Ivan Lopez wrote:
-I have configured my Asterisk server and my Snom phones to monitor
extension state using the SUSCRIBE/NOTIFY instructions on the WIKI.
-Current Version of Asterisk:
Asterisk CVS-Nv1-2-0-rc2-11/16/05-21:16:33
-I get the subscriptions:
asterisk*CLI sip show subscriptions
On Wed, December 7, 2005 2:24, Marc-andre Poupier said:
Also I have another question about the voicemail system, is it possible
in my message to say HI you have reached me blah blah if you need to
speak to so and so press on this number to reach him and the call would
be transferred back to an
C F wrote:
Yeah, it shoud NOT work 100% of the time (maybe not even 50%)
So, are there any IP faxes?
--
Best regards,
Bartosz Piec
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Hi all,
can anyone tell me when (or how) * starts generating ring tone when a
call is made. The reason I ask this is I have an E1 coming from a PBX
into my * box (CVS 19/07/2005). I have some intermittent problems.
1. Sometimes no ringtone is generated, so I dial a number, and the
person
Hi,
I have a problem with loading res_perl ins asterisk
1.2 stable:
res_perl.so]Dec 5 03:06:53 WARNING[11563]:
loader.c:325 __load_resource: /usr/lib/asterisk/modules/res_perl.so: undefined
symbol: ast_bridge_callDec 5 03:06:53 WARNING[11563]: loader.c:554
load_modules: Loading module
On 11/25/05, Denny Schierz [EMAIL PROTECTED] wrote:
Is there anybody, who has asterisk 1.2 with misdn running?can anybody help?
Hi Denny,
I have it running. One card is in TE mode and it is connected to
outside line and one card is working in NT mode and I have Siemens ISDN
base station
Hi All
We are evaluating Nextone MSC for use as a outbound
proxy for our asterisk.
Asterisk version 1.0.9
We use Canreinvite=yes in asterisk.
When two end points are connecting via MSC, asterisk
issues reinvite to both the end points but thereafter
no media flows between endpoints.
Has anyone
Hi
I
am looking at implementing few features on Asterisk.
-Custom
Call Routing
-Ringing
Line Preference
Can
anyone tell me how to configure these?
I
am using a SJ Phone .
Appreciate
your help
Thank
you
Regards
Sudhir
Confidentiality Notice
The information contained in this
Can you explain scenarios in a bit detail?
--- [EMAIL PROTECTED] wrote:
Hi
I am looking at implementing few features on
Asterisk.
-Custom Call Routing
-Ringing Line Preference
Can anyone tell me how to configure these?
I am using a SJ Phone .
Appreciate your help
Thank you
Thanx for your help.
I will do some home work and thenget back to you.
Do you think the problem is of the asterisk version??
main issue:
* after running for some time the phone logs out (gets out of registeration).
Regards
Amna
On 12/6/05, Dan [EMAIL PROTECTED] wrote:
Hi,I really like DIAX and
I did use IAX2 but sound quality wasn't that good which codec are you using
with IAX2 ?
*** REPLY SEPARATOR
***On 12/6/2005 at 9:22 PM Alvaro Parres wrote:
Why using SIP instead of IAX2 ???
Only a question becouse i always use IAX
On 12/6/05, Waldo
Hi,
Where can I find Asterisk modules description? For example, I need to
know what is app_zapateller, app_zapbarge, app_zapscan and chan_zap
--
Thanks,
Eugene Prokopiev
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Hey folks,
I have my linuxbox connected to a PBX through a digium te110p card, E1 line.
The asterisk is set up to be the timing master for the line.
recently run into the following error message when starting asterisk:
The message:
-
== Primary D-Channel on span 1 down
Dec 7 11:34:02
Hi
Cab Asterisk accept h323 RAS packet( registration) using OH323 channel.
--
Thank You,
Code Lover
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Hi,
Exactly I want to switch landscape to portrait. Where can I switch it
and what (bash script) can I use for it?
Cheers
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Are there any GDK users on the list?
Do you know if its possible to disable voicemail on a
per extension basis such that it returns busy rather than diverting to
voicemail?
I have the manuals but I cant see any reference to
this.
Regards
Steve
The information
We're about ready to go ahead with a nice 6 line (maybe later 8) ISDN setup with
[EMAIL PROTECTED] and the quad Junghanns card.
Before we do, could anyone confirm for me that BT's ISDN2e lines do actually
provide Asterisk with the DDI number? We need to be able to route incoming
calls based on
While I'm not sure about the 1.2 ChangeLog the 1.2.1 that was released
recently has an real Changelog that is linked at the top of Asterisk.org.
http://ftp.digium.com/pub/asterisk/ChangeLog-1.2.1
It looks like they put some effort into this one so I hope it continues for
future releases.
Hi
I'm not sure if I can ask it here or on Dev. But the following
I'm trying to make a webinterface for calling/hanging up/transfer.
Thats all going very well.
But I'm wondering about the following. I'm trying to catch a busy
call from the CLI or the manager.
If I call someone and it takes to
Its all there
show commands
show functions
or
just read the configs.
internetworking with other devices is where the labor is
On 12/6/05, Douglas Garstang [EMAIL PROTECTED] wrote:
I was going to bite my tongue on my response to this, but keeping quiet is
driving me nuts.
If this is
If I remember correctly an E1 has two D-Channels. Check your notes on
what channels 31, 32 really do.
On 12/7/05, Laszlo Megyer [EMAIL PROTECTED] wrote:
Hey folks,
I have my linuxbox connected to a PBX through a digium te110p card, E1 line.
The asterisk is set up to be the timing master
Hi all,
I am looking for a cheap VoIP to GSM provider (most notably to GSM
networks in The Netherlands), but so far the cheapest I have found is
VoipGATE(.com, not .nl), and their prices are slightly (if using the Econo
package) more expensive than the normal ISDN/PSTN rates
The cheapest
On Wed, 2005-12-07 at 13:58 +0300, Eugene Prokopiev wrote:
Hi,
Where can I find Asterisk modules description? For example, I need to
know what is app_zapateller, app_zapbarge, app_zapscan and chan_zap
Start asterisk. To get an overview of all available apps in the console
type:
show
Hi,
I will do some home work and then get back to you.
Do you think the problem is of the asterisk version??
main issue:
* after running for some time the phone logs out (gets out of
registeration).
I am not aware of such an issue, but I cannot exclude it before test
it.
Unfortunately I
On Wednesday 07 December 2005 07:42, Andrew Latham wrote:
Its all there
No, it's not. There are TONS of configuration options that are not described
adequately, or described wrong. We fix them as we come across them but yours
is one of the WORST replies in this thread. Source and
E1 PRI has only one D channel and it is correctly defined as channel 16
in this config. You probably just need a T1/E1 crossover between
asterisk and the PBX. Just cross pin one with four and two with five.
I have even cut a cat5 cable, stripped the wires and crossed the pins
appropriately and
Bartosz Piec wrote:
C F wrote:
Yeah, it shoud NOT work 100% of the time (maybe not even 50%)
So, are there any IP faxes?
Sort of. There's IAXmodem, which should be run on the same box as
Asterisk itself (even thought it connects to * via IAX2); combine that
with HylaFAX and you can
Hi all,
can anyone tell me when (or how) * starts generating ring tone when a
call is made. The reason I ask this is I have an E1 coming from a PBX
into my * box (CVS 19/07/2005). I have some intermittent problems.
1. Sometimes no ringtone is generated, so I dial a number, and the
We are looking for a high density PRI-to-SIP gateway for
our call center and IVR applications. The device must take in a
channelized DS3 and output SIP g729a to multiple Asterisk servers. We have
looked at the Cisco AS5400XM, Lucent APX 1000 and Quintum Tenor CMS (fronted by
an Adtran M13).
Would like to find out if it is possible to setup a VoIP network with
asterisk and 9 x polycom IP 501 or 600 handsets, the main difficulty
is
that there is no cabling in place, and it isn't possible to run
cabling
(heritage building, need to demolish walls + half the ceiling to get
the
I have a client we provided with 2 Sipura 3000 units for FXO on his two
voice lines, he also has a fax machine connected to it's own line, and
he has several Polycom phones on the network. This all works great. He
has a shop in the building that is connected by a single network cable.
In the
Hello all,
I am running Asterisk with Digium E1 card with zaptel, libpri, asterisk
cvs v1-2. My server is interfaced with EWSD v16 using a PRI on E1. I am
running into a problem that at my telco's end alot of trunks are getting
BPRM (Block permanant) status. I am not sure why EWSD is blocking
If I understand correctly what you are trying to do, you wish to rotate the
image for viewing. This is usually handled at the client level - most PDF
viewers handle this with ease.
If you really want to rotate every inbound fax, you could add a line to the
bash file to rewrite the ORIENTATION
2005/12/6, C F [EMAIL PROTECTED]:
Yeah, it shoud NOT work 100% of the time (maybe not even 50%)
Then... ¿there is not any way to connect a hardware fax to an asterisk pbx?
--
Alejandro Vargas
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2005/12/7, rommel malana [EMAIL PROTECTED]:
easier for me. Can the asterisk act as a softswitch for managing call
traffic for other voice gateways?
If you mean connecting asterisk to one or more voip providers and
route calls to and from them, yes, it can do it and dones not need any
module.
2005/12/7, xcel [EMAIL PROTECTED]:
I did use IAX2 but sound quality wasn't that good which codec are you using
with IAX2 ?
The sound quality doesn't depend on the protocol but the codec you are
using and the bandwidth you are giving to it.
--
Alejandro Vargas
I tried G711 and GSM and in both cases call quality degraded when the softphone was conferencing more than 2 people (note: not a meetme room).- WaldoOn Dec 7, 2005, at 5:45 AM, xcel wrote: I did use IAX2 but sound quality wasn't that good which codec are you using with IAX2 ? *** REPLY
Buy the O'Reilly Asterisk book. It describes them in one of the
apendixes.
- Waldo
On Dec 7, 2005, at 5:58 AM, Eugene Prokopiev wrote:
Hi,
Where can I find Asterisk modules description? For example, I need
to know what is app_zapateller, app_zapbarge, app_zapscan and chan_zap
--
try with ${DIALSTATUS} -
http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS
you could start an agi(which receives the dialstatus-variable) and let this agi
write to your webapp-db...
regarding recall you should have a look at callfiles -
try with ${DIALSTATUS} -
http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS
you could start an agi(which receives the dialstatus-variable) and let this agi
write to your webapp-db...
regarding recall you should have a look at callfiles -
We're about ready to go ahead with a nice 6 line (maybe later
8) ISDN setup with [EMAIL PROTECTED] and the quad Junghanns card.
Before we do, could anyone confirm for me that BT's ISDN2e
lines do actually provide Asterisk with the DDI number? We
need to be able to route incoming calls
this can be usefull for you.
http://www.voip-info.org/wiki-IAX+versus+SIP
I guess that for Voice Over IP only, IAX is by far a better choice,
since SIP is designed for any kind of session, not just voip calls. IAX
is most efficient using bandwidth, you just need 1 port, and works
behind firewalls
On Tue, 6 Dec 2005 20:50:25 +0100, '[EMAIL PROTECTED]' wrote:
I wan't users that are within the same area code
to be able to dial each other using just their
extension as well as using area code +
extension. Users in other area codes are of
course only available through area code +
Subject:
[Asterisk-Users] RE: OH323 user configuration
From:
Code Lover [EMAIL PROTECTED]
Date:
Wed, 7 Dec 2005 14:06:40 +0300
To:
asterisk-users@lists.digium.com
Hi
Cab Asterisk accept h323 RAS packet( registration) using OH323 channel.
--
Thank You,
Code Lover
Asterisk oh323 is an
Doug,
We are in the process of implementing a large asterisk cluster for a client
of ours using 6 servers. We tried sharing the sip contact information using
the realtime architecture, but that did not work. Instead we resorted to
using the Realtime Static configuration and many dialplan tricks
There are two things about Attended Transfer
1. When you place onhold a user and want to dial another one
when the other one is busy you are automatically back to the
original call.
Is it possible this automatically back to the call to be skipped,
the user to here busy and
he must
suppose i want to do something on call hangup
how can i detect it
in the extensions.conf?
Regards
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Hi,
If you were to lead someone (with a UI) through the process of
configuring a a Digium T1/E1 card with asterisk and a T1/E1 trunk from a
provider, would the following questions cover most scenarios? i.e. given
the following questions and assumptions, would the configurations below
work
Jerry Geis wrote:
Jason,
I added:
nat=never
qualify=no
and I still cant get a UIP200 to ring when calling it after using 1.2?
Any other suggestions?
Well, we're using UIP200's (BS 4.63 firmware) with 1.2. There is a bug
(5780) re: rfc2833 g729 in 1.2 tarball but that doesn't seem to be
Quoting Patrick Lidstone \\(Personal E-mail\\) [EMAIL PROTECTED]:
Yes, but you need to subscribe to one or more MSNs to do so. MSNs are
available with ISDN2e or Business Highway, but not Home Highway. The
distinction is a marketing, rather than technical, one. If you need a block
of
Patrick Lidstone (Personal E-mail) wrote:
We're about ready to go ahead with a nice 6 line (maybe later
8) ISDN setup with [EMAIL PROTECTED] and the quad Junghanns card.
Before we do, could anyone confirm for me that BT's ISDN2e
lines do actually provide Asterisk with the DDI number? We
need
Anybody have a manual or link for a manual for the AMP? english or spanish
thaks in advance
__
Visita http://www.tutopia.com y comienza a navegar más rápido en Internet.
Tutopia es Internet para todos.
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Dear All,
Any one knows where to buy did's from Japan or Exchange them with US and UK did's ?
I need them for re-selling on didx.net-- Rehan Ahmed AllahWalahttp://www.SuperTec.com - Tommrow's Technology, Today.
http://www.didx.net - DID Number Exchange and Peering Service.
Quoting John Daragon [EMAIL PROTECTED]:
Patrick Lidstone (Personal E-mail) wrote:
We're about ready to go ahead with a nice 6 line (maybe later
8) ISDN setup with [EMAIL PROTECTED] and the quad Junghanns card.
Before we do, could anyone confirm for me that BT's ISDN2e
lines do actually
I've tried to configure the services-key on my Polycom 501 to run a
SpeedDial-entry in [MACADRESS]-directory.xml (which would call a
asterisk-extension that starts SayUnixTime) but i have not been able to
accomplish my goal. Whe configuring the SpeedDial-function in sip.cfg VolUp
is started
This might be an obvious question, but should you be using a crossover
cable?
Information on setting up Nortel to TDM card links can be found at:
http://www.pham.org/asterisk/asterisk-meridian-a1.pdf
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web:
So, I'd like to get some feedback on how it
might work if we
simply put a wireless access point at each workstation, and
used the 4 port switch to connect to the PC + polycom handset.
In my experience, wireless signals have a really poor range in elderly
buildings - they're usually built of
Title: Message
Anyone out there
have any experience with setting up asterisk and a Lucent TNT...I need some help
with setup and configuration...I have no experience with Lucent...I have
referenced the wiki on voip-info.org but does not do much good since i also need
help configuring the
Can you do a ISDN message trace in LD 96 on the M1 when you try to bring up
the D-Channel?
LD 96
enl msgo 10
enl msgi 10
Make sure you later do a
dis msgi 10
dis msgo 10
To shut it off.
You should see good info there.
-Original Message-
From: Anthony Rodgers [mailto:[EMAIL
I can't seem to get my outgoing connections to work with IConnecthere. At one
time it did with v1.0
I can register and receive calls just fine. But can't make them.
Ultimately, the trace ends with a 400 Bad Request error when you do a SIP
debug.
Has anyone got it to work with v1.2? Don't know
The ringtone on your Grandstreams is indeed set in the phone itself. I think
they hold up to 4 ringtones (default, custom 1 2 3) which can be configured
either per line or different rings on different caller ID. Grandstream
have a freely available utility to convert PCM ringtones into the
Dogers wrote:
Quoting John Daragon [EMAIL PROTECTED]:
snip...
When you say ringtones, do you mean sounds like a UK phone when it
rings, or sounds like a UK phone when we ring someone else ?
It does actually sound okay when we ring someone else, but when it rings, it has
the long single
I am compiling a app file writting in C for asterisk and I am getting the following errors: ../include/asterisk/file.h:27:2: #error You must include stdio.h before file.h!In file included from app_akEventsProxy.c:17:../include/asterisk/file.h:56: error: syntax error before '*'
jourdan lemieux wrote:
I am compiling a app file writting in C for asterisk and I am getting the
following errors:
../include/asterisk/file.h:27:2: #error You must include stdio.h before
file.h!
Any help please on this!!
How much clearer can that be? Your source file is out of date
It seems that Asterisk 1.2.1 is on the Digium FTP, but no posts to the
users lists, nothing in the wiki?
Everybody still asleep? Looking forward to the changelogs :)
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I'm looking for a provider that can offer VoIP origination and
termination in the domestic US and Puerto Rico. To be more exact,
Toll Free numbers origination is a must. I'm looking for a block of
100 domestic toll free numbers and 100 local DIDs. Estimated traffic
is about 100K
Anyone out there have any experience with setting up asterisk and a Lucent
TNT...I need some help with setup and configuration...I have no experience
with Lucent...I have referenced the wiki on voip-info.org but does not do
much good since i also need help configuring the E1's...
any help would
We are proud to announce that Asterisk 1.2.1 has been released!
This release of Asterisk contains a number of bug fixes over version
1.2.0. See the ChangeLog at
http://ftp.digium.com/pub/telephony/asterisk/ChangeLog-1.2.1 for more
details.
It is available from the ftp.digium.com FTP servers, as
Hello all,
I know the TDM cards (and I assume the TE cards) provide a timing source to
be used for IAX trunking etc., but is it possible to use a BRI card running
under zaphfc as a timing source, or should one run ztdummy as well?
Thanks in advance.
Regards,
Chris
--
C.M. Bagnall, Director,
Its also on www.asterisk.org.
Jared Armstrong
-Original Message-
From: Remco Barende [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 07, 2005 12:39 PM
To: Asterisk Users List
Subject: [Asterisk-Users] Asterisk 1.2.1 released
It seems that Asterisk 1.2.1 is on the Digium FTP, but
http://ftp.digium.com/pub/telephony/asterisk/ChangeLog-1.2.1
On 12/7/05, Jared Armstrong [EMAIL PROTECTED] wrote:
Its also on www.asterisk.org.
Jared Armstrong
-Original Message-
From: Remco Barende [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 07, 2005 12:39 PM
To:
Hello all:
I'm currently running the latest version of asterisk, and using
chan_bluetooth
I am using the usb/bluetooth dongle from Compusa:
http://www.compusa.com/products/product_info.asp?product_code=312330pfp=SEARCH
I am also using the M2500 Plantronics headset - 29.95 from Frys
things are
On 12/7/05, Bartosz Piec [EMAIL PROTECTED] wrote:
C F wrote:
Yeah, it shoud NOT work 100% of the time (maybe not even 50%)
So, are there any IP faxes?
Sure, it's called email.
___
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Yes there is, using TDM, but not VoIP.
On 12/7/05, Alejandro Vargas [EMAIL PROTECTED] wrote:
2005/12/6, C F [EMAIL PROTECTED]:
Yeah, it shoud NOT work 100% of the time (maybe not even 50%)
Then... ¿there is not any way to connect a hardware fax to an asterisk pbx?
--
Alejandro Vargas
Hello,
I'm trying to figure out how to setup live recording of a phone call.
I've read all the docs at the wiki, but can't seem to figure out how
to implement it.
I'm running asterisk 1.2
I have the Polycom IP500 SIP phones.
In a perfect world, I would dial something to start recording, and
Yes, according to the document link that you provided and most other
sources, a T1 crossover cable is required to connect the Nortel Meridian to
an Asterisk server. Here is a summary of the important settings that I
have:
Nortel Meridian Asterisk Digium TE405P
I've got an account that's looking at doing some cable/VoIP
integration. They were wondering if it were possible to set up
something like this:
PSTN (T1) - Asterisk - (some VoIP protocol, probably SIP) - Siemens
soft switch - their product
It sure sounds nice in theory, but I've never
Anyone out there have any experience with setting up asterisk and a Lucent
TNT...I need some help with setup and configuration...I have no experience
with Lucent...I have referenced the wiki on voip-info.org but does not do
much good since i also need help configuring the E1's...
any help would
Is is possible to record video and audio when using SIP with video?
Regards,
Chris___
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Update on this... And it is still not solved.
This is actually fairly interesting. I have two installations at a
construction company. They are both running similar class machines (I
was wrong in my initial post) they are:
System A
2.4 ghz Celeron
1 gb RAM
IDE Drives
[EMAIL PROTECTED] 1.13
On Wednesday 07 December 2005 11:10, John Voss wrote:
I can't seem to get my outgoing connections to work with IConnecthere. At
one time it did with v1.0
I can register and receive calls just fine. But can't make them.
Ultimately, the trace ends with a 400 Bad Request error when you do a SIP
The device that interfaces to the PSTN is the interface that must cancel
echo. If I read your post correctly, that is the SAP-3000 and the
Audiocodes boxes in your case.
Jeff Busch wrote:
Update on this... And it is still not solved.
This is actually fairly interesting. I have two
Guys, Im wondering, is anybody in Mexico using any kind of door phone with
asterisk?
Please drop me a note.
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Jeff:
I'm not an Asterisk person but I play one on TV :-) Actually I just use
it for voicemail and ancillary services from my SER proxy so take this
message with the appropriate caution.
I'd look at and possibly tweak parameters in the phone.conf file.
echocancel, txgain and rxgain look
Ken D'Ambrosio wrote:
I've got an account that's looking at doing some cable/VoIP
integration. They were wondering if it were possible to set up
something like this:
PSTN (T1) - Asterisk - (some VoIP protocol, probably SIP) - Siemens
soft switch - their product
It sure sounds nice in
I'm not in Mexico, but I'm sure what I use here works in Mexico as
well (BTW, it's on the wiki). I have successfuly used:
Valcom
VikingElectronics
doorfonebell
I might have spelling wrong on the last one. Too lazy to look it up on
the Wiki :)
On 12/7/05, Anton Krall [EMAIL PROTECTED] wrote:
Correct.
The issue is that most of the echo is between internal stations. SIP -
SIP.
The users with the system using the sipura's don't report any echo when
calling outside the office or receiving a call.
The users with the system using the audiocodes report an echo for the
first 1 - 2
Hi. I just upgraded my asterisk server to a better
P4 2.6 Ghz.
I thought everything went smooth until someone
tried the Meetme.
I seems the ztdummy won't compile on the new
server.
I am running Mandriva 2006 on the new
server.
After downloading in /usr/src/ I uncommented the
#ztdummy to
Hi
We're trying to migrate our platform from 1.0 to 1.2 and we're seeing
some oddness in app_queue.
We use local_channels a lot for things like persistent agents,
call-forwarding on agents and such. Now on our 1.2 server we notice that
the queue is listing all members as 'Invalid' (thus any
On Wed, Dec 07, 2005 at 09:20:38PM +0100, Insider KT wrote:
Hi. I just upgraded my asterisk server to a better P4 2.6 Ghz.
I thought everything went smooth until someone tried the Meetme.
I seems the ztdummy won't compile on the new server.
I am running Mandriva 2006 on the new server.
Hi list:
i have an asterisk box behind the NAT ,when i try to
send calls through Sip to the voip provider server the
call is answered but in a one way calling,I hear the
voice of the other side just for 4 seconds and then
stop but the call do not hangup.
my sip.conf is:
[voip provider]
type=peer
Insider KT wrote:
Hi. I just upgraded my asterisk server to a better P4 2.6 Ghz.
I thought everything went smooth until someone tried the Meetme.
I seems the ztdummy won't compile on the new server.
I am running Mandriva 2006 on the new server.
After downloading in /usr/src/ I uncommented the
On Wed, Dec 07, 2005 at 09:20:38PM +0100, Insider KT wrote:
Hi. I just upgraded my asterisk server to a better P4 2.6 Ghz.
I thought everything went smooth until someone tried the Meetme.
I seems the ztdummy won't compile on the new server.
I am running Mandriva 2006 on the new server.
After
Hi all,
I'm finding with Asterisk 1.2.1 (and 1.2.0) that when connecting over an
unauthenticated IAX2 connection (ie: as [guest] in iax.conf), a codec
will always fail to be negotiated (see trace snippet below).
The problem appears to be specific to only unauthenticated IAX2
connections.
We have the PanCode
Door Phones which works with asterisk via the ata box. If you
are interested please contact me off list. I am sure it will work
anywhere.
C F wrote:
I'm not in Mexico, but I'm sure what I use here works in Mexico as
well (BTW, it's on the wiki). I have successfuly used:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dogers
Sent: 07 December 2005 16:24
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] UK ISDN2e with DDI?
Quoting John Daragon [EMAIL
Hi,Have you followed the instructions outlined in README.udev ?HTH,KunalOn 12/8/05, Insider KT
[EMAIL PROTECTED] wrote:
Hi. I just upgraded my asterisk server to a better
P4 2.6 Ghz.
I thought everything went smooth until someone
tried the Meetme.
I seems the ztdummy won't compile on the
List users,
I am experiencing segmentation faults in AstManProxy. If anyone could
help me identify their source, it would be appreciated.
The pertinent information is below. Please let me know if you need any more.
Asterisk Version
Asterisk ABE-A.2-beta
AstManProxy Version
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
David Cook
Sent: 07 December 2005 21:26
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] UK ISDN2e with DDI?
Try adding the following to your handset
I was thinking ethernet-over-power myself, but I haven't tried it yet
That plus those 8 port netgear POE switches might work well.
PaulH
- Original Message -
From: Chris Bagnall [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
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