Hello,
I would like to know which phones are IAX compatible.
Thank-you
Marios Moutzouris
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.338 / Virus Database: 267.10.7/70 - Release Date: 11/8/2005
___
Hi Newbie, or would you prefer to be called VoIP(y)? :-)
Thanks for the advice, It's great to hear from somebody that has
suffered in the same way :-)
I've cc'd in the dev and user lists mostly so that others looking for
the same issue (FXO PCI Master Abort) can find some info! - hope you
For example TEK SIP-IAX 323.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dr. Marios
Moutzouris
Sent: Wednesday, August 17, 2005 8:18 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] IAX compatible phones
On 17 Aug 2005, at 02:26, Don Fanning wrote:
I've surmized that it's Voipbuster having issues. Paid up another euro
on the second account and it works fine. When their support gets
better, I'll have them work on the other account.
I've had similar flakyness with Voipbuster. Sometimes the
Hi Newbie, or would you prefer to be called VoIP(y)? :-)
Thanks for the advice, It's great to hear from somebody that has
suffered in the same way :-)
I've cc'd in the dev and user lists mostly so that others looking for
the same issue (FXO PCI Master Abort) can find some info! - hope you
On Tue, Aug 16, 2005 at 02:40:36PM -0400, hugolivude wrote:
I'm running RedHat 9 with a TDM400 (2FXO, 2FXS).
I'd like to give my Asterisk users the option of cleaning up their
voicemail mailbox from their Windows PCs. I set up Samba and added
all the users with restricted access to their
Ma Zhiyong schrieb:
...
Trace shows that the fax is received successfully.
Aug 17 12:01:10 VERBOSE[19571]: -- Executing RxFAX(Zap/94-1,
Hi,
sorry, I don't know the solution to your problem, but I would like
to know, how did you get that trace?
I'm looking for a reliable way to
Thank you for your answer.
I didn't register on the domain of the Eyebeam software, actually I don't
understand how to do that!
I bouught 5 eyebeam activation keys and I am trying with the first 2 of
them
On the Eyebeam side (both eyebeam), I only enabled the Basic H.263 codec,
no other.
If, on
We are testing our Asterisk server prior to deployment. The server has
a 4 port T1 Digium adapter (WCT4XXP) with 3 Long Distance (LD) T1s and
one PRI for local calls.
We are using sipp from two different stations routing a test number out
the LD lines and another test number out the PRI line.
I took a look at the NEAX brochures available from NEC's website. I
may be wrong but I don't think you could change the way dtmf tones are
sent from the PBX, but you should be able to send them out of band
(with RTP, as per RFC 2833) from the cisco to the asterisk box.
Generally, out of band dtmf
They're using the same hosted servers with different billin schemes.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Burton
Sent: Tuesday, August 16, 2005 11:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
On Wed, 17 Aug 2005, Tzafrir Cohen wrote:
On Wed, Aug 17, 2005 at 06:57:19AM +0200, Remco Barende wrote:
After upgrading a CentOS 3.x box to CentOS 4.1 (both x86_64 with an
Athlon64) I also wanted to get the latest bristuff. Unfortunately
bristuff without florz causes the box to kernel panic
Hi!
I've read in the archives that there are problems concerning Nikotel calls
being disconnected after two minutes. I had the same problem yesterday. Is
there a fix? There was only a giving up statement after the last e-mail in
the archive, I'm about to do that too.
Here's my sip.conf entry
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Wed, 17 Aug 2005, Steve Underwood wrote:
Roger Schreiter wrote:
How can I enable asterisk to fax to itsself?
Well, it won't be the normal operation, but when allowing clients
to fax, it can happen by chance, that someone faxes to another
user on
On Tue, 2005-08-16 at 23:53 -0700, Pudenz, Duane wrote:
We are testing our Asterisk server prior to deployment. The server has
a 4 port T1 Digium adapter (WCT4XXP) with 3 Long Distance (LD) T1s and
one PRI for local calls.
We are using sipp from two different stations routing a test number
I'd like to have the ringing a caller hears to be
more like a 'british' ring when I am calling an internal extension. The
phones I'm calling already do this, now I'd like to find a way to make the same
thing happen for the caller who waits...
Any ideas?
Chris Coulthurst
[EMAIL PROTECTED]
Hi,
I am very new to Asterisk. I wanted to know how to retrive the Voicemails. I could see some voicemails assosiated with some extensions.
Any ideas??
How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos. Get Yahoo!
Hi!
I'm searching for a 1-800 number that simply plays music for a long time
(3mins) and no one picks up. I've bothered the ATT lines so far when trying
out my SIP-PSTN connection but then always someone answered :-)
Anyone have a number?
Christoph
On Wednesday 17 August 2005 10:29, Sharadindu Mohanty wrote:
Hi,
Hi!
Any ideas??
Yes, I do it in the following way. In extension.conf add this line:
exten = ,1,VoiceMailMain(s${CALLERIDNUM})
exten = ,2,Hangup()
Here any extension can call and then automatically gets directed to
Take
this as an example
[from-sip]
exten
= 2000,1,Dial(SIP/2000,20)
exten
= 2000,2,Voicemail(u2000)
exten
= 2000,102,Voicemail(b2000)
exten
= 2000,103,Hangup
exten
= 2001,1,Dial(SIP/2001,20)
exten
= 2001,2,Voicemail(u2001)
exten
= 2001,102,Voicemail(b2001)
exten
= 2001,103,Hangup
try bankone, their 1800 waiting is long
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Christoph
Eicke
Sent: Wednesday, August 17, 2005 4:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] 1-800 number
Hi!
I'm
More info
On 8/17/05 3:34 AM, Christoph Eicke [EMAIL PROTECTED] wrote:
Hi!
I'm searching for a 1-800 number that simply plays music for a long time
(3mins) and no one picks up. I've bothered the ATT lines so far when trying
out my SIP-PSTN connection but then always someone answered :-)
On Wednesday 17 August 2005 10:45, Michael K. Rodriguez wrote:
More info
I don't quiet understand your mail ;-)
Do you want more info from me?
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Thanks for the hint, do you know where to buy it (cheap) and the
price for it?
Thanks,
Roland
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of VoIP
Newbie
Sent: Wednesday, August 17, 2005 6:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
I am running asterisk at home but have a strange phenomena that is going on
with my flash panel I am using two ips an internal and an external public ip
address on my box. If I go to the page on my asterisk external ip address
the displays the flash panel everything is fine, but on the internal
I have several Asterisk servers installed and one SER server which will
act as a gateway to PSTN, en redirect server.
I was thinking to implement it the following way:
- Register all the * servers at SER (is this neccessary?) - this works
via register=asterisk:[EMAIL PROTECTED] in sip.conf
-
Hi!
I'm trying to start asterisk at boottime. Since SuSe has no rc.local like in
Redhat linux, I need asterisk starting script to /etc/init.d/rc3.d -directory
(I assume it is like that if i want automated asterisk startup).
Do you have any experience how this is implemented in SuSe, and if you
Hi All,
i have a HFC card running in ptp mode. I set overlapdial to yes and
immediate to no in /etc/zaptel.conf. DID works, but the timeout for
immediate=no is much to low. Calls from GSM or via Speed Dial work fine,
but you hardly can dial the digits fast enough to reach the extension
(im using
I did the same way but it is asking for some password and mailbox. I think mail box is extension no but what abt password?
Can i overide this procedure?
ThanksChristoph Eicke [EMAIL PROTECTED] wrote:
On Wednesday 17 August 2005 10:29, Sharadindu Mohanty wrote: Hi,Hi! Any ideas??Yes, I do it in
Hi All,
so i have a better description: DID does not work with match as you go
dialing, all at once is ok..
have a nice time,
Hari
Am 17.8.2005 schrieb Harald Klein [EMAIL PROTECTED]:
Hi All,
i have a HFC card running in ptp mode. I set overlapdial to yes and
immediate to no in
Hi,
This procedure will work under one condition --
your user names are same as your extension numbers. I have same problem. I was
giving phones alphanumeric user names, like "phone1".
When VoicemailMain is called with ${CALLERIDNUM},
it is actually called as VoiceMailMain("phone1"). As a
Great idea, thanks! I'd never heard of externnotify. I shudder to
think of how many other cool features I'm missing! I'll let u know
how it goes.
Cheers,
Hugh
On 8/16/05, Chris Coulthurst [EMAIL PROTECTED] wrote:
My suggestion would be, use the externnotify=/usr/bin/myapp feature in
Is there a way around this w/o giving everyone root privileges!
Do you want to allow every user to delete another user's voicemail?
If not, how do you sync voicemail users and samba users?
I want each user to see, read and write (delete) their own voicemail
ONLY (i.e. a user shouldn't be
I'd not bother with using the flash based 3 way calling. Instead I'd
setup an account with an ITSP and make the outbound calls via IP,
preferabbly via IAX2. That way to can reach out to as many people as
your bandwidth allows. Simply. Conveniently.
Add one IP based DID and you can let
Hi!
Then I get compile-errors.
Greets, Christian
Johann Steinwendtner schrieb:
Christian Wengel schrieb:
Hi!
I tried install-misdn.tgz from http://www.beronet.com/download/ ,
some minutes ago. Also I switched to an older kernel (2.6.8), but I
get the same error.
I think that I made the
Are you referring to the sip.conf setting or something in the phone's
config? Sip.conf already reflects rfc2833.
Craig Bruenderman
Network Advocates, Inc.
300 Envoy Circle
Suite 300
Louisville, KY 40299
Main: 502-412-1050
DID: 502-992-5929
Fax: 502-412-1058
Mobile: 502-548-1100
I have been reading with great interest the posts on trouble shooting
echo cancellation with *. Is it just coincidence that all of this
discussion has been with analog lines. Are PRI's susceptible to echo
problem like POTS lines.
Thanks for clearing this up.
Alan
Hi all!
I'm new to asterisk and I'm trying a simple config with:
- Debian GNU/Linux (unstable)
- last version of Asterisk
- a X100P card
I have a problem with dial out from a SIP software phone (XLITE) to a
public number (ex. my mobile phone), asterisk start the call, but nothing
happen...
If I
I have a PRI with quite a nasty echo problem that I cannot seem to get
rid of. I've tried all of the echo cancellation settings and tweaked
gains to hell and back but still get echo. I am convinced it can only be
addressed by hardware echo cancellation but that's not an option unless
I replace my
In addition to my previos e-mail.
'callerid' filed in sip.conf or iax.conf
(depends where user is defined) must be set to"
callerid "User Name" EXT
Where EXT is a number that will be picked up by
VoiceMailMain and will be used as a mailbox number.
Rudolf
- Original Message -
Alan Bunch wrote:
I have been reading with great interest the posts on trouble shooting
echo cancellation with *. Is it just coincidence that all of this
discussion has been with analog lines. Are PRI's susceptible to echo
problem like POTS lines.
Alan,
I have experienced echo on our
Bartek Kania wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Wed, 17 Aug 2005, Steve Underwood wrote:
Roger Schreiter wrote:
How can I enable asterisk to fax to itsself?
Well, it won't be the normal operation, but when allowing clients
to fax, it can happen by chance, that someone
Hello,
I use iaxcomm-latest from the iaxclient.sf.net page (binary
release) on linux, also tried Mac OS X version with the same result
and Asterisk 1.0.9 from Debian. Iaxcomm has a huge latency -- tens of
seconds, constantly changing over time. It was run on two different
machines, always to a
Hey Guys,
Wanted a Suggestion..Howz this Xorcom Asterisk?I am using it and till now its fine as currently it is in testing stage with 3-4 users.
Any Ideas???
ThanksSharadindu Mohanty
How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos. Get Yahoo!
I've done this without any problems. I changed from 10
digits to 11 digits and I'm still able to use all of
the cards.
--- Nate Kapi [EMAIL PROTECTED] wrote:
I currently have my astcc databases card lenghts at
7 digits long. I
would like to expand this to 10 digits now though.
Will I screw
I have experienced pretty nasty echo on my PRI w/TE110P. The echo was
only coming from other POTS lines, because cell phones already have
echo cancellation, and other PBX's had the same. I resolved the
problem by turning on the AGGRESSIVE option and it works fine now, and
we haven't noticed a
You could just add the line asterisk to /etc/init.d/boot.local
Angus
- Original Message -
From: [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, August 17, 2005 11:27 AM
Subject: [Asterisk-Users] Automatic
We like it alot. It makes rapid deployment of asterisk boxes a breeze.
Brent
On 8/17/05, Sharadindu Mohanty [EMAIL PROTECTED] wrote:
Hey Guys,
Wanted a Suggestion..Howz this Xorcom Asterisk?I am using it and till now
its fine as currently it is in testing stage with 3-4 users.
Any
We have a web interface where users can update their dialplan online
(not in production yet). The web page modifies the mySQL record.
It seems that some options are not re-read when caching is on, for
example, changing the caller ID value in the sip table has no effect
until a reload (or
Hello Everyone!
I'm wondering if the following is possible with asterisk...
What i'm trying to do is find a program or a solution that can help me set
appointments for a delivery company...
the program should call a person asking them if the following time is suitable
for a delivery... if they
Hi list,
anybody any example how to use it? I did not find any hint in the wiki
nor in the mailinglist archive :-(.
I want to use one button showing my agents the actual state (logged in
or logged off)
Thank you
Gerd
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Asterisk-Users mailing list
Hi all,
I dont know why, but if I plug my PC inside the 'PC' slot on my polycom,
this is not working. (Polycom IP600 is online on the net.)
I'm using normal network cables. (I see jumpers behind the phone... do I
need to play arround with that?)
Any help would be appreciated.
--
Alexandre
Yes, you could do that with Asterisk and Cepstral/Festival.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, August 11, 2005 6:36 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] is this possible
Yes, its possible and not too difficult. You can start here to see what
you can do with Call Files:
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
And a simple example of this in action is the perl wake up call application:
THe wiki doesn't seem to have any user reports.
If your using them, how are the working, better, worse about the same.
Also what hardware seems to be stable with them installed.
Alan
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-BEGIN PGP SIGNED MESSAGE-
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On Wed, 17 Aug 2005, Steve Underwood wrote:
Bartek Kania wrote:
If the call really dialed out through a PSTN port and back in it
should work. It is was a pure internal connection between 2
processes it will not. The timing for these programs comes
Alexandre Leclerc wrote:
Hi all,
I dont know why, but if I plug my PC inside the 'PC' slot on my polycom,
this is not working. (Polycom IP600 is online on the net.)
I'm using normal network cables. (I see jumpers behind the phone... do I
need to play arround with that?)
Any help would be
Hello there.
Does anyone have idea how to setup these two to work
together? I'm really going insane with this
combination... Any .conf files or something?
Cheers,
Vedran.
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Hello, I have such a problem. I have an * configured as a peer connected
to the gateway to PSTN.
While calling to the switched off cell phone, the gateway sends to the *
the SIP message 180 with the SDP part, and also a lot of rtp packets
containing the operator's in band info.
But *
There is a .php wakeup agi on voip-info too. I don't think it would be
that difficult to modify it to your needs
On Wed, 2005-08-17 at 06:54, Tim Pushor wrote:
Yes, its possible and not too difficult. You can start here to see what
you can do with Call Files:
you could declare the phone names as variables..
PHONE1=SIP/phone1
PHONE1VM=12345
On Wed, 2005-08-17 at 03:31, Rudolf Ladyzhenskii wrote:
Hi,
This procedure will work under one condition -- your user names are
same as your extension numbers. I have same problem. I was giving
phones
Hey Vedran
I did this a while ago but to put you on the write track you have to
register your gatekeeper (gnugk) with Asterisk as a gateway specifying a
prefix, let's say for arguments sake '0'. Then any numbers dialled on your
GK-managed H.323 network, that start with a zero, are routed to the
Man, I would really be grateful if you could put me out of my misery and
send me something, I don't know where's anything anymore in the config files
or anything. Too much editing those in the past 16 hours, I guess..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
There is a different approach to this;
Put a priority 'a' in the extension dialplan that goes to
Voicemmailmain(${EXTEN})
Users then dial there own extension from any location and press the *
key once voicemail picks up.
This method seems to emulate what most people are already used to.
If you
Hi,
I am having a strange problem. When ever I made a call, one leg is IAX and
other leg is OH323. The call establish fine but anybody talking from OH323 leg
side, I hear broken sound in IAX side. Something is wrong with RTP. Is it
something do with FRAME set in OH323? if so, what will be the
It seems that some options are not re-read when caching is on, for
example, changing the caller ID value in the sip table has no effect
until a reload (or expiration), so at least in some cases
rtcahcefriends
makes realtime notsorealtime.
No. It is doing exactly what it says it
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent: Wednesday, August 17, 2005 9:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] realtime caching
It seems that some
Hi Bartek, I posted the exact same problem last week - I found that if I
connected two Asterisk systems together via a PRI crossover cable and talk
txfax to rxfax then you get a T4 state timeout. I tried connecting ports
one and two together on a TE410p and also connecting a TE410p to a
On Wednesday 17 August 2005 7:27 am, [EMAIL PROTECTED] wrote:
Hi!
I'm trying to start asterisk at boottime. Since SuSe has no rc.local like
in Redhat linux, I need asterisk starting script to /etc/init.d/rc3.d
-directory (I assume it is like that if i want automated asterisk startup).
Do you
If you don't mind sharing, what was the vendor that worked great? Thanks!
On 8/17/05, VoIP Newbie [EMAIL PROTECTED] wrote:
I bought 3 from 3 different vendors. One of them has echo issue.
Another one has an issue regarding PCI master abort. Only one really
works fine for me. These 3 cards use
pruning breaks asterisk on high loads
at least on all 5 of our servers.
all using different versions and custom.
What you can do is use sip prune realtime name to remove just the
single peer/user from memory. And you can force a reload of that peer
from realtime by using sip show peer
When a user dial voicemail and just hangs up or enters the wrong
password 3 times asterisk will crash.
We are using Cisco 7960G with SIP
My asterisk is CVS-HEAD built on 2005-08-02 23:47:59 UTC
Any help would be great!!!
Thanks
___
Asterisk-Users
Chris Mason (Lists) a écrit :
Alexandre Leclerc wrote:
Hi all,
I dont know why, but if I plug my PC inside the 'PC' slot on my polycom,
this is not working. (Polycom IP600 is online on the net.)
I'm using normal network cables. (I see jumpers behind the phone... do I
need to play arround
It's in the wiki:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20snom
About halfway down the page where it says: SNOM SUBSCRIBE/NOTIFY
support for monitoring extension states
--
Tom Hayden
Astoria Telecom, LLC
www.astoriatelecom.net
On 8/17/05, Gerd Mueller [EMAIL PROTECTED]
It was fixed a while ago, download new code. There is a bug in the
tracker on it.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Hall, Eric M.
Sent: Wednesday, August 17, 2005 9:23 AM
To: Asterisk Users Mailing List - Non-Commercial
quickly this looks like a incompatible codec.. or unrecognized..
show codecs on CLI
show show
262144 (1 18) (0x4) videoh261 (H.261 Video)
524288 (1 19) (0x8) videoh263 (H.263 Video)
1048576 (1 20) (0x10) video h263p (H.263+ Video)
does it ?
On
Aug 17 08:37:06 WARNING[14731]: Don't know any of 0x8 formats
is from what you pasted btw..
Don't know any of 0x8 formats
is
524288 (1 19) (0x8) videoh263 (H.263 Video)
meaning it downst understand it or find it
On 8/17/05, Jimmy Smith [EMAIL PROTECTED] wrote:
quickly
On Wed, Aug 17, 2005 at 02:11:09PM +0100, Angus Comber wrote:
You could just add the line asterisk to /etc/init.d/boot.local
Excerpt from /etc/init.d/boot.local
# Here you should add things, that should happen directly after
# booting
# before we're going to the first run level.
Do not
Does anyone know if the current TDM400 card can take DID digits from the LEC?
If so is there any reference to how to set this all up? As I get my current
service from my LEC over an IAD, so would be sweet to just have trunks, not
each channel specific to a number.
Also if the above is
Damon Estep wrote:
I may have answered my own question, is it true that realtime extensions
are still queried every call, and only chan_sip is effected by
rtcachefriends?
Damon
True. RealTime Exensions are queried every time. There is no caching of
extensions.
If you turn on
Jimmy Smith wrote:
pruning breaks asterisk on high loads
at least on all 5 of our servers.
all using different versions and custom.
You should bug report this if you have a backtrace. Kevin and I worked
on the pruning stuff (well, he coded and i tested) for a while and
seemedly got it
Thanks for the feedback
Just for a background, one of the reasons for redundancy (notice the
quotes ;-) is that the PC is setup as a kiosk style application in which
we do a shell replacement with the Windows Explorer, so instead of a
desktop, the user gets a dedicated application which is very
Hi,
I'm trying to disable the internal dial plan of an Avaya 4602 with SIP
firmware 1.1 but couldn't find how to do it.
Even if I configure a custom Dial Plan it keeps adding other builtin
rules to my dial plan.
Ex:
Configured dial plan:
DialPlan
Greetings --
Many of you have downloaded and tried out Astmanproxy, a multi-threaded
C-based proxy for Asterisk's Manager Interface. It has been under
development since April 2005 and was presented at the Madrid Astricon in
June, and will also be presented at Astricon in Anaheim in October.
Greetings.
Has anyone made this work with BootROM 2.6.2 and app 1.5.2?
I've tried sending DHCP options 128, 144, 157 and 191 containing a
single digit (the VLAN ID) with the phone's 'Fixed' setting for DHCP VLAN
discovery. Different DHCP data types don't seem to help, as I've tested
with raw
Hi,
Im using asterisk 1.0.6 and I would let media path be connected directly between the
phones without going through Asterisk. I have to it with an AtCom320 (with
pa168s chip).
I just saw and tryied to
do what this page
Is there a way to cause an Iaxy to do distinctive ring?
Clint
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To UNSUBSCRIBE or update options visit:
It looks like you are sending calls out over one port. To help you out, we
will need to look at your extensions.conf and zapata.conf. My hunch is that
you are dialing out using something like
Dial(zap/g3/${EXTEN},20,) where the group of channels you're using is on one
port of your Digium
I will answer you, the same somebody told me at IIRC.
A watch has more processor power than a Iaxy...
So, in few words: No.
I already tried to have a lot ot things (callpickup, distinctive ring,
changing the time of flash pulse) and nothing...
El Miércoles, 17 de Agosto de 2005 11:33,
2005/8/17, Andrès Tello Abrego [EMAIL PROTECTED]:
I will answer you, the same somebody told me at IIRC.
A watch has more processor power than a Iaxy...
Uuuuh... well, I feel stupid but... what is the meaning of laxy ?
'cause... a watch... ;o)))
sorry for my ignorance...
Best regards,
YLB.
Colin,
Is there any reason why you couldn't just set up a T1 card and
channel banks (as many as needed) and use your exisiting agent
phones via zap channels?
Tom
On Aug 17, 2005, at 11:59 AM, Colin Stefani wrote:
Thanks for the feedback
Just for a background, one of the reasons for
Hi:
I was running TDM12B. Both FXS and FXO were working
fine. Then all of the sudden FXS had problems. When I
pick-up the phone and dial any number, FXS doesn't
respond. I just keep hearing the normal signaling line
tone comming from the FXS. I changed the FXS module
and it had the same problem.
I have a tdm 3xfxs and 1xfxo, aslo I have a setting with 1 snom 190 and 2 xten
line.
I can call from the snom to the ptsn line at the fxo port ok.
I can call from the ptsn to the xten lite phone.
I can call from the xten lite to snom
but
what I CAN`T do is;
Call from xten to ptsn. When I
Hi:
I would like to know what are the issues I need to
look for in a chipset board so I can make sure it
works fine with digium cards and Asterisk . Is intel
board 865 fits the description?
Regards;
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail
A iaxy, is a CPE device that provides VOIP capabilities to normal phones,
using the iax protocol...
So is a little hardware, for telephony usages, which doesn't have a lot of
features, and is't so cheap...
El Miércoles, 17 de Agosto de 2005 11:54, Yoann Le Bihan escribió:
2005/8/17,
Sorry it took me so long to keep on this thread. But I
got a quation Rich. Can the impedance missmatch kill
the dial tone completely?
This is, when I plug my X100p clone card to my line
the dial tone just goes away. I check this by using an
analog phone that is also on the line.
Is it possible
[EMAIL PROTECTED] ~]# netstat -naptu | grep asterisk
tcp0 0 0.0.0.0:20000.0.0.0:*
LISTEN 9231/asterisk
udp0 0 0.0.0.0:27270.0.0.0:*
9231/asterisk
udp0 0 0.0.0.0:45200.0.0.0:*
9231/asterisk
udp0
On Wed, Aug 17, 2005 at 07:48:29AM -0400, hugolivude wrote:
Is there a way around this w/o giving everyone root privileges!
Do you want to allow every user to delete another user's voicemail?
If not, how do you sync voicemail users and samba users?
I want each user to see, read and
I have one Asterisk system working with a Junghanns BRI card and another
working with a Digium TDM card with an Intel D865 motherboard.
Angus
- Original Message -
From: jonny hashem [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, August 17, 2005 6:14 PM
Hi:
I was running TDM12B. Both FXS and FXO were working
fine. Then all of the sudden FXS had problems. When I
pick-up the phone and dial any number, FXS doesn't
respond. I just keep hearing the normal signaling line
tone comming from the FXS. I changed the FXS module
and it had the same problem.
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