Once you've finished drooling over the UTStarcom you can start drooling
over the Linksys WIP330
http://ces.engadget.com/2006/01/07/linksys-wip330-in-da-house-but-you-cant-have-one/
VERY nice phone in my opinion.
Regards,
Rupert G
Philip Edelbrock wrote:
We're getting our feet more and
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Yeah, that should theoretically work, but I've got about 60 cisco phones
that don't respond to the check-sync.
If you ever make it work, please anounce it on the group.
--
Tomislav Parcina
[EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
It does with the latest BETA firmware. But it dosn't
seem to work to well. It stops working and the phones
have to be rebooted.
works good, as long as asterisk doesn't get restarted. then you need to
reboot the phone. it's a bug.
trixter aka Bret McDanel wrote:
I havent looked, I am sure that its there somewhere on grandstreams site
but where is the latest beta located?
all info can be found on http://www.voip-info.org/wiki/view/GXP-2000
___
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In article [EMAIL PROTECTED],
Somesh S Shanbhag [EMAIL PROTECTED] wrote:
Hi Asterisk-users,
I am working with Aterisk Manager API's.
I can login successfuly with the following.
char buff[256];
strcpy(buff, Action: Login\r\nUsername: admin\r\nSecret: unix\r\n\r\n);
send(msock,
On Mon, Jan 09, 2006 at 11:10:59PM -0800, Somesh S Shanbhag wrote:
Hi Asterisk-users,
I am working with Aterisk Manager API's.
I can login successfuly with the following.
char buff[256];
strcpy(buff, Action: Login\r\nUsername: admin\r\nSecret: unix\r\n\r\n);
send(msock, buff,
Kevin Bockman wrote:
If you are using 1.2, I would use native (codec, not MP3). There
should be an example in the sample config file in
/usr/src/asterisk/configs/musiconhold.conf.sample - I don't see it on
the Wiki. It should be there, somewhere. Must be buried. For this
option, you
hi,
My apologies for repeating this question, but I hoped re-frasing it
might help.
I would like to assemble an PABX larger than what you possible can put
inside one Asterisk box. What is the best way to do this? Can it be done
at all with Asterisk? Any ideas or hints would be apreaciated.
http://lists.digium.com/pipermail/asterisk-users/
May 2016? November 2007? Woot? Some kind of delayed Y2K bug?
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Has anyone tried out Hitachi IPC-5000 ?
It looks nice and it's a bit exensive, but I would still like to hear
how does it behave around Asterisk.
Ivan
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To
I have tried Zyxel P2000W , it works very fine.
- Original Message -
From: Asterisk-User [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, January 10, 2006 3:23 AM
Subject: RE: [Asterisk-Users] Recommendations on
We have a billing system that depends on the CDRs. We had a guest that
made a one minute call to a local cellphone, this call went out Zap
channel through our channel bank. The CDR recorded a 200 minute call,
but I checked with the Telco's records and it had terminated after one
minute. What
On Tuesday 10 January 2006 13:06, Jean-Michel Hiver wrote:
http://lists.digium.com/pipermail/asterisk-users/
May 2016? November 2007? Woot? Some kind of delayed Y2K bug?
Randal Law lives in future.
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Philip Edelbrock wrote:
We're getting our feet more and more wet with VOIP at work. We want to
experiment with a good wireless (as in WiFi) phone. What would be a
good phone to impress my boss with?
depending on the needs, Kirk phones are very nice. They are DECT, but
they've got an own
Jean-Michel Hiver wrote:
http://lists.digium.com/pipermail/asterisk-users/
May 2016? November 2007? Woot? Some kind of delayed Y2K bug?
More like misconfigured client machine.
The 2016 ones are from: Randal Law
The 2007 ones are from: CW_ASN - Gus
:)
/me wonders what would happen if my
Robert La Ferla wrote:
Chris Albertson wrote:
Second even if there were one the mpg123 process is not long lived. A
new one is started for each
MOH session.
I hate to say it but there is a problem where the mpg123 process never
terminates. This occurs with the latest
Geoff Manning wrote:
Our users are experiencing some unacceptable delay when trying to have a
conversation. The delay is so noticeable that they keep stepping on each
others words and resort to calling the customers via cell phone.
We've had some pretty bad delay in the past when customers
[EMAIL PROTECTED] wrote:
We had phones in Perth hooked up to an Asterisk box in Melbourne, and the call
was fine - so I know it can be done.
I'm currently in Italy and I've had a few conversations that lasted a
few minutes before the person at the other end realised I was in Italy
:) (from
Hi all,
I am trying to
configure Asterisk using Mysql database. Well I am using Asterisk version 1.0.3.
And I have the asterisk-addons which contains the file res_config_mysql.c to
create the res_config_mysql.so file which used as Mysql Realtime Configuration
Driver. But on compilation
Hi,
We have recently setup a Live Demo of DRUID our Asterisk management interface
product. Also I'd like to thank all of you that took the time to download the
trial edition and give us your feedback. WE've tried to incorporate as much
of that feedback into our new updated release.
Feel free to
Forwarded to OCLUG, LUGIE UUASC which have members that have expressed
interest in asterisk.
Mike
Kerry Garrison wrote:
The SoCal Asterisk Users Group will be meeting at the Heritage Park Public
Library on the corner of Walnut and Yale in Irvine on the 3rd Thursday every
month. The
Is it possible to increment a call group (Max_Calls) from one server to
the next server?
I'm trying to set a maximum call limit between two facilities (2
Asterisk servers) using Set Group. I want the limit to be set at 4.
The problem with the code that I have in place, is it allows for 4 on
Don't waste your money. It works with Asterisk but it's a pain to setup
and use. It's too expensive but at least the firmware is starting to
get there.
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Asterisk-User
Sent: 10 January 2006 11:24
To:
Rich Adamson wrote:
There will be a delay associated with any sip-to-sip call, but it
should not be all that noticable unless both the talker and listener
are in the same room.
Are you sure this is a delay problem, or might it be a half-duplex
problem?
If any of the hardware mentioned
Rupert Gregory a écrit :
Once you've finished drooling over the UTStarcom you can start
drooling over the Linksys WIP330
http://ces.engadget.com/2006/01/07/linksys-wip330-in-da-house-but-you-cant-have-one/
VERY nice phone in my opinion.
I dunno... it looks like a cell phone, except it's
Hi
I have a proxy server running and i want to have a sipura IP phone behind
it.
it does not work, but it works when it's behind nat, not proxy. is there a
place in Ip phones to give a proxy address.
please help me to configure this.
Regards
Kani
Another, much cheaper option is to get DECT phones and connect them to
IAXy's:
DECT-PHONE ((( * ))) DECT-BASEIAXy[=IAX2=]Asterisk- TheWorld(tm)
That's pretty much what I do at the moment, except that I connect the
phone to a Fritz!FonBox integrated 2 FXS SIP / DSL modem router.
Hi
I just discovered an interesting product line. Not tested yet...
http://www.paesys.com/fr/lecteurs_VoIP_WiVoip_VideoIP_GSM.htm
In french, sorry...
Any feedback ?
---
Guillaume de Lafontaine
___ D W A M ___
-Original Message-
From: Jean-Michel Hiver
Hi
I have a proxy server running and i want to have a sipura IP phone
behind
it.
it does not work, but it works when it's behind nat, not proxy. is
there a
place in Ip phones to give a proxy address.
please help me to configure this.
Regards
Kani
I guess when you say proxy you do
There will be a delay associated with any sip-to-sip call, but it
should not be all that noticable unless both the talker and listener
are in the same room.
Are you sure this is a delay problem, or might it be a half-duplex
problem?
If any of the hardware mentioned is operating
we had the same problem, so we switched it to madplay (do a search in
the wiki)
and that issue stopped.
Best Regards
Greg Cirino
[EMAIL PROTECTED] Virus Spam Free
and you can't do better than that!
http://www.cirellemail.com
Cirelle Enterprises Inc.
25 Indian Rock Rd #421
Windham NH,
Matt Riddell (IT) wrote:
Geoff Manning wrote:
Our users are experiencing some unacceptable delay when trying to
have a conversation. The delay is so noticeable that they keep
stepping on each others words and resort to calling the customers
via cell phone.
We've had some pretty bad delay
Kevin Bockman wrote:
If you are using 1.2, I would use native (codec, not MP3). There
should be an example in the sample config file in
/usr/src/asterisk/configs/musiconhold.conf.sample - I don't see it on
the Wiki. It should be there, somewhere. Must be buried. For this
option, you will
Just a quick guess tells me its hangup detection failure.
On 1/10/06, Chris Mason (Lists) [EMAIL PROTECTED] wrote:
We have a billing system that depends on the CDRs. We had a guest that
made a one minute call to a local cellphone, this call went out Zap
channel through our channel bank. The
Too bad this wasn't a couple weeks ago. We just sold a huge lot of unused
Polycom IP300's for $99/each.
---
Date: Mon, 09 Jan 2006 15:28:28 -0500
From: Ken D'Ambrosio [EMAIL PROTECTED]
Subject: [Asterisk-Users] Decent sub-$100 SIP phone.
To: Asterisk Users Mailing List -
I was wondering, how does the PCI bus effect echo and latency?
I know a large part of the echo issues out there have to do with the pci
bus latency, but are there any suggestions on the hardware side to
minimize this? For example, a 533mhz vs 800mhz fsb, would it have any
effect on
Robert La Ferla wrote:
How can you convert mp3 to gsm? mencoder? Do you have an example?
Once again, documented on the wiki :-)
sox is your friend; it can convert between pretty much all available
codecs, except G.729 and G.723.1.
___
Our users are experiencing some unacceptable delay when trying to
have a conversation. The delay is so noticeable that they keep
stepping on each others words and resort to calling the customers
via cell phone.
We've had some pretty bad delay in the past when customers have been
Figured it out :)
Basically, you have to have a file called syncinfo.xml in the tftp root
directory, with the following contents:
SYNCINFO
IMAGE VERSION=* SYNC=1/
/SYNCINFO
Also, in SIPDefault.cnf or the phone's configuration file, stick:
sync: 0
somewhere so the phone's sync value doesn't
The Austin Asterisk User Group is meeting next Monday. See
aaug.bybent.com for more information on the location. Topic and date
aren't correct on the page, but the location and time are.
--
Chris Tooley
512-646-1507
[EMAIL PROTECTED]
___
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On Tue, Jan 10, 2006 at 01:16:51PM +0100, Matt Riddell (IT) wrote:
I tried to get my wife to killall mpg123 processes from the console, but
the machine was probably in the Asterisk console rather than the command
line.
!killall mpg123
should work from the asterisk CLI
--
Tzafrir Cohen
Rich Adamson wrote:
Be carefull with vlan assumptions. If two or more vlans exist across
multiple switches, how do you know if another vlan hasn't consumed all
available resources leaving little (or none) for your phone vlan?
Hint: look for discarded packets in or out on the physical ports
Hi,
I have similar problems. Did you find out what is the problem?
We use TE205P and Octro Bri card. At the Octo Bri there ISDN Router connect
which dial out over the TE205P card.
Regards
Miloš Kocbek wrote:
Hi
I have octo bri card connected to 4 telco lines and 4 alcatel PBX lines.
Hi!
We have some problems with calls that get disconnected in the middle of a call.
We are using Asterisk 1.2.1 with a TE410P (2.gen firmware).
When the call is disconnected Asterisk writes this to the log:
Jan 9 14:56:17 DEBUG[4404] dsp.c: ast_dsp_busydetect detected busy, avgtone: 300,
On Tue, Jan 10, 2006 at 05:17:02PM +0400, Jean-Michel Hiver wrote:
[snippage]
I dunno... it looks like a cell phone, except it's not one. It would be
nice if it was a dual GSM / wifi phones which transparently switch to
VoIP when you have a strong enough signal. This way, it would provide
On Mon, Jan 09, 2006 at 03:50:55PM +0100, Armin Schindler wrote:
On Mon, 9 Jan 2006, Louis-David Mitterrand wrote:
I am now using a cross cable and the green led lights up on the Diva
port when plugging the phone in.
When dialing from the phone I get no debug or trace at the asterisk
Hi all,
I want to
configure Asterisk using Mysql Database. But on compilation of asterisk-addons
I am getting some errors. I have pasted the errors in the pastebin.
Please checkout this link. http://pastebin.com/499106
Also please do let me know which are packages required for
Morten Isaksen wrote:
We have some problems with calls that get disconnected in the middle of a
call.
We are using Asterisk 1.2.1 with a TE410P (2.gen firmware).
When the call is disconnected Asterisk writes this to the log:
Jan 9 14:56:17 DEBUG[4404] dsp.c: ast_dsp_busydetect detected busy,
Hi,
I'm struggling to
get busydetect to work.
I'm using asterisk
1.2.1 and a digium TDM04B (4 port FXO) card.
I've set
busydetect=yes, busycount=6 and busypattern=300,200 in zapata.conf and i've
modified zondata.c with a busy setting of 620+480, 300/200 which is the
busysignal received
I've been trying to use the VMAuthenticate function in 1.2+. This
function is supposed to behave[s] the same way as the Authenticate
application, but the passwords are taken from voicemail.conf.
The problem is that it always gives the comedian mail prompt and
requests the mailbox number, even
Are you supplying the context?
On 1/10/06, Gil Kloepfer [EMAIL PROTECTED] wrote:
I've been trying to use the VMAuthenticate function in 1.2+. This
function is supposed to behave[s] the same way as the Authenticate
application, but the passwords are taken from voicemail.conf.
The problem is
Someone can also give solutions for IP Phones/gateways - grandstream,
sipura, linksys?
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Thanks for your
suggestion Steve.
I have done as you advised and set busypattern=300,200 to match the sample I recorded.This hasn't worked though, asterisk doesn't seem to detect the busy signal.Does asterisk require a the signal to be in a certain power range? The signal I getis very
Be carefull with vlan assumptions. If two or more vlans exist across
multiple switches, how do you know if another vlan hasn't consumed all
available resources leaving little (or none) for your phone vlan?
Hint: look for discarded packets in or out on the physical ports of
those
Same here. I believe there's some funkiness (spelling?) with lights staying
on when calls are transferred to another extension. Rebooting the phone
and/or asterisk is required for me sometimes.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Hello everyone
We have recently built a web conferencing tool that uses IAX2 for the
voice. We make use of Asterisk 1.2.1 running on a Debian
Sarge and have built the conferencing tool using VB,
so unfortunately it will only run on Windows.
We are now looking for people to help us
I would look at using serveral machine splitting up the load using one
4 port card in each.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, January 10, 2006 6:03 AM
To: Asterisk Users Mailing List -
Alexander Lopez a écrit :
Check out Redfone.
http://www.red-fone.com
It converts PRI or T1 to TDMoE (TDM over Ethernet)
Isn't TDMoE considered deprecated?
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We do have a PRI on order but have an immediate need so band-aid in the
mean time. This circuit has actually been hooked to a channel bank for
years with just analog phone and I stuck Asterisk in the middle about 3
months ago. I setup the standard S extension and it is handling
incoming no
Hi, does anyone knows a good (comercial/oss) web client for asterisk? i
mean similar to a softphone but using some kind of web interface,
ideally a would create all the user/pass in asterisk, the customer logs
in using a web form and he can make calls using the web interface,
something similar
Wow, I agree with Alexander... putting that many lines in a single box
is one rather large single point of failure...
Aaron
Alexander Lopez wrote:
I would look at using serveral machine splitting up the load using one
4 port card in each.
-Original Message-
From: [EMAIL PROTECTED]
On 1/10/06, Gil Kloepfer [EMAIL PROTECTED] wrote:
The problem is that it always gives the comedian mail prompt and
requests the mailbox number, even though I provide the mailbox
number already.
On Tue, Jan 10, 2006 at 10:00:44AM -0500, C F wrote:
Are you supplying the context?
Actually I
Hello!
After upgrading my production machine to 1.2.1(used to be 1.2.0) on friday
i experienced strange behaviour yesterday, i received
deadlock-avoided-messages and channels refusing to hangup on span1(used
for inbound calls), both messages in all cases paired:
Jan 9 17:40:01 WARNING[30003]
David Sampson wrote:
One other question - how do I get outgoing calls to select last
available channel instead of first?
There is an explanation of this in the sample extensions.conf.sample in
/usr/src/asterisk/configs using r/R/g/G
Kevin
___
Anyone have or know where I can go to get the astcc
voice prompts in spanish ? Thanks.
Dovid
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Original Message
Subject: RE: [Asterisk-Users] Recommendations on a WiFi phone for *?
From: Asterisk-User [EMAIL PROTECTED]
Date: Tue, January 10, 2006 5:23 am
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Has anyone tried
Use capital G instead of lowercase g for groupdialing.
Michael
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] Im Auftrag von David Sampson
Gesendet: Dienstag, 10. Januar 2006 16:20
An: Asterisk Users Mailing List - Non-Commercial
Original Message
From: Jean-Michel Hiver [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com Sent: Tuesday, January 10, 2006 2:17
PM Subject: Re: [Asterisk-Users] Recommendations on a WiFi phone for *?
Rupert Gregory a écrit :
Try adding searchcontexts=yes to the voicemail.conf file. I had the
same problem sometimes on my system until I turned that on. (If you've
already got that, I'm lost)
Aaron
Gil Kloepfer wrote:
On 1/10/06, Gil Kloepfer [EMAIL PROTECTED] wrote:
The problem is that it always gives the
http://www.paesys.com/en/index.htm for the english versionOn 10/01/06, Guillaume de Lafontaine
[EMAIL PROTECTED] wrote:HiI just discovered an interesting product line. Not tested yet...
http://www.paesys.com/fr/lecteurs_VoIP_WiVoip_VideoIP_GSM.htmIn french, sorry...Any feedback
Does anyone know of an American made option like the ISDN Guard?
I'm looking for something that will listen for a heartbeat from
Asterisk and if it fails, flip the PRIs over to a backup box.
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Rusty Dekema wrote:
How far (physically) is the Asterisk server location from the
location of the phones? Have you tried pinging the Asterisk server
from the network to which the phones are connected?
As a rule of thumb, If the two sites are within 2500 miles of each
other and the network
Hi Crhistian. Please try activating the debug output, so may be some
helpfull message will help us. Hopefully some developer will look at
this message and try to help. I have checked the source code but have
not concluded the source of the problem May be with the verbose output
i can help you
The question was actually if Asterisk could support this and still act
as one entity?
I would in this case use 2 PC's with 16 E1's each, but this is as far as
I can see 2 separate PABX's. Are there a possibility to make these 2 (3
or 4 or whatsoever) PC's act as one entity so I can connect a
http://www.paesys.com/en/GSM_Wi-Fi_phone_for_SIP_voice_and_data_GTEK_PWG500.htm
Yeah I've seen that one... to bad it looks like such a brick :(
Cheers,
Jean-Michel.
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it seems to me that you need to upgrade your libpri, but i tought that
your chan_zap compilation should have died if you have not the correct
version of libpri. Anyway, what version do you have of libri? just
upgrade libpri and recompile chan_zap, or all asterisk just in case.
regards
On 1/9/06,
On Tue, 10 Jan 2006, Louis-David Mitterrand wrote:
On Mon, Jan 09, 2006 at 03:50:55PM +0100, Armin Schindler wrote:
Here is the mlog with an attempt to Dial(CAPI/DIVA2/146472131):
[C:4] 22:0167:202 - D-X(003) 02 01 7F
[C:4] 22:0168:202 - D-X(003) 02 01 7F
[C:4] 22:0169:202
Sorry Micheal, i have seen similar posts before, but no answers. May
be the best way to go is contact digium support.
Good Look
On 1/9/06, Michael Loftis [EMAIL PROTECTED] wrote:
We've been having lost dialtone problems on one of our analog station
ports. Just before rebooting this time I
Oh, wow, ok. Doesn't look like the problem is with your WAN then! (Assuming that the ping times stay like that when the network is at its normal load.) -RustyOn 1/10/06,
Geoff Manning [EMAIL PROTECTED] wrote:
Rusty Dekema wrote: How far (physically) is the Asterisk server location from the
Sorry about the unrelated questions about cisco phones, but does anyone
know how to set the second line up as a speed dial in the config file?
Or is that specifically a per-user basis setting?
Aaron
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MvPhone wrote:
Hi,
You can put max of 2 4xE1 cards per dual CPU box. That means that you
end up with 4 dual CPU boxes
4 boxes * 2 cards * 4 E1 = 32 E1's
I sell affordable quadE1 cards that work well with asterisk. You can
check on pbxhardware.com
As for the dual CPU boxes it depends what
http://www.sacaug.org/
At our new location 7-9pm Janurary 12th. No fee to attend
Exit Certified
8950 Cal Center Drive
Suite 110, Bldg. 1
Sacramento, California, USA
95826
map links on the sacaug webpage.
We will be doing an install of astlinux with simple configuration as
well as
Hi all,
I installed asterisk stable cvs 1.2 and chan_capi 0.4.0 PRE1, with one AVM
Fritz Card ISDN connected to a Telecom NT1 Plus
I configured asterisk via AMP.
No problem in making calls.
If I try to ring the ISDN Phone Number, I don't see anything on the
asterisk Console,
I I activate the capi
On 1/10/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Morten Isaksen wrote: We have some problems with calls that get disconnected in the middle of a
call. We are using Asterisk 1.2.1 with a TE410P (2.gen firmware). When the call is disconnected Asterisk writes this to the log: Jan 9 14:56:17
Well, this product from signate uses infiniband...
It has 4 slots for quad e1/t1 per slot.
http://www.signate.com/pdf/TelephonyServer.pdf
Just read the PDF. Obviously this is not an x86 Pc. I wonder if you want to build your own or were looking for a beast like this.
BTW, any real world comments
Dear all,
Any one has experience in CAGI script setgroup? Please
let me know a bit of command detail.
Thanks,
Ray
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I suggest you use the newer chan_capi-cm (loadable from sourceforge.net).
Armin
On Tue, 10 Jan 2006 [EMAIL PROTECTED] wrote:
Hi all,
I installed asterisk stable cvs 1.2 and chan_capi 0.4.0 PRE1, with one AVM
Fritz Card ISDN connected to a Telecom NT1 Plus
I configured asterisk via AMP.
No
Can you post your voicemail.conf as well?
On 1/10/06, Gil Kloepfer [EMAIL PROTECTED] wrote:
On 1/10/06, Gil Kloepfer [EMAIL PROTECTED] wrote:
The problem is that it always gives the comedian mail prompt and
requests the mailbox number, even though I provide the mailbox
number already.
Is there a Monitor application called in your dialplan? It might have a
basename parameter that configures this. Or you could maybe call
ChangeMonitor yourself but I don't know how to configure the timestamp
portions of the filename.
Tim Litwiller wrote:
Mojo with Horan Company, LLC
Greetings. I am trying to get AMP up and going on my Asterisk server. I
can access the admin pages on my asterisk server via a web browser. I can
add and edit things via the web browser and it edits the database
accordingly. Everything seems fine except when I try to run 'amportal
start'.
Title: Message
The problem
:
an asterisk box
with 2 fxo
First fxo just
receive calls from pstn (ulaw)
Second fxo receive
and send call to mobile network thru a sipbox(ulaw)
Calls to pstn are
sent to a pstn provider accepting only g729
Internal calls
doesn't care of
Hi ALL,
Is it possible to use symbols # and * in the
dialplan for pattern matching?
I am doing a "follow me" dial plan, and wanted that
my users could dial everything in one shot.
But, exten =
888*XXX*XXX,1,NoOp(Follow Me from XXX to XXX)
doesn't seem to work...
Thank you
Dov
On Tue, Jan 10, 2006 at 05:43:12PM +0100, Armin Schindler wrote:
On Tue, 10 Jan 2006, Louis-David Mitterrand wrote:
[C:4] 22:0188:202 - D-X(003) 02 01 7F
[C:4] 22:0189:202 - D-X(003) 02 01 7F
[C:4] 22:0190:202 - D-X(003) 02 01 7F
[C:4] 22:0191:201 - MDL-ERROR(G)
Thank you.
I already downloaded and installed it (they are dated 07-01-2006, version
0.6.3, correct ?)
I maked clean, make and make install.
Nothing changed, dial out perfect, dial in: (capi debug on)
asteriskge03*CLI capi info
Contr1: 2 B channels total, 2 B channels free.
asteriskge03*CLI
Actually got my hands on one, it's not that bad size wise. About the size of
a big smartphone
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel
Hiver
Sent: January 10, 2006 11:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
TDMoE is stable and stale, There is no more development planed or needed as it
only opens up a pipe between two ethernet points using Layer 2.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jean-Michel Hiver
Sent: Tuesday, January 10, 2006
Hi,
I was wondering if anyone has had a problem adding the delete
field to the voicemail_users table. I have no
problems adding other fields e.g.
alter table voicemail_users add column hidefromdir
varchar(3) NOT NULL default no;
However when I do
alter table voicemail_users
On Tue, Jan 10, 2006 at 06:52:43PM +0100, Louis-David Mitterrand wrote:
On Tue, Jan 10, 2006 at 05:43:12PM +0100, Armin Schindler wrote:
On Tue, 10 Jan 2006, Louis-David Mitterrand wrote:
[C:4] 22:0188:202 - D-X(003) 02 01 7F
[C:4] 22:0189:202 - D-X(003) 02 01 7F
[C:4]
Louis-David Mitterrand wrote:
It seems ptmp is required for NT mode to work with this phone. I tried
plugging the phone directly on the telco BRI and it worked fine too.
Phones are generally only for ptmp mode. Only PBXs use ptp mode. (Here
in Germany the ptp mode is even called PBX mode)
Check out the Cisco SIP IP Phone Administrator Guide, Appendix D -
speed_line and speed_label
Do a google for Cisco SIP IP Phone Administrator Guide, easy peasy nice n
easy.
Sorry about the unrelated questions about cisco phones, but does anyone
know how to set the second line up as a
speed
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