On Tue, Sep 26, 2006 at 08:43:11PM -0700, Nick Ellson wrote:
>
> The link is not working at OpenVox.
There's a "download" link in the bottom of the page, that leads to:
http://www.openvox.com.cn/members_downloads.php .
That page has the "A1200P device driver" as a download item (not just
for mem
On 2006-09-23 12:43:32 -0700, "Kevin P. Fleming" <[EMAIL PROTECTED]> said:
- Matt Riddell (IT) <[EMAIL PROTECTED]> wrote:
Also, are you referring to newer ones than the 1.4 downloads that
were
available a couple of days ago or do you mean people running the 1.2
versions?
The versions that
On 2006-09-24 17:51:51 -0700, "Tom Lynn" <[EMAIL PROTECTED]> said:
I'm keeping my Qwest line for this purpose.
Me too, but I hate paying them every month! I also do terminate some
locals calls that way though...
Also if all the power goes off this might still work ;~)
__
Two things come to mind. First, (I'm not familiar with the SonicWall, so this may be way off), could it have suddently decided that your Voip provider's IP address is a threat? From what I understand, Cisco uses some technology such as this. If it thinks there is a threat, it starts blocking thi
Look for pagerbody and pagersubject.
On 9/26/06, Mike Diehl <[EMAIL PROTECTED]> wrote:
Hi all.I currently get an alpha-page via email from Asterisk when I get a newvoicemail message. This is in ADDITION to getting the message emailed to me.
I can iss in voicemail.conf that I can change the text of
It's in voicemail.conf as well.
On 9/26/06, Mike Diehl <[EMAIL PROTECTED]> wrote:
Hi all.I currently get an alpha-page via email from Asterisk when I get a newvoicemail message. This is in ADDITION to getting the message emailed to me.
I can iss in voicemail.conf that I can change the text of the
I didn't see it as making fun of anyone. I, for one, was curious about it. I suspected it was some type of translation issue, whether it was a word in another language that doesn't translate or what. I know there are many concepts in English and in other languages that just doesn't translate cor
Maybe you could try an asterisk forum in spanish in order to get better
results using your native language.
DiegoF escribió:
hola a todos, tengo una duda, ye he resuelto algunas pero otras
llegan, bueno como habia dicho quiero conectar una pbx a una te110p,
la pbx me ofrece señalizacion r2 eur
Jay R. Ashworth wrote:
On Tue, Sep 26, 2006 at 05:33:15PM -0500, DiegoF wrote:
hola a todos, tengo una duda, ye he resuelto algunas pero otras llegan,
/
hello to all, I have a doubt, ye I have solved some but others arrive, go
Thanks all, I have it now :)
--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.
On Tue, 26 Sep 2006, Nick Ellson wrote:
The link is not working at OpenVox.
Nick
--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
N
When I reloaded my asterisk I saw these lines, which I have noticed before:
[Sep 27 11:46:09] WARNING[27468]: chan_sip.c:12039
add_realm_authentication: Format for authentication entry is
user[:[EMAIL PROTECTED] at line 797
[Sep 27 11:46:09] WARNING[27468]: chan_sip.c:12039
add_realm_authenti
The link is not working at OpenVox.
Nick
--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.
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Barry Fawthrop wrote:
Hi all
I didn't change anything that's my point
It has be running and working just fine then at 4:32 pm yesterday I
could not make or recieve VoIP calls via our VoIP Provider
They say the Invite packet was being rejected and thus there was no
"real" connection even thoug
Hi,
When I issue a show queues command, it shows something below.
2600 has 0 calls (max unlimited) in 'leastrecent' strategy (1s
holdtime), W:0, C:491, A:29, SL:100.2% within 120s
Does anyone know any reference to explain the meaning of the about
information?
1s holdtime: what is that h
I have one extension that rings in many places. It has just come to my attention that I can only monitor 4 devices within a hint.
Ex:
exten => 132,hint,SIP/DEVA&SIP/DEVB&SIP/DEVC&SIP/DEVD
if I add SIP/DEVF, DEVF is not monitored.
Is anyone else monitoring more than 4 devices, and if so, wh
Hi all.
I currently get an alpha-page via email from Asterisk when I get a new
voicemail message. This is in ADDITION to getting the message emailed to me.
I can iss in voicemail.conf that I can change the text of the email message,
subject, and sender.
But, how do I change the text of the al
Steve Totaro wrote:
I set caller ID to a unique identifier before sending to a transfer
partner or overflow call center. This makes it much easier to match
CDRs and get stats on the outcome of calls once they leave our center.
It is a very valuable and legitimate use. Am I committing a crime
Kristian Kielhofner wrote:
Jay R. Ashworth wrote:
But gratuituously making easy something that very few people have a
legitimate need to do, which undermines something that -- even if you
do only make the resaonable assumption that you know which phone, and
not which person, is calling -- is use
>
> How do I use priority "n" correct?
>
First, which version of * are you using? Hopefully something recent.
If you've got 1.2.x then you can use n and labels. Check this out:
http://www.voip-info.org/wiki/index.php?page=Asterisk+priorities
Tinker with it - you'll be surprised at how easy i
Jay R. Ashworth wrote:
But gratuituously making easy something that very few people have a
legitimate need to do, which undermines something that -- even if you
do only make the resaonable assumption that you know which phone, and
not which person, is calling -- is useful and productive... is pro
On Tue, Sep 26, 2006 at 05:33:15PM -0500, DiegoF wrote:
>hola a todos, tengo una duda, ye he resuelto algunas pero otras llegan,
>/
>hello to all, I have a doubt, ye I have solved some but others arrive, good
*Oh*.
*That's* w
On Tue, Sep 26, 2006 at 07:17:57PM -0400, Kristian Kielhofner wrote:
> Quite frankly, it is not my fault that the general public and several
> institutions like banks, etc have poorly implemented systems on
> THEIR end that ASSUME that CNID is gospel and use it for all kinds
> of authentication pu
Jay R. Ashworth wrote:
On Tue, Sep 26, 2006 at 05:28:12PM -0400, Kristian Kielhofner wrote:
2) Get a telco that lets you set any CID. I don't know if I just look
trustworthy or something, but I have had no problems whatsoever getting
several LECs and CLECs in multiple states to let me set an
How do I use priority "n" correct?
Here is the current example:
exten => 615,1,Dial(${PHONE_615},60,tr)
exten => 615,2,Voicemail,[EMAIL PROTECTED]
exten => 615,103,Voicemail,[EMAIL PROTECTED]
and:
exten => 617,109,GotoIf($["${DIALSTATUS}" :
"(CHANUNAVAIL|CONGESTION)"]?110:999)
exten => 617,11
Nicolas,
We use a TE110P from digium. We make the same procedures oriented in that website. the only change was in signaling as i've said previously.
My alcatel aready have an E1 ISDN installed from local carrier. After asterisk is setup, we change cables from carrier to asterisk, and our spa
Hi,
Someone has a Grandstream GXV 3000 to run a small test with
me? I have one GXV 3000 setup and I can’t get video from that videophone
with Eyebeam.
Best regards,
Chris HARIGA
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I want to make the context [default] as an alarm, for not having
set-up correct.
I am looking for a way to get incoming calls via ENUM or via names (e.g.
sip:[EMAIL PROTECTED]) into a defined context. How can I do that?
bye
Ronald
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What a confused message, isn't it?
As far as I could understand, if you're getting a RJ45 for conection,
you won't need any kind of adaptor. For coaxial cable, you'll need a
balun. That's all layer 1 talk - physic layer
Yes, you need to know a lot more about your pbx to proceed with the
connecti
Great idea but I am looking for more informative answer than 'might
accidentally happen something'.
But anyway here is the result for your pleasure:
calling with context1:
-- Executing NoOp("Zap/32-1", "CONTEXT IS: context1 DIAL") in new stack
-- Executing Wait("Zap/32-1", "1") in new stac
Hi all,
I'm having trouble with Background DTMF detection, and would appreciate
any suggestions.
A call comes in to a Sangoma A200D PSTN line. A standard menu welcome
is used. Most of the time, callers have to wait until the message
completes in order to have their selection recognized. P
hola a todos, tengo una duda, ye he resuelto algunas pero otras llegan, bueno como habia dicho quiero conectar una pbx a una te110p, la pbx me ofrece señalizacion r2 europea en cable rj45 o coaxial. ese tipo de señalización me sirve para la tarjeta te110p, ademas, alguno de esos dos tipos de conexi
Try a
different, (larger or smaller) room with different acoustical
characteristics. You may be talking, the audio is transmitted from a primary
source - you - but then it may pick up the reflections of your
voice bouncing off of the walls in the room, and the phone may be picking
that up a
On Tue, Sep 26, 2006 at 05:28:12PM -0400, Kristian Kielhofner wrote:
> 2) Get a telco that lets you set any CID. I don't know if I just look
> trustworthy or something, but I have had no problems whatsoever getting
> several LECs and CLECs in multiple states to let me set any CID I want.
> Lo
Hi Francesco
Yes it is
SonicWALL Enhanced has an option called 'Persistent NAT'... Is it turned on?
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| Just wondering if anyone has come up with a reliable method for
| remotely monitoring Asterisk boxes.
|
| I need to be able to check if Asterisk is actually providing service
| (registering clients, processing calls), not just answering to pings.
|
| In the past I have used sipsak in a cron sc
I have two similar Billion HFC cards working on a bri-stuffed Asterisk
0.99 server which has run my phone system in the UK for 2 years
without fault. The hardware is getting old, and I thought it wilse to
upgrade at my own speed, rather than in response to a failure.
I've installed Asterisk 1.2.
Barry Fawthrop wrote:
Hi all
I didn't change anything that's my point
It has be running and working just fine then at 4:32 pm yesterday I
could not make or recieve VoIP calls via our VoIP Provider
They say the Invite packet was being rejected and thus there was no
"real" connection even thoug
I'm having speaker phone echo issues with my grandstream phones 100. i understand that the echo'ing issue is only obvious because of the round trip latency and that traditional phone lines have echo's too but because there is such a slight delay, it can be mistaken for side tone, which is perfe
Shawn Kelley wrote:
Hi all,
I've searched around and haven't found much of an answer to my issue. Any
advice from you would be appreciated.
Problem: Need to take an inbound call from our PRI and forward it to another
PSTN user via the PRI, sending the original callers id with it.
I know this can
Greeting Everyone,
Just wondering if anyone has come up with a reliable method for
remotely monitoring Asterisk boxes.
I need to be able to check if Asterisk is actually providing service
(registering clients, processing calls), not just answering to pings.
In the past I have used sipsak in a c
Hello, We have test this configuration but we think it's a problem with the Alcatel.how are you doing to make the trunk between alcatel and Asterisk?We use a card PRA recommended by an Alcatel's technician and you?
ThanksNicolasOn 9/26/06, Frederico Madeira <[EMAIL PROTECTED]> wrote:
I'm trying the
On Tue, September 26, 2006 22:21, Barry Fawthrop wrote:
> Hi all
>
> I didn't change anything that's my point
> It has be running and working just fine then at 4:32 pm yesterday I
> could not make or recieve VoIP calls via our VoIP Provider
> They say the Invite packet was being rejected and thus t
On Tue, Sep 26, 2006 at 03:03:23PM -0400, ?lvaro Palma wrote:
> I recently moved a TE406P card from an Intel D865GBF motherboard (where
> it worked fine), to an Intel D101Ggc card, and now I can't get the spans
> to got up correctly. All I get is an endless burst of:
As much of a pain as it is,
Hi all
I didn't change anything that's my point
It has be running and working just fine then at 4:32 pm yesterday I
could not make or recieve VoIP calls via our VoIP Provider
They say the Invite packet was being rejected and thus there was no
"real" connection even though SIP SHOW PEERS has us
On Tue, Sep 26, 2006 at 12:49:34PM -0500, Lacy Moore - Aspendora wrote:
>Yes, it is possible. But, your Telco has to support this. Your Telco has
>to give you the ability to set your caller ID. Some providers (and it
>sounds like yours may be one of them) only allow you to use number
On 18:32, Tue 26 Sep 06, Andrea Spadaccini wrote:
> Well, how does Asterisk interact with those devices? Is there a
> chan_gsm_pci?
It's using chan_zap.
junghanns.net created an extra zap driver for it, same as
with their quad/octobri and zap_hfc stuff.
So asterisk will see it as Zap/
--
Michie
Ok, so does anyone know who the contributor of the new moh code is into
Asterisk 1.4? I'll email them directly.
Doug.
> -Original Message-
> From: Douglas Garstang
> Sent: Tuesday, September 26, 2006 8:31 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [aster
> "DG" == Douglas Garstang <[EMAIL PROTECTED]> writes:
DG> I'd like to know if anyone has sucessfully managed to run multiple
DG> instances of Asterisk on the same system. - Did you run each
DG> instance as a separate user? - Did you have any install or config
DG> problems? - It looks like the
On 9/26/06, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
Next thing to do, I guess, is to run:strace ztcfgto see which device exactly is accessed . Though /dev/zap/ctl is the
usual suspect.
[EMAIL PROTECTED] zaptel-1.4.0-beta1]# strace ztcfgexecve("/sbin/ztcfg", ["ztcfg"], [/* 25 vars */]) = 0unam
Barry Fawthrop wrote:
Hi all
Anyone using a sonicwall firewall ?
I have been and then suddenly it drops UDP packets because SIP is no
longer on port 5060 but some random assigned port ?
Why ?
Yes, have multiple clients with asterisk behind a sonicwall.
I don't understand from your wording
I recently moved a TE406P card from an Intel D865GBF motherboard (where
it worked fine), to an Intel D101Ggc card, and now I can't get the spans
to got up correctly. All I get is an endless burst of:
== Primary D-Channel on span 4 up
== Primary D-Channel on span 2 up
!! Got a UA, but i'm in st
Here is an output from a 1.4.0-Beta2
voipgw*CLI> show channeltypes
TypeDescription Devicestate
Indications Transfer
-- --- ---
---
Agent Call Agent Proxy Channel
Hi,
Is there way a way to restrict access to certain menus, such as the
following:
0 Mailbox options
1 Record your unavailable message
2 Record your busy message
3 Record your name
4 Record your temporary message (new in Asterisk v1.2)
Thanks in advance,
Jack
__
I would also recommend either the Polycom IP4000,
or the Clearone MAXIP, both of which are SIP native. If cost is an issue,
you can also take an inexpensive Polycom analog conference phone, such as the Voicestation
100, and SIP enable it using a Linksys SPA-1001 analog adapter. For about
Hi all.
As a convieneince to my users, I'm trying to strip off the leading 1 and
areacode from incoming calls. However, when I do, the src field in the CDR
is also stripped. I'd like the CDR to reflect the "connonical" form of the
incoming number.
Any way do to this?
TIA,
Mike Diehl.
__
On Tuesday 26 September 2006 13:57, C F wrote:
> Andrew what does "show channeltypes" give you?
*CLI> show channeltypes
TypeDescription Devicestate Indications
Transfer
-- --- --- ---
I spent quite a bit of time debugging the
7935/7936, and it is an issue inside the firmware that Cisco knows how to work
around in CallManager. There are better conference phone options available, and
development on chan_sccp is basically dead at this point anyway, so I don’t
see this one e
Hi guys!Can someone give advice on nice H323 IP phones brands?? I'm looking for some H323 IP phones for a customer. Diving in theinternet found the Uniden - TVUNIDEN_UIP300, but haven't ever heard about them. Can someone give feedback experince about it??, configease, sound quality, visual appe
Thanks All
I have those settings already enabled
It is rejecting the SIP INVITE packet not even getting to Voice at all
The VoIP provider shows a registered with a good Qualify time 55 ms but
not calls come in due to the Invite packet being rejected
Why and why would it suddenly do this noth
Andrew what does "show channeltypes" give you?
On 9/26/06, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote:
On Tuesday 26 September 2006 13:02, Steven wrote:
> That is the metermaid patch. It has been included into 1.4 as far as I
> know.
I do not see "DevState" in my "show application" output, so
How about trying Noop(CONTEXT IS: ${CONTEXT}) before you post it? but
now that you have posted just try it and report back.
On 9/26/06, Artifex Maximus <[EMAIL PROTECTED]> wrote:
Hello,
For example I have this dialplan:
[context1]
exten => s,1,Noop
exten => s,n,Dial(...)
exten => s,n,Playback(
There
seems to be three tiers in my experience:
1.
Only your DID's
2.
Arbitrary, but the pilot number of the PRI will appear if you suppress your
Caller ID
3.
Completely arbitrary, including <--this is the fa
shizzle
So you
want 2) or 3) but definitely it is a telco thing. You need t
On Tuesday 26 September 2006 13:25, Lacy Moore - Aspendora wrote:
> http://forums.digium.com/viewtopic.php?t=891&highlight=shared+line
Direct link for those of us who can't stand forums:
http://bugs.digium.com/view.php?id=5779
-A.
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Besides for what Lacy answered, have you tried NOT playing with
setting CID? Just do a blind xfer, or just use dial whatever on the
DID itself. If that doesn't work then like Lacy said your provider
might be blocking it.
On 9/26/06, Shawn Kelley <[EMAIL PROTECTED]> wrote:
Hi all,
I've searched
I am thinking of using a mini atx 1u server with a digium zaptel
(wcte11xp) installed to act as a SIP gateway. This way any of my
asterisk servers can forward calls to any gateway (seperated by about
3miles of fiber). Has anyone else tried this? I would just load a
basic asteisk config and zap
Yes, it is possible. But, your Telco has to support this. Your Telco has to give you the ability to set your caller ID. Some providers (and it sounds like yours may be one of them) only allow you to use numbers which you are authorized to use (such as your DIDs).
On 9/26/06, Shawn Kelley <[
Hello,
For example I have this dialplan:
[context1]
exten => s,1,Noop
exten => s,n,Dial(...)
exten => s,n,Playback(${CONTEXT})
exten => s,n,Hangup
[context2]
include => context1
[context3]
include => context1
Then I make dial-out call files with context2, context3, etc. What is
the value of $
Steven,
If you are trying to do this on a stock Asterisk system (and I can certainly understand why you would want to), then what I have implemented will definitely not work. I couldn't find anyway to do this on a stock system. Upgrades are going to be a nightmare with all the patches that have
;exten => 799,hint,DS/mmgc
Lacy, What is the DS/mmgc?
The DS is what the DevState patch adds. I actually got to this point by following this thread:
http://forums.digium.com/viewtopic.php?t=891&highlight=shared+line
After I implemented these changes, I had the DevState on the system.
Hi all,
I've searched around and haven't found much of an answer to my issue. Any
advice from you would be appreciated.
Problem: Need to take an inbound call from our PRI and forward it to another
PSTN user via the PRI, sending the original callers id with it.
I know this can be done since we curr
On Tuesday 26 September 2006 13:02, Steven wrote:
> That is the metermaid patch. It has been included into 1.4 as far as I
> know.
I do not see "DevState" in my "show application" output, so I would say no,
it's not in 1.4.
-A.
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That is the metermaid patch. It has been included into 1.4 as far as I know.
I am hoping to use that for parking slot BLFs on the phones.
My extension for day/night mode is not a real channel, so I am hoping to set
the hint value manually.
--
--
Steven
http://www.glimasoutheast.org
"C F"
Barry Fawthrop wrote:
Hi all
Anyone using a sonicwall firewall ?
I have been and then suddenly it drops UDP packets because SIP is no
longer on port 5060 but some random assigned port ?
Why ?
Which Sonicwall model? Some (like the TZ170) have special VOIP settings,
like "Enable consistent
;exten => 799,hint,DS/mmgc
Lacy, What is the DS/mmgc?
-- -- Steven
http://www.glimasoutheast.org
"Lacy Moore - Aspendora" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]...
Is
it possible to manually set the hint status of a virtual extension via the
Barry Fawthrop wrote:
Hi all
Anyone using a sonicwall firewall ?
I have been and then suddenly it drops UDP packets because SIP is no
longer on port 5060 but some random assigned port ?
Why ?
Thanks all
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Barry Fawthrop wrote:
Hi all
Anyone using a sonicwall firewall ?
I have been and then suddenly it drops UDP packets because SIP is no
longer on port 5060 but some random assigned port ?
Why ?
SIP is still on 5060, but the AUDIO (which is RTP) is on a dynamically
negotiated port. Now you
Hi all
Anyone using a sonicwall firewall ?
I have been and then suddenly it drops UDP packets because SIP is no
longer on port 5060 but some random assigned port ?
Why ?
Thanks all
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asteri
Ciao Michiel,
> > > http://www.junghanns.net/en/GSM-PCI_produkt.html
> > >
> > > If they are as stable as the quad/octo BRI cards they have
> > > it's a real winner.
> >
> > Where can I see the prices of this cards?
>
> My supplier has them listed as:
> UnoGSM: 900 euro
> DuoGSM: 1200 euro
> Qu
Did anyone success to install X100P Clone card on Asterisk to work with Japan stanadards for Analog line over ISDN TA. I can't make call when the call is ringing is OK and in the moment when the call is pick up the line is droped. I have hang ups all the time. I can't make call. Did anyone else hav
On 10:25, Tue 26 Sep 06, Tomislav Par?ina wrote:
> In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> > 1/2/4 simslot pci card:
> > http://www.junghanns.net/en/GSM-PCI_produkt.html
> >
> > If they are as stable as the quad/octo BRI cards they have
> > it's a real winner.
>
> Where can I
IIRC, there was a dev status for the local channel being worked on the
bug tracker.
Ok, here is the link:
http://bugs.digium.com/view.php?id=5779
On 9/26/06, Lacy Moore - Aspendora <[EMAIL PROTECTED]> wrote:
> Is it possible to manually set the hint status of a virtual extension via
the dialpl
On Tue, Sep 26, 2006 at 06:03:46PM +0300, Tzafrir Cohen wrote:
> On Mon, Sep 25, 2006 at 08:58:52PM +0100, Julian Lyndon-Smith wrote:
> > Jay R. Ashworth wrote:
>
> > voicemail.conf doesn't, as it needs to be modified by app_voicemail for
> > password changes.
>
> An alternative is to use an ext
marek cervenka wrote:
T38 passthrough doesn't seem to work in trunk at the moment.
that's true
http://bugs.digium.com/view.php?id=7679
http://bugs.digium.com/view.php?id=7844
t.38 in asterisk 1.4
http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38
I've taken the code in Openpbx so
On Wednesday 20 September 2006 21:40, Douglas Garstang wrote:
> We stuck OpenSER in between the phones and Asterisk, and pointed our phones
> towards the OpenSER boxes for SIP registrations and subscriptions. When
> OpenSER received a REGISTER or SUBSCRIBE message, it would use the send()
> command
Here's what we set in menuconfig when building Linux kernels for
multi-processor systems:
Processor Type and Features --->
->Symmetric multi-processing support
->Timer frequency (1000 HZ)
Device Drivers --->
Character devices --->
<*> Enhanced Real Time Clock Support
Re
I'm trying the same in Alcatel 4200 and i solved changing ignaling from pri_cpe for pri_net.-- Frederico Madeira[EMAIL PROTECTED]
2006/9/26, Sylvain ZUCCA <[EMAIL PROTECTED]>:
Hi,
can you send logs from alcatel 4400 ? just log in with account mtcl and launch "t3" to see traces from the PBX
Bes
Rich Adamson wrote:
Eric "ManxPower" Wieling wrote:
Use IP addresses instead of hostnames in your Asterisk config. It
sucks, but that is the only way I know of.
Eric Bishop wrote:
When we loose Internet access (DNS) Asterisk basically halts until
Internet
comes up even for internal registrat
I want to be able to playback a certain soundfile for
all parties in a call to hear.
How would I do that?
Eric
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Pato Valarezo wrote:
Lacy Moore - Aspendora wrote:
Wherever you have your exten => s,1,Answer statement, replace with:
exten => s,1,Wait(30) ; or however long you want to wait to give
someone else the chance to answer
exten => s,n,Answer
then continue on.
Asterisk will then wait 30 secon
Hang onI have a 480i on my desk. The deskset definitely has a hold key. The programmable keys make a VM key really easy too.
The cordless handset is limited by the number of buttons, but there are keystrokes for hold and a number of other functions. I wouldn't say that the cordless could
Do anybody knows?
Rgds, Ricardo.
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On Tue, Sep 26, 2006 at 08:45:49AM -0400, Michel Vaillancourt wrote:
> Tzafrir Cohen wrote:
> >
> > what's the contents of /etc/zaptel.conf ?
> >
> pbx1:~# cat /etc/zaptel.conf
> #
> # Zaptel Configuration File
> # This file is parsed by the Zaptel Configurator, ztcfg
> #
> loadzone = us
> defaul
On Mon, Sep 25, 2006 at 08:58:52PM +0100, Julian Lyndon-Smith wrote:
> Jay R. Ashworth wrote:
> voicemail.conf doesn't, as it needs to be modified by app_voicemail for
> password changes.
An alternative is to use an external script to modify that file.
--
Tzafrir Cohen sip:[EMAIL PROT
1 - Eclipse situation - What is inside fastagi-mapping.properties ? Are
you using the sample HelloAgiScript from asterisk-java ?
2 - Command line situation - what's the command line you are using ?
[]'s,
Edmilson Santana
Unitech Tecnologia de Informação (http://www.unitech.com.br/)
[EMAIL
Hi,
can you send logs from alcatel 4400 ? just log in with account mtcl and launch "t3" to see traces from the PBX
Best Regards.
2006/9/26, et pourquoi pas ? epp <[EMAIL PROTECTED]>:
Hi everybody, I have a problem and maybe you have a idea for me. I want to connect a asterisk and a alcatel 440
Eric Bishop wrote:
Hi All,
When we loose Internet access (DNS) Asterisk basically halts until
Internet comes up even for internal registrations and calls. We are even
running a caching DNS server on the Asterisk box but this does not seem
to help. Any suggestions?
We just went through the s
I had a problem on one box where /var/run/asterisk/ did exist and had the correct
non-root permissions.
There was a typo in /etc/asterisk/asterisk.conf.
was: astrundir =>
/var/run
changed to: astrundir =>
/var/run/asterisk
I do not remember which version of asterisk this
was or if it was
Steven wrote:
I found this command if your Cisco switches support it:
"auto qos voip trust"
You set this on each interface.
It automatically prioritizes all SIP and skinny traffic, but not iax.
There is also "auto qos voip cisco-phone". This one can detect a Cisco phone
and prioritize it.
I ju
Eric "ManxPower" Wieling wrote:
Use IP addresses instead of hostnames in your Asterisk config. It
sucks, but that is the only way I know of.
Eric Bishop wrote:
When we loose Internet access (DNS) Asterisk basically halts until
Internet
comes up even for internal registrations and calls. We a
Hi All,When we loose Internet access (DNS) Asterisk basically halts until Internet comes up even for internal registrations and calls. We are even running a caching DNS server on the Asterisk box but this does not seem to help. Any suggestions?
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