Hello all,
I use Fedora 3, and I have compilation
problem of SpanDsp , when installaing spandsp, tiff warnings and errors generated and libunicall generate unresolved
references.
Is
the problem in my libtiff installation, spandsp.h or in tif_dir.h and tiffiop.h?
Are tif_dir.h and
Thanks.
How do I implement this command, could not see much info on the wiki about how I
actully use it?
Mike
On Tue, 07 Dec 2004 16:45:51 -0600, Christopher L. Wade
[EMAIL PROTECTED] wrote:
Mike Dent wrote:
Hi,
now I've got caller id working on my BT line in the UK, I'd like to
I am having lots of problems with dropped calls from my asterisk.
My Setup is:
Dell PowerEdge 1750 with 2.4Ghz Single Zeon
1GB Ram
2 x 36 GB SCSI Drives
The server is co-located
I connect to my providers via uLaw and IAX2.
My remote users are mainly are IAX2 softphone users.
I also have a DDI
I don't know if this would help. But the latest version have two login
options now.
One is the administrator and the other one is a simple user.
If you're not seeing the other forms there probably you're logged in as
a simple user.
HTH
-Original Message-
From: [EMAIL PROTECTED]
-Original Message-
From: Dan Goscomb [mailto:[EMAIL PROTECTED]
Sent: 07 December 2004 15:38
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] SIP URLs
I have set up an asterisk server and can successfully call
between extensions using SIP.
i wish to be able to call other sip
Cheers
i realised this last night and now have SER set up
i can call between phones on SER, and to extensions handled on asterisk
(for example voicemail). However... if i dial an extension which used to
be assigned to a SIP phone, it tells me the user is on the phone... is
there any way to get
Hi,
I use Fedora 3, and I have compilation problem of SpanDsp , when
installaing spandsp, tiff warnings and errors generated and libunicall
generate unresolved references.
Which is the same reason why spandsp is not packaged for Debian yet. The
p in tiffiop.h means private, i.e. the t4.c
Hello *'s,
I have recently installed CentOS v3.3 and i have latest stable
Asterisk's source code ,i compiles it shows no error but when i am
looking for sip.conf,zapata.conf ,i am amazed the /etc/asterisk folder
was empty i am compling several times but no luck what's the problem i
compiled in the
-Original Message-
From: Adnan Ahmed [mailto:[EMAIL PROTECTED]
Sent: 08 December 2004 10:52
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk's Empty Folder
Hello *'s,
I have recently installed CentOS v3.3 and i have latest
stable Asterisk's source code ,i compiles it
You need to do a:
make install
and then
make samples
to install sample conf files.
Jim
On Wed, 8 Dec 2004, Adnan Ahmed wrote:
Hello *'s,
I have recently installed CentOS v3.3 and i have latest stable
Asterisk's source code ,i compiles it shows no error but when i am
looking for
Yes and no, what if Asterisk itself crashes? You would have a cluster full
of machines, but no running server :)
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Kanwar Ranbir Sandhu
Verzonden: dinsdag 7 december 2004 22:02
Aan: Asterisk Users Mailing List
Jim Radford wrote:
You need to do a:
make install
and then
make samples
to install sample conf files.
Jim
On Wed, 8 Dec 2004, Adnan Ahmed wrote:
Hello *'s,
I have recently installed CentOS v3.3 and i have latest stable
Asterisk's source code ,i compiles it shows no error but when i am
hi:
i would greatly appreciate it if somebody can refer me
to asterisk consultants.
__
Do you Yahoo!?
The all-new My Yahoo! - What will yours do?
http://my.yahoo.com
___
Asterisk-Users mailing list
It should be a mpg123 problem, not a spandsp problem.
Stop asterisk, make clean, make install and start asterisk again.
Have fun.
Ariel Batista [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]...
I have compiled Spandsp without any problems. I got no errors I have also
done the patch
Network Partners, Inc.
You can find them at http://www.routers.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of chawki
hammoud
Sent: Wednesday, December 08, 2004 5:34 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] asterisk consultants
hi:
i
Yup. Asterisk was built to handle the media stream, what you are looking
for is aproxy, which is what SER is best at doing. But then you may have
a problem in billing.
Iqbal
Tracy R Reed wrote:
On Tue, Dec 07, 2004 at 08:44:50PM -0800, Gonzalo Gasca Meza spake thusly:
I have just setup
hi
for call forwarding, I was told by the telco to set the call forward
number on the PRI,
how can I do this?
roy
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options
Hi,
I've got a big problem.
I have 2 ISDN cards windbond I can make calls from sip phones on
ISDN lines but only from one card channel B1 to isdn phone,
and what ever ttyI* I choose only one of cards is choosen.
How to bind /dev/ttyI to another ISDN card?
Kind Regards,
C.
On December 8, 2004 02:40 am, Altus Snyman wrote:
Is there someone that's got asterisk working well with a A101/E1 card
Apparently they don't have RBS support?
I'm using an A101u and it seems to work fine connected to a Carrier Access
Access Bank I (24 FXS).
-A.
On December 7, 2004 05:35 pm, Philipp von Klitzing wrote:
With 24 analog extensions 2 PRI seems rather unlikely... :-) So let's
assume 2xBRI = max. 4 simulatenous calls.
Yes that was a bit of a stupid question. :-)
As Peter S. pointed out I think he really wants to go for a Quad-BRI card
I want to use it in my pbx as a bri card for incoming and outgoing calls
in asterisk
How did you get it working with asterisk,drivers ens.
Please Help
Andrew Kohlsmith wrote:
On December 8, 2004 02:40 am, Altus Snyman wrote:
Is there someone that's got asterisk working well with
If there are any Vonage employees on this list, I would like to
discuss a non Asterisk issue with them. Please contact me
directly.
..don
support at microtechniques.com
White Plains, NY
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[EMAIL PROTECTED]
I would be grateful if anybody could tell me what I should
tell Verizon
in NJ so they would enable disconnect supervision for my lines.
Apparently remote hangup notification or disconnect
supervision or
calling party control is NOT the magic phrase for them. Although
disconnect
If you really added /var/local/lib, that would eb your trouble. It
should be /usr/local/lib.
Steve
Ariel Batista wrote:
I have compiled Spandsp without any problems. I got no errors I have
also done the patch without getting any error. I have tried pre4 and
pre6 version with same problem. I
Does the IAXy support AutoDial on off-hook?
It seems like it easily can since when you take the phone off hook
asterisk recognizes it. I got this device here that's supposed to
auto dial but it doesnt consistently work.
Thanks.
--
MBM
___
I guess this topic is now closed. The problem was that Asterisk 0.7.2 (yeah
yeah, this is a busy production system) has buggy handling of overlapping
extensions ( 224 versus 2246 ). Asterisk 1.0.3 allows such numbers to
coexist and -does- wait to see if the dialing is completed.
In
What that suggests is that dtmfmode=inband is being used by your *
system for outbound calls. BV will use whatever their default happens
to be when sending tones to you (or simply passing them via the
audio channel).
Someone else posted something not to long ago relative to a cell
provider
I would be grateful if anybody could tell me what I should
tell Verizon
in NJ so they would enable disconnect supervision for my lines.
Apparently remote hangup notification or disconnect
supervision or
calling party control is NOT the magic phrase for them. Although
chawki hammoud wrote:
hi:
i would greatly appreciate it if somebody can refer me
to asterisk consultants.
Hi,
Here you are
http://www.voip-info.org/wiki-Asterisk+consultants+Europe
Laurent
--
[EMAIL PROTECTED]
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On the surface, that sounds like an * problem, not sprint.
What are you using to interface to sprint (analog, bri, T1), which
cards in your * box (and associated config files)?
A fairly standard telco operating approach is _not_ to
provide any answer
supervision, and * works just fine
[ == snip ==]
I have used the same settings as you have, and I have no problem
connecting various sip devices with different codecs to my Asterisk server.
In fact, I use XTen Lite and I can disable/renable codecs as I please. Under
[general], I have disallow=all, allow=g729, allow=gsm. SIP
Hi
Here in the UK telephone numbers vary in length. When PrivacyManager kicks in
it seems to only listen for the first 10 digits.
Is it possible to have it take any number of digits followed by # to
indicate the end of the number?
Thanks
Mike
___
Reid Forrest wrote:
On the surface, that sounds like an * problem, not sprint.
What are you using to interface to sprint (analog, bri, T1), which
cards in your * box (and associated config files)?
A fairly standard telco operating approach is _not_ to
provide any answer
supervision, and * works
On the surface, that sounds like an * problem, not sprint.
What are you using to interface to sprint (analog, bri, T1), which
cards in your * box (and associated config files)?
A fairly standard telco operating approach is _not_ to
provide any answer
supervision, and * works
I seem to remember reading some patches on asterisk-cvs that actually
corrected the protocol selection logic and made it work the way it
should.
Just wanted to let people know about it in case they have other
problems.
Yes, those patches had two major effects:
Where does one get these
One of the suggestions I got for dealing with my analog FXO woes was to
look into using a Voicetronix card instead of the Digium. They're more
expensive but if it worked better it may be reasonable. They indicated
they were compatable with the CVS HEAD version of *.
Can someone share with me
One of the suggestions I got for dealing with my analog FXO woes was to
look into using a Voicetronix card instead of the Digium. They're more
expensive but if it worked better it may be reasonable. They indicated
they were compatable with the CVS HEAD version of *.
Can someone share with me
It's an analog POTS line connected to a TDM400 interface in
the * box. Config
files are set for kewlstart.
I've never heard of this problem. Analog FXO ports on Asterisk are
considered answered when Asterisk finishes sending the DTMF.
Do you have callprogress=yes in
Matthew Boehm wrote:
Where does one get these patches?
They are in CVS HEAD, and I don't think they will apply to the 1.0.x
series... there are too many changes.
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[EMAIL PROTECTED]
Hi,
--- Norman Zhang [EMAIL PROTECTED] wrote:
I'm following, http://www.voip-info.org/wiki-Asterisk+phone+sjphone.
However, I cannot find the SIP tab. Would someone please give me a few
pointers? The screen capture can be seen at URL below
The wiki talks about an older version of SJPhone.
On December 8, 2004 07:52 am, Altus Snyman wrote:
I want to use it in my pbx as a bri card for incoming and outgoing calls
in asterisk
How did you get it working with asterisk,drivers ens.
The A101u is a T1/PRI card, not BRI.
-A.
___
Asterisk-Users
Kevin P. Fleming wrote:
Matthew Boehm wrote:
Where does one get these patches?
They are in CVS HEAD, and I don't think they will apply to the 1.0.x
series... there are too many changes.
cvs co -r v1-0 asterisk # will get you 1.0.3 plus the patches (including
the codec case problem)
--
I am
Hello
I'm planning using meetme module with SIP videophones (Videotel Hard
Phones), I have tried the meetme module with the v option but I haven't get
any video... can someone get the video mode running in meetme ?
Thanks a lot
Nicolas
___
In addition to the suggestions with the quality of the phone lines, I had
the same problem when I had an IRQ conflict with two cards in the same *
box. I actually fixed the problem by just swapping the cards between the two
slots they were in. Two TDM cards and just flipped them and they both now
Ok, thanks to Clive, got this working, however what I would like it to
do is if the
PrivacyManager application kicks in, is only let the user leave a VM
(after entering
their number)
BUT if they do not activate PrivacyManager the call is handled differently?
Thanks
Mike
On Wed, 8 Dec 2004
On Wed, 8 Dec 2004, Rich Adamson wrote:
A fairly standard telco operating approach is _not_ to provide any answer
supervision, and * works just fine without it for lots of folks.
Answer supervision allows you to dial several recipients at the same time
and bridge the first one to answer.
Dec 8 10:10:54 NOTICE[620]: chan_zap.c:7494 pri_dchannel: PRI got event: HDLC
Bad FCS (8) on Primary D-channel of span 1
Dec 8 10:10:54 NOTICE[620]: chan_zap.c:7494 pri_dchannel: PRI got event: HDLC
Bad FCS (8) on Primary D-channel of span 1
Dec 8 10:10:54 WARNING[620]: chan_zap.c:7246
Hi guys.
You're doing a great work.
A couple of questions.
We're going to install Asterisk in our branch office.
The configuration will start with 5 PSTN lines, 11 analog telephones and
four voip phones.
I'd very much like to use a channel-bank for connecting the analog lines
and phones to
Christopher L. Wade schrieb:
Please understand that Digium (Mark) is the reason * exists.
That's clear to me.
Selling a $10 winmodem for $80 is the reason Digium exists.
But the difference between $10 and $80 is really to much for me for just
playing around like I'm doing with asterisk. If I would
Hi
We use the Voicetronix cards in SA and have tested them
extensively against the Digium.
I found that the Digium has many more settings that you may
tweak to get the best out of it, but have found the Voicetronix
to perform allot better. The Voicetronix has an onboard DSP which
helps with the
On December 7, 2004 11:42 am, Andrew McRory wrote:
For a few weeks we have been getting errors that drop our PRI. The telco
says the the line is clean and that our equipment is the problem. We're
currently running Asterisk CVS-HEAD-12/03/04 but several versions
have been tried in an attempt to
for call forwarding, I was told by the telco to set the call forward
number on the PRI,
how can I do this?
To answer my own message, I need to set the REDGNO (0x74) number to the
originating number in the PRI SETUP. Example can be found here:
http://pastebin.ca/2783
Does anyone know how I can
Hi,
I'm new to the post and am looking at ways to see how i can get Asterisk
into my office for VOIP integration and also as backup PBX system.
So I want to know if Asterisk works with 3COM 1102 phones. This will
turn out to be a key decision point.
Any help is greatly appreciated.
regards
On Wed, 2004-12-08 at 03:34 -0800, chawki hammoud wrote:
hi:
i would greatly appreciate it if somebody can refer me
to asterisk consultants.
Please search the wiki at voip-info.org first:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20consultants
Regards,
Patrick
I hear ya. I guess what I was getting at was that I couldn't see much
hope of getting Asterisk to natively support proprietary sets,
It is not that I want to use proprietary phones, I want the benefits that
native digital phones will provide. This will allow Asterisk to work more
effectively
Hello
I'm planning using meetme module with SIP videophones (Videotel Hard
Phones), I have tried the meetme module with the v option but I haven't get
any video... can someone get the video mode running in meetme ?
Thanks a lot
Nicolas
SIP Hard VideoPhones
http://www.call.fr
In fact Sangoma does have full RBS support. If you require assistance in
configuring please contact Sangoma at 905-474-1990 X 118 or
[EMAIL PROTECTED]
Doug Vilim
Sangoma Technologies
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Altus Snyman
Sent:
Hi,
I'm new to the post and am looking at ways to see how i can get Asterisk
into my office for VOIP integration and also as backup PBX system.
So I want to know if Asterisk works with 3COM 1102 phones. This will
turn out to be a key decision point.
Any help is greatly appreciated.
regards
In the process of turning up a new pri. Zttool indicates the T1 is
ready with no alarms.
asterisk*CLI pri show span 1
Primary D-channel: 24
Status: Provisioned, Down, Active
Switchtype: National ISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
T200
It's the RTP Stream
Asterisk by default uses ports UDP 10,000 to 20,000
RTP = Audio
Open them on your firewall.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Aken
Sent: 07 December 2004 15:21
To: Asterisk Users Mailing List - Non-Commercial
I'm going to install asterisk with four digium cards.
Can anyone mention a brand that carries boards with 4 compatible pci
slots?
Thanks
Shoval Tomer,
IT Manager,
SofTov Advanced Systems, Ltd.
Office: +972-3-9230686 ext. 179
Fax: +972-3-9216642
Mobile: +972-54-8000200
Hi Nicolas,
There doesn't seem to be any interest in using asterisk and video.
I posted a $1,000 bounty to get video meet me working without a single
reply.
I have now just bumped this to $2000
http://www.voip-info.org/tiki-index.php?page=Asterisk+bounty+Meet+Me+vid
eo+conferencing
This is a
On December 8, 2004 10:56 am, Rich Adamson wrote:
Status: Provisioned, Down, Active
What does the Down mean in the above status line?
It means that your D channel isn't up, this is normal if * isn't running (or
the telco hasn't turned up the D channel yet)
-A.
I thought the standard for the UK was 11 Digits in length, (save some old
0845, 0800, 0870 numbers), but most of these are transported to normal 11
digit numbers.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mike Dent
Sent: 08 December 2004
Has anyone seen an issue with SIP phone (polycom 500)
dropping calls at irregular intervals with no errors in the asterisk log files?
I am having this issue as described and it is a complete pain in my rear to
trouble shoot because when I call my cell phone I can get a call to last over
30
I'm having a similar issue to this in that the USB ports on the system are
ohci based, not uhci, therefore ztdummy will not work. My system is also
running the SMP kernel so zaprtc will not work.
To me it looks like the only good solution is a hardware timer, even if it's
as simple as an x100p
In article [EMAIL PROTECTED],
Peter Svensson [EMAIL PROTECTED] wrote:
The RHEL 3 (and thus WhiteBox, TaoLinux etc) kernels also work well.
Are they 2.4 or 2.6 kernels?
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] -
Brian West wrote:
Not to mention that allow=all works now too. Not to mention per user/peer
codec prefs.
Well, I did mention the new prefs support, but you're right, I forgot
about allow=all now working properly as well.
Basically, it now works the way it was always expected to work :-)
Thanks
David J Carter [EMAIL PROTECTED] wrote:
I thought the standard for the UK was 11 Digits in length, (save
some old 0845, 0800, 0870 numbers), but most of these are
transported to normal 11 digit numbers.
The UK number plan contains 8, 10 and 11 digit numbers if you count
the leading zero access
On Wed, 2004-12-08 at 18:07 +0200, Shoval Tomer wrote:
I'm going to install asterisk with four digium cards.
Can anyone mention a brand that carries boards with 4 compatible pci
slots?
What 4 cards are you thinking of installing? Most people seem to run
into trouble after the second concurrent
On Wednesday 08 December 2004 16:51, David J Carter wrote:
I thought the standard for the UK was 11 Digits in length, (save some old
0845, 0800, 0870 numbers), but most of these are transported to normal 11
digit numbers.
there are lots of areas in the UK where there are still 5 digit
I'm going to install asterisk with four digium cards.
Can anyone mention a brand that carries boards with 4 compatible pci
slots?
Why ???
if your wanting the pci 4 port fxo/fxs boards you should be looking at
a t1 card channel bank, its less expensive far superior quality
chawki hammoud wrote:
hi:
i would greatly appreciate it if somebody can refer me
to asterisk consultants.
Chawki,
If you look around, I think most of the people on this list are
Asterisk consultants!
http://www.voip-info.org/wiki-Asterisk+consultants
--
Kristian Kielhofner
Hi Shoval -
I'm going to install asterisk with four digium cards.
Can anyone mention a brand that carries boards with 4 compatible pci
slots?
I ended up getting a Dell PE1600SC. It is a tower computer with 6 PCI
slots (2 32-bit, 2 PCI-X, 2 64-bit), but it can be rack-mounted with a
special kit
On December 7, 2004 11:42 am, Andrew McRory wrote:
For a few weeks we have been getting errors that drop our PRI. The telco
says the the line is clean and that our equipment is the problem. We're
currently running Asterisk CVS-HEAD-12/03/04 but several versions
have been tried in an
Garrett Smith
Sales Executive
[EMAIL PROTECTED]
B2 Technologies
454 Sonwil Drive
Buffalo, NY 14225
(716) 250-3408 Direct
(716) 630-1548 Fax
(716) 903-9495 Cell
AOL IM: B2sales
Specializing
in New and Used equipment from vendors including Cisco Systems, Juniper,
-Original Message-
From: Steven Critchfield [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 08, 2004 6:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] pc
On Wed, 2004-12-08 at 18:07 +0200, Shoval Tomer wrote:
I'm going to
Simply because I'm afraid of buying a used channel-bank on ebay.
Never seen one up close.
I'm quite certain I can handle installing 4 PCI cards in linux better
then I can figure out how to work with a channel-bank.
Sure hope digium will pick up this gauntlet (provide a way to connect
numerous
Sorry for the slightly OT question...
Anybody have configuration files for a Polycom IP 400 MGCP version, or
be able to point me in the right direction for being able to create them?
Thanks!
Kris
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[EMAIL PROTECTED]
On Wed, 2004-12-08 at 19:09 +0200, Shoval Tomer wrote:
-Original Message-
From: Steven Critchfield [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 08, 2004 6:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] pc
On Wed,
On Wed, 2004-12-08 at 19:22 +0200, Shoval Tomer wrote:
Simply because I'm afraid of buying a used channel-bank on ebay.
Never seen one up close.
I'm quite certain I can handle installing 4 PCI cards in linux better
then I can figure out how to work with a channel-bank.
Sure hope digium
Can you recommend a channel bank make and model that will support all
(or most) of Asterisk's features and can be installed by a newbie?
I'll search for it on ebay
-Original Message-
From: Steven Critchfield [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 08, 2004 7:22 PM
To:
Hi,
I bought the Digium Asterisk Developers kit a few
weeks ago, got the x101p and a tdm400p cards,
both seemed to be working fine up until today, now
the tdm400p seems to have gone to the great PBX in the sky ..
If I left the reciever on the analog phone I don't
get a dial tone and
On December 8, 2004 10:56 am, Rich Adamson wrote:
Status: Provisioned, Down, Active
What does the Down mean in the above status line?
It means that your D channel isn't up, this is normal if * isn't running (or
the telco hasn't turned up the D channel yet)
Okay, the d-chan is now up.
Have you tried different phones? This sounds a little like an issue I
had with older Siemens phones we had around and since you appear to be
in Europe you could be seeing similar problems. Oh, these worked
intermittently.
They work now with a patch that changes ringing frequency in wcfxs.c
On December 8, 2004 12:45 pm, Rich Adamson wrote:
It would appear the call failed due to Info. element nonexist or
Invalid number format.
is your pridialplan=unknown and do you know how your telco is expecting calls
from you? I've seen some telcos refuse calls if Caller ID Name was set too
Rich Adamson wrote:
On December 8, 2004 10:56 am, Rich Adamson wrote:
Status: Provisioned, Down, Active
What does the Down mean in the above status line?
It means that your D channel isn't up, this is normal if * isn't running (or
the telco hasn't turned up the D channel yet)
Okay, the d-chan
Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
User (0)
Ext: 1 Cause: Info. element nonexist or not implemented
(99), class =
Looks like the switch on the telco end might be confused about the number
pattern; try adding:
pridialplan=unknown
to
Colin Anderson wrote:
Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location:
User (0)
Ext: 1 Cause: Info. element nonexist or not implemented
(99), class =
Looks like the switch on the telco end might be confused about the number
pattern; try adding:
Ok, So I removed the Qualify=2000 from my
sip.conf file and set NAT=no and I changed the connection negotiation in my
Polycom config to 3600 (default) and now I havent had a dropped call
yet. Does anyone know which of these if any might be the cause or if there is
something else I am still
I don't really understand why do many people set their pridialplan. The
text PRI Dialplan: Only RARELY used for PRI. in zapata.conf.sample is
not there to just take up space in the file.
Some telco's do, some telco's don't. The text Rarely used for PRI is IMO a
location-centric and
Why ???
if your wanting the pci 4 port fxo/fxs boards you should be looking at
a t1 card channel bank, its less expensive far superior quality
Simply because I'm afraid of buying a used channel-bank on ebay.
Never seen one up close.
I'm quite certain I can handle installing 4 PCI cards in linux
Colin Anderson wrote:
Some telco's do, some telco's don't. The text Rarely used for PRI is IMO a
location-centric and carrier-centric statement. As another poster suggested,
PRI's are picky, and some carriers are super-picky so it's always best to
dot your i's and cross your t's when dealing with
Has anyone used 3COM 1102 with Asterisk.
regards
KIRAN
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How can you set sip accounts into groups? And also, when you dial a group
such as Zap/g1 how can you change the behavior of the calling? Can you make
it force all phones in that group to ring at once? Also if sip accounts
cant be set in groups is there any other simple trick to force a ring
Kiran Kumar Vangaveti wrote:
Has anyone used 3COM 1102 with Asterisk.
Asking three times in one day isn't going to get you an answer.
Try the mailing list archives:
http://www.google.com/search?hl=enlr=q=site%3Alists.digium.com+3COM+phonebtnG=Search
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I'd have to agree with the channel bank plan. You don't even need to
buy a used channel bank. You can get a new Rhino for $1500:
http://www.channelbanks.com/
I gather you've had experience with Rhino.
How does it work with Asterisk?
Does it provide all features? Caller-id, echo
I just set up a
Polycom 500 on *. Every few calls I make, the call connects and the
receiving party can hear me (thru Broadvoice), but I still get ringing on my
end, as if they never picked up. * logs look just fine. Does any one
have any suggestions? Thanks.
Adam S. RobinsExecutive
Hi,
we are based in Ireland
I've tried 3 different handsets - a philips elegance, an Audioline Tel 35 and a
generic BT telephone, all give the same result.
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Date: Wed, 8 Dec 2004 18:51:10 +0100
From: Wilson Pickett [EMAIL PROTECTED]
Subject: Re:
w00t... what to pull out of our hats next?
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming
Sent: Wednesday, December 08, 2004 10:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
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