Andrew Kohlsmith wrote:
On Tuesday 09 May 2006 15:23, Josep Aguilar wrote:
Can anybody help me on configuring utstarcom F1000 on asterisk?
Is there a way to do it or is it impossible?
There's not much to it. Tell it that your asterisk box is your SIP register
server and outbound proxy,
For a linux newb who needs to wake up ? How does one do this ?
Copy/create a app_wakeme.c in the source directory then compile asterisk ?
How do I call it in dialplan ?
(first)
% cd /usr/src/asterisk/apps
(if you're running 1.2)
% wget
How do I monitor the whole conversation on a Zap channel without
answering it - the channel is hanging up, I think it's because it's not
answered.
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hi
all,
i want to build a
extension that when i call 46-50 that ONE a account is ringing i have
this:
exten =
[46-50],1,Set(LANGUAGE()=de)exten =
[46-50],2,CDR(userfield)=INTERNexten =
[46-50],3,MusicOnHold(0.5)exten = [46-50],4,SIP/1000144|60|wWexten
= [46-50],5,Hangup
but it is not
working.
If ztcfg -v shows your card it's working, it's OK. probably kudzu
doesn't that your card is already configured
I am sorry cause i post this questions is not related to your problem,
but i am having problem detecting my TDM400P which is a TDM400P problem.
I manage to installed the card with
Hi Adibar,
Thank you. I have also received some
recipes from Sam.
I have sent the codes to the client
and waiting for him to confirm the good news.
It is amazing sometimes how slow some things happen.
I'll let you know.
Regards,
Benchev
Hi Benchev
Mine is working now. It was set to a
Well,
Bellsouth gave me a box of filters that have two RJ-11 jacks. One for
the DSL modem and one for a phone. The instructions specified that
every phone be connected to a filter. The DSL modem would then be
connected to the DSL jack along with one of the phones. The modem
should not be
try
exten = [46-50],1,Set(LANGUAGE()=de)
exten = [46-50],2,CDR(userfield)=INTERN
exten = [46-50],3,Answer
exten = [46-50],4,MusicOnHold(0.5)
exten = [46-50],5,SIP/1000144|60|wW
exten = [46-50],6,Hangup
René Enskat [Teamware GmbH] wrote:
hi all,
i want to build a extension that when i call
Daren J. Howell DTCommunication wrote:
Have Asterisk connected to a H323 compatible legacy PBX using QSIG
protocol and IP trunks.
I can call to Asterisk, and from Asterisk using X-Lite softphone but
whenever either end picks up, the calls disconnects.
Try restricting both ends to one
Thanks for the replies guys (Chris Peter)...
I think I've confused matters by not explaining things properly!
My ISP has my internet connection on a private IP address - so my LAN has an
address (192.168.42.*) and my internet connection has an address 10.100.x.x.
That is then NAT'd again onto
Board 1 was _really_ board 1 for 2 years ... don't think they just swaped by themselves.
Any way ... i replaced board 1 and 3 and now it's everyting back to normal...
thx for the help...
On 5/9/06, Rich Adamson [EMAIL PROTECTED] wrote:
Bogdan Tocu wrote: The outline is like this : Board 1
hi,try thatexten =
_[46-50],1,Set(LANGUAGE()=de)exten =
_[46-50],2,CDR(userfield)=INTERNexten =
_[46-50],3,MusicOnHold(0.5)exten = _[46-50],4,SIP/1000144|60|wWexten
= _[46-50],5,HangupChristian GansbergerACCM, Vienna
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From what I have tested, using cisco phones and 1.2.5, the original callerID
is not kept when making a transfer.
Any other ideas?
Joe
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On 5/10/06, Joseph Rothstein [EMAIL PROTECTED] wrote:
From what I have tested, using cisco phones and 1.2.5, the original callerID
is not kept when making a transfer.
Any other ideas?
We use SPA, snom and aastra phones, and I had assumed that this was a
limitation of the SIP protocol. I would
What do you see on the asterisk console?
do you see it setting the language etc or does it not match the pattern?
try
exten = [46-50],1,Set(LANGUAGE()=de)
exten = [46-50],2,CDR(userfield)=INTERN
exten = [46-50],3,Answer
exten = [46-50],4,MusicOnHold(0.5)
exten = [46-50],5,SIP/1000144|60|wW
It seems it must be in thix way:
_4[6-9]
But this is not very confortable if you have 4x and 5x numbers :)
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Alasdair
Gow
Gesendet: Mittwoch, 10. Mai 2006 11:54
An: Asterisk Users Mailing List -
i have realtime
running over oracle database when i have some _ extensions in the database the
asterisk won't accept them.
Here i tried to call
number 47.
the extension for
this one in the db is: _4[6-9]
so the second select
should found something with sqlnavigator i find the row but asterisk
Hello,
I 've installed both cacti and res_snmp for
monitoring.
Does res_snmp is able to send snmp traps when hardware
is out of service or others status ?
Harry
___
Faites de Yahoo!
Jürgen,
The TAE jack sounds like a great idea. In my house all of the phone and
data cabling is home-run to a punch-down block in a Comm closet. The
single DSL/POTS filter is located there along with the modem router and
a SPA-3000. Other than a nearby lightning strike destroying my filter,
On Wednesday 10 May 2006 02:00, Brian Capouch wrote:
Asterisk's sip.conf. The password must be numeric, that's the only
trick.
I don't believe that last sentence is correct.
I've got several of those phones, all of them with alphabetic passwords.
. . original (i.e. non-G) model.
My two
I have restricted the asterisk server to G711 to match the
choice on the PBX, and still same result.
I have read that either endpoint have to be either a master
or slave to communicate to each other. I see in the logs that both are shown to
be a slave. The pbx side has to be set to slave.
Hey,
Im running Asterisk 1.2.2 and im having problems with the audio when
bridging calls between the zap interfaces and sip. zap to zap work
fine, as do sip to sip (but asterisk isnt in the media stream, as it
doesnt need to be) and terminating the call and playing a test message
via either sip
2006/5/8, Senad Jordanovic [EMAIL PROTECTED]:
Why would you need to create your own?
Many reasons:
1. not relying on already busy open source developers
Free projects are getting better that closed ones because the
programmers don't need to re-invent the weel. Yo don't need to relay
on
Hi all,
I have a Cologne Chip Designs GmbH ISDN network controller and I want to
terminate voip calls via this ISDN card.
My question is:
How I must to wire the ISDN equipment with my ISDN card? With normal cable or
crossover? How I can to check if ISDN card is linked with ISDN equipment?
In
On 5/9/06, Alasdair Gow [EMAIL PROTECTED] wrote:
If you are in the same call group *8# should pick it up I think.
It should work if a channel is ringing but not if answered. Some
answering machines have the same problem, you can't pick up once the
message is being recorded (from the same
Show application dial
You may be interested in one of:
g- Proceed with dialplan execution at the current extension if the
destination channel hangs up.
G(context^exten^pri) - If the call is answered, transfer both parties to
the specified priority. Optionally,
And again I'll say... calleveryone.com for all your RELIABLE
termination needs. And again... don't go by the rates on the page...
those are the end-user rates... call them for wholesale rates.. they
will be competitive to voipjet, and you get phone support and quick
response time. Come on
陈帆 wrote:
there have only 30 channel for E1,,
He has defined the correct 30 channels for his E1.
On 5/10/06, *Steve Underwood* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Edwin Lam wrote:
hi folks.
does any one have experience setting up E1 PRI in Shanghai,
Does anyone know what would cause a 1.2.6 or 1.2.7 asterisk system to
do the following?
After inserting a zaptel card the system starts up and when I log into
the asterisk CLI I can't do a show sip, show zap, or show iax2
commands.. they just don't exist. Likewise, a help shows no such
Commands are
iax2 show ...
zap show ...
sip show ...
On Wed, 2006-05-10 at 13:32, Matt wrote:
Does anyone know what would cause a 1.2.6 or 1.2.7 asterisk system to
do the following?
After inserting a zaptel card the system starts up and when I log into
the asterisk CLI I can't do a show
Have you configured the zaptel card correctly?
On Wed, 2006-05-10 at 13:32, Matt wrote:
Does anyone know what would cause a 1.2.6 or 1.2.7 asterisk system to
do the following?
After inserting a zaptel card the system starts up and when I log into
the asterisk CLI I can't do a show sip, show
I use Plainvoip.. And I know a lot of
the community does.. Rates are inexpensive and quality is excellent.
brian
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julius Barber :: GringoTel.com
Sent: Tuesday, May 09, 2006 3:40
PM
To:
Juergen K. Zick wrote:
But slowly, we are getting completely off-topic on this list. I doubt
that changing to static IP will solve to decribed problem, because it is
a line mismatch problem on the physical layer of the connection. And
these will not go away unless you change the wiring !
Straight cable for TE mode and Xover for NT mode
you have to make a call, than L1 should go up
I have a Cologne Chip Designs GmbH ISDN network controller and I want to
terminate voip calls via this ISDN card.
My question is:
How I must to wire the ISDN equipment with my ISDN card? With
We use SER to front several Asterisk systems. Phones register on SER,
which also acts as a load balancing and failover proxy for the Asterisks.
Phone account details are held in MySQL, which Asterisk could access but
does not currently do so. At present, call routing is done on the
Asterisks
Hi
all.
I am attempting to
setup Asterisk to allow me to press *1 while in a call to use automon to record
the call but have had absolutely no success. Is there a trick to
this?
In
extensions.conf
[globals]
DYNAMIC_FEATURES=automon
[default]
On 5/9/06, Jeroen Zwarts [EMAIL PROTECTED] wrote:
I have a problem with the Bristuffed version of Asterisk. I have tried
Bristuff-0.3.0-pre-1m,n,o,p (Asterisk 1.2.6 to 1.2.7.1) but they all have
the same problem it seems:
Hi Jeroen, any progress made yet? I noticed I'm experiencing the same
How do you do it on a PRI and what do I ask my provider (Bellsouth) for
if they permit it?
Eric ManxPower Wieling wrote:
Tim Litwiller wrote:
Not, on your question - but you brought up something I would really
like to do and I was told it wasn't possible.
how do you do the transfer to
I have been having a strange issue with my Asterisk 1.2.1 server. I have a
TDM400 for the three POTS lines I have and I can receive calls without any
problems. But sometimes (not everytime, but 70%) when I dial out of those
lines it drops a number and of course gives me the telco error
this should work:
-call comes in
-you answer with a ZAP FXS device
-you hook flash
-dial a valid internal extension or if your dialplan permits it dial an
outside line (if you have 1 FX0 port or PRI)
If you have a PRI you can arbitrarily set the caller ID in your dialplan
-hang up. Caller is
2006/5/10, Dave Morrow [EMAIL PROTECTED]:
I am attempting to setup Asterisk to allow me to press *1 while in a call to
use automon to record the call but have had absolutely no success. Is there
a trick to this?
May be a problem with the way you are sending the dialtones. Try
sending as data.
I have been having a strange issue with my Asterisk 1.2.1 server. I have a
TDM400 for the three POTS lines I have and I can receive calls without any
problems. But sometimes (not everytime, but 70%) when I dial out of those
lines it drops a number and of course gives me the telco error
David,
You need to use the 'g' option with Dial().
g- Proceed with dialplan execution at the current extension if the
destination channel hangs up.
--johann
David L. West wrote:
In the following macro, a call is dialed and control branches according
to DIALSTATUS, much
Hi!
I want to make a call to/from Panasonic IP pbx thru asterisk via H323.
H323 is working nice, I can call/receive using netmeeting
-- Executing Dial(OH323/[EMAIL PROTECTED], OH323/[EMAIL
PROTECTED]|15|tr) in new stack
-- H.323 call to [EMAIL PROTECTED] with codec(s) alaw
--
I cant imagine anyone using voipjet as their only or main provider. And I'll
say again, 3.9 cents for an ITSP is the most expensive I have found.
Business grade termination is typically much less than that with top notch
companies like https://www.nexvortex.com/ at 2.5c.
-Original
- Original Message -
From: stoffell [EMAIL PROTECTED]
To: Jeroen Zwarts [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Wednesday, May 10, 2006 3:37 PM
Subject: Re: [Asterisk-Users] Bristuffed Asterisk: Hangup problems
On
Hello.I have a MinITX motherboard with only one pci slot and one onboard ethernet interface, I have a TDM04B card plugged into that motherboard and the /proc/interrupts:CPU0 0: 169626332 XT-PIC timer
1: 1270 XT-PIC i8042 2: 0 XT-PIC cascade 8: 4 XT-PIC rtc12: 170166219 XT-PIC eth0, wctdm14:
[EMAIL PROTECTED] wrote:
2006/5/8, Senad Jordanovic [EMAIL PROTECTED]:
Why would you need to create your own?
Many reasons:
1. not relying on already busy open source developers
Free projects are getting better that closed ones because the
programmers don't need to re-invent the weel.
Thanks, Wes, G was just what I
needed. Here's where I am:
[macro-stdexten]exten =
s,1,Set(cname=${CALLERID(number)[EMAIL PROTECTED])exten =
s,n,Set(CALLERID(number)=${cname})exten =
s,n,Dial(${ARG2},20,gtTwWG(OnAnswer,1,1))exten =
s,n,Set(savestatus=${DIALSTATUS})exten =
I will agree 3.9c is quite expensive for termination. Most providers
hover around the 1 to 2c mark. 3.9c is just a way for them to cover all
of their overhead. I have found a lot of providers even at 1c can be
very stable and offer good services.
-Original Message-
From: [EMAIL
On Wednesday 10 May 2006 16:02, Woodoo People .pGa! wrote:
Straight cable for TE mode and Xover for NT mode
you have to make a call, than L1 should go up
I connected the card to ISDN provider's equipment via crossover cable, but
portinfo report:
Port 1: TE-mode BRI S/T interface line (for
To answer your question on how I do the hook flash transfer here it is :in the globals section of extensions.conf put all your cell phone number like this :[globals]MartCell=5141234567Then add this macro in your extensions.conf :[macro-cell_user]exten = s,1,Playback(Call_Transfer)exten =
Aaron Daniel wrote:
Has anyone done any work with implementing dbsecret for chan_sip?
No. That would only be useful for DUNDi, but chan_sip dial strings
cannot contain the secret (unlike chan_iax2 dial strings), so it would
not be useful to pass it to the remote party, and thus they would not be
It happens by default. I don't know what you would ask BellSouth.
Lenwood S. Sawyer III wrote:
How do you do it on a PRI and what do I ask my provider (Bellsouth) for
if they permit it?
Eric ManxPower Wieling wrote:
Tim Litwiller wrote:
Not, on your question - but you brought up something
I am having problem diagnosing a call problem. On both a Cisco phone
and a Linksys 942 I am only getting one side of the call when connected
over a WAN link or internet connection. I have set nat=yes and qualify
in sip.conf and the phone registers fine. I can hear the other end, but
they do not
I've defined my dialplan as showed below. My internal lines are numbered
as 12345XX, and internal users can call another by the entire 7-digits
extension, or by just last 2 digits.
[invalid]
exten = _X.,1,Playback(pbx-invalid)
exten = _X.,2,Hangup()
[internal]
include = invalid
exten =
Antonio,it changes slot of tdm04b and restarts the server.
I wait to have helped.
RegardsJosué
2006/5/10, Antonio Almodóvar [EMAIL PROTECTED]:
Hello.I have a MinITX motherboard with only one pci slot and one onboard ethernet interface, I have a TDM04B card plugged into that motherboard and the
I am planning to deploy Asterisk Business edition. Does this edition
have a web module administration?
Thanks
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Kerry,
Do you have a reading problem? Both times that I have tried to help
people out by suggesting a company I have personally used and have had
good luck with, you reply and say that the rates are horrible. If
you would read my e-mails you would see that the 3.9 cents is NOT for
wholesale
ok then.. Where are their wholesale prices?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Wednesday, May 10, 2006 1:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] voipjet down?
Kerry,
Do you
Whoa, down boy! Kerry is just saying that people who are using voipjet are most likely not in the market for wholesale termination. That being the case, the higher rates are going to make these users shy away. It's great that you've found their rates to be comparable for wholesale termination, and
Antonio,it changes slot of tdm04b and restarts the server.
Since he said in the email that the machine only had 1 PCI slot, I
don't think he can do that
Check in your BIOS, some let you assign a specific IRQ to the PCI slot
hth
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Alvaro:
I dont think such a thing exists for a simple reason. I you think
things right, you will be able to end with a good combination of
contexts.
In your case, having:
[internal]
include = invalid
exten = _XX,1,Dial(SIP/12345${EXTEN}) ; Short alias for internal lines
exten =
(apologies if this is a dup but my original message never showed up
and I mailed it this morning)
For a linux newb who needs to wake up ? How does one do this ?
Copy/create a app_wakeme.c in the source directory then compile asterisk ?
How do I call it in dialplan ?
(first)
% cd
Sigh. Nevermind. Looks like I need to use the
Manager interface to get what I need, but I guess I'll be learning a new
technology every day for a while. ;)
"David L. West" [EMAIL PROTECTED] wrote in
message news:[EMAIL PROTECTED]...
Thanks, Wes, G was just what I
needed. Here's
OK. You lost me.
David Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodatasolutions.com
Tel: (519) 963-3020
Fax: (519) 451-6615
Lead, follow or get out of the way!
This message has originated from Autodata Solutions. The attached
material is
On Wednesday 10 May 2006 16:02, Woodoo People .pGa! wrote:
Straight cable for TE mode and Xover for NT mode
you have to make a call, than L1 should go up
I connected the card to ISDN provider's equipment via crossover cable, but
portinfo report:
Port 1: TE-mode BRI S/T interface line
Whoa, kid, do you have a behavioral problem? No reason to get nasty.
You're only confirming the long-held suspicion that you're in bed with
calleveryone. Friendly references are one thing, but defending it at
the expense of courtesy, that's suspicious.
I don't think their customer service
I am trying to use QSIG to interoperate with legacy PBXs.
I am looking to see whether any one knows whether Call Hold, Call Transfer, MWI
works with QSIG support in Asterisk.
Thanks in advance.
--dp
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I'm not in bed with CEO. All I do is use them... and want to see
them get more customers. Sorry if I sounded upset... it just seems
like Kerry never reads the e-mail every time this issue comes up.
On 5/10/06, Jay Milk [EMAIL PROTECTED] wrote:
Whoa, kid, do you have a behavioral problem?
If you want someone would will give cheap termination to end users, go
use voipjet or whatever you want.
If, on the other hand, you want some reliable cheap wholesale
termination, go check out voipjet.
glad to know theres options out there !
-
Shidan
Hey
Everyone,
We are in the
process of reviewing headsets for use with our GXP-2000s. I'm looking for
some feedback as to which headsets people are using, the pros and cons of those
headsets, and if they would recommend them to someone else.
Any help would be
appreciated...
-
Jason
Split the contexts up even more. Keep in mind the SIP users you setup can all
start in a different context and you can have your incoming zap calls start in a
different context.
Ie, make a context that includes ability to dial internally and outside. Then
make another context that just
Hello,
I'm experiencing some problems with AstTAPI driver. Dialing works just fine,
but 'Hangup' from Outlook doesen't.. actually that's not the problem as fact
that Outlook doesen't detect end of conversation - once the call is
terminated 'manually' via the phone Outlook still 'thinks' that call
Thank you for replying.Icannot assign specific IRQ to the PCI or ethernet slot via bios, I can assing IRQ to usb, serial,... but not to the PCI or Ethernet. Maybe both devices must share the same pci, I don't know.
Anyone who has this motherboard?
___
On Thu, May 11, 2006 at 12:01:08AM +0200, Olivier Krief wrote:
Hello,
I tried to read Bristuff source code to understand the way calls are bridged
from one BRI port to another (as HFC cards have active channel switching
capability).
Doing so I looked at zaphfc.c file which seems to be the
On Wed, May 10, 2006 at 08:53:36AM +0200, Anthony Azzopardi wrote:
How do I monitor the whole conversation on a Zap channel without
answering it - the channel is hanging up, I think it's because it's not
answered.
If the channel is not answered, there is no (useful) audio in it to
monitor.
Hi Gary -
I had to make asterisk use ODBC for everything when I converted to
ODBC message storage. res_odbc and res_mysql wouldn't co-exist.
Hmm. That's odd. I've got the voicemessages table accessed through
res_odbc (of course), and the voicemail_users table accessed directly
through
I had similar problems when I first started to play with it. I've gotten
Omniis TSP for Astrisk to work just fine. http://www.omniis.com/asttapi But
i don't know the version im using 0.0.8
Terrelle
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tomislav
You didn't mention which mini-itx board you had so I don't know if this
is relevant but I'm running an EPIA 5000 (fanless 533mhz) with a TDM11B
and, well, I never checked /proc/interrupts just because it seems to
work great ;) I'll try to check tonight what the bios offers and report
back :)
Antonio Almodóvar wrote:
I've tried modifying parameters in the bios and I didn't managed to
change the irq.
Does anyone have a machine like mine?
Have anyone changed the irq in order to not sharing irq's?
You can't change interrupt for the card in the BIOS. Disable all the
unnecessary
On Thu May 11 2006 07:16, Jason Adams [EMAIL PROTECTED] wrote:
Hey Everyone,
We are in the process of reviewing headsets for use with our GXP-2000s.
I'm looking for some feedback as to which headsets people are using, the
pros and cons of those headsets, and if they would recommend them to
Hi guys,
Currently I just purchased the TE205P for the installation of my ISDN30 setup. I
compiled, setup the card correctly I think. When I do the ztcfg -vvv it shows
all 31
channels clear, with 30 bchannels using for voice ranging from 1-15, 17-31, and
16
is the dchannel. However, when I do a
Hi, test with dmesg and past in list the log´s
Regards
Josué
2006/5/10, azyuky [EMAIL PROTECTED]:
Hi guys,Currently I just purchased the TE205P for the installation of my ISDN30 setup. Icompiled, setup the card correctly I think. When I do the ztcfg -vvv it shows all 31
channels clear, with 30
Leo Ann Boon wrote:
Antonio Almodóvar wrote:
I've tried modifying parameters in the bios and I didn't managed to
change the irq.
Does anyone have a machine like mine?
Have anyone changed the irq in order to not sharing irq's?
You can't change interrupt for the card in the BIOS. Disable
I upgraded recently to Asterisk SVN-branch-1.2-r25165M
the commandline
asterisk -rx 'sip show peers'
returns with the first line:
on
Is that a bug, or how can I omit it?
I used:
asterisk -rx 'sip show peers'|grep OK|sort | tee /dev/tty |wc -l; echo
registered at ELMIT
which results
Hi, every one
I'd like to introduce some new feature of our
products.
mg3000-r
fxo gateway
provides
more feature to work with asterisk.
1.play
asterisk ivr with no interuption.
when the
mg3000-r received call from co line, it wouldn't conect instantly.instead, it
start call to
You always recommend CEO when ever anyone asks about a service provider when
nobody has asked about wholesale. Very very few people are interested in
wholesale pricing. If you are a consultant you want to recommend a company
to your clients that is the best fit for their needs. If you are an end
You could install any number of interfaces but it does not
come with one.
Kerry
GarrisonDirector of Technical ServicesTech Data
Pros - Orange County's Mobile IT Service
Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com
From: [EMAIL PROTECTED]
Hi Josue and list,
This is part of my dmesg log. Also one other thing, I don't know why the red
light
on span/port 1 of my card keep flashing.. is it normal that way? Or it's
supposed to
mean something? Obviously I have no experience with ISDN before so this is my
very
first try.
Found
Go into the BIOS and disable every possible device like
USB, COM, Serial, etc. But odds are, you are screwed with that
motherboard.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Antonio
AlmodóvarSent: Wednesday, May 10, 2006 8:01 AMTo:
Asterisk Users
Hi, every one
I'd like to introduce some new feature of our
products.
mg3000-r
fxo gateway
provides
more feature to work with asterisk.
1.play
asterisk ivr with no interuption.
when the
mg3000-r received call from co line, it wouldn't conect instantly.instead, it
start call to
Hi all. I posted this earlier but
nevergot any advice that helped. If anyone knows how to get this
going, I'd appreciate some
advice.
I am attempting to
setup Asterisk to allow me to press *1 while in a call to use automon to record
the call but have had absolutely no success. Is there a
I have contiuned to fight this problem all day, and still have not
found a solution. I did get DTMF tones to the other end, but no voice.
Any tips on where to look?On 5/10/06, Bruce Reeves [EMAIL PROTECTED] wrote:
I am having problem diagnosing a call problem. On both a Cisco phone
and a Linksys
Hi,
I am looking for voip providers in bay area, any suggestions?
My requirements are:
handling around 2000 calls a day (incoming) and around 1000 calls a day outgoing. I have a Asterisk PBX server to take care of routing calls to appropriate deparment. So I am looking mainly for IAX2 or SIP
How do you know it is doing nothing -- does it beep when you hit *1 --
are you sure it is seeing the *1 -- wat version of asterisk do you
have, etc. etc.?
on Wednesday 05/10/2006 Dave Morrow([EMAIL PROTECTED]) wrote
Hi all. I posted this earlier but never got any advice that helped. If
ESF/B8ZS is a T-1 config.
azyuky wrote:
Hi Josue and list,
This is part of my dmesg log. Also one other thing, I don't know why the red
light
on span/port 1 of my card keep flashing.. is it normal that way? Or it's
supposed to
mean something? Obviously I have no experience with ISDN before
Have you looked at CBeyond? I like their T1 SIPConnect
product.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nitin
GuptaSent: Wednesday, May 10, 2006 7:04 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject:
[Asterisk-Users] VOIP provider
I know and it's crazy isn't it because this is what i have in my zaptel.conf
bchan=1-15
bchan=17-31
dchan=16
span=1,1,0,ccs,hdb3,crc4
loadzone=sg
defaultzone=sg
The configuration is set to conform to my ISDN30 E1 specifications. And the
jumper
on the digium card is closed to enable E1 mode.
I'm trying to add ooh323c to my asterisk 1.2.7.1 install and did an svn
update of asterisk-addons and followed the readme in asterisk-ooh323c
and I get through the .configure with no errors. But make causes:
rpath /usr/local/lib -L./ooh323c/src -version-info 1:1:0 -lpthread
make: rpath: Command
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