In article [EMAIL PROTECTED],
Doug Lytle [EMAIL PROTECTED] wrote:
There is a link on Groklaw for the following article:
Open source companies to watch
Digium makes the second entry on the list. Link below:
http://www.networkworld.com/news/2006/082806-open-source.html?ts
jurgen ha scritto:
Hi,
The problem happens when I record a call using MixMonitor. Even though
it's recording natively in g729, a single call uses 2 decoders and one
encoder! The only explanation I can think of for that is that
MixMonitor is transcoding the g729 streams to something else,
Hi,Taliax has unlimited calling plan per month. You can see WWW.TELIAX.COMRegards,Chandra."Eric \"ManxPower\" Wieling" [EMAIL PROTECTED] wrote: Steven M. Sawczyn wrote: Greetings, I finally got my Asterisk server up and running and now am in the process of looking for a provider to use as a SIP
Hi, Taliax has unlimited calling plan per month. You can see WWW.TELIAX.COM Regards, Chandra."Steven M. Sawczyn" [EMAIL PROTECTED] wrote: Greetings, I finally got my Asterisk server up and running and now am in the process of looking for a provider to use as a SIP trunk. Unfortunately, I'm
Andy Chung (Power-All) wrote:
Hi all,
I have three Asterisk servers behind a SER. I want to know how to
configure the Dispatcher module of SER to achieve load balance for the
Asterisk servers. I have visited
http://www.openser.org/docs/modules/1.1.x/dispatcher.html, is there any
web sites
Hi,
currently I use version 1.1.0.16 for my GXP-2000 which works really
fantastic. The only drawback I see is the addressbook.
Is the firmware 1.1.1.9 stable enough to use the phone in normal
environment? The webpage http://www.voip-info.org/wiki/view/GXP-2000
says that there it is possible to
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Of course we care. Turns out that schedule was unrealistic, and when we start
the next cycle we will regroup and decide if we either stretch out the cycle
or reduce the amount of new functionality that gets added during the cycle.
OK,
Giorgio Incantalupo wrote:
Hi Tommaso,
have you tried to search for noise suppression? I remember some phone
has a function to automatically suppress it so the caller does not
hear anything and thinks the other party has hung up.
Giorgio Incantalupo
Tommaso Calosi wrote:
I have this
Dear friends,
Does anyone know how do i convert hex to int in the dialplan. I want to do
this:-
Take the sip call-id in hex, use CUT to extract the first part , and convert it
to an int. But the math function ony takes arguments as int. Can anyone suggest
how to do that?
eg:-
exten =
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I don't know. Do you use Asterisk? That makes you part of the team.
Have you tested the trunk version? Provided assistance testing out
patches waiting for completion?
Really, once all the new features have been completed, it will
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I don't know. Do you use Asterisk? That makes you part of the team.
Have you tested the trunk version? Provided assistance testing out
patches waiting for
Hello
I just want to have a confirmation, line status detection (with digium
TDM400P) is highly not reliable outside of US.
With busydetect=yes and callprogress=yes I can experience very strange
phenomenons (randomilally occurs) like pick up not detected or hang up not
detected.
I'm in Israel
Hi,
Does anybody know if asterisk 1.4 will support comfort noise? Or if there is
a patch for it now?
If it will be in 1.4 any idea of release date?
Thanks,
Dean Bath.
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asterisk-users mailing
I have a Snom 360 phone and I'm configuring it for use with Asterisk
1.2.9 and Freepbx 2.1.1
On my PBX there are:
1) Some SIP phones
2) One digium quadri primary ISDN interface (TE410P)
3) Two Rhino Channel Banks
4) 25 Analogue Phones on every channel bank
How I can configure function keys on
Hi all
Two questions.
We have a multi language voicemail setup.
Unfortunately I did not find a way to localize the email notification sent to
the customer. How can one do this? For the moment messages are hard-coded in
german.
The System Locale is 'C'.
emaildateformat=%A, %d %B %Y um
Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I don't know. Do you use Asterisk? That makes you part of the team.
Have you tested the trunk version? Provided assistance testing out
patches waiting for completion?
Really, once all the new features
Hello!
On 8/29/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Trying to add faxing to asterisk but get a compile error. Any ideas? Is
it broken for Asterisk 1.2.11 or was it me again :-)
app_rxfax.c:105: error: structure has no member named `column_resolution'
app_rxfax.c:105: error:
Hi all,
I have 2 agents (1234, 4321) and a PSTN phone (9876). When 1234
makes a call to 4321, 4321 will have a callerid 1234 on his screen.
Now, 4321 has forwarded his call to PSTN phone. When 1234 makes a
call to 4321, it will forward to PSTN phone. However, caller display
can't show
Hello!
Thanks for your answers!
Everything works fine now there was some problem at my provider. I
compiled and use rxfax successfully.
bye,
Zsolt
On 8/15/06, Marco Mouta [EMAIL PROTECTED] wrote:
Hi,
I didn't try that way, only tx fax in call file. But my experience is when u
r working with
Cisco released last Aug 23 the latest SIP firmware for Cisco 7960G. Any
info?
SIP Flash Image for 7940/7960 IP Phone v8.4(0) - Non CallManager
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To UNSUBSCRIBE or
the configuration is
this :
NT
PRI
TD405P TE
A
-- B
(Asterisk)
A make a call to
B.
A can display the ID
(caller ID , example John) of B ?
these information
are exchanged in the call progress ?
B can change the
called number and communicate this change to A whene the
SER/OpenSER can get around call forwarding/transfer problems. You just need
to account for those SIP dialogue's that can be problematic, and bypass
using the dispatcher module for those situations.
One thing to remember is to replicate usr-loc info that is cached in memory,
otherwise
Thanks a lot!Muchas gracias amigo!EstebanOn 8/29/06, Matt Riddell (IT) [EMAIL PROTECTED]
wrote:-BEGIN PGP SIGNED MESSAGE-Hash: SHA1Steve Edwards wrote:
It's not clear if the OP wanted 1) information on how to analyse the core file or 2) provide information to the bug tracker for others
Hi all,
this is probably a weird question and something I'm not doing right
but I got this bizarre thing going on here. When I boot the system
with the SMP kernel and compile (*) with the smp kernel source
(actually even if I don't compile, but as long as I boot into the SMP
kernel), I get this
Packet8 is unlimited usa, or a more expensive plan for unlimited global.
You have the use an ata however.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling
Sent: Wednesday, 30 August 2006
Sorry I was not clear William.
In the actual code, the exten marked 'old' is commented out and only
'new' is active. Then I reload. But only the single 120 instrument rings.
Larry
William Piper wrote:
The whole thing.
Both (old and new) have the same exten and the same priority, you can't
Sorry I was not clear Rushowr.
In the actual extensions.conf as used, the 'old' line is commented out
so only 'new' is active. Then I reload. However, only the single 120
line rings instead of all.
Larry
Rushowr wrote:
Then entire OLD extension must be removed so the new one will match
Teliax is not unlimited but has a cap of 2500 minutes per month.***
Softcap of 2500 Minutes (including 1000 minutes of toll-free inbound, if applicable).On 8/30/06, Crazy Boy
[EMAIL PROTECTED] wrote:Hi, Taliax has unlimited calling plan per month. You can see
WWW.TELIAX.COM Regards,
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all,
I mentioned this back in February, and there wasn't much response (John
Novack was the only one who responded.) I assumed it was due to the
fact that nobody was really sure. :) So, I dropped the idea and haven't
re-visited it until today --
What
cost do you pay per month for the 2500 minutes?
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Tom
VileSent: 30 August 2006 13:54To: Asterisk Users Mailing
List - Non-Commercial DiscussionSubject: Re: [asterisk-users] does
anyone
Hi all
I've had a number of servers, all generally running Asterisk 1.0.9-1.0.11.1,
with TDM cards for analog lines. They have been in production use for many
months, handling incoming calls, and also allowing daily inter-server calls
over IAX (transfers, extension calls etc)
All of a sudden,
$24 per monthOn 8/30/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
What
cost do you pay per month for the 2500 minutes?
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of Tom
VileSent: 30 August 2006 13:54To: Asterisk Users Mailing
List -
How many simultaneous calls?
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
Sent: Wednesday, 30 August 2006
9:16 AM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] does
anyone offer truly
Doesnt matter I just checked, only
2.
Also the soft-cap for residential is 1500
mins for $24.99
2500 soft-cap is for corporate with $44 a
month (but has 4 lines)
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
Sent: Wednesday,
In article [EMAIL PROTECTED], Chris Earle [EMAIL PROTECTED] wrote:
Hi all
I've had a number of servers, all generally running Asterisk 1.0.9-1.0.11.1,
with TDM cards for analog lines. They have been in production use for many
months, handling incoming calls, and also allowing daily
Hi!
I'm trying to install a A104d.
1. LSPCI detects the card:
# lspci
...
00:1f.2 IDE interface: Intel Corporation 82801FB/FW (ICH6/ICH6W) SATA
Controller (rev 03)
05:04.0 Class affe: Sirrix AG security technologies Sirrix.PCI4S0 4-port
ISDN S0 interface (rev 02)
05:09.0 Network controller:
Hi! Problem solved. I just removed the wanrouter modules and tried
again. This thime there were some more modules loaded and the card is found:
]# wanrouter hwprobe
---
| Wanpipe Hardware Probe Info |
---
1 . AFT-A104-SH : SLOT=9 : BUS=5
Sangoma provides EXCELLENT support.
I would try them
I just installed a A101, and had some problems, but the hwprobe found
the card OK.
You MIGHT want to try different PCI slots before contacting them
My problem was somewhat different, and was fixed by a reboot of the
machine between
Hi UsersI'm new to Asterisk PBX.Mainly i'm using the openser for call routing and Asterisk as PBX and Voicemail generating.let see my secnario ---UAC -- ser Asterisk(for voice mail only and extension and PBX Purposes
SER system ip is 192.168.2.75:5060Asterisk is in
Anything is possible. The biggest challenge with OpenSER is getting past the
horrible documentation and the cryptic, one line responses to questions asked
in the mailing list.
-Original Message-
From: Adam Linford [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 30, 2006 5:33 AM
-Original Message-
From: Jeremy McNamara [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 30, 2006 1:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SER Dispatcher Load Balance How-To?
Andy Chung (Power-All) wrote:
Hi all,
I
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 29, 2006 11:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] SER Dispatcher Load Balance How-To?
Well, it really depends on what he's using the
On Wed, 2006-08-30 at 08:34 -0600, Douglas Garstang wrote:
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 29, 2006 11:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] SER Dispatcher Load Balance
If you want a good explaination of SER and how to use it start here.
http://siprouter.onsip.org/doc/gettingstarted/
They have GREAT pre-written configs and walk you through ever part of SER. I
was about scrap SER before I found these tutorials.
Natambu Obleton
Network Engineer
FastTrack
On 14:23, Wed 30 Aug 06, [EMAIL PROTECTED] wrote:
Dear friends,
Does anyone know how do i convert hex to int in the dialplan. I want to do
this:-
Take the sip call-id in hex, use CUT to extract the first part , and convert
it to an int. But the math function ony takes arguments as int.
Hi
i have a small problems with my asterisk connected to phonesystems :
Now i have this message:
-- SIP read from 62.39.136.151:5060:
SIP/2.0 403 Cant accept register from myself
Via: SIP/2.0/UDP 84.14.xx.xx:5060;branch=z9hG4bK38f74bd7;rport=5060
From: sip:[EMAIL PROTECTED];tag=as42b95c05
To:
On Sat, 2006-06-17 at 15:49 -0500, Lacy Moore - Aspendora wrote:
Can't be done using the 7960 with SIP, unless you are talking about
just monitoring that phone. You can monitor a 7960, but you can't
show the status of other phones on a 7960 with SIP.
Do you know wether it can be done with a
You can get as many minutes and channels as you require from TelIAX. You
just have to call them to customize the account.
Start by setting up the Corporate Account, then call them to customize it
to your needs.
Erv Bauman
NISCOMM
+1-412-567-0343 ext. 150
11 Aldred Lane
Pittsburgh, PA 15227
Hello,
I would like to somehow get the presence of IAX2 and SIP users from
Asterisk Manager API or using any other means.
I tried watching for PeerStatus event, but it seems unrealiable
(http://bugs.digium.com/view.php?id=7833).
I tried defining hint for user and sending ExtensionState event,
Hi pals, im trying to save some money in
cellphones calls, so i bought a GSM gateway and a Sipura SPA3000
gateway.
The GSM gw is currently working, and now im trying to configure the
SPA, but every call i send, i get a 503 service unavailable.
Im using this extension to match cell calls and
At 05:02 AM 8/30/2006, you wrote:
Packet8 is unlimited usa, or a more expensive plan for unlimited global.
You have the use an ata however.
I think you'll find they're only unlimited until you abuse
them! Most seem to have a 2000-3000 minutes/month limit written
somewhere in the fine print
I don't know then, I do the same exact thing:
exten = _352688,3,Dial,SIP/202SIP/214|20
Perhaps try sending everything in that context exactly as it is typed let us look at it.
I'mpretty sure you have something configured incorrectly.
Thanks,
bp
On 8/30/06, Larry Alkoff [EMAIL PROTECTED]
Hello,
I'm looking for an agent managing dialplan/software/agi/whatever that
independent from asterisk queue management. I already tried this
http://www.voip-info.org/wiki/view/Agents+without+agent+channel
with no success but a lot of warning. I'm using asterisk 1.2.10 and
the dialplan above
Google is your friend:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+ExtensionState
bp
On 8/30/06, Juraj Bednar [EMAIL PROTECTED] wrote:
Hello,I would like to somehow get the presence of IAX2 and SIP users fromAsterisk Manager API or using any other means.
I tried
The
one we use here works out at $0.0286 cents per min, but has unlimited amount of
lines,we use one account for our call centre and we have had up to 40
calls in the call queue, and it works fine. Not sure if they do USA numbers but
could find out if needed. We also use one account for all
I found this, which looked interesting:
http://wiki.onmac.net/index.php/Triple_Boot_via_BootCamp
Also, Apple released a new version of BootCamp that supports the Xserve on
Aug 16. If it'd work, and you could shoehorn a PRI card into it, man
wouldn't that make a nice Asterisk box? And at $2999,
hi everybody,I bought few units for evaluation but we were not able to upgrade the firmware to 1.4 ,
it's currently set at 1.2,
when we go to the webadmin page,
whether we try to change the IP of the tftp server or the firmware name and set values, the reply is always
Invalid IP address
Like what? I haven't tried the non-Call Manager version yet. The Call Manager version seems to work fine with Asterisk. Haven't run into any issues yet. I wish there was a softkey for DND, but that hasn't seemed to be in any SIP version. I thought maybe the CallManager version would have this.
On
Steve,
VoiceEclipse has a US unlimited plan for $20/month. Two inbound numbers
that can be in different area codes. I have not figured out how to
recognize which number the inbound call came in on, but, right now, that
is not that important to me. Others have had other problems. Research
On Wed, 2006-08-30 at 16:25 +0100, Conrad Wood wrote:
On Sat, 2006-06-17 at 15:49 -0500, Lacy Moore - Aspendora wrote:
Can't be done using the 7960 with SIP, unless you are talking about
just monitoring that phone. You can monitor a 7960, but you can't
show the status of other phones on a
Make sure all of the lines you are ringing are registered up and
running. I noticed this when I did a paging extension. I rang about 40
phones and the second it saw one offline it failed only ringing one phone.
William Piper wrote:
I don't know then, I do the same exact thing:
exten =
Hi
a small question:
I have one Asterisk Server with:
VoIP Provider gateway for incomming/outgoing call
5 VoIP Phone
(i name it Master)
i want add a another Asterisk server but only connected to:
5 new VoIP Phone
To the master for incoming/outgoing call (in g729)
It's
Hi Michael,
Thanks a lot. I am working on an agi script and it does it. Thanks a lot again.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
All science is either physics or stamp collecting.
-- Ernest Rutherford
Michiel van Baak wrote:
Seems to be working ok on my handset for the past couple of weeks.
No major bugs, registration, xml services and MWI works etc..etc..
Have not given it a thorough testing though.
Regards,
Nathan.
On 30/08/2006, at 6:51 PM, Hermann Wecke wrote:
Cisco released last Aug 23 the latest SIP
For DND press Call Forward All (CFwdAll softkey)
then Messages button on the SCCP version. I havent seen the SIP version
of 7961G.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora
Sent: Wednesday, August 30, 2006
Douglas Garstang wrote:
What about transfers and forwards?
if your system is designed properly, it doesn't matter which Asterisk
box actually processes the call.
Jeremy McNamara
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Westany, the Asterisk voice experts, announce their first Russian voice for
the Asterisk PBX. Tamara, a Russian female voice, is the latest addition to
Westany¹s growing catalogue of proven, meticulously-crafted voice prompt¹
suites for Asterisk, Freepbx, trixbox, Bicomsystems and Amp.
Produced
William I found and fixed the problem. Your comment gave me the kick to
persevere. Thank you very much.
My exten line had a comment at the end that contained a close paren.
That apparently screwed up the context line - although it shouldn't
have. Now all three extensions ring.
Note my
Does anyone have any experience with this device? Does it interface nicely
as a FXS / FXO for use with Asterisk?
smime.p7s
Description: S/MIME cryptographic signature
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asterisk-users mailing
Hi all,
I've compiled/installed both * and Speex but I'm getting an error upon *
startup:
...Aug 30 11:23:34 WARNING[27652]: loader.c:325
__load_resource: /usr/lib/asterisk/modules/codec_speex.so: undefined symbol:
speex_preprocess_ctl
Aug 30 11:23:34 WARNING[27652]: loader.c:554
How about creating some documentation?
-Original Message-
From: Ira [mailto:[EMAIL PROTECTED]
Sent: Monday, August 28, 2006 12:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Asterisk with PABX
At 11:16 AM 8/28/2006, you wrote:
Actually
Here's what we do:
[agent-login]
exten = s,1,NoOp(${AgentUser})
exten =
s,2,AddQueueMember(${AgentContext}|${AgentChannel}|${AgentPenalty})
exten = s,3,Wait(1)
exten = s,4,Playback(agent-loginok)
exten = s,5,Hangup
exten = s,103,RemoveQueueMember(${AgentContext}|${AgentChannel})
exten =
Giorgio,
I believe the syntax for mISDN is mISDN/port:channel/number. In other
words, replace your - with a :.
On 8/25/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi,
I have a quadBRI beronet ISDN card. Is there anybody who knows how to
choose the channel to make calls? I tried with
Westany speaks biz
CP
On Aug 30, 2006, at 9:50 AM, Stuart wrote:
Westany, the Asterisk voice experts, announce their first Russian
voice for
___
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To
Francisco Seratti wrote:
Hi pals, im trying to save some money in cellphones calls, so i bought a
GSM gateway and a Sipura SPA3000 gateway.
The GSM gw is currently working, and now im trying to configure the SPA,
but every call i send, i get a 503 service unavailable.
Im using this extension
Ninneman, Tj wrote:
Hi all,
I've compiled/installed both * and Speex but I'm getting an error upon *
startup:
...Aug 30 11:23:34 WARNING[27652]: loader.c:325
__load_resource: /usr/lib/asterisk/modules/codec_speex.so: undefined symbol:
speex_preprocess_ctl
Aug 30 11:23:34
On Wed, Aug 30, 2006 at 05:50:53PM +0100, Stuart wrote:
Westany, the Asterisk voice experts, announce their
[ snip product description, that ommited a price tag of 124$ ]
There¹s simply no substitute for knowledge and experience.
Reading list descriptions also helps. This list is not
Noc Phibee wrote:
Hi
a small question:
I have one Asterisk Server with:
VoIP Provider gateway for incomming/outgoing call
5 VoIP Phone
(i name it Master)
i want add a another Asterisk server but only connected to:
5 new VoIP Phone
To the master for
Hello,
Nobody has replied on this message.
Isn't there anybody that has any input?
Best regards,
Henrik Woffinden
Henrik Woffinden wrote:
Hello,
I'm fairly new to Asterisk.
Installation went fine, and things seem to work, but I have 1 problem.
Hardware:
2 HFC ISDN cards (1 in TE mode
I think this is a known problem that was fixed in v1.3.
I think you need to do this upgrade using a 'put' install via tftp client
rather than trying to configure it to 'get' from a tftp server. It's been
awhile so my memory is a bit foggy. I used pumpKIN.
http://kin.klever.net/pumpkin/
Hi,
I have an Ascom Eurit 133 ISDN base station with 2 cordless handsets.
I can receive calls excellent on these phones, but when I dial out
Asterisk can't see what number I want to dial, and it routes me to the
s extension. That rather unlucky for an outgoing call not to know the
number you
Hi,
I have a few
questions on the Polycom 501. I am using latest
firmware.
1) When I press the
"Call List" button (on the left row of buttons), I get the call lists (as
expected). When I press the "Directory" button, I get the choice between
Directory and Call lists. How can I make this
Hi,
I have not tested yet, but maybe Dial(Zap/g1) would work;
Guess this would ring everthing on Group 1...
Best regards,
Martin Polainer
Am Mittwoch, 30. August 2006 21:45 schrieb Henrik Woffinden:
Hello,
Nobody has replied on this message.
Isn't there anybody that has any input?
Best
Hi,
I've just tested that... And no, nothing on the channel rings.
Henrik Woffinden
Martin Polainer wrote:
Hi,
I have not tested yet, but maybe Dial(Zap/g1) would work;
Guess this would ring everthing on Group 1...
Best regards,
Martin Polainer
Am Mittwoch, 30. August 2006 21:45
Dave Fullerton escribi:
Francisco Seratti wrote:
Hi pals, im trying to save some money in
cellphones calls, so i bought a GSM gateway and a Sipura SPA3000
gateway.
The GSM gw is currently working, and now im trying to configure the
SPA, but every call i send, i get a 503 service
I may have to do something like that to be able to setup some way to temporarily close our office. I haven't really found anything else that would have a visual indicator that the system is on temp. closed mode. I can manually set a database entry (which I already do), and I know I can add an
You will find here all the info that you need to
make the SPA3000 to work with Asterisk:
- Original Message -
From:
Francisco
Seratti
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, August 30, 2006 5:39
PM
Subject: Re:
Sorry,
http://voxilla.com/PNphpBB2-viewforum-f-14.html
Cheers,
- Original Message -
From:
Francisco
Seratti
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, August 30, 2006 5:39
PM
Subject: Re: [asterisk-users] Sipura 3000
and
good idea! I tried but it doesn't work either...[08/30/06 23:12:33] UDP packet receive failed[08/30/06 23:12:33] Invalid opcode (0) during transfer received[08/30/06 23:12:34] Sending '
firmware.st' to '192.168.0.101'[08/30/06 23:12:34] UDP packet receive failed[08/30/06 23:12:34] Invalid opcode
Franciso, can you make a call to the outside world, from the FXS port
and going out the FXO port?
I mean, without Asterisk in between. (The SPA300 can be configured that way)
I'm asking because I remember having trouble with the SPA recognizing
that the FXO line was alive when I plugged in a
At 10:20 AM 8/30/2006, you wrote:
Well, I started writing a tutorial for programming dial plans and
sent it to two people who claimed interest, never heard back from
either so I stopped. It's hard to know if what I write would be
useful to anyone, so I don't want to just post it without
Hi List,I am working with an Asterisk server running on Fedora Core 4. It has two TDM400P cards installed. There are 6 trunk ports and 2 (unused) analog line ports. There are 5 Polycom SoundPoint 501 SIP phones connected to the server, and a Linksys 24-port powered switch connecting everything.
Hello,
Google is your friend:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+ExtensionState
not today.
I mentioned in my original mail, that ExtensionState is unrealiable
too. Sometimes I quit my softphone and I see extension as Idle
(status 0), sometimes I log in
OK, changed the register interval for the Linksys PAP2T to 10 times
longer and the output described earlier on the CLI also appears to
follow the same schedule.
I guess I'll have to check a Linksys list to see what could be
causing this and if I should expect things to get worse.
On Aug
On Aug 30, 2006, at 2:58 PM, Mike wrote:
Hi,
I have a few questions on the Polycom 501. I am using latest
firmware.
1) When I press the Call List button (on the left row of
buttons), I get the call lists (as expected). When I press the
Directory button, I get the choice between
Hi, we have few cisco's...is there a way to push the queue information to the phone ?thanks in advance,jean-louis2006/8/24, Brodie Macleod
[EMAIL PROTECTED]:I know this isn't answering your question, but what I did for queue
notification was use softkeys on the phones that call a PHP script on
Just post it. be sure to wear asbestos. someone is sure to take
offense. someone else just as surely will silently find it useful
Ira wrote:
At 10:20 AM 8/30/2006, you wrote:
Well, I started writing a tutorial for programming dial plans and sent
it to two people who claimed interest, never
Douglas Garstang wrote:
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 29, 2006 11:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] SER Dispatcher Load Balance How-To?
Well, it really depends on what
OK, so I had an issue where I needed to add a w when dialing
out my POTS line. But now when the calls go out my VoIP providers the w makes
the call fail. I am using freePBX and the only place I found to change this was
in the extensions.conf which makes it global. Am I missing something
Not in this case -- this is simply the phone doing something
configurable when it receives a plain old ring on the line. We're not
necessarily talking about the old phones in which the changed voltage on
the line is actually shaking the bell around -- the phone would be smart
enough to see
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