From: "Yuan LIU" <[EMAIL PROTECTED]>
How can I fix this? Or does fxotune only tune TDM400? (My TDM400P shows a
mere 1.2% echo.) Could it do "authentic" X100P?
I just didn't want to accept fxotune.c's claim about working only with TDM.
Several other users indicated that they were not able
On Thu, 14 Dec 2006, Gregory Duchatelet wrote:
> > It looks like 107 is busy ;-)
> > Please increase verbosity, like
> > set verbose 5
> > capi debug
> > to see what is happening.
>
> Hi Armin,
>
> Verbose was at 30 :)
> 107 is not busy since i can call it from 102, which is another internal
My SM56 (Motorola X100P clone) has echo as hight as 38%, according to
fxotune -d. But when trying to take action, it fxotune simply says it
can't.
./fxotune -i3 -
Running with parameters:
doset=0
docalibrate=1
dodump=0
startdev=1
stopdev=252
calibt
Joao Pereira ha scritto:
Do you know if it has 802.1x authentication as it is defined in
EDUroam ( http://www.eduroam.org/ ) ?
I never found a WiFi phone working with 802.1x I tested ZyXel
Prestige 2000 but the sound was bad and it doesnt support 802.1x :(
Thanks
Joao Pereira
Well, I hav
Last week I asked about MWI indicators on wireless phones that would work
with Asterisk. I sent a message off to Panasonic asking them about it
because in their ads they specifically stated that the indicator works
with and requires phone company voicemail subscription.
The is the model TG5631.
S
What do they provide you? You normally wouldnt install asterisk "on their
system" unless you are leasing a server from them.
On 12/15/06, Zeeshan Zakaria <[EMAIL PROTECTED]> wrote:
Does anybody know how to setup bandwidth.com trunk on asterisk. They
provide bandwidth services to asterisk.org, b
Does anybody know how to setup bandwidth.com trunk on asterisk. They provide
bandwidth services to asterisk.org, but don't know how to setup up asterisk
on their system.
--
Zeeshan A Zakaria
___
--Bandwidth and Colocation provided by Easynews.com --
as
Matt wrote:
So you are saying that the card is on it's own IRQ and is not sharing
anything with anything? I realize the eth0 and usb are sharing, but
am not too concerned about that.
What's your zttest result and did zttool reported any irq misses? If
zttest is mostly >99.98%, then the zap devi
this is ata , simple flash button works,
thx
> If using a VOIP phone there should be a button. If using an ATA the
> instructions should be in the manual of the ATA (you also may be able to
> look in the web interface of the device). I forgot how to do it if you are
> using ZAP. Have a look on th
I am not sure how far I will go with that, but
I did a capture and explained in detail what is the problem,
I hope that somebody there will forward it to high level support maybe,
who knows, it is so hard to get help when something strange happens,
from my experience with broadvoice, everything w
I have the following configuration:
VoIP Provider <> Asterisk <> Samsung PBX <---> PSTN
^
Asterisk has a few VoIP extensions connected, but most of the extensions
are hanging of the Samsung PBX.
Asterisk has 1 NT and 1 TE interface, which are connected to a TE an
So you are saying that the card is on it's own IRQ and is not sharing
anything with anything? I realize the eth0 and usb are sharing, but
am not too concerned about that.
On 12/14/06, Leo Ann Boon <[EMAIL PROTECTED]> wrote:
Matt wrote:
> I see that the digium card doesn't share the IRQ howe
You might want to take a look at the new 4 port FXO from Grandstream
I haven't had one yet to evaluate but assuming it works it is very price
competative and off-loads all the analog (TDM) stuff from your PC
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
> I have been using
Sounds like you have a disconnect supervision problem.
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
> We currently have a pri coming into our asterisk system. Most of the
> time, the did numbers that we call into it work great. However,
> occationally, we get fast busies,
If using a VOIP phone there should be a button. If using an ATA the
instructions should be in the manual of the ATA (you also may be able to
look in the web interface of the device). I forgot how to do it if you are
using ZAP. Have a look on the wiki.
- Original Message -
From: "Barto
Matt wrote:
I see that the digium card doesn't share the IRQ however Digium
has recommended diabled USB still... additionally the Digium card is
on 169 which isn't a valid IRQ.. how can I find out what it is sharing
with?
the tdm card is not sharing an interrupt with your USB. It's your LAN
Howard Lowndes wrote:
How old is your mobo?
I have that same problem and I think it because the TDM card will only
work with PCI 2.2 or later and, although lspci finds the card, udev is
not installing the zap devices.
Which is why those in the know who don't care to hear Digium's stock
an
Did you do a reload ? Also when you say you commented it out you mean that you
commented out the register statement ?
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, December 10, 2006 5:02 PM
Subject: [asterisk-users] chan_sip.c:5
I have been using the sangoma A200 with echo cancelation and I have been
real happy.
- Original Message -
From: "Todd- Asterisk" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, December 14, 2006 3:23 PM
Subject: [asterisk-users] (no s
Any ideas?
Did anyone experience something like that?
Thx
Yes, unfortunately, all the time. There answer is if it works with a sip
softphone client than it's not their problem. It does work with the
softphone client.
-Original Message-
From: Bartosz Wegrzyn - maillists [mailto:[EMAIL
On Fri, Dec 15, 2006 at 12:05:13AM +1100, Rudolf Ladyzhenskii wrote:
> Hi, all
>
> I am building a new server. Have installed FC 6 and put in TDM400 card.
>
> Checked out latest asteriusk code, run make install in zaptel directory.
> So far all is fine.
>
> Now I am trying to install the drivers
The Fedora Extras rpm is tiny because it has nothing really of help
in it. It's missing the modules.
I've had some success on Fedora Core 6 using the ATrpms repository,
which has the zaptel-kmdl package for most variations of kernels
included in FC6.
Simon
On 14 Dec 2006, at 22:31, Yuan
> It looks like 107 is busy ;-)
> Please increase verbosity, like
> set verbose 5
> capi debug
> to see what is happening.
Hi Armin,
Verbose was at 30 :)
107 is not busy since i can call it from 102, which is another internal
phone. All internal phones are busy for Asterisk...
Here is the lo
I am trying to get asterisks to work with http://www.voiptalk.org 's IAX
service. I have configured asterisks as per their instructions and am
using the x-lite soft phone. When I get an incoming call the softphone
rings but the caller (from pstn) gets a recorded message saying the
number is unable
Yuan LIU wrote:
From: "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]>
Yuan LIU wrote:
Another bizarry: If I run the Echo application from the console, I
can hear a very long delay (upward to 1,000 ms). I can run the same
application from a GrandStream phone (on the same LAN) and hear
little
From: "Rudolf Ladyzhenskii" <[EMAIL PROTECTED]>
I guess, modules are mot there. Running find / -name zaptel* did not
find any modules.
Be careful here - wildcard expansion takes place locally unless you quote
the string:
$ find / -name 'zaptel*'
Of course search from / is suboptimal as you a
From: "john beaman" <[EMAIL PROTECTED]>
StripLSD is obsolete:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+StripLSD
StripMSD is being phased out: http://bugs.digium.com/view.php?id=5673
John Beaman
Telecom Specialist
Voice Telecommunications Services Department.
Good Samaritan Nat
From: "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]>
Yuan LIU wrote:
Another bizarry: If I run the Echo application from the console, I can
hear a very long delay (upward to 1,000 ms). I can run the same
application from a GrandStream phone (on the same LAN) and hear little
delay. What cou
Yuan LIU wrote:
All of StripMSD, StripLSD, etc., are missing when I downloaded
asterisk-1.2-current.tar.gz, which explodes into 1.2.13. Are the strip
club deprecated? What replacement functions should I use?
See README.variables in the Asterisk source.
___
Rob Schall wrote:
We currently have a pri coming into our asterisk system. Most of the
time, the did numbers that we call into it work great. However,
occationally, we get fast busies, but we noticed those busies were not
due to anyone being on the line, etc...
Any ideas what could cause this? I
From: "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]>
[Goodies skipped]
-= Info about application 'ZapRAS' =-
[Synopsis]
Executes Zaptel ISDN RAS application
[Description]
ZapRAS(args): Executes a RAS server using pppd on the given channel.
The channel must be a clear channel (i.e. PRI so
Thanks for suggestion.
Just tried that Was surprised with download size of only 72k. Anyway,
command work, but I still have same ptoblem.
I tried to run modprobe -- it failed.
Tried to run service zaptel start and get:
service zaptel start
No functioning zap hardware found in /proc/zaptel, loadi
StripLSD is obsolete:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+StripLSD
StripMSD is being phased out: http://bugs.digium.com/view.php?id=5673
John Beaman
Telecom Specialist
Voice Telecommunications Services Department.
Good Samaritan National Campus
605-362-3331
>>> [EMAIL PRO
Yuan LIU wrote:
Another bizarry: If I run the Echo application from the console, I can
hear a very long delay (upward to 1,000 ms). I can run the same
application from a GrandStream phone (on the same LAN) and hear little
delay. What could possibly be wrong? If it were interrupt overload,
I
All of StripMSD, StripLSD, etc., are missing when I downloaded
asterisk-1.2-current.tar.gz, which explodes into 1.2.13. Are the strip club
deprecated? What replacement functions should I use?
Yuan Liu
___
--Bandwidth and Colocation provided by Eas
We currently have a pri coming into our asterisk system. Most of the
time, the did numbers that we call into it work great. However,
occationally, we get fast busies, but we noticed those busies were not
due to anyone being on the line, etc...
Any ideas what could cause this? Is this a congestion
Another bizarry: If I run the Echo application from the console, I can hear
a very long delay (upward to 1,000 ms). I can run the same application from
a GrandStream phone (on the same LAN) and hear little delay. What could
possibly be wrong? If it were interrupt overload, I'd hear lots of cr
Jordan Novak wrote:
My need to do this through asterisk is simply the ability to provide me
access with no additional cost to my customer. It seems like a nice
thing to include as long as authentication is done well. I have
worked on a dozen or more types of switches and all of them have
suppo
Matt wrote:
Hi,
I have an IBM xSeries server... and the digium card is sharing IRQ
with USB and giving me crackling audio.
cat /proc/interrupts
It brings up these results:
0: 10566547IO-APIC-edge timer
1: 9IO-APIC-edge i8042
2: 0 XT-PIC cascade
Hi,
Philipp von Klitzing posted this solution in Dec. 2005
Answering machine mimic: Listen while caller is leaving voicemail for
you; with pick-up option
Is there any other way to listen while caller is leaving a voicemail for
you?
Thanks
Fernando
___
Joao Pereira wrote:
Hello
how can I distinguish all the calls that arrive to my Asterisk starting
with: 351217588XXX ?
I want match the first 9 digits does Asterisk has any function for
this?
exten => _51217588XXX,1,Whatever
___
--Bandwidth and
[EMAIL PROTECTED] wrote:
Do anybody know, if there is a way to connect 2 zap-channels with
Hardware TDM Switching?
It's called DACS. See the /etc/zapata.conf config file sample.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-us
I can dial out via FWD because the login is in the session.
Vonage requires a register even for outbound.
My understanding is that they log the register and then any call from that IP
is from that user.
This is why I can't dial out vonage.
The root cause is that sip is not registering at all.
I
This may not be vonage related as it appears that I can not register with any
sip servers.
I tried FWD and also get a black "sip show registry"
Could it be a firewall issue?
I am running IP tables on the computer which is on the internet with no NAT.
Asterisk 1.2.13
I have allow outbound all.
Al
From: "Rudolf Ladyzhenskii" <[EMAIL PROTECTED]>
Now I am trying to install the drivers.
# modprobe zaptel
FATAL: Module zaptel not found.
Fair enough, no zaptel driver is found on the system.
Is there are any known problems with FC6? I did not have much trouble
running on FC3 before.
I'm not
On Thu, Dec 14, 2006 at 11:13:29AM -0500, Matt wrote:
> Hi,
> I have an IBM xSeries server...
Which model, exactly? With which customizations?
> and the digium card is sharing IRQ
> with USB and giving me crackling audio.
Do you actually get any interrupts from the USB?
>
> >>>cat /proc/int
Hello,
When in conversation, how can I put somebody on hold?
thx
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Hello,
I have two broadvoice accounts.
Lately, very often my broadvoice accounts are in unregistered state.
When I log into asterisk I see:
voip*CLI> sip show registry
Host Username Refresh State
sip.broadvoice.com:5060 [EMAIL PROTECTED] 120 Request Sent
sip.broadvoice.com:5060 [EMAIL PROTECTED]
From: Steve Prior <[EMAIL PROTECTED]>
The feature request # is 4542, but I don't know any associated bug number,
nor with what phones other people had to tweak. My phone is a GE
27935GE3-B. (Don't know what possessed me to say GM:-)
Yuan Liu
Just gotta ask - you did plug in the power suppl
At 05:23 AM 12/14/2006, you wrote:
Should I just get 2 or 3 X100P cards? Or do
I need the Sangoma A20200 or even the A20200D (Echo cancelation)...
When I started down this path I choose the TDM04 and have always had
occasional echo issues, not bad and not often, but it annoys the wife
and o
At 03:40 AM 12/14/2006, you wrote:
I tested ZyXel Prestige 2000 but the sound was bad and it doesnt
support 802.1x :(
Wow, I've always been impressed with the sound from my Zyxel 2000W.
Ira
___
--Bandwidth and Colocation provided by Easynews.com -
Title: Message
Hi All, Is it
possible to show an agent's queue status on the phone? For example, in
our current non-asterisk PBX, if a
member of a call queue does not answer the phone when a queue call is sent to
them, they go to a 'not ready' status, and this is indicated on their phone. So
Jeremy wrote:
> What kind of luck are people having with the Web-MeetMe control? The
> condition of the page on the voip-info wiki makes me a bit nervous
about
> putting Web-MeetMe into a production environment. Use of MeetMe has
> really taken off here since installation and I need a scheduling
You might check out http://sourceforge.net/projects/web-meetme and version
2.1.0, I had to tweak it a little, but it has worked well for people to
schedule their own meetme conferences.
On 12/14/06, Porier, Jeremy M. <[EMAIL PROTECTED]> wrote:
What kind of luck are people having with the Web-Me
Sorry to bring back a post from the grave, but what do you feel is the worst
problem you are having with AgileVoice? Is their support easy to reach?
On 11/16/06, Chris Mazuc <[EMAIL PROTECTED]> wrote:
Andrew Joakimsen wrote:
> Chris:
>
> We were evaluating AgileVoice currently, could you please
right, but who have production and tested code of application level
encryption for SIP and IAX for SECURE(!) trunks?
turby
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pavel Jezek
Sent: Thursday, December 14, 2006 6:15 PM
To: Asterisk Users Mailing Lis
Yuan LIU wrote:
The feature request # is 4542, but I don't know any associated bug
number, nor with what phones other people had to tweak. My phone is a
GE 27935GE3-B. (Don't know what possessed me to say GM:-)
Yuan Liu
Just gotta ask - you did plug in the power supply connection on the
bo
What kind of luck are people having with the Web-MeetMe control? The
condition of the page on the voip-info wiki makes me a bit nervous about
putting Web-MeetMe into a production environment. Use of MeetMe has
really taken off here since installation and I need a scheduling and
provisioning syste
I see that the digium card doesn't share the IRQ however Digium
has recommended diabled USB still... additionally the Digium card is
on 169 which isn't a valid IRQ.. how can I find out what it is sharing
with?
On 12/14/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
compile kernel without u
tunneling small rtp packets through vpn has big overhead,
better to use application level encryption - encrypted iax or srtp.
PJ
[EMAIL PROTECTED] wrote:
joao,
you can use ssh tunel, pptp or vpn for any sip/iax trunks or users.
turby
-Original Message-
From: [EMAIL PROTECTED]
[mai
Ale wrote:
hi all,
I'm trying to send text messages to Snom 300 to show the credit
remaining during the call...
Sending a MESSAGE directly to the phone via udp i'm able to update
the text on the display... but not during the conversation.
I read about AOC, but i can't find any documentati
From: Tzafrir Cohen <[EMAIL PROTECTED]>
> >Have you tried toe boost ring voltage option then recompile Zaptel?
> >It is normally set to a fairly low voltage
> >
> >John Novack
>
> Thank you so much! I googled a bit about how to change ring voltage and
> only found an old and suspended feature req
joao,
you can use ssh tunel, pptp or vpn for any sip/iax trunks or users.
turby
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pavel Jezek
Sent: Thursday, December 14, 2006 4:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
compile kernel without usb support or unload usb modules
turby
ps
your tdm card don't share the irq, your network card share the irq...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Thursday, December 14, 2006 5:13 PM
To: Asterisk Users Ma
Hi,
I have an IBM xSeries server... and the digium card is sharing IRQ
with USB and giving me crackling audio.
cat /proc/interrupts
It brings up these results:
0: 10566547IO-APIC-edge timer
1: 9IO-APIC-edge i8042
2: 0 XT-PIC cascade
8:
as I know, only preliminary support:
0005413: [patch] Secure RTP (SRTP)
http://bugs.digium.com/view.php?id=5413
Joao Pereira wrote:
Can I do the encrypted trunk in SIP? Does Asterisk supports it?
Thanks
Joao Pereira
Pavel Jezek wrote:
http://www.voip-info.org/wiki/view/IAX+encryption
Can I do the encrypted trunk in SIP? Does Asterisk supports it?
Thanks
Joao Pereira
Pavel Jezek wrote:
http://www.voip-info.org/wiki/view/IAX+encryption
Joao Pereira wrote:
Hello
I would like to define a trunk from my Asterisk to a VoIP provider,
but I want to make it secure, because its
On 12/13/06, Aaron Daniel <[EMAIL PROTECTED]> wrote:
Does anyone have the pickup application working? I'm attempting to get
I did have it working.
The problem I'm having is in the fact that my phones register with mac
addresses instead of extensions, so I'm unsure as to what to put in the
perfect!!!
its now working this way:
exten => _.,4,GotoIf($[ "${EXTEN:0:9}" = "351217588"] ? 20:10)
Thanks a lot
Joao Pereira
Ove Aursand wrote:
Use ${EXTEN:0:9}
Regards,
Ove
Joao Pereira wrote:
Hello
how can I distinguish all the calls that arrive to my Asterisk
starting with: 351217588X
Todd- Asterisk wrote:
Hello everyone! I'm planning on setting up a new system shortly and
can't pick the right card... We will have 2 or 3 lines coming in and 7
extensions (GXP2k's). Should I just get 2 or 3 X100P cards? Or do I
need the Sangoma A20200 or even the A20200D (Echo cancelation).
did you use T38 with you patton smart node 2400 ?
why Patton are very good GW and fax must work.
you must also check that the clock source is the Primary and not the
internal clock...
2006/12/14, Jerry Jones <[EMAIL PROTECTED]>:
Or any of a number of gateways that do this. Off the top of my h
On 14 Dec 2006, at 13:32, Jordan Novak wrote:
My need to do this through asterisk is simply the ability to
provide me access with no additional cost to my customer. It seems
like a nice thing to include as long as authentication is done
well. I have worked on a dozen or more types of switc
Or any of a number of gateways that do this. Off the top of my head
you can get one from CarrierAccess, Vega, Audiocodes, Mediatrix,
Adtran, and others.
Just try to be very careful as they all have their strengths and
weaknesses and you need to evaluate how they would fit your needs.
Best i
On Thu, Dec 14, 2006 at 07:15:30AM -0600, Russ Price wrote:
> Yuan LIU wrote:
> >A configuration string "boostringer" was mentioned in several messages,
> >including one concerning TDM400P, all without indicating the applicable
> >configuration file. This has no apparent effect on TDM400P wherev
My need to do this through asterisk is simply the ability to provide me
access with no additional cost to my customer. It seems like a nice
thing to include as long as authentication is done well. I have worked
on a dozen or more types of switches and all of them have supported this
or had the capa
Hi all,
I recently installed asterisk 1.2.4 on a HP DL140 G2 server and co-located it.
My only problem with the box is that there
is a noticeable delay in the processing of agi scripts compared to any other
install of asterisk I have.
Has anyone got any ideas why this is happening and any guide
Hello everyone! I'm planning on setting up a new system shortly and
can't pick the right card... We will have 2 or 3 lines coming in and
7 extensions (GXP2k's). Should I just get 2 or 3 X100P cards? Or do
I need the Sangoma A20200 or even the A20200D (Echo cancelation)...
I was thinking
Yuan LIU wrote:
A configuration string "boostringer" was mentioned in several messages,
including one concerning TDM400P, all without indicating the applicable
configuration file. This has no apparent effect on TDM400P wherever I
tried.
That would go in your /etc/modprobe.conf which controls
Hi, all
I am building a new server. Have installed FC 6 and put in TDM400 card.
Checked out latest asteriusk code, run make install in zaptel directory.
So far all is fine.
Now I am trying to install the drivers.
# modprobe zaptel
FATAL: Module zaptel not found.
Fair enough, no zaptel driver
On 12/14/06, Tobias Wolf <[EMAIL PROTECTED]> wrote:
Hmmm, there is really not much to share. Most of the code handles
Authentication or other stuff, like informing another server that a new
user has entered an conf-room, or updating databases.
Mostly I look an the CallerId to decide if this shou
Try
Exten => _351217588XXX, 1, Dial ( ... )
Thanks and Regards
--Sandeep Kalra
Ph: +91-120-4342000-X-2966
: +91-120-4342966 (direct)
M- 9810683168
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joao Pereira
Sent:
I think, Nokia E60/61/70 currently supports 802.1x
Joao Pereira wrote:
Do you know if it has 802.1x authentication as it is defined in
EDUroam ( http://www.eduroam.org/ ) ?
I never found a WiFi phone working with 802.1x I tested ZyXel
Prestige 2000 but the sound was bad and it doesnt supp
For PRI you have 3 main solutions. This is the order of stability (and
pricing):
1. Digium or Sangoma cards use the computer processor and that could be
bad if you have huge traffic through the PRI
2. Eicon Diva cards have their own processor, which releases the PC
processor and gives more s
Use ${EXTEN:0:9}
Regards,
Ove
Joao Pereira wrote:
Hello
how can I distinguish all the calls that arrive to my Asterisk
starting with: 351217588XXX ?
I want match the first 9 digits does Asterisk has any function for
this?
Thanks
Regards
Joao Pereira
_
Hello
how can I distinguish all the calls that arrive to my Asterisk starting
with: 351217588XXX ?
I want match the first 9 digits does Asterisk has any function for this?
Thanks
Regards
Joao Pereira
___
--Bandwidth and Colocation provided by Easy
Do you know if it has 802.1x authentication as it is defined in EDUroam
( http://www.eduroam.org/ ) ?
I never found a WiFi phone working with 802.1x
I tested ZyXel Prestige 2000 but the sound was bad and it doesnt support
802.1x :(
Thanks
Joao Pereira
[EMAIL PROTECTED] wrote:
No, the Gig
Mochamad Susantok wrote:
I have already use smokeping, and great for measure latency and packet
loss, but not voip packet especialy, or you has been modified smokeping ?
I have not modified it. I take it that if the network has considerable
latency, so will VOIP. It has been my experience tha
Hi all,
First let me say thank you for Lee Howard, you definitely found my problem
on sending faxes!
I'm using hy-email2fax to send faxes, and i notice that is there the problem
is starting, as the subject of my .eml file contains only the phone number
but then some how hy-email2fax is not dete
RR schrieb:
> And what would someone have to do to sweet-talk you into sharing this
> AGI ;)
Hmmm, there is really not much to share. Most of the code handles
Authentication or other stuff, like informing another server that a new
user has entered an conf-room, or updating databases.
Mostly I lo
hi all,
I'm trying to send text messages to Snom 300 to show the credit
remaining during the call...
Sending a MESSAGE directly to the phone via udp i'm able to update the
text on the display... but not during the conversation.
I read about AOC, but i can't find any documentation about Ast
Hello
How many simultaneous conversations g.729a should one expect with a WRAP board
running Asterisk?
Has anybody tried this?
Kind Regards
Jon Leren Schøpzinsky
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>Has anyone hooked up * as an extension/trunk of an Avaya system that has
>around 2 ISDN30e's.
I'm currently running an Asterisk box between my ISDN30 PRI and my
Argent Office (pre Avaya takeover of Network Alchemy but still the same
box as the Avaya IP Office).
All it took was a two PRI digium
On 12/14/06, Tobias Wolf <[EMAIL PROTECTED]> wrote:
Actually you don't need 2 different extension, but two different
parameter-sets for the meetme-App. So, you have to implement some logic
that detects, if the calling user has to be marked or not. It's your
choice if you do this by dialplan logic
Thanks Tzafrir and Marco for the info.
If I want to unload modules during start-up, I have to edit my
/etc/asterisk/mudules.conf and add something like;
noload => app_test.so
or I can unload them immediately at CLI using Mr. Cohen suggestion.
Regards.
> /etc/asterisk/modules.conf
>Marco
> I wonder if anyone can help me with this. I have 4 sites running
> Asterisk and these are linked via IAX trunks and ADSL lines. Calls
> coming into any of these sites are received locally and forwarded to a
> central operator. E.g. Call comes in on site A and is forwarded to
> the operator on
Savoy, Kevin - Williston, ND schrieb:
> I'll give this a try but seems silly to require 2 different extensions
> for one conference room. Thanks for the input.
>
Actually you don't need 2 different extension, but two different
parameter-sets for the meetme-App. So, you have to implement some logic
http://sipp.sourceforge.net/
SIPp is a free Open Source test tool / traffic generator for the SIP
protocol
On 12/13/06, Andre Luiz Martins Rodrigues <[EMAIL PROTECTED]> wrote:
Hello peoples,
I need to do a test of urgent stress. It know as much as connections
simultaneous my equipment is
Hello,
Do anybody know, if there is a way to connect 2 zap-channels with Hardware
TDM Switching?
Thanks
Nico
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Hey everyone !
This config worked !
; zapata.conf
[channels]
language=en
context=from-pstn
switchtype=euroisdn
pridialplan=local
signalling=pri_cpe
usecallerid=yes
hidecallerid=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
group=
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