Hi * users,
Is that possible to make voicemail audio file (that is attached to forwarding email) as MP3 file, rather than WAV?
TIA
Kuni
--
Kuniyoshi Murata
English-Japanese Interpreter Macintosh Webcast Specialist
[WebSite]
Hello
How do you configure Polycom for presence please ?
Harry
--- Alvaro Parres [EMAIL PROTECTED] a écrit :
Hi list, i have the next problem:
I create 3 hints.. (111 (SIP/111), 112 (SIP/112),
and 102 (ZAP/35) )
the SIP/111 is a GrandStream ATA
the SIP/112 is a Polycom 301
the ZAP/35 is
http://www.hylafax.org/
Harry
--- Doug Lytle [EMAIL PROTECTED] a écrit :
Jason Brashear wrote:
Receiving faxes with Asterisk.
Is there a good resource for learning how to set
this up?
www.soft-switch.org
Doug
--
Ben Franklin quote:
Those who would give up Essential
Gervais de Montbrun ha scritto:
**keepalive = 5
set the keepalive to 60 or more
speeddial = 500,500,[EMAIL PROTECTED]
that phone should not be able to display a hint status so
speeddial = 500,500
This is what is displayed in the console when I try to call the 12SP
from the ATA
Hi Seong,
The creditlimit mean the amount of credit you authorize the CardHolder to go
in negative. You should have try ;-)
Rgds,
/Areski
On 11/11/05, Ah khng [EMAIL PROTECTED] wrote:
Hi all,
I'm glad to hear that areskicc v3 have been released. But i have a
problem to use a2billing as
Hi:
I have been posting this issue for over a month and I
am not irritated. I appreciate all the users who
helped before and I thank in advance any offer for
help including reasonable paid time.
You may sent me your price me list at
[EMAIL PROTECTED]
Regards;
chawki
--- Rob Lith [EMAIL
Hi Abdock,
1# You can set a context in iax.conf [mytrunkiax] with the username,
secret, host, etc.. and then use the name mytrunkiax in A2Billing it
will dial using this trunk.
This will allow you to configure the willing codec.
2# Directly use in the Edit trunk, username:[EMAIL PROTECTED], I
Sorry,
Here are some files
Harry
--- BJ Weschke [EMAIL PROTECTED] a écrit :
This is good debugging info you've listed below,
but this isn't a sip
debug/trace.
To do that, first verify in your logger.conf file
you have the following line:
full = notice,warning,error,debug,verbose
Morning all,
I'm trying to rewrite my dialplan macros into AEL. How does one handle
result-dependent branching (e.g. VoiceMail will branch to n+101 if mailbox
doesn't exist) in AEL? Or is there a better way of doing this?
Thanks in advance.
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur
Bukoka Budoka a écrit :
Hi,
i installed A2billing according to instructions. GUI works fine, i
entered a new card and i put the appopriate context in my
extensions.conf:
[callingcard]
exten = _123,1,Answer
exten = _123,2,Wait,2
exten = _123,3,DeadAGI,a2billing.php
exten = _123,4,Wait,2
Here are some other files.
Why asterisk send sip OPTION message to agents ?
Harry
2005-11-11 11:23:10 WARNING[1639]: chan_sip.c:1045
__sip_xmit: sip_xmit of 0x81cf940 (len 477) to
192.168.0.20:-1 returned 5060: Operation not permitted
On Wed, 9 Nov 2005, Nir Simionovich - CTO wrote:
but hey, I spend my nights debugging boards and
sending back remarks to Intel on how to make their boards better for
Asterisk.
Heh,
A customer of mine discovered that the onboard sound hardware on Intel
Desktop boards created an echo -
On Wed, 9 Nov 2005, Kevin Bockman wrote:
Waldo Rubinstein wrote:
One T1 is with one carrier, who provides timing signal.
The other 3 T1s are from a different carrier, all sharing the same
timing signal.
Based on this, I have in /etc/zaptel.conf something like:
I expect that
Hi All,
I've a very strange error.
I've configured a Cisco gw with * and when an incoming call is arriving from
the Cisco to * asterisk will always put the call in the default context (ignoring
the part in the [Cisco])
I'm attaching my conf files:
[general]
port = 5060 ; Port
Im willing to help for free contact me via msn messenger my id is
[EMAIL PROTECTED]
On 11/11/05, chawki hammoud [EMAIL PROTECTED] wrote:
Hi:
I have been posting this issue for over a month and I
am not irritated. I appreciate all the users who
helped before and I thank in advance any offer
Hi there
I am using an IAX2 softphone built from the IaxClient library dialing into
Meetme conferences. It works fine most of the time, but sometimes calls are
being dropped and this error is given:
Chan_iax2.c:1480 attempt_transmit: Max retries exceeded to host 146.18.3.5
on IAX2/[EMAIL
Hi!
I am running Asterisk (1.0.8) on gentoo (2.6.13-gentoo-r5), with AVM C2
ISDN card. capiiinit works OK, capiinfo shows card is up and running
with CAPI OK, but asterisk refuses to load the capi-cm module
(chan_capi-cm, 0.5.4) giving the warning
CAPI not installed, CAPI disabled!.
Any hints of
On Wed, 9 Nov 2005, Kyle Hagan wrote:
We purchased a new Dual Xeon 3ghz, 2gb ram to upgrade our 3ghz Pentium
1gb ram, that has been having load issues due to our growing company.
We are having problems... We use a predictive dialer that we custom
programmed in perl. It basically drops,
Stephen Arulraj wrote:
Has anyone got these SIP firmwares for the Siemens IP Phones? Would
appreciate it.
Thanks and regards,
Stephen
Stephen
You can download a copy from my website here:-
http://chaz6.com/static/files/sip_v2_3_14.app
Regards
--
Chris Hills
I.T. Services
North East
Hi!
I am running Asterisk (1.0.8) on gentoo (2.6.13-gentoo-r5), with AVM C2
ISDN card. capiiinit works OK, capiinfo shows card is up and running
with CAPI OK, but asterisk refuses to load the capi-cm module
(chan_capi-cm, 0.5.4) giving the warning
CAPI not installed, CAPI disabled!.
Any
On Friday 11 November 2005 02:25, Jacques Beyers wrote:
I have installed Asterisk 1.0.9 on FC3. I have recently installed a Digium
TDM400P and a red Digium FXO module, which I hope to connect to the PSTN so
I can make outbound calls. The FXO card is installed in port 1, and the
telephone jack
When the polycom ip300 phone (1.6.2) send registration
SUBSCRIBE message is sent to buddies from directory
file so NOTIFY is received from these one.
When I want to change status the ip phone don't send
NOTIFY to subscriber unlike SER which is a proxy!!!
Why?
Harry
--- harry gaillac [EMAIL
Great! tnx matt!On 11/11/05, Matt Florell [EMAIL PROTECTED] wrote:
It's CVS v1-0. Digium has said that they will do a release of 1.0.10at the same time they release 1.2.I highly recommend upgrading to this if you are still on the 1.0 tree.It has a lot of bug fixes, and the new v2 firmware telco
Armin Schindler wrote:
Correct permissions to access /dev/capi20 for Asterisk?
Duh! Of course it had to be something as trivial as that! Thanks!!
Julf
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Hello,
I try to setup presence with polycom ip phones ip300
(1.6.2) .
I added buddies in directory files all is right for
registration subscription notification but when i want
to change status notify message is not sent to
subscribers ?
I don't understand !
Regards
Harry
I have installed Asterisk 1.0.9 on FC3. I have recently installed a Digium
TDM400P and a red Digium FXO module, which I hope to connect to the PSTN so
I can make outbound calls. The FXO card is installed in port 1, and the
telephone jack is inserted into port 1.
No matter how I try, I
Message: 5Date: Fri, 11 Nov 2005 08:11:09 +0100From: Stefan-Michael. Guenther (in-put GbR)
[EMAIL PROTECTED]Subject: [Asterisk-Users] Softphone with Lotus Notes support?To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]Content-Type: text/plain;charset=utf-8Hi,has anyone of you
I have installed Asterisk on SuSE 10.0 with an active firewall/NAT filter.
The server has connection to my own Intranet (private IP) and to Internet
Everything works well for clients behind and in front-of the firewall
but they can not communicate with each other. Signalling gets through
but the
We had a Rev I card that did not work. We sent it
back to Digium and had it reflashed back to H.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob
LithSent: Friday, November 11, 2005 1:40 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
Aíeee Caramba, intro_prompt is really to have a customized message at
the initiation of the call. please read the comment above in the
configuration file.
Try from shell to run the script and press enter until you get back to
the shell,
then you will see perhaps if an error occur on the AGI. if
A Linksys WRT54G runs for about $60. I think it supports QoS out of the
box, but flash it with 3rd party firmware (i.e. Sveasoft) to get a bunch
of extra features.
Note: The latest version, version 5, of the WRT54G only has half the
memory of the previous versions and there is no 3rd party
[DISA-context]
exten = 204117733,1,DigitTimeout(8)
exten = 204117733,2,ResponseTimeout(15)
exten = 204117733,3,DISA(no-password,default)
exten = 204117733,4,Hangup
We succesfully implement a dialup gateway with the following.
What I now wish to do is to be able to make multiple telephone
Hi Peter -
When you set up the DHCP pool in Cisco you need to use syntax like:
-- option 66 ascii a.b.c.d
Thanks!
I guess maybe I didn't explain very well. I did get this far, and this
seems to work well, if I manually set the phone to read an ascii string.
I'm being really picky here,
Hi Gabor,
I'm not sure about USB ISDN adapters but I'm using an AVM Fritz ISDN
PCMCIA card with asterisk and chan_capi very sucessfully. The notebook
is in a remote location and is solar powered.
You'll find that these cards are cheap on Ebay - its a German card and
you'll probably find
What version are you running, and is your [Cisco]
definition the last one in the file? I have the same problem with 1.0.7,
and the ugly fix I came up with was to add a dummy entry as the last sip
entry.
B. J.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Sent: Friday, November
n+101 feature is deprecated and is no longer supported in Asterisk.
All applications are modified to set exit status variable. Use something
like
VoiceMail(b${EXTEN});
if(${VMSTATUS} = FAILED) {
Noop(mailbox doesn't exists);
}
On Fri, 2005-11-11 at 10:11 +, Chris Bagnall wrote:
Asterisk sends OPTIONS message if the device have qualify=NNN option
set.
On Fri, 2005-11-11 at 11:24 +0100, harry gaillac wrote:
Here are some other files.
Why asterisk send sip OPTION message to agents ?
Harry
2005-11-11 11:23:10
In the wiki it states:
When an extension is dialed, the command tagged with a priority of 1 is
executed, followed by command priority 2, and so on. This goes on until:
the call is hung up,
a command returns a result code of -1 (indicating failure),
a command with the next higher priority
Is there a way to have a separate MOH/Media server for playing music
and/or audio prompts/files?
I have an * box where calls come in and sit in a queue until an agent
is available. I noticed that at the end of the day, I end up with a
bunch of zombie mpg123 processes for calls that were
Hi,
Thanks Dave, gracias Jose
Luis ;-).
Once everything is
configured, the mobile phone connected via bluetooth.. I've got a segmentation
fault when trying to dial from sip to bluetooth:
CLI Nov 11 16:53:34 NOTICE[]:
/usr/src/chan_bluetooth/chan_bluetooth.c:2041 try_connect:
On Sat, 5 Nov 2005, Shane DeRidder wrote:
I've been scouring the mailing list archives for an answer to this, and
cannot find one. I'm hoping someone else out there has run into this.
Communication between the TNT and Asterisk seems to be operating
properly, but I'm unable to accept or
Trying to get this working on FreeBSD 5.4.
zaptel-0.10_1 driver from ports.
Digium TDM400 ( don't remember all these codes,
but the card has fxs module in the 1st socket, and fxo module
in the 4th one.
/usr/local/etc/zaptel.conf
loadzone=us
defaultzone=us
fxsks=4
fxoks=1
When I start the
I have a script that doesn't quite fit your needs, but does send out email
reminders for on a regular basis, and runs as a daemon. If you are
interested, please let me know and I will send it to you. A little warning,
this was one of my first major perl scripts, so it may be a little ugly
and
Hi,
Does anyone know if GPS data is available from a cell phone (GPS cell
phone) in a similar fashion as CallerID. I saw a past posting where the
GPS data is emailed - which just seems strange... Being able to
integrate such data into a dial plan could lead to all sorts of
applications.
Hi.
I`m using asterisk 1.0.9 and it works fine.
I didn`t patch the asterisk, only I followed the README of
chan_bluetooth.
Regards,
José Luis
El vie, 11-11-2005 a las 14:58 +, Victor Alvarez escribió:
Hi,
Thanks Dave, gracias Jose Luis ;-).
Once everything is configured, the mobile
I have an * box where calls come in and sit in a queue until an agent
is available. I noticed that at the end of the day, I end up with a
bunch of zombie mpg123 processes for calls that were once on hold and
this seems to be eating up memory.
There should not be several zombies remaining,
I have one also that just does nag paging. It looks up the extension in
the db and gets the pagers to notify. Sets x number of attempts and if a
user checks his messages it will clear the remainder of the pager
attempts. Written in perl. Not a daemon, uses the run_external_notify.
Sends one
Hi everyone.
I have a small problem with my Asterisk setup?!?
I am trying to connect to another endpoint through my asterisk server.
The packet going in is just like i want it, but the packet going out of
asterisk at to the other endpoint is missing a part in the header?
it looks like this:
To:
Do you know anywhere to find information
about this?
MVH
Amund
Nygaard
A NOVO Norge AS
Fra:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] På vegne av Dean Collins
Sendt: 10. november 2005 15:27
Til: Asterisk
Users Mailing List - Non-Commercial Discussion
Emne:
Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com on November 11, 2005 at 10:10 AM -0400 wrote:
set the keepalive to 60 or more
OK. I set this to 120
that phone should not be able to display a hint status so
speeddial = 500,500
Thanks. I've made the
Hi everybody,
I have this issue: one particular Read command seems not work and return an
empty string immediatelly.
This is CLI output(partial)...
-- Goto (ask_aster,s,1)
-- Executing Read(SIP/2000-0b6d, aster|asterisco|2|skip) in new stack
-- Accepting a maximum of 2 digits.
-- Playing
Hello List!
I wrote something to allow me to easily interact/configure Asterisk thru a WEB
interface. Over time I added several things to it. I thought 'you all' might
get some use from it. I call it 'astwebmgr'.
You can get it here: http://www.micpc.com/astwebmgr
It is written in PHP
Hello,
I forgot to mention the version details..
Asterisk CVS-v1-0-11/08/05-01:22:43
spandsp-0.0.2pre21
libtiff-devel-3.6.1-8
libtiff-3.6.1-8
Could this be a problem with my provider cos they support only alaw and
ulaw ?
Regards
Dushyanth
Hey all,
Iam trying to receive fax's over a
Hi does any one have the sip.ld file of a SoundStatios IP 4000
Thanks.
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People,
Im tring to use 2 e-1 in Brazil. In order to get R@ signaling,
I compilled libdsp, unicall and stuff following www.soft-switch.org comparing with
another site (Dezert of Zazamora, in Mexico). Asterisk is running fine
with Asterisk but I cant make calls.
The guys on the telco
Mabio Coelho wrote:
People,
I’m tring to use 2 e-1 in Brazil. In order to get R@ signaling, I
compilled libdsp, unicall and stuff following www.soft-switch.org
http://www.soft-switch.org/ comparing with another site (Dezert of
Zazamora, in Mexico). Asterisk is running fine with Asterisk but
Steve,
The line is good, because we looped in my end and everything is ok.
Regarding the card, it might be faulty, but I doubt it because I've tested
with two cards.
My config files might be broken, that is why I've posted my config files in
my previous post but it got stripped by the list. So
Hi list, i have the next problem:
I have conifgured hint for all my extension ( SIP and ZAP) but at the console
i send show hints and always all the channels are idle..
My config files:
at extension.conf
...
[sip-test]
exten = 101,hint,ZAP/35
exten = 101,1,Dial(ZAP/35)
exten =
Mabio Coelho wrote:
Steve,
The line is good, because we looped in my end and everything is ok.
Regarding the card, it might be faulty, but I doubt it because I've tested
with two cards.
My config files might be broken, that is why I've posted my config files in
my previous post but it got
Hello,
Asterisk don't support IM presence because of no proxy
function in chan_sip !
Regards
Harry
--- harry gaillac [EMAIL PROTECTED] a écrit :
When the polycom ip300 phone (1.6.2) send
registration
SUBSCRIBE message is sent to buddies from directory
file so NOTIFY is received from
I've tried to leave the wanpipe configuration as vanilla as possible. I
just turned of the hardware HDLC (that is because I've been told that if
Hardware HDLC turned off, Sangoma cards are 100% compatible with
digium/tormenta2 cards).
Here is my wanpipe configuration:
[devices]
wanpipe1 =
Hi,
Asterisk is 1.09, I've tried to
change that like you suggested but no luck.
When I'm doing sip debug, its look
like it always go to the default sip context.
I've a second sip host definition and
that works, exactly the same configuration just different IP.
Could that be a
anyone using a high availability server set up for Asterisk ? I saw IBM
had some kind of solution at VON but was too busy exhibiting to check it
out. :(
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On 11/11/05, Chuck Bunn [EMAIL PROTECTED] wrote:
Hi,
Does anyone know if GPS data is available from a cell phone (GPS cell
phone) in a similar fashion as CallerID. I saw a past posting where the
GPS data is emailed - which just seems strange... Being able to
integrate such data into a dial
I'd love to see both of these scripts, if only to help me get started
crafting my own. Can you guys post them to the list so that others
will be able to find them in the archives?
Tom
On Nov 11, 2005, at 10:24 AM, James Armstrong wrote:
I have one also that just does nag paging. It looks
It's more like a research project going to proof of concept.
Was very interesting tho.
-bill
On 11-Nov-05, at 12:23 PM, Matthew Simpson wrote:
anyone using a high availability server set up for Asterisk ? I
saw IBM had some kind of solution at VON but was too busy
exhibiting to check it
Hi,
I am wondering if it is possible to tweak IAX2 protocol to packetize audio
data
more efficiently. I would like to try setups where multiple audio frames
(gsm)
are combined into single UDP packet. I know that it will incur delay in
audio
streams but I don't care. Primary concern is to lower
Gervais de Montbrun ha scritto:
**I did this in the console and the output is below. It does not seem
to say much to me about audio.
Dunno why, but the phone is not sending an open receive channel ack. In
fact it does ot open the rtp media port so the channel don't know where
to send (udp
On Friday 11 November 2005 13:00, Branko Samardzic wrote:
I am wondering if it is possible to tweak IAX2 protocol to packetize audio
data
more efficiently. I would like to try setups where multiple audio frames
(gsm)
are combined into single UDP packet. I know that it will incur delay in
On Fri, 2005-11-11 at 12:41 -0500, BJ Weschke wrote:
On 11/11/05, Chuck Bunn [EMAIL PROTECTED] wrote:
Hi,
Does anyone know if GPS data is available from a cell phone (GPS cell
phone) in a similar fashion as CallerID. I saw a past posting where the
GPS data is emailed - which just seems
I am running into issues with this same setup and would like to
update the wiki with information on connecting an avaya legend to an
asterisk server via a T100P. Please post your experiences with the
legend and asterisk so we can compile a great list of step by step
instructions for the
In Canada, Bell is pushing a CDMA-based geolocation service as a
subscription add on to your plan. Unfortunately, you are required to use
their crappy web app although one could probably hook the data with some
well-crafted wget's and grep's
-Original Message-
From: Austin Denyer
Why would you have all those modules loaded on an asterisk server?? Do
you *REALLY* even have a PCMCIA slot on your server? Do you need USB? Or
parallel port? Do you use IPv6 with asterisk (not supported AFAIK)??
even bluetooth and stuff is running!
I know I need to remove alot of things,
Might be sharing a brain today... Here is my config as it stands:
/etc/zapata.conf
span=1,1,1,esf,b8zs
bchan=1-23
dchan=24
loadzone= us
defaultzone = us
/etc/asterisk/zaptel.conf
switchtype = 5ess
signalling = pri_net
channel = 1-23
On the merlin side, I have:
ADS1 SLOT
Hi all
iam new to this VoIP
iam just looking to deploy
IP PBX Services
I have 4 port gateway to call Around the world
i want to setup with Asterix the followings
local user authentication and billing
user authenticated need to route to 4 port gateway
call record track
call start time, end
On Wed, 2005-11-09 at 12:45 +, Are wrote:
We want to intergrate AstBill with a Groupeware or CRM but want input
what people will prefeer.
On our list today we have
http://www.sugarcrm.com/crm/
http://www.vtiger.com/
http://www.egroupware.org/
A couple more worth looking at. Don't
hello,
http://www.egroupware.org/ would be a good choice (
open source).
--- Patrick [EMAIL PROTECTED] a écrit :
On Wed, 2005-11-09 at 12:45 +, Are wrote:
We want to intergrate AstBill with a Groupeware or
CRM but want input
what people will prefeer.
On our list today we have
Hi everyone !
I'm running Asterisk 1.0.9 and testing it with a GrandStream BudgeTone 101, a Linksys PAP2-NA Gateway, and 2 X-Ten Lite clients.
Everything works fine, except the BudgeTone is not showing the name of the calling extension only shows the extension number.
In the sip.conf file i
Hi,My information is that Asterisk/Portaone Radius behind a NAT cannotsend start accounting packet to SIP, so no call accounting...confirm, anyone?
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Hmmm, I tested this quite a bit as per below...
Sorry if this seems lame, but you are using FTP right? Because FTP is the
default, not TFTP (even though you use the DHCP TFTP option to set the FTP
server address).
Peter
Case 1 (by FTP from current bootrom and application versions):
Good Afternoon,
The next Asterisk Users Group meeting has been scheduled for tomorrow,
November 12th at 11:30am.
Meetings are held monthly on the second Saturday of each month, excluding
July and December.
Sound Choice Communcations is located in Bloomington Minnesota, just 1/2
mile west
Hi Peter -
Hmmm, I tested this quite a bit as per below...
Sorry if this seems lame, but you are using FTP right? Because FTP is the
default, not TFTP (even though you use the DHCP TFTP option to set the FTP
server address).
Thanks Again! I haven't tried yet with the 3.x bootrom series.
Can anyone recommend a source for IAX2 phones located in the USA?
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To
On 11 Nov 2005, at 12:49, Paul Davidson wrote:
**
As someone who uses and develops Notes and Asterisk on an almost
daily basis, I can tell you two things:
1. Technically, all softphones 'support' Lotus Notes- if Notes knew
how to pass them a number,
2 SIP phones on Y data connector on 1 ethernet -
will that cause problems ?
thx in advance
_
Dont just search. Find. Check out the new MSN Search!
http://search.msn.click-url.com/go/onm00200636ave/direct/01/
At least use a hub or switch (preferred)
But if you MUST use a Y connector make sure the adapter meets the
International Data 10T
Standard
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of A_ Navone
Sent: Friday, November 11, 2005 3:54 PM
To:
We are considering Quantumvoice as a provider -
They are telling us they will give us 1 line number but we can have 5
concurrent incoming and outgoing line numbers. Charge is about $45 +
extras - this seems considerable less expensive than the competition
which seem to focus on.
My second
I was testing Broadvoice few weeks before Hurricane Wilma here in FL.
Since then, I had been since the landline (Bellsouth), and I had to
'remote callfwd' the BS # to my broadvoice #.
So, from my impression, is ok for my needs (I got a weird no ringback
problem that I kind of solved with a
Julio Arruda wrote:
I was testing Broadvoice few weeks before Hurricane Wilma here in FL.
Since then, I had been since the landline (Bellsouth), and I had to
'remote callfwd' the BS # to my broadvoice #.
So, from my impression, is ok for my needs (I got a weird no ringback
problem that I
I have the same problem but I did not think it was a problem. I don't
think the display supports alpha characters.
Carlos Prieto wrote:
Hi everyone !
I'm running Asterisk 1.0.9 and testing it with a GrandStream BudgeTone
101, a Linksys PAP2-NA Gateway, and 2 X-Ten Lite clients.
In my experience, no it does not support alpha only digits
-Original Message-
From: Paul [mailto:[EMAIL PROTECTED]
Sent: Friday, November 11, 2005 2:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Non-numerical caller id in Budgetone 101 Ip
Turn on full logging with loglevel=255 in unicall.conf, and send me a
log when a channel locks up.
Steve
Thank you for your answer.
In the below log, loglevel=255, the Unicall/2 is locked up, it stay in
Bad State. It starts work well, but at about 8:26:48 it's locked up
until the next reload
Second post
I have installed Asterisk on SuSE 10.0 with an active firewall/NAT filter.
The server has connection to my own Intranet (private IP) and to Internet
Everything works well for clients behind and in front-of the firewall
but they can not communicate with each other. Signalling gets
I am trying to test whether a callerid number is a valid ten digit
number. I'm a total novice with regular expressions.
I've tried:
exten = s,n,GotoIf($[${CALLERIDNUM} : \d{10,10}]?label)
But CLI gives an error. Can someone please show me what the correct
syntax would be to do this?
Thanks,
Title: [Announce] Web-MeetMe v1.4.0
New Features-
- Weekly recurring meetings with the same room and pin numbers.
Any conflict in the conference number as identified before
the conference is added, allowing the submitter to change
the conference room number
- Database storage of
Hi Andrew,
thanks for your prompt response. However, I am not sure whether IAX trunking
can be of any benefit on 33.6kbps link. It shows significant bandwith
reduction
on 2 and more simultaneous calls. My question relates to single call over
such
link.
Current measurement say that gsm call
I've been jacking with this for a while but don't
understand all thatI'm reading...
The problem is sometimes I get ANI II digits from
the phone company. These will be two digits that prefix ANI- so some
callerid might arrive as only "00" or "007147391234", "00714", "714"
or
How about you stop pulling your hair out and let me send you one of the
56k modems I have sitting on my desk. heh
On Fri, 11 Nov 2005 13:00:15 -0500, Branko Samardzic
[EMAIL PROTECTED] wrote:
Hi,
I am wondering if it is possible to tweak IAX2 protocol to packetize
audio
data
more
On Wednesday, November 09, 2005 5:57 PM, Jon Reynolds wrote
Hello,
I have an AAH-1.5 with a TMD400P with four lines, 8
Grandstream GXP-2000
phones, I am having echo issues on the GXP-2000 side.
I have evaluated a similar setup as yours involving the Granstream 2000.
I was able to isolate two
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Still looking for any advice with this. I had given up with the upgrade
process (to SIP.. tftp won't send the files for some reason) but I can't
even get this to work with sccp. It doesn't seep to ever finish booting.
My understanding is that after
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