Hi,
- Original Message -
From: Juan J. Sierralta P. [EMAIL PROTECTED]
Cool. It is posible to use the GSM phone as a DIAX headset ? At least
there is posible to transmit audio using Bluetooth.
Unfortunately not, because the GSM phone does not support Audio Gateway
profile (just
Greeting to all,
Hello my is James Jones. I am one of the new Tech Support people at
Broadvoice. I have signed up for the Asterisk mailing list to better
understand some of our customer's need and to learn more about Asterisk and
what it can do. I can help answer any general questions
So the kid installed it in his house and wrote an article about it.
Hey, it's free publicity.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Wednesday, June 09, 2004 5:39 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] PC
On Wed, 2004-06-09 at 17:03 -0400, Karl J. Vesterling wrote:
ANother workaround I have through of would be to use 99 for outgoing,
but with no transfer options in the Dial() setting. This of course
implies you're expecting to interact with a remote IVR.
Last week I had to spend all day
There appear to have been some changes made recently to the way VoiceMail
works.
Previously if you pressed 7 whilst a message was playing, it would delete the
message.
Now if you press 7 whilst a message is playing it takes you to a menu and
then you have to press 7 again to delete the message.
The point is that I was using an Alcatel 4200 legacy PBX
and it can not interact with IVRs unless you tap a code
before the first interaction,
Ugh, yes.. I remember those.. nice ring tones and stuff, the phones weren't
all THAT bad to use.. but that pressing something before you can send DTMF
The internal call parking SUCKS .. that's why tony and I wrote
valetparking but nobody seems to have liked it so we gave up trying to
give it away since everyone was so down on it.
Hehe, I was just adding valetparking to the asterisk wiki software addons.
Maybe you write a page there that
Hello Martin,
how would you like to integrate? PRI (E1) or BRI (ISDN)?
Besides of making calls with VoIP from PC to PC, we'd like
that our people abroad could dial company internal extensions
through Asterisk using a SIP client. On a second approach,
the same people abroad could dial the
Jose R. Ortiz Ubarri wrote:
Hi:
Is DynExtebDB module still working??
Don't bother. That application should have never been written.
Jeremy McNamara
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Hi Patrick,
could you please give us a feedback if that have worked?
Because I have hacked the source to disable fax..
Thanks
Felix
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Nicolas Gudino
Sent: Wednesday, June 09, 2004 8:48 PM
To:
Hi Dan,
could you support alaw/mlaw? Is that a big problem?
Regards
Felix
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Hi,
-Original Message-
Yes. The issue here is that billing information is never correct in
such a scenario, since the call duration on the registered asterisk
machine (the one that is kicked from the path) is no longer correct.
To fix this a notransfer=yes is mandatory, but that
Maybe, maybe not... Depending how one designs the GUI!
No, I think that GUIs though needed, do limit flexibility because the
information density is limited on the user-system direction (they are
better on the System-user end, however). However, this is NOT an
argument not to package
Yes, you are right!!
However, GUI for newbie's will help some people to overcome the first
hurdles, and then plunge into more advanced stuff!
One thing quote a lot of companies do is outsource the initial
configuration, because they simply don't have the technical skills
initially. But what
Hi all,
I just saw this article about this new offer from Lingo.com:
http://www.techweb.com/wire/story/TWB20040607S0008
$20 monthly plan with unlimited local and long-distance calling in North
America (US Canada) and Western Europe. Plus first three months free and
free equipment. It doesn't
Eric J Merkel wrote:
Then you won't be able to us it for terminating voice. Sorry :(
Eric
Oh well, it was worth a though.
Thanks for the answers though. I knew I should have gone for the 5350.
If you get 'voice' cards for the AS5300 it'll work, and if you had gone for
the 5350 you
Hi Felix,
- Original Message -
From: ePyron Felix Deierlein [EMAIL PROTECTED]
could you support alaw/mlaw? Is that a big problem?
DIAX is based on iaxclient library which is another project.
I have just done some modifications to comply with my app,
but not added any new feature,
Hello Brent,
Wednesday, June 9, 2004, 7:13:52 PM, you wrote:
BF On Wed, 9 Jun 2004, Alessio Focardi wrote:
Asterisk with one HFC isdn card, using the zaptel driver bristuff
All works ok, but voice coming in/out of the isdn card is out of sync,
squelky and disrupted, UNTIL I PUT SOME LOAD TO
Sure... So, this issue is sort of a bug and it really needs to be
implemented then!
I'm afraid its not that simple. Unless I'm misunderstanding the
concepts of
IAX(2) design, it does not support such behaviour _by design_. Who
knows what would break if someone hacked our desires in
Hi.
Thanks for tipping me off with the new firmware. I installed it and tested
the codec. Has more delay but seems to be better quality than what I was
using before.
Anyways, that didn't fix the SIP Registration Failure that I am getting.
Any ideas?
S.
-Original Message-
From: [EMAIL
Chris Bond wrote:
Yes, you are right!!
However, GUI for newbie's will help some people to overcome the first
hurdles, and then plunge into more advanced stuff!
One thing quote a lot of companies do is outsource the initial
configuration, because they simply don't have the technical skills
Tony Hoyle wrote:
Steve Underwood wrote:
I didn't say one patent covered all the world. I said the patents on
codecs exist all over the world. WIPO is simplifying this a bit, but its
still pretty expensive to get a patent everywhere. I know of no country
where the key aspects of a codec
Internet telephony (VoIP): Regulators and industry debate 'irreversible'
trend
The Commission is weighing up its options on how to regulate internet
telephony. Major telecoms operators are already proposing services to
avoid being squeezed out of the market.
More at:
On Thu, 2004-06-10 at 11:41 +0200, Philipp von Klitzing wrote:
telephony. Major telecoms operators are already proposing services to
avoid being squeezed out of the market.
I dont know if you can call Free/OneTel a major yet, but here in France
they offer VoIP to all connected to their ADSL
Hi,
I would like to set up a high density FAX/PBX server, am looking at using
Eicon Diva E1 card with Asterisk and Hylafax sharing channels, is this
possible. I know Extensions can be reserved for voice OR fax with the
combination of chan_capi used for * voice, capi4hylafax on fax but then
I have configured several ISDN cards in Brazil in Cisco Routers. There
is a configuration called compand-type (ulaw alaw) (Cisco). They are
different between US and Brazil. The sound is very distorted when in the
wrong configuration. The difference is between 56 bit and 64 bit ISDN.
Maybe that s
Hi All,
I am trying to Register Asterisk PBX to a SIP Server. But SIP Server
gives the following response to Asterisk: 400 Bad Request .
Asterisk sends the Register Message to SIP server with the
URI: sip:
domain_name_sip-server. BUT
URI should be of the format: sip: user@
Dear Jimenez,
You have to configure a dial-peer in Cisco box. A 2611 with a NM-HDV-E.
It works. The configuration is something like:
[Cisco]
dial-peer voice 8000 voip
protocol sipv2
codec g711
dest pattern 4... (Whatever says your dialing plan)
session target ipv4:(ip address of
On 10/06/2004 at 09:04 Dan wrote:
Hi,
- Original Message -
From: Juan J. Sierralta P. [EMAIL PROTECTED]
Cool. It is posible to use the GSM phone as a DIAX headset ? At least
there is posible to transmit audio using Bluetooth.
Unfortunately not, because the GSM phone does not
Randy Ackers wrote:
Tony Hoyle wrote:
Steve Underwood wrote:
I didn't say one patent covered all the world. I said the patents on
codecs exist all over the world. WIPO is simplifying this a bit,
but its still pretty expensive to get a patent everywhere. I know
of no country where the key
Hi Andy,
- Original Message -
From: Andy Powell [EMAIL PROTECTED]
Any chance of getting this to work with Nokia phones Dan?
No chance unfortunately..
Nokia does not support the extended AT commands set needed to control phone
keyboard and display.
This is one of the reasons I like
Karl J. Vesterling wrote:
It would be nice to have an option where dialing two #'s would send a
single #.
I too have had difficulty with this.
My workaround is to use my cell when calling an IVR.
ANother workaround I have through of would be to use 99 for outgoing,
but with no transfer options
Hi,
I apologies for reposting this message, I am getting
no where in solving this issue. And I am sure it is
something very simple.
I have two Firefly clients configured, If I use
iax.conf to specifiy the accounts everything seems to
work as expected.
However If I use mysql, I can register
Simon Brown [EMAIL PROTECTED] wrote:
There appear to have been some changes made recently to the way VoiceMail
works. Previously if you pressed 7 whilst a message was playing, it would
delete the message. Now if you press 7 whilst a message is playing it
takes you to a menu and then you have
So simple question, without googling:
Where can the g723.1 and g723.1a be found... so a 'EU-patch' can be make.
I'm able to host it in Amsterdam.
Greetings,
Stefan de Konink
On Thu, 10 Jun 2004, Vlasis Hatzistavrou wrote:
Randy Ackers wrote:
Tony Hoyle wrote:
Steve Underwood wrote:
Hi Dan
On 10/06/2004 at 14:01 Dan wrote:
Hi Andy,
- Original Message -
From: Andy Powell [EMAIL PROTECTED]
Any chance of getting this to work with Nokia phones Dan?
No chance unfortunately..
Nokia does not support the extended AT commands set needed to control phone
keyboard and
On Thursday 10 June 2004 03:03, Holger Schurig wrote:
Maybe you write a page there that describes valetparking more?
Yes, please! ValetParking is supposed to do practically everything yet there
is next to no documentation on how to make it do _anything_ -- Let's get some
killer documentation
Hello,
I have the whole library (G.723.1 and G.723.1b) downloaded from ITU, but
it doesn't compile with Asterisk out-of-the-box.
So, unless someone else can provide a library which compiles with *,
we'll have to tinker with the ITU source code (if it is possible at all).
Best regards,
Vlasis
I have a problem in that when you use IAX2 for trunking and have 2 nics one
is used to connect directly to 2nd Asterisk server how do we get the outside
Nic card to take IAX connections? Is there any way to get this working via
two paths? There is only one bindipaddr=10.1.1.1 for internal trunk
Randy Bush wrote:
Time for Duane to start implementing DNS SRV, since it's from now on is
turned on by default in CVS head.
Unless you're planning on breaking other standards my A records will
keep on working just fine :)
except you (likely to be ex-) customers will have problems reaching
more
They don't provide soft accounts. You need to use their D-Link box which
connects back to them using MGCP. Overall service is reasonable, acceptable
for home users but definitely not good enough for business use. I am just
about to send their units back.
Thanks,
Wojtek
- Original Message -
This email is intended to document an issue for anyone searching the archives.
We had a problem yesterday with _all_ iax2 and sip sessions; no reasonable
conversation could be established due to extremely choppy audio in one
direction only (outbound from * to distant sip phones and distant *
Hi,
i have a problem with iax2 and ringtone.
Here is the call path
pstn - asterisk - iax - firefly or any iax phone.
My problem is when i receive a call on my iax phone, the ring sound is very distort
and bad.
If i open my sip phone, and receive a call from my pstn, the ring is like dring
Hi,
i have a problem with iax2 and ringtone.
Here is the call path
pstn - asterisk - iax - firefly or any iax phone.
My problem is when i receive a call on my iax phone, the ring sound is very distort
and bad.
If i open my sip phone, and receive a call from my pstn, the ring is like dring
FWIW, I like the valetparking feature but the documentation sucks rocks --
you
had to describe it at least three times before I *started* to understand
its
utility and features above and beyond normal parking.
Perhaps the problem isn't so much that the people were down on it as that
they
I replaced the .c file with a note
read the .c file. :)
bkw
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin
Sent: Wednesday, June 09, 2004 9:32 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Sending # and Asterisk Transfer Conflict
I was
Dear friends,
I have an HFC ISDN card that I have set up in NT mode.
Will this be enough to connect to an ISDN Pbx and pretend to be and
ISDN line ?
Tnx for any help ?
--
Best regards,
Alessio mailto:[EMAIL PROTECTED]
On Thursday 10 June 2004 09:50, brian wrote:
The docs on my site were fine... they explained and gave examples of how to
use it.
Hmm I did not see them. Are they still up? I saw you said you took down
the .c, so I am assuming they're gone too?
-A.
The reference code does not pack or unpack the bits. It needs additional
work to make a usable codec. This is true of most reference codec
implementations. The bit packing arrangements depend on the application
of the codec, so they are often not specified as part of the codec.
Regards,
Steve
Randy Ackers wrote:
Tony Hoyle wrote:
Steve Underwood wrote:
I didn't say one patent covered all the world. I said the patents on
codecs exist all over the world. WIPO is simplifying this a bit,
but its still pretty expensive to get a patent everywhere. I know
of no country where the key
On Thu, 2004-06-10 at 23:27, Rich Adamson wrote:
This email is intended to document an issue for anyone searching the archives.
We had a problem yesterday with _all_ iax2 and sip sessions; no reasonable
conversation could be established due to extremely choppy audio in one
direction only
I've been avoiding commenting on this thread because I haven't studied
the code enough, or my current problem, but anyway, here is my 0.02c
worth...
I found the documentation to be OK, and the app seems to do some
fantastic things, which the current call parking can't do. However, the
real reason
http://asterisk.gnuinter.net/files/digium/asterisk-ng/db1-ast/
I 'stole' the sources from them, compiled (and working) after uncommenting
the makefile. Tobad only the Budgettones work with the codec and not the
Cisco's :(
Stefan
On Thu, 10 Jun 2004, Vlasis Hatzistavrou wrote:
Hello,
I have
Hi
I am in the UK and am looking for a device that will allow me to connect
two sim cards (read wireless lines) to either the port on the back of my
fritz card or any other connection direct to the PC that provides a
usable telephony interface.
I will even plug two devices into a windows box
Fyi,
Successfully compiled Asterisk on an Apple G4 PPC with Yellow Dog Linux -
without any source modifications.
Worked fast and smooth.
- Darren
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http://www.bkw.org/archives/000291.html
the docs are still up.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
Sent: Thursday, June 10, 2004 9:04 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Sending #
Hi Chris,
CL All I want is two GSM lines that look like voice modems to
CL the PC and provide full telephony interface, that is DTMF
CL both ways CLI and a few other bits and pieces.
We use the Nokia 22:
http://www.nokia.com/nokia/0,,56024,00.html
They have worked well providing both telephony
Why???
Is there another way to do Dynamic Extensions???
--
JO
Jeremy McNamara wrote:
Jose R. Ortiz Ubarri wrote:
Hi:
Is DynExtebDB module still working??
Don't bother. That application should have never been written.
Jeremy McNamara
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Asterisk-Users
Chris Lee [EMAIL PROTECTED] wrote:
[...]
I am in the UK and am looking for a device that will allow me to
connect two sim cards (read wireless lines) to either the port on
the back of my fritz card or any other connection direct to the PC
that provides a usable telephony interface.
My initial
Hi,
I have compiled and installed app_prepaid module. But have
problem when connect to postgres database. I guess so because after key
in card number, it always play prepaid-no-aaa voice file.
Anyone succeeded in configuring the app_prepaid for prepaid
calling service for asterisk?
Hello
I have heard that i can put a file in a certain directory to get * to
initiate a call.
Is this true ? if so where would i look ?
Best Regards
Simon Garvey
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[EMAIL PROTECTED]
Thanks, I figured it out and replaced it with the proper file.
-Original Message-
From: brian [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 10, 2004 9:51 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Sending # and Asterisk Transfer Conflict
I replaced the .c file with a note
Hi all
Does anybody know if it is possible to change the tones on a 7960 ?
I guess there must be some way to edit the dial/busy/congestion tones ?
/Mike
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On advice from others I dropped gnophone in favor of iaxComm.
I am operating on an IBM T30 laptop Redhat Linux 2.4.20-8 with an Intel
i810 audio chipset (comes in the laptop).
I am using the Gnome desktop.
There is no reference to alsa or oss to be found.
All audio components function fine.
Hi everybody...
I'm having an odd problem with voice mail on a recent CVS of * where it
appears not to detect a hangup on FXO and * will keep treating the call
as new and continue leaving voicemails until the max has been reached.
It will then continue trying to leave voice mails and basically
Hi Jean,
It seems that no one on the list is interested in IAX,
I have posted a couple of basic questions but no ones
seems to want to answer. I guess everyone is busy
right now.
Anyway back to your question. When you say the
ringtone , do you mean the rinback tone (what the
caller hears) or
Thanks to the lack of documentation, I decided to
write my own AGI script (working but no where near
complete)
Look forward to replies and guidence on this topic.
Umar.
--- Yang Tao [EMAIL PROTECTED] wrote:
Hi,
I have compiled and installed app_prepaid module.
But have problem
What I do is create a file from setting in a DB.
These extensions are included in the extensions.conf
with a #include line.
I can be done with a little perl scrit and a cron.
After recreating the file all you have to do is reload the extensions.
For eficiency, I create a temp file, and diff
Very interesting,
Would like to hear what sort of performance you get
out of it.
I was considering linux on a sun box. Anyone done that
?
Umar.
--- Darren Sessions [EMAIL PROTECTED] wrote:
Fyi,
Successfully compiled Asterisk on an Apple G4 PPC
with Yellow Dog Linux -
without any source
Yes you can, I have never used it but here is a
link
http://www.voip-info.org/wiki-Asterisk+auto-dial+out
Umar
--- Simon [EMAIL PROTECTED] wrote: Hello
I have heard that i can put a file in a certain
directory to get * to
initiate a call.
Is this true ? if so where would i look ?
Alessio Focardi wrote:
BF You can try doing different things with it, but I know that I am
currently
BF set to level 3 rather than 5 as default with RedHat.
I checked hdparm googling around, what parameter have you set to 3
instead of 5 ?
I'm pretty sure this is a confusion.
I think this must
Storer, Darren wrote:
Hi Chris,
CL All I want is two GSM lines that look like voice modems to
CL the PC and provide full telephony interface, that is DTMF
CL both ways CLI and a few other bits and pieces.
We use the Nokia 22:
http://www.nokia.com/nokia/0,,56024,00.html
They have worked well
Hi Simon,
SG I have heard that i can put a file in a certain directory
SG to get * to initiate a call. Is this true ? if so where
SG would i look ?
It *really* is time that you got to grips with voip-info.org. There are many
gems in there; I typed in auto dial out and pressed the search button,
The directory is /var/spool/asterisk/outgoing. see
http://www.voip-info.org/tiki-index.php?page=Asterisk%20auto-dial%20out
for information on how to use the auto-dial out features...
Steve Rosebush
[EMAIL PROTECTED]
PSTN: 1-248-724-4452 FWD: 63420
IAXTEL: 1-700-356-6191
-- all extension 201
Hi, all
This is certainly very good news!
http://www.neowin.net/comments.php?id=21119category=main
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Simon wrote:
Hello
I have heard that i can put a file in a certain directory to get * to
initiate a call.
Is this true ? if so where would i look ?
The WIKI: http://www.voip-info.org bookmark it
Jeremy McNamara
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[EMAIL
/var/spool/asterisk/outgoing
On Thu, 2004-06-10 at 15:27, Simon wrote:
Hello
I have heard that i can put a file in a certain directory to get * to
initiate a call.
Is this true ? if so where would i look ?
Best Regards
Simon Garvey
On Thu, 2004-06-10 at 16:27 +0100, Simon wrote:
Hello
I have heard that i can put a file in a certain directory to get * to
initiate a call.
Is this true ? if so where would i look ?
The rumours are true! You would look in the ever-so-helpful Wiki:
Hi,
Anyone having problems receiving DTMF on incoming iConnectHere
lines? They disappeared for us sometime in the last 12 hours...
And, yes, we've restarted * and rebooted our * machine.
Michael Swan
Neon Software, Inc.
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Look in the asterisk source directory for a file called sample.call
Read it and it'll give you all thed details
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Simon
Sent: Thursday, June 10, 2004 10:28 AM
To: Asterisk-Users
Subject:
On Thu, 2004-06-10 at 10:27, Simon wrote:
Hello
I have heard that i can put a file in a certain directory to get * to
initiate a call.
Is this true ? if so where would i look ?
Google would have been a good starting point. Next would have been to
exercise some curiosity in the source
On Thu, 2004-06-10 at 10:54, Chris Hirsch wrote:
Hi everybody...
I'm having an odd problem with voice mail on a recent CVS of * where it
appears not to detect a hangup on FXO and * will keep treating the call
as new and continue leaving voicemails until the max has been reached.
It will
Adam Hart wrote:
I've also added support for SIP via TCP and the ability to change the
SIP port
It complains every time you click OK in the Options page about Changing
SIP port requires restart, even if you never looked at the SIP page
(and don't even have any SIP networks configured).
hi, Simon, you can look at this
http://www.voip-info.org/tiki-index.php?page=Asterisk%20auto-dial%20out
Simon wrote:
Hello
I have heard that i can put a file in a certain directory to get * to
initiate a call.
Is this true ? if so where would i look ?
Best Regards
Simon Garvey
Good day,
I've got around to installing an X100P card in my computer to try out
asterisk. I noticed (and people who were testing with me also noticed) that
when dialing from my SIP soft phone to the PSTN, the ringer tone changes
after 2-3 seconds, precisely when the Zap channel takes over the
Hi,
I read to use NAT clients behind firewalls to use nat=yes and qualify=xxx to
keep a nat connection open. I am using sip clients behind a firewall that
is symmetric and my * box on public internet. Are these the only two
options that I need in my configuration? Isnt the qualify command
I think the problem it's in your dialplan, extensions.conf:
; voicemail management
[voicemail]
include = misc
exten = 6245,1,VoiceMailMain2()
exten = 6245,2,Hangup()
Check the last line, I have the same problem and was because I wrote wrong the Hangup
instruction...
Regards!
Date: Thu, 10 Jun
I use Nokia 32s.
I don't know what a fritz
card is but they can act either as an FXO or as an FXS device. Beware though, if
you use them off a port that expects a telephone set (like an ATA or so) you'll
need a special cable to program the 32 properly - the cable is
pricey.Acting asanFXO,
Since updating * via CVS earlier this week, I've been having problems
with cdr_mysql. Prior to that time my queries and cdr all worked fine.
Now, even though my queries still work, I get the messages similar to
this:
ERROR[1211374384]: cdr_addon_mysql.c:203 mysql_log: Failed to insert
into
Hello all,
I am a relative asterisk noob so please bear with
me if my questions are obvious. What I'm trying to do is get our analog PBX (A
Merlin Legend) connected to VoIP. From all my googling and reading voip-info.org
(and this list) it seems very possible. I just wanted to describe my
Pablo Endres wrote:
For eficiency, I create a temp file, and diff from the previous version
(so I don't reload if I don't have to).
Why bother doing that much processing? Just set a flag somewhere that
determines weather or not you need a reload.
Jeremy McNamara
I have a setup where I am using asterisk a SIP proxy. My ATA is behind
NAT. Asterisk user is setup with nat=1 and canreinvite=yes. The call
sets up and I can get one way media. The media works ok from the ATA
behind the NAT to the external SIP endpoint, but I cannot get media back
through the
See also:
http://www.adslguide.org.uk/newsarchive.asp?item=1723
- Original Message -
From: Senad Jordanovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 10, 2004 5:18 PM
Subject: [Asterisk-Users] BT is moving to IP ONLY
Hi, all
This is certainly very good news!
Their Syntegra trading turrets already have begun the migration. Now, if I
can get my hands on one and getting it to work with *, I'll be set.
-cj
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Senad Jordanovic
Sent: Thursday, June 10, 2004 12:18 PM
On Thu, 2004-06-10 at 02:04, Dan wrote:
Unfortunately not, because the GSM phone does not support Audio Gateway
profile (just Headset profile).
It can connect only with the headset.
..but.. you can use the Bluetooth headset for DIAX and the GSM phone as
CallerID/Dialer.
.. and all this even
BF You can try doing different things with it, but I know that I am
currently
BF set to level 3 rather than 5 as default with RedHat.
I checked hdparm googling around, what parameter have you set to 3
instead of 5 ?
Alessio Focardi wrote:
I'm pretty sure this is a confusion.
I think
This is a wonderful idea. I like the app_im concept a lot.
I'd make a few additions though. Like the ability to have festival read
the Away message as the Voicemail message. I'd definitely change my
voicemail more often if I could do it by changing my Jabber away
message.
I would suggest that
yes the bell to notify, when it is to iax, the bell sound is very bad. With sip it's
fine, the ringback is good with both technology
Regards
JF
De: Umar Sear [EMAIL PROTECTED]
Date: 2004/06/10 jeu. PM 12:00:33 GMT-04:00
À: [EMAIL PROTECTED]
Objet: Re: [Asterisk-Users] Iax2 ringtone
On Thu, 2004-06-10 at 12:04, Umar Sear wrote:
Very interesting,
Would like to hear what sort of performance you get
out of it.
I was considering linux on a sun box. Anyone done that
?
It surely gives a new life to those old suns, usually linux runs faster
than slowlaris. I have
At 09:43 AM 6/10/2004 -0700, you wrote:
Hi,
Anyone having problems receiving DTMF on incoming iConnectHere
lines? They disappeared for us sometime in the last 12 hours...
And, yes, we've restarted * and rebooted our * machine.
Hi,
I'll follow up on my own question. :-)
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