Try [EMAIL PROTECTED]
Thanks!
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> programming dept
> Sent: Friday, January 21, 2005 7:45 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Stumped on LD
Daniel Nyström wrote:
Hi again folks! ;)
As before, I will transform one E1 30 Channel PRI into 30 FXS channels using
Adit 600.
Now I'm into choosing server platform. And the two opponents are:
* Dell PowerEdge 750 w/ SCSI RAID (or even SATA RAID1)
* FujistuSiemens PRIMERGY RX100 S2 (SATA RAID1)
are you looking to do actual pstn to voip termination? if so, then you are
gonna need ss7, cama and imt trunks - things which asterisk doesnt
necessarily support.
now if you just want to buy pri/t1 from the local telco and sell voip
services off an asterisk server that gets back to the pstn over t
Just got my 7960 that I picked up
from ebay. It looks like it
has a SKINNY image instead of SIP.. where
can I get a SIP image ?
And how do I unlock the phone.. it is stuck at configuring IP, configuring CM based on the
old settings.. tried **# but nothing happens !
__
I think you need to look at a few other factors.
1. Some IP phones are really flakey (had some serious issues with a
couple of vendors MGCP Business line package).
2. Line power - Cisco uses one standard, other phones use another... but
Cisco is the 900# gorilla in the powered switch market... yo
> Which codecs do you use for the second call?
>
>
>> One limitation of the sipura 2000 is that you can not use both ports at
>> the same time with the G729 codec, I belive this may be due to the
>> sipura
>> having an smal CPU that can not handle the load of 2 G729 codecs.
>>
>> Other limitations
I have an stanaphone number for incoming calls, I am using asterisk and it
works fine at first, but after a few hours the incoming calls fail they go
directly to the stanaphone mailbox.
I believe the registration probably times out, or maybe it is the dynamic
IP changing.
Is there a way to fix th
On Thu, 20 Jan 2005 22:19:24 -0500, Nabeel Jafferali
<[EMAIL PROTECTED]> wrote:
> Hello.
>
> Short of buying a (no doubt) expensive one designed specifically for the
> Cisco 7960, what are my options for using headset with this phone? Is
> there some kind of adapter to buy so I can use standard
>
on 1/20/05 17:22, Ty Carter at [EMAIL PROTECTED] wrote:
> OK.. I'm up to my eyes in LD BS!
>
> I can't for the life of me understand how any carrier, either VoIP or
> traditional service provider can make heads or tails of how to hand off an *
> based call to an LD provider. Every provider I tal
Andrew Kohlsmith wrote:
On January 21, 2005 10:13 am, [EMAIL PROTECTED] wrote:
Hello, I have two TE-405Ps that I am having trouble with.
I'm using an Intel 865 motherboard with a Celeron D processor. Kernel
2.4.26, Slackware 10.0.
9: 0 XT-PIC t4xxp
That is a bad
Webmin modules are usually written in perl but there is always the
possibility that the perl invokes some compiled binaries. I would always
ask if 100% source is included. Otherwise you can end up paying for
something that someday prevents you from upgrading the overall linux system.
Don't forg
>
>
>
>
>>-Original Message-
>>From: [EMAIL PROTECTED]
> [mailto:asterisk-users->[EMAIL PROTECTED] On Behalf Of
> [EMAIL PROTECTED]
>>Sent: Friday, January 21, 2005 9:28 AM
>>To: Asterisk Users Mailing List - Non-Commercial Discussion
>>Subject: Re: [Asterisk-Users] Webmin Module for Asteri
Mike Dent wrote:
Hi Glenn,
What do you mean by "provisioning"?
loading the config files, with proxy servers, usernames, passwords, etc.
cheers,
glenn
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Florian Overkamp wrote:
You can even make your own adapter if it has to be really cheap :)
Instructions for making an adaptor:
http://www.mml.uni-hannover.de/einhorn/headset/index_e.html
cheers,
glenn
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I have a couple of DID's through VP Connect and have been having sound
quality issues on incoming calls. During the call, the calling parties
voice sometimes sound like it is crackling, in other words it is not
very crisp. I would liken it to listening to a radio with a blown
speaker. This sound de
Which codecs do you use for the second call?
One limitation of the sipura 2000 is that you can not use both ports at
the same time with the G729 codec, I belive this may be due to the sipura
having an smal CPU that can not handle the load of 2 G729 codecs.
Other limitations are the lack of GSM, an
Just add:
include => [your extensions context here]
inside your IVR context
Example:
include => default
On Fri, 21 Jan 2005 15:52:15 -0800, Michael Levenson
<[EMAIL PROTECTED]> wrote:
> I have searched and I have my IVR working when it has to fork off to another
> application but how do I g
yes
On Fri, 21 Jan 2005 17:38:27 -0600, Henry Devito <[EMAIL PROTECTED]> wrote:
> Hi, I have not implemented any of the spa-2000's yet. Do they work ok with
> asterisk? Is the 2000 capable of having 2 FXS extensions off each one or is
> it two fxs ports with the same extension?
>
> __
> Hi, I have not implemented any of the spa-2000's yet. Do they work ok
> with
> asterisk? Is the 2000 capable of having 2 FXS extensions off each one or
> is
> it two fxs ports with the same extension?
>
> ___
> Asterisk-Users mailing list
> Asterisk-
I have searched and I have my IVR working when it has to fork off to another
application but how do I get it to allow callers to dial the extension
directly instead of going though the directory?
[mainmenu]
exten => s,1,Answer
exten => s,2,SetMusicOnHold(default)
exten => s,3,DigitTimeout,5
fixed, sorry. (Theew was a mailer problem and I missed the err check.)
/ed
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jeff R
Glassman
Sent: Friday, January 21, 2005 6:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Aste
Hi, I have not implemented any of the spa-2000's yet. Do they work ok with
asterisk? Is the 2000 capable of having 2 FXS extensions off each one or is
it two fxs ports with the same extension?
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Same here!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael
Levenson
Sent: Friday, January 21, 2005 6:26 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Bellster - IAX-based interchange -- lets
youcallanyw
On Fri, 21 Jan 2005 09:42:15 -0800 (PST)
R A <[EMAIL PROTECTED]> wrote:
> Have some bady working asterisk with oracle?
I will use odbc.
I did use odbc with MySQL without problem.
regards
Charlie
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On Fri, 21 Jan 2005 15:26:06 -0800, Michael Levenson
<[EMAIL PROTECTED]> wrote:
> I get a 500 Internal Server Error when I try to register. :(
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Ed Guy
> Sent: Friday, January 21, 2005 2:55 PM
> To: As
I am having a problem compiling zaptel on my Arch Linux (linux 2.6.10) system:
CC [M] /home/sbn/src/zaptel-1.0.3/wcfxs.o
/home/sbn/src/zaptel-1.0.3/wcfxs.c: In function `wcfxs_interrupt':
/home/sbn/src/zaptel-1.0.3/wcfxs.c:473: sorry, unimplemented: inlining failed
in call to 'wcfxs_proslic_che
I get a 500 Internal Server Error when I try to register. :(
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed Guy
Sent: Friday, January 21, 2005 2:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Bellster - IAX
Ouch, sorry to hear that you bought 20 of those damn things.
I'd recommend you provision them up with a static. The IAXy doesn't
really use dhcp, it uses a mix breed of bootp and dhcp which works on
only certain implementations of dhcp with varying degrees of success.
Since it is acting more alon
On Fri, 21 Jan 2005 13:28:46 -0600, Leif Madsen wrote:
>On Fri, 21 Jan 2005 11:26:12 -0700, Steve Murphy <[EMAIL PROTECTED]> wrote:
>> I didn't get any response at all to my last "request for status" on
>> IAXTEL.
>>
>> So, when this happens, I attribute it to one of a number of things:
>>
>> 1.
I've a couple of standard telephone headsets here but as we know they
do not work with the Cisco headset socket.
Does anybody know the pinout of a standard telephone headset?
thanks
Mike
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Fellow Asterisk Users,
The original Free World Dialup vision was to facilitate world wide calling
using each others phone lines. The service grew significantly, but is
pretty
much limited to Internet calling and contacting other ITSPs.
Now, Jeff Pulver has created Bellster(tm) - Half Napster/Ha
I am not sure if this is the place for Digium user-to-user discussion, but...
We have deployed many (20+) IAXy's in the field. At a couple of locations, the IAXy's have just stopped working after 1 or 2 days use. No lights go on, no DHCP lease is renewed as far as we can tell, and of course no
Robert Rozman wrote:
Hi,
if I change Iaxphone settings to qualify=yes it says it's unreachable.
Can iax2 clients be monitored with qualify option ? Is this problem related
to iaxphone ?
Anyone sucessfully using iax qualify feature ?
It was just fixed in iaxclient-cvs this week, I think..
Don't k
This is not burn out. This is time to jump to a higher paying job
Brandon Patterson
> I'd guess he's tired of it. It is fairly rare to see people in his
> type of position to stay in it for more than 4-6 years. They just seem
> to burn out.
>
> It is too bad, however. In most cases, he was a fr
I'd guess he's tired of it. It is fairly rare to see people in his
type of position to stay in it for more than 4-6 years. They just seem
to burn out.
It is too bad, however. In most cases, he was a friend to VoIP.
Brian
On Fri, 21 Jan 2005 15:28:40 -0500, dean collins <[EMAIL PROTECTED]> wrote
> Jens, thanks for the feedback.
No problem - but I think I didn't help.
> >>I've added a ZAPHFC card to my CAPI based system. Calls coming in via
> >>ZAPHFC do not forward the caller id to the SIP phones. Calls coming in
> >>via CAPI do forward the caller id to the SIP phones.
>
> > I think
http://techstore.doit.wisc.edu/product.asp?login=D&itemnum=C36738
On Fri, 21 Jan 2005 14:44:23 -0500, Nabeel Jafferali
<[EMAIL PROTECTED]> wrote:
> > http://www.mml.uni-hannover.de/einhorn/headset/index_e.html
>
> Umm.. did you read my question? I said I had seen that page, but what do
> I do if
>-Original Message-
>From: [EMAIL PROTECTED]
[mailto:asterisk-users->[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
>Sent: Friday, January 21, 2005 9:28 AM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [Asterisk-Users] Webmin Module for Asterisk (and third
Make sure that in your SIPDefault.cnf you have
# Proxy Server (change for your settings)
proxy1_address: 192.168.6.10
# Proxy Server Port
proxy1_port: 5060
further comments inline:
Julian.
eamonn doyle wrote:
This I am sure is a very easy question, but I can't seem to find the
answer.
Here is the s
Hi, is there a way of rotate Mastr.csv in a daily basis?, i mean an asterisk
way, or should i use logrotate or similar?
thanks
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To UNSUB
We have a EuroISDN30, linked directly to * via a TE405P card, on span 3.
(span 1 is linked to another EuroISDN, and span 2 to a pbx, which we are
decommissioning).
After a couple of days, I notice that incoming calls are getting more
and more "busy" tones. These calls are not being handled by *
This I am sure is a very easy question, but I can't seem to find the
answer.
Here is the scenario:
cisco 7940g phone has SIP 6.3 firmware applied
the file SIP.cnf does not seem to have a place for it:
image_version: P0S3-06-3-00
#line 1 settings
line1_name: "5010" ; Line 1 Ext
you might try level3 or broadwing they are both very large carriers like
att/worldcom and provide voip termination
- Original Message -
From: "Ty Carter" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
; <[EMAIL PROTECTED]>
Sent: Thursday, January 20,
Hi again,
Should the include # iax_additional.conf have the # in front of it?
Here's the include...
[201]
username=201
type=friend
secret=password
qualify=no
notransfer=yes
mailbox=201
host=dynamic
context=from-internal
callerid="Mike"<201>
That's it...
Mike
- Original Message -
From
Looking to see if anyone has a WElltech 3804 Config that they would
be willing to share.
I will update/place them on the wiki if I get one that
works.
thanks
ron
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- Originele Bericht -
Van: Remco Barende
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Datum: Friday, 21 January 2005, 21:24
Onderwerp: Re: [Asterisk-Users] zaphfc no callerid incoming to SIP phone but visible in logfile
>>> Ive added a ZAPHFC card to my CAPI based system.
Hi,
if I change Iaxphone settings to qualify=yes it says it's unreachable.
Can iax2 clients be monitored with qualify option ? Is this problem related
to iaxphone ?
Anyone sucessfully using iax qualify feature ?
Regards,
Rob.
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Hi Mike -
I am using Firefly for the softphone (IAX option).
IAX.CONF
bindport=4569
bindaddr=0.0.0.0
delayreject=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
jitterbuffer=yes
mailboxdetail=yes
#include iax_additional.conf
Well, I don't see any definitions for devices. I'm assuming that
they'r
Stereorize was added to cvs head lastnight.
SetVar(MONITOR_EXEC=stereorize)
Before the monitor call and it will do it for ya too :)
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Jeffrey C. Ollie
> Sent: Friday, January 21, 20
I've added a ZAPHFC card to my CAPI based system. Calls coming in via
ZAPHFC do not forward the caller id to the SIP phones. Calls coming in
via CAPI do forward the caller id to the SIP phones.
-- Accepting call from '1729731418' to '807440' on channel 0/1, span 1
-- Executing Answer("Zap/1-1", "")
Breaking News - Powell
resigns as FCC Chair
Federal Communications Commission (FCC)
Chairman Michael Powell announced his resignation today (January 21, 2005) and
will step down in March. In a brief statement on the FCC's website (www.fcc.gov) Powell
said, "Having completed a bold and aggr
I am using Firefly for the softphone (IAX option).
IAX.CONF
bindport=4569
bindaddr=0.0.0.0
delayreject=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
jitterbuffer=yes
mailboxdetail=yes
#include iax_additional.conf
Hope this helps.
Mike
- Original Message -
From: Noah Miller <[E
On Sat, 2005-01-22 at 02:47, C F wrote:
> Any T extensions set?
Yes there are, but it not going down that path because they all do
things - like voicemail.
> Maybe autofallthrough=yes and absolutetimeout
Where would the first be set, and the second is not set anywhere.
>
>
> On Fri, 21 Jan 20
Jens, thanks for the feedback.
>>I've added a ZAPHFC card to my CAPI based system. Calls coming in via
>>ZAPHFC do not forward the caller id to the SIP phones. Calls coming in
>>via CAPI do forward the caller id to the SIP phones.
I think you didn't set usecallerid=yes in your zapata.conf?
Added i
Anyone has an example of how a working record for agress and rates
tables should look?
I'been trying all the thinkable patterns, obviously not the right ones,
for the last two days.
Tkx, LTenorio
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Well you have two options. Cut the end off the headset you have and
put dual jack plugs on it. Or buy a chassis socket to match your plug
and wire it up.
Guess you will have to do a little research to find out the pinout of that.
Mike
On Fri, 21 Jan 2005 14:44:23 -0500, Nabeel Jafferali
<[EMAIL
You are right that according to theri web site it does, however it
just doesn't work.
The following is an email I received from them:
>From me:
Thanks for your reply, I figured it out. However I have another
problem, is the Conference button suppose to work? Also when putting
someone on hold if I h
With the help of some people on IRC I've gotten further with using
GStreamer:
gst-launch interleave name=int ! audio/x-raw-float,buffer-frames=256 !
audioconvert ! wavenc ! filesink location=x.wav { filesrc location=vm-
youhave.gsm ! audio/x-gsm,rate=8000 ! gsmdec ! audioconvert ! buffer-
frames-c
Hi, this was my fault, you are right, i tried with a X-lite Professional and
the conference (3-way call) is working now, i guess the phone BT-100 doesnt
support it, i dont have a BT102D, so i can tell if it works too,
bye
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
I'm using T100P with CAC AB II, only FXS ports.
After upgrading, asterisk stoped sending caller IDs to the phones.
Even inside - port to port.
I got 2 errors in the debug:
__zt_exception: Exception 23, channle 2 (i'm ringing to channel 2)
zt_handle_event: Didn't finish Caller-ID spill. Canceling.
_
I have a toll-free number inbound from sixTel. This gets answered by my IVR
system. If they choose "technical support", I have it dial my SIP phone
(SJPhone) and my cell phone (through sixTel) at the same time. If I answer
on the SIP phone, all is well. If I try to answer on my cell phone, they
I connect to the PSTN using cisco as5400 gateways, this cisco devices have
E1's to a DMS300 switch. I mean, i configured sip channels (in and oout) to
these gateways, i dont have any special hardware in the asterisk server.
thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL P
> http://www.mml.uni-hannover.de/einhorn/headset/index_e.html
Umm.. did you read my question? I said I had seen that page, but what do
I do if I want to use a single 2.5mm jack headset (aka cell phone
headset), as opposed to a dual 3.5mm jack headset (aka computer
headset)?
Thanks anyways.
--
N
Chirstopher,
What version of PIX software are you using? I had to make no special NAT
configuration on the * server when using a PIX 525 firewall. I was
actually impressed that it is a completely SIP aware firewall in that it
will handle all the header re-writing. I am using PIX version 6.4 with
On Fri, 2005-01-21 at 10:42 -0500, Nabeel Jafferali wrote:
> do you know how I could make that adapter if I
> wanted to use a single 2.5mm connector headset (like the kind used with
> cellphones and cordless phones)? Any idea what the pinout for that would
> be?
>
http://www.mml.uni-hannover.de/e
Paul Rodan wrote:
The BT100's do support conferencing, most SIP phones do. But how does your
Asterisk connect you to the PSTN? Through a Zap interface? If so, what kind;
or through a VoIP provider like BroadVoice, NuFone, LookieLoo, VoipJet,
VoicePulse?
You basically need to make sure your Asteri
On Fri, 21 Jan 2005 11:26:12 -0700, Steve Murphy <[EMAIL PROTECTED]> wrote:
> I didn't get any response at all to my last "request for status" on
> IAXTEL.
>
> So, when this happens, I attribute it to one of a number of things:
>
> 1. No-one knows.
> 2. No-one cares.
> 3. Everyone knows, but are
I've been trying to setup asterisk with an Internet Line Jack card for
sometime. I've been successful in configuring asterisk to handle incoming
calls, make calls between sip phones, call the asterisk demo, and even
answer the phone with a sip client. I've been using x-lite as a sip
client for my
Mark had made a post recently (last week or so maybe) -- could have been in
IRC too... (it starts to blur together) that he was aware of the IAXTEL
problems and that they were working on the issues.
Details are hazy... But then I drink alot too, so everything is hazy...
(that's the point)
Tom
Hi All!
Let me explain the problem. When using the Originate
command from the manager api, the dialstatus variable returns results
for whichever phone picks up first, and in this case it is the IAX/2
connection. It doesn't matter if Zap/G2/XXX is set as the channel,
or an extension either.
The BT100's do support conferencing, most SIP phones do. But how does your
Asterisk connect you to the PSTN? Through a Zap interface? If so, what kind;
or through a VoIP provider like BroadVoice, NuFone, LookieLoo, VoipJet,
VoicePulse?
You basically need to make sure your Asterisk server has acce
Title: Message
I have
it running on my Windows 2000 mahcine using STI products and don't have much of
a problem. I would guess that it might be something on the workstations instead
of the AstTAPI. Also, might be a little faster, easier, cheaper to just upgrade
your existing workstations? Ju
[EMAIL PROTECTED] wrote:
Hi, i cant make a three way call using grandstream phones (BT-100) and
asterisk using sip, is this supported or i need a zap interface?
The BT101 cannot to supervised transfers or 3-way calling.
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Ast
Just got a reply from them. $300 Commercial $100 personal. They gave me
a login to look at the demo. Didn't see anything I wanted to buy.
David
On Fri, 2005-01-21 at 09:34, C F wrote:
> Interesting, because I also called them, and I was able to get a
> price, they told me $300 per license, for bu
Sorry about the previous HTML posts, keep forgetting to change the default
in my email program.
I just wanted to let any potential developer know that this is a different
offer than the other one about an Agents web interface. I work for multiple
companies, and 1 request is for 1 company, and anot
Folks,
Has anyone connected Asterisk to a peer that uses multiple IP's?
Here's my situation:
Signalling IP - 1 address.
RTP - 2 addresses.
for a total of 3 IP's, all on the same subnet.
Thanks in advance,
...Paul
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I didn't get any response at all to my last "request for status" on
IAXTEL.
So, when this happens, I attribute it to one of a number of things:
1. No-one knows.
2. No-one cares.
3. Everyone knows, but are too busy to reply.
At any rate, my investigative side kicks in and I began searching thru
Hi, i cant make a three way call using grandstream phones (BT-100) and
asterisk using sip, is this supported or i need a zap interface?
thanks
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Looks like these features and more developer services are needed, not all
Asterisk related.
One of my employers has authorized me to look for a developer capable of
programming in Java/Perl/PHP (of course C, C++ and knowledge of Asterisk is
a huge plus) so if you're looking for part-time work (wit
On Fri, 2005-01-21 at 09:55 -0800, Asterisk List wrote:
> On Thu, 20 Jan 2005 22:41:08 -0600, Steven Critchfield
> <[EMAIL PROTECTED]> wrote:
> >
> > Shouldn't you contact your vendor for support and not a different
> > vendors support channel?
> >
> As far I know, although Digium hosts the aster
Chris,
Wanted to give you some insight on how my Asterisk is setup behind
by PIX. It works great with remote SIP UA's registering to
Asterisk on the Public IP address, or behind VPN.
I have Fixup protocol enabled on TCP and UDP, just to be safe ;-)
fixup protocol sip 5060
fixup protocol sip udp
Hi,
I think you didn't set usecallerid=yes in your zapata.conf?
Another way is to set the callerid in your extensions.conf via
exten => 807440,2,SetCIDNum(0${CALLERIDNUM}). So you also have a 0 in
front of the displayed number - nice for callback.
regards
Jens
> Hello,
>
> I've added a ZAPHFC
No word on this post. Please, can anybody
help me? Is there a known issue with AstTAPI and Windows 2000? Or AstTAPI w/
Amicus Attorney?
If we’re willing to hire a developer
to help us fix AstTAPI, is there a developer out there willing to help us? Cost
is of less importance than time ri
Hi,
> -Original Message-
> > You can even make your own adapter if it has to be really cheap :)
>
> I saw that on the Wiki a few moments after I posted the initial query,
> but I had a question: do you know how I could make that adapter if I
> wanted to use a single 2.5mm connector heads
A quick patch test yields a mixture of hunks failing and succeeding. I
guess my approach will be to build it and try it with broadvoice first.
I certainly don't want to spend any time analyzing code that is not needed.
I already have a 1.0.2 debian package set that works with broadvoice. I
gues
Did you made changes to sipsak or to the Nagios plugin?
In case of sipsak would think about submitting the patches to the author (me)
to make it public available for everybody?
Hi Nils,
The changes were to the Nagios plugin. I have added it to the Wiki at:
http://www.voip-info.org/tiki-index.
On Thu, 20 Jan 2005 22:41:08 -0600, Steven Critchfield
<[EMAIL PROTECTED]> wrote:
>
> Shouldn't you contact your vendor for support and not a different
> vendors support channel?
>
As far I know, although Digium hosts the asterisk-users list and
supports the Asterisk development, Asterisk is stil
Hi!
There's any way to set up a call using
G726 (sip peer) receive it on Asterisk convert it to G711Mu to send it
to PSTN broadband termination?
I've put the following in
sip.conf:
disalow=all
allow=gsm
allow=g726 (my TAs use G726
32K)
best regards,
Helder
___
Hello,
I've added a ZAPHFC card to my CAPI based system. Calls coming in via
ZAPHFC do not forward the caller id to the SIP phones. Calls coming in
via CAPI do forward the caller id to the SIP phones.
Any and all help is greatly appreciated.
The (hopefully relevant) conf file excerpts are:
extens
Hi all
Have some bady working asterisk with oracle?
thanks in advance
wert
Do you Yahoo!?
Yahoo! Search presents - Jib Jab's 'Second Term'--- Begin Message ---
Hi all
Have some bady working asterisk with oracle for the CDR?
thanks in advance
wert
--- End Message ---
At 07:22 AM 1/21/2005 -0600, you wrote:
I'm having the exact same issue on a brand new Dell Poweredge 700, using
FC2. It locks the machine totally.
-Original Message-
From: Michael Loftis [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 20, 2005 7:06 PM
To: Asterisk Users Mailing List - No
What are the advantages in using mISDN over other solutions?
If I knew why it was a good idea (like does it have better sound quality than
alternatives?) then I would put the time in to test it, and also improve the
Wiki.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTE
Hello world.
A Colombian LUG has published an article written by Diego A. Asenjo G.,
a young engineer from the also young VoIP enterprise Avatar ltda.
(http://www.avatar.com.co), about DeStar, a web frontend for Asterisk.
The article pretends to inform about this project and atract some users
an
Interesting, because I also called them, and I was able to get a
price, they told me $300 per license, for bulk every 5th license is
free.
On Fri, 21 Jan 2005 11:43:13 -0500, Ferguson, Michael
<[EMAIL PROTECTED]> wrote:
> Same here.
> I called them yesterday plus email and still no reply.
>
> --
Would you please be so kind and share your config and/or phones used,
with the list so we can help you?
On Fri, 21 Jan 2005 12:07:57 -0500, [EMAIL PROTECTED]
<[EMAIL PROTECTED]> wrote:
> Hi,
>
> I hope this isn't a double-post...but here goes. I have setup an * box using
> WBEL, and I have * up
On Fri, 2005-01-21 at 09:23 -0700, Mike Dewey wrote:
> morn all,
> I am trying to compile 1.0.4 and SpanDSPpre10 to compile and have problems
> patching the apps makefile. Anyone want to help a little?
>
> # patch < apps_makefile.patch
> patching file Makefile
> Hunk #1 succeeded at 41 (offse
Apply the patch manually. The changes are not that significant.
-Brian
On Fri, 21 Jan 2005 at 09:23 Mike Dewey ([EMAIL PROTECTED]) wrote:
> morn all,
> I am trying to compile 1.0.4 and SpanDSPpre10 to compile and have problems
> patching the apps makefile. Anyone want to help a little?
>
Hi Mike -
I hope this isn't a double-post...but here goes. I have setup an * box
using WBEL, and I have * up and running. The problem I have is that
when I dial an extension I cannot hear anything. It's not my sound
card either. I can see the call going through on the CLI and I see
where it goe
I want to put a single voice-mail box on a remote server, where I have
metered bandwidth. Before I do this, I want to make sure it's feasible.
Could someone confirm the following math for me?
G.711, at 64kpbs has a rated network load of 88kbps.
So for each second of conversation, about 11KB are
Andrew Kohlsmith wrote:
Thunderbird, Eudora, hell even Pine I think.
Thunderbird works very well but you have to enable it, since it doesn't do it
by default. View -> Sort by -> Threaded
-A.
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