlein wrote:
> Hello,
> I was wondering if anyone out there is successfully running
> Asterisk 1.2 svn w/ Centos 4.3. I had an experience over the last two
> weeks that has me scratching my head and muttering strange things in the
> wee hours of
On Sun, 2006-05-21 at 14:28 +0200, Olivier Krief wrote:
> Hi,
>
> I can't upgrade an old 7960 from SCCP 3.0 to SIP 7.5. Could you
> help ?
>
> From
> http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml#sccptosip2,
> I got the following:
> 1. Copy the de
On Mon, 2006-05-15 at 17:40 -0300, Gustavo Souza Queiroz wrote:
>
> Hello,
> I´m have a CCM 3.3 and Asterisk in my LAN.
> I need connect my Asterisk in my CCM 3.3.
> You can a help me?
I hate to say it, but your best bet is to upgrade to CCm 4.0 and use
SIP.. It is a free cisco upgrade assum
René Enskat [Teamware GmbH] wrote:
> Somebody know if the AVM C4 Quad ISDN card is supported by the current
> asterisk version?
It is supported.
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___
--Bandwidth and Colocation provi
On Sun, 2006-04-30 at 11:53 +0800, Ronald Wiplinger wrote:
> One of my user is praising Skype!!!
>
> I cannot figure out anymore what I can improve!
>
> This users sip show peers is jumping from 65 msec to 1800 all the time.
> Of course his voice quality is like a morse code with dashes or dots
Try to upgrade asterisk to version 1.2.4
Are you using OH323 or H323 ?
I had same problem with 1.2.1 using H323(addon) , Installed 1.2.4 and OH323
and everything worked fine.
Cheers,
Oliver
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alejandro
Dear List,
We have Romania Bucharest DID’s available with area
code 4021 and 4031
For more information go to www.didx.org
Best Regards,
Oliver
Vermeulen
World Venture
Group Telecom
Tech
/ Admin
Corporate Address:
Str Avionului Nr 35/bl16J/3
Bucharest, 014333
On Mon, 2006-04-10 at 22:42 -0400, Andres wrote:
> Steven wrote:
> You heard wrong. We have multiple PRIs from XO and they DO NOT send
> caller name. We have discussed the issue with them on several
> ocassions. The sales people will say whatever they want, but the tech
> people who actually
On Thu, 2006-04-06 at 18:57 -0700, Jay Wilton wrote:
> Hello people,
> 370 Watts maximum output / 9.6 Watts/phone = 38 phones
> Does this logic hold water or change with line loss?
>
> Thank you,
> JJW
>
All I can say is that if you "oversubscribe" POE devices to a cisco
switch, they have the te
OK - I know this is expected behavior, but I am stuck.
Transferring calls from the * IVR to another SIP PBX ringing multiple
extensions simultaneously with call-forwarding set on a phone obviously
goes directly to the forwarded # since that phone answers first.
I need a way to make it where if it
On Wed, 2006-04-05 at 17:54 -0400, Jimmy Smith wrote:
> does one know how to program so i can have 2 lines on one sip account
> on that phone ?
>
> im runnign my own asterisk
>
> do i need 2 local accounts ? one for each line ? that rebounds to same
> SIP forp VOIP provider ?
Yes.
___
On Mon, 2006-04-03 at 13:59 -0500, Doug wrote:
> At 22:16 3/30/2006, Bill Gibbs wrote:
> >Use the codec command in your dial-peer. Or a voice-class so you can
> >have multiple supported codecs.
>
> Thanks, Bill.
>
> Could you please give an example of a voice-class
> entry in the dial-peer file
Hi All,
We are offering Romania Bucharest DID’s
NXX : 4021+ and 4031+
We have plenty available on http://www.didx.org/
Regards,
Oliver
Vermeulen
World Venture
Group Telecom
Tech
/ Admin
Corporate Address:
Str Avionului Nr 35/bl16J/3
Bucharest, 014333
On Tue, 2006-03-28 at 13:08 -0500, Technical Support wrote:
> You can't reliably run a real-time application (like asterisk) on a
> virtual machine. You will get better performance from an old PC than
> a VM on a new top-end PC. Sorry
>
> MD
H, I would have to say a properly configured GSX
Bob McDowell wrote:
> Can you chain these to get more that 42 buttons? I need about 60...
>
>
> Bob McDowell
42+12 is fairly near the 60 target.
--
Best regards
Peer Oliver Schmidt
PGP Key ID: 0x83E1C2EA
___
--Bandwidth and Colocat
On Fri, 2006-03-24 at 13:07 +1100, James Harper wrote:
> I haven't done any sort of research, but I've been told that GSM+DECT
> phones are available, and while having them seamlessly switch network
> types during a call probably isn't possible, they can function as a
> cordless handset.
>
> Can
On Thu, 2006-03-23 at 02:17 +, john wrote:
> Hi,
> Does anyone know how to define speeddials in XML for the 7970 sip
> firmware?. I've played with the SEP.cnf.xml file that was posted
> previously but can't find a way to do it. I can define them on the
> phone usually (seems a bit buggy) but
On Wed, 2006-03-22 at 11:52 +0100, Paul Brown wrote:
> Hi,
>
> I couldn't find the 7970 SIP image on the cisco.com site. Is it hidden :-)
>
> Any pointers would be appreciated
http://www.cisco.com/cgi-bin/Software/Tablebuild/doftp.pl?ftpfile=cisco/voice/ip-7900ser/cmterm-7970_7971-sip.8-0-2-0.co
On Wed, 2006-03-22 at 09:22 -0500, Ron Joffe wrote:
> On Wednesday 22 March 2006 00:33, Nathan Alberti wrote:
> >
> > Here is a dump of the configuration options, you will see there is a
> > few new, these are also documented on the wiki.
> >
> Nathan,
>
> How did you go about obtaining the dump ?
Hi All,
Anybody knows how to terminated calls using Grandstream
Ht488 and the FXO port ?
I can ring the FXO port fine , rings 1once then give me dial
tone.
Thanks,
Oliver
Vermeulen
World Venture
Group Telecom
Tech
/ Admin
Corporate Address:
Str Avionului Nr
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