Hi,
As I am the developer of DIAX
- Original Message -
From: "Robert Rozman" <[EMAIL PROTECTED]>
there is already iax softphone called diax
(http://www.laser.com/dante/diax/diax.html) that can be controlled over
bluetooth on some phones. The thing that is missing is to be able to use
cellu
I use Digium's Licensed Codec and I have no problems in routing calls to
either E1 or T1 interfaces.
But ...beware of the Pitfalls in using non-standard G729 Codecs.
I used a couple of sets before and here are the problems I found (I have not
used Daniels codec though):
1) Calls are too noisy and
I have calls coming in via SIP (a DID) and I want to forward them right back
out to my cell.
If I do it in one step,
(as if 2125551212 was the DID, and 202111 was my cell number)
exten => 2125551212,1,Dial(SIP/${PROVIDER}/1202111,60)
The call comes in via sip, my system sends the invite
On Sat, 2004-09-25 at 09:49, Michael Bielicki wrote:
> 64bit it :)
>
> [EMAIL PROTECTED] root]# cat /proc/cpuinfo
> processor : 0
> vendor_id : AuthenticAMD
> cpu family : 15
> model : 5
> model name : AMD Opteron(tm) Processor 244
Any idea to the number of channel
Dear Valued OnDO users,
OnDO PBX v1.3 now supports 100 concurrent calls
Brekeke is excited to announce our new OnDO PBX v1.3
with increased concurrent call capacity that is 4 times
greater th
Peter Svensson wrote:
On Sat, 25 Sep 2004, Steve Underwood wrote:
I wouldn't do that, if I were you. Distributing the source code for
educational and evaluation purposes won't get anyone into trouble with
the patent issues. I think (not sure) that Intel's copyright on the code
is OK, since Da
I tried to install chan_sccp (make; make install) but after that when
asterisk starting:
[chan_sccp.so]Sep 25 06:34:28 WARNING[16384]: loader.c:242
ast_load_resource: /usr/lib/asterisk/modules/chan_sccp.so: undefined
symbol: __use_ast_pthread_create_instead__
Sep 25 06:34:28 WARNING[16384]: loader
On Sat, 25 Sep 2004, Steve Underwood wrote:
> I wouldn't do that, if I were you. Distributing the source code for
> educational and evaluation purposes won't get anyone into trouble with
> the patent issues. I think (not sure) that Intel's copyright on the code
> is OK, since Daniel is distribu
Hi all,
I am new to asterisk. Is there any way to get dtmf in
between an active conversation. For eg. If phone1 has
made a connection with phone1, then is it possible to
handle all the dtmf send between them in
extensions.conf.
Thanks
Jibu
__
Has anyone tried loading the PAP2-NA firmware on the PAP2 Vonage model?
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Hello
all,
I am currently using
asterisk in a call center configuration.
I have created a
queue where our customers listen to music while an agent picks up. Pretty
standard stuff.
I have a total of 5
agents who are able to sucessfully sign-in and sign-out of the queue by using
set
Eric Wieling wrote:
On Fri, 2004-09-24 at 20:33, Steve Underwood wrote:
It is very difficult to be legally correct with this. The IP holders
don't have simple programs for selling licences in small quantities. If
you buy licences from Digium, they deal with the IP issues on a larger
volume ba
On Fri, 2004-09-24 at 20:33, Steve Underwood wrote:
> It is very difficult to be legally correct with this. The IP holders
> don't have simple programs for selling licences in small quantities. If
> you buy licences from Digium, they deal with the IP issues on a larger
> volume basis. Unless you
I am having my colo set up 2 PRI's for my new asterisk implementation. They
asked the following...
##SNIP##
What type (NI2, NTI, 4ESS, or 5ESS) and whether they want to be USER or
NETWORK.
If the equipment is flexible, NI2, with us as NETWORK is preferred.
##SNIP##
We are using a digium quad
HI Martin:
Just one question???
I am receptive to your problem however, I am one step back, I have a
cisco 3920 router with ISDNT1 connected PRI card. I having a major
struggle dealing with the so called routing on this card using "show
isdn history". The server is rejecting my calls. Through ma
you will probably have to d a combi and still buy the digium license
which should allow you to use xyz lines of any g729 version that is
g729b
On Sat, 25 Sep 2004 09:33:51 +0800, Steve Underwood <[EMAIL PROTECTED]> wrote:
> Danny Zak wrote:
>
> >Hello TELUX,
> >
> >could anybody post something m
Arkadi Shishlov wrote:
I expropriated the right to rip Daniel's disclamer for use in my
email too..
DISCLAIMER:
You might have to pay royalty fees to the G.729 patent holders for using
their algorithm.
For easier testing I prepared codec_g729.so binaries and associated
libraries and put them on the
Im also interested in a couple of these... plesase email me if you are
selling or post over a link!
Lethol
On Fri, 24 Sep 2004 22:02:52 -0400, William Suffill
<[EMAIL PROTECTED]> wrote:
> Anyone here have any pointers of where to get 1 of the PAP2-NA. Given
> all the talk about it I'd be curious
Greetings all,
I have asterisk running on a high speed connection, sending and receiving
calls from a service provider.
I can receive calls from the PSTN into voicemail, etc., and it works fine.
However, when I try to redirect the call back out to the PSTN (for example
sending it to my cell phon
Anyone here have any pointers of where to get 1 of the PAP2-NA. Given
all the talk about it I'd be curious as to testing one myself .
-- William
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To
It's probably something akin to the older H.323 VxWorks based units where
the hardware is custom designed, but the software is the same or very
similar. No sense reinventing the wheel if someone has code written
already and is willing to license the code.
Does anyone know anything about the ESS V
Danny Zak wrote:
Hello TELUX,
could anybody post something more about being legaly correct using
this codec and the corresponding "royalty's".
It is very difficult to be legally correct with this. The IP holders
don't have simple programs for selling licences in small quantities. If
you buy li
Sorry, just dual Xeon's, not P4's!
Scott M. Stingel
>Hi-
>I've run extensive load testing with both single and dual P4's and
>Xeon's (all at least 2.8GHz), and I've got 6 installed IVR systems of
>this size in various configurations.
Hmm, I was under the impression that it was impossible to
Andres wrote:
Ryan Wilkins wrote:
This begs the question, again, that someone else posted originally..
what about loading SPA-2000 or PAP2-NA firmware in the PAP2? If it's
the same hardware, there shouldn't be any reason not to try it.
Thats the first thing I'm going to try when we get our
Ok, I got 1 7910 up and I get dial tone when I go off hook, The other 7910
is battery, dead air. Using the default configuration files what extension
do I dial for voicemail? I'm confused. I've programmed many PBX's but it
is hard for me to get my mind around * definition of extensions.
-
Hi,
Thanks alot to Mark and his team for bringing
a great voip project and a great conference.
I am a developper in the *BSD projects and will go back to
these with the intention to help in providing a better availability
of '*' for these OS's and also working the zaptel4BSD.
cheers and see you s
Hi-
I've run extensive load testing with both single and dual P4's and Xeon's
(all at least 2.8GHz), and I've got 6 installed IVR systems of this size in
various configurations.
Hmm, I was under the impression that it was impossible to run dual P4 CPUs.
I thought Intel programmed instruction in t
Hi-
I've run extensive load testing with both single and dual P4's and Xeon's
(all at least 2.8GHz), and I've got 6 installed IVR systems of this size in
various configurations.
Asterisk can run 4 E1's (120 channels) in an IVR scenario, but just barely.
With this many simultaneous calls, you may n
I'm wondering if tweaking [rx|tx]gain would improve my fax reception success
rate.
Running ztmonitor when receiving a fax shows 4 "octos" and an * on the RX
side and nothing on the TX side.
At the end of the page, there's a burst where RX goes to about 1/2 and TX
goes to about 2/3 of the range
We use mostly dual opterons and they don't seem to notice the quad E1's on them
On Fri, 24 Sep 2004 12:57:09 -0400, steve szmidt <[EMAIL PROTECTED]> wrote:
> On Friday 24 September 2004 12:46 pm, Christian Victor wrote:
> > Hi Régis,
> >
> > > We're going to build an IVR system with a TE405P and
64bit it :)
[EMAIL PROTECTED] root]# cat /proc/cpuinfo
processor : 0
vendor_id : AuthenticAMD
cpu family : 15
model : 5
model name : AMD Opteron(tm) Processor 244
physical id : 0
siblings: 1
stepping: 8
cpu MHz : 1791.799
cache size
works fine for me too...
[EMAIL PROTECTED] ~]$ cat /proc/cpuinfo
processor : 0
vendor_id : GenuineIntel
cpu family : 15
model : 2
model name : Intel(R) Xeon(TM) CPU 2.80GHz
stepping: 9
cpu MHz : 2787.439
cache size : 512 KB
[...]
processor
Hi,
I am looking for some help with my Asterisk system. I
have a server setup with one digium card with a single
connection. It is running on a 2000+ Athlon system. We
are using CISCO 30 VIP and 12 SP phones. I also have a
SIPURA to which I can connect analog phones. The phone
system is far below
I *think* he meant a PC application that complements an analog phone adding
all features a high end VOIP phone would have and more, allowing someone to
use cheap, but high quality analog POTS phones as extensions, but getting all
the extremely nice features that a high end VOIP phone would have,
Yes, in fact all my telephony goes through my computer here (asterisk)
:)
Or what is it that you meant in particular?
> -Original Message-
> From: TELUX [mailto:[EMAIL PROTECTED]
> Sent: Friday, September 24, 2004 4:51 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Su
You don't need a serial cable to send the AT commands. Bluetooth
provides a "virtual serial port" which makes the modem commands
available wirelessly.
> -Original Message-
> From: Stefan de Konink [mailto:[EMAIL PROTECTED]
> Sent: Friday, September 24, 2004 4:46 PM
> To: Asterisk Users
Yep, right on -- once it's a channel, you can run any extension you like
-- including your IVR prompts and voicemail. Whether your extensions
are SIP, H.323 or zaptel won't even matter.
> -Original Message-
> From: Damjan [mailto:[EMAIL PROTECTED]
> Sent: Friday, September 24, 2004 5:07
Hi Danny,
dont mix ppp and p2p...
pp = point to point
pmp = point to multipoint
(both are ISDN connection configurations)
ppp = point to point protocol
a higher level protocol for data transmission etc.
Of which type your ISDN connection is, is usually easy to decide looking
at your phonenumbers
A
Jonathan Augenstine a écrit :
I am new to Asterisk and I am investigating setting up a very large
Asterisk server farm. I have found a lot of good information on this
topic on the Wiki pages. I am drinking from the fire hose and I
thought that I read somewhere on Wiki a caution about a potenti
PPP is Point To Point Protocol. It was designed to allow TCP/IP over a
modem/dialup connection which a ISDN modem is capable of. Now to allow
combining the two b channels to appear as one data channel with the same IP
address and so forth, that's where the Multilink PPP comes in. It's an
extensi
Hi Danny,
- how can i checke the number that is being dialed by the caller to
reachh the * box (so one of the 4 msn's). I have seen dialplans
making use of the CALLERIDNUM; but what do i need to query for the
called num ?
Here is a working extensions.conf section separating calls based on the MSN
Hi,
I'm a lab manager / supervisor at our labs. We've had Asterisk in
use for over a year directly hooked to the PSTN - a no brainer for
configuration (although I had to fix some AT&T specific things in libpri).
Right now I have two big challenges. One is to hook our box up lineside to
a
> ON: Why would one prefer bluetooth over wires if one still needs a
> serial cable to send AT commands?
You can send AT commands via BT too... while sending audio.
BT is a great standard.
> For an incomming call, GSM to
> GSM-Asterisk it would make sense, but I never saw a bluetooth headset
Funcionó, es asi como dijiste, gracias Horacio
- Original Message -
From: "Horacio J. Peña" <[EMAIL PROTECTED]>
To: "M. Willigs" <[EMAIL PROTECTED]>
Cc: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Friday, September 24, 2004 11:41 AM
Subject: Re: [Ast
Hello;
we are looking to replace our current PBX with a *-box; it is
connected to ONE ppp isdn connection that is terminated by the NC. We
got on this box 4 msn's configured.
currently we are working with pstn fxo's behind the PBX; it works but
we can't use the CSID information behind it. We wa
Has anyone started on CTI development?
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Jay Milk wrote:
That's exactly it! The asterisk box acts as a handset for the phone and
uses AT-commands for call-origination and progress.
OFF: Although the reversed thing, having a console with a bluetooth
headset would also sounds very ok.
ON: Why would one prefer bluetooth over wires if one
Hello.
I was hoping someone might be able to help me with the following problem:
When an incoming call comes into to * I would like it to attempt to find
the first extension in a group of extensions that isn't busy and to send
the call to that extension. Should that extension not be picked up or
Is there a way to change the format for the file name when using the
command monitor-format = in the queues.conf config file Similar to the
ChangeMonitor() command?
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That's exactly it! The asterisk box acts as a handset for the phone and
uses AT-commands for call-origination and progress.
> -Original Message-
> From: Damjan [mailto:[EMAIL PROTECTED]
> Sent: Friday, September 24, 2004 2:43 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] GSM
Hi all,
I bought a couple phones for really cheap just for a simple
solution. I’m trying to get a
few 7910 to work with *. I’m
just not sure how to get them to work.
The 7910 just sits there “configuring IP” Here is a copy of my skinny.conf. the extensions.conf is default. I just
Hi,
I have in one Asterisk box different numbers, all are coming
thru the T1.
We need to know exactly the CallerID (whose calling I have
it) and witch number was called.
I take a look on cdr module and I didn’t find that
information. I think I’m not the only one with this problem L
A
On Fri, 24 Sep 2004, John Bohman wrote:
> Same problem here..
> Redhat 9 2.4.20-31.9
John,
Let me look into it, and rebuild the RPMS. Perhaps I have a header
missing issue.
> John B
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Gre
I expropriated the right to rip Daniel's disclamer for use in my
email too..
DISCLAIMER:
You might have to pay royalty fees to the G.729 patent holders for using
their algorithm.
For easier testing I prepared codec_g729.so binaries and associated
libraries and put them on the web:
http://kvin.lv/
//www.readytechnology.co.uk/open/g729
>>
>>
>>
>>
>> ___
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On Sat, 25 Sep 2004 07:42:14 +1200, Richard Scobie
<[EMAIL PROTECTED]> wrote:
> These are generated by an FXS module. I do not know why, as I and a
> number of others see them on a semi regular basis - in my case 6 or so
> times a month and very occasionally there will be a burst of four, the
> sa
Your extensions.conf and console output might help us out quite a bit.
Thank you,
Steve Maroney
On Fri, 24 Sep 2004, Paul Oster wrote:
> I'm in the process of turning up a PRI in one of my markets and have
> run into a problem I have never seen before. I am unable to place a
> local outgoing ca
terisk-users
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__ NOD32 1.876 (20040924) Information __
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> As for how BT transmits Audio:
>
> www.bluetooth.org
> www.bluez.org
>
> How Linux utilizes Bluetooth:
> http://www.google.com/search?hl=en&ie=UTF-8&q=linux+bluetooth
> www.bluez.org
>
> For how to write a channel, I suppose a seasoned linux programmer would
> know by looking at the sources fo
Gabriel Gunderson wrote:
I've installed a TDM04B and a TDM40B. I haven't plugged any lines
into them yet but I'm starting to see this in my logs...
[EMAIL PROTECTED] asterisk]# grep alarm /var/log/messages
Sep 20 09:13:22 webster kernel: Power alarm on module 1, resetting!
Sep 22 11:07:07 webster
Greetings,
Try a couple of basic things here.
CLI> sip debug
This will let you see if SIP and RTP are really talking. If you
need help with the SIP messages let me know
Often there is a CODEC miss match. Asterisk prompts are in GSM,
what is your phone producing?
Will this run on and AMD based machine ?
Umar
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Daniel
Pocock
Sent: 24 September 2004 17:50
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Free G.729 ready for download
DISCLAIMER: This code is free (I am no
[EMAIL PROTECTED] wrote:
I think they are mistaken about the $8. The min Cisco maint contract is
a category 1 (like you have) and it is $90. As far as I know, there is
no way to buy just the software, you have to buy maintenance to get
software upgrades.
Christopher Jacob wrote:
Hey all,
I am
Hi all,
Is there any basic information available for app_conferense?
Does it suport SIP and other codecs
Any installation guide
Thanks
Umar
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tom Ivar
Helbekkmo
Sent: 24 September 2004 06:55
To: Asterisk User
Hi,
The instructions at http://www.opencall.org/installing-mfcr2.html and
the code at ftp://ftp.opencall.org/pub now include interfacing to
Asterisk. I have been building and testing with the current * CVS code.
I still need to work through the national variants, and get some of the
them better
I'm interested in the g729 diff you posted...
I've applied the patch, but I don't seem to have the prerequisites to
compile it... I tried downloading the other code available from
Intel, but even the 'eval' version won't install without a FlexLM
license (damn license managers...). Am I heading t
I think they are mistaken about the $8. The min Cisco maint contract is
a category 1 (like you have) and it is $90. As far as I know, there is
no way to buy just the software, you have to buy maintenance to get
software upgrades.
Christopher Jacob wrote:
Hey all,
I am trying to get the Cisco
On Fri, 2004-09-24 at 12:53, Anton Tinchev wrote:
> Whicch version of zaptel and Zapata should I use with 1.0?
I suspect 1.0.0 versions available via FTP on Digium's server are the
best.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ru
Thanks but I just purchased that card last week and the fxo is on port 4 not port 1.Rich Adamson <[EMAIL PROTECTED]> wrote:
If I recall correctly, the problem with fxo port 1 is a hardwaredesign issue with early TDM cards. Call digium support to confirm.> I had a similar pro
Hey all,
I am trying to get the Cisco SIP image loaded on my new 7960. The Wiki and
several emails in the list archives say the cost is approx. $8.50 per phone
per year. The problem I have is the Cisco is giving me part number
CON-SNT-PKG1 which costs $90.00. I believe this covers phone support as
I just signed up for Broadvoice, and used a similar network configuration that
I have on stanaphone, voipjet, and others.
My asterisk box is behind a vanilla Linux masquerade (netfilter/ipchains)
firewall. The SIP and IAX services have been working fine in both directions
for the other SIP ter
If I recall correctly, the problem with fxo port 1 is a hardware
design issue with early TDM cards. Call digium support to confirm.
> I had a similar problem but not exactly same: when telco lines are
> plugged into the FXO ports, initially "zap show channel 1" says it is
I've installed a TDM04B and a TDM40B. I haven't plugged any lines
into them yet but I'm starting to see this in my logs...
[EMAIL PROTECTED] asterisk]# grep alarm /var/log/messages
Sep 20 09:13:22 webster kernel: Power alarm on module 1, resetting!
Sep 22 11:07:07 webster kernel: Power alarm on m
Yep, they took over one of the conference rooms, and basicly everyone from
digium is there. They had planned on having calls routed to them there but
there were lots of problems with the internet connection at the hotel, so it
didn't work very well. Today is the developer day(last day of Astricon).
CSID
is caller sending ID. This is what
number you are sending from the PBX to the local carrier.
-Original
Message-
From:
Paul Oster [mailto:[EMAIL PROTECTED]]
Sent:
Friday, September 24, 2004 12:02 PM
To:
Henry Devito
Subject:
Re: [Asterisk-Users] Local Outbound Calls
Back in the office post-astricon. 1.0.0 running in the lab.
YIIIHAA!
THIS GUY rocks. Thanks to Mark for *, Steve and Olle for the conference
and to ALL community members. Everyone using * is contributing in one way
or another.
See y'all next year
Jason Kawakami
www.optellabs.c
I am new to Asterisk and I am investigating setting up a very large
Asterisk server farm. I have found a lot of good information on this topic
on the Wiki pages. I am drinking from the fire hose and I thought that I
read somewhere on Wiki a caution about a potential problem with running
Aster
| On Sep 24, 2004, at 10:12 AM, Cirelle Enterprises wrote:
|
| > Has anybody been able to get in touch with anybody at digium today?
|
| I suspect that they're all at Astricon.
|
|
| Scott
Ah... didn't realize astricon was still going on.
Greg
___
On Sep 24, 2004, at 10:12 AM, Cirelle Enterprises wrote:
Has anybody been able to get in touch with anybody at digium today?
I suspect that they're all at Astricon.
Scott
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I had a similar problem but not exactly same: when telco lines are
plugged into the FXO ports, initially "zap show channel 1" says it is
"Onhook" but I cannot make outgoing calls. Once I unplug the telco
line and re-plug it, or after there is an incoming call, "zap show
channel 1" says "Offhook" b
Heres a patch for the app_valetparking not working with music on hold.
This patch was made against the version at
http://www.bkw.org/app_valetparking.c.
As you can see, the original author of app_valetparking simply forgot to
copy the chan->musicclass to the new masq'ed channel. I'm not entire
Has anybody been able to get in touch with anybody at digium today?
Regards
Greg Cirino
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Has anyone had any experience with connecting
asterisk and Verso's new SIP stack in their Class 5 Call Manager?
I am hearing theres incompatabilites, but I can not
get anything directly from Verso themselves.
Their Call Manager is supposed to support XTens
softphones, so I would think that as
It has happened at two different locations with two different cables/plug. Also when I plug to a normal phone it works ok.
"Michel Belleau (malaiwah.com)" <[EMAIL PROTECTED]> wrote:
Hi.
Did you check out the plug? Is it wired correctly?
I once had this problem because there was two li
On Friday 24 September 2004 01:53 pm, Anton Tinchev wrote:
> Whicch version of zaptel and Zapata should I use with 1.0?
One should always try to use the same version. CVS will give you all the files
you need.
--
Steve Szmidt
"They that would give up essential liberty for temporary safety
de
On Friday 24 September 2004 12:46 pm, Christian Victor wrote:
> Hi Régis,
>
> > We’re going to build an IVR system with a TE405P and 4 E1. We’re sure
> > that the 120 channels will be filled by 120 simultaneous calls during
> > peak, so we want to have the good server to manage this.
> >
> > We won
Whicch version of zaptel and Zapata should I use with 1.0?
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DISCLAIMER: This code is free (I am not charging you to use it), but
you might have to pay royalty fees to the G.729 patent holders for using
their algorithm.
I finished this last Saturday and have had it on an Asterisk machine for
5 days without a crash, so I'm hoping that means it's safe to
Has anyone else experienced a problem with app_queue
where after a time, calls can still come into asterisk, but once they enter a
queue, they just get silence, any calls in the queue get frozen in it, and
never get sent to an agent, yet calls can be made in or out of the phone
system.
Hi Régis,
We’re going to build an IVR system with a TE405P and 4 E1. We’re sure
that the 120 channels will be filled by 120 simultaneous calls during
peak, so we want to have the good server to manage this.
We wonder a lot of things and maybe you could help us.
- Are you ever build a similar sys
*IF* I didn't already have the phone...
CellSocket $100
FXO Port $80
Non-BT Phone $0 (after rebates for new service)
$180
BT Dongle $10
BT-Phone $75 (after rebates for new service)
$85
But aside from all that, many
As for how BT transmits Audio:
www.bluetooth.org
www.bluez.org
How Linux utilizes Bluetooth:
http://www.google.com/search?hl=en&ie=UTF-8&q=linux+bluetooth
www.bluez.org
For how to write a channel, I suppose a seasoned linux programmer would
know by looking at the sources for existing channels.
- Original Message -
From: "William Suffill" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]>
Sent: Friday, September 24, 2004 11:56 AM
Subject: Re: [Asterisk-Users] Asterisk 1.0 released
| Cirelle did you delete the .version file in
Hello.
I trying to use SER with Asterisk together. I have a question
regarding the RTP path. If i make a call from one of my endpoints
registered in SER Server, and that call in particular is forwarded to
Asterisk and then to a PSTN-GW, Does the media goes through Asterisk?? is
there a w
What are you sending for the CSID? Dialing LD goes through the CLEC and may
be excepting your call no matter what the CSID is. The local switch may be
rejecting you because the CSID you are sending is not what they are
expecting. I had a the same experience on a legacy phone system.
-Origin
Asterisk and Cisco 79XX series configuration:
http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx
Christopher Jacob wrote:
I am in the process of ordering a support contract from Cisco for my new
7960 phone, but I would really like to get it up and running. At the risk of
being flamed off this
Just tried it with 7,10, and 11 digit dialing, and got the expected
error from the switch, "the number you have dialed is not a long
distance number, there is no need to dial the digit one before the
number..."
Good suggestion, but that doesn't appear to be the problem.
On Fri, 24 Sep 2004 11:25:
Hi.
Did you check out the plug? Is it wired
correctly?
I once had this problem because there was
two lines on that same plug (red/green and yellow/black) and plugging it into a
fxo would short them out and make then off-hook.
Michel Belleau
De :
[EMAIL PROTECTED]
[m
On Fri, 24 Sep 2004 10:27:03 -0500, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> I'm curious if there is a way I can do some kind of balancing if an
> outgoing
> connection is already being used?
> I was thinking about using the System command with a python script to
> keep
> an inventory of what
Cirelle did you delete the .version file in the src tree on your box?
I doubt cvs is 2 wks behind since I got cvs commit emails this
morning. I believe make update will remove the .verision for you too
which will fix that issue.
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