Greetings...things got way better for us when we:
0. Opted for voip gateways
1. Eliminated all hubs for switches
2. Eliminated all viruses (I hate PCs)
3. Recabled and seperated our voice from our data network
Three months later we just can't be happier!
Todd
"Terence Parker" <[EMAIL PROTECTED
Hello,
I have been trying to get my coders to work without a conversion. I have
read all the available asterisk documentation and support groups without
any luck. Here is my issue. (Please feel free to ask questions if you do
not understand what I am talking about.)
I am using Cisco ATA-186 se
Does anyone know of any inexpensive alternatives to the four
port analog module offered by Digium ($305) what work
seamlessly with asterisk?
Thanks
On Sun, Jan 04, 2004 at 07:38:16PM +, WipeOut wrote:
>
> Also a failover system would typically only be 2 servers, if there were
> a cluster system there could be 10 servers in which case five 9's should
> be easy..
>
Err, no. five 9s is *never* easy.
Does your telco provide you with SLAs
I think your problem comes from a misunderstanding of how the calls are
placed. With your canreinvite=no in the ATA section, you end up with the
ATA negotiating with asterisk for a call leg. Then you have asterisk
negotiating for the other call leg. Since the RTP stream is going
through asterisk, i
Asterisk Newbie writes:
> Does anyone know of any inexpensive alternatives to the four port analog
> module offered by Digium ($305) what work seamlessly with asterisk?
>
> Thanks
While we're on the subject, I'm looking for a 4-port FXO PCI card
supported by *.
--
"Speed, it seems to me, pro
John Coll wrote:
Half an hour's research and reading tells me that PCMU and PCMA are G.711.
http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+grandstream+budgetone
Updated.
/O
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digi
On Mon, 2004-01-05 at 01:38, Asterisk Newbie wrote:
> Does anyone know of any inexpensive alternatives to the four port
> analog module offered by Digium ($305) what work seamlessly with
> asterisk?
In telephony terms, that is inexpensive. I think the norm is ~$150 per
port or more. I think on a p
Steve Totaro wrote:
I just signed up for Voicepulse with a DID. I can register with
Voicepulse and dialout just fine. Only problem is that when I dial my
DID from my POTS line I just get a fast busy and nothing in the console.
Any ideas?
I signed up for a VP DID several months back. It too
Good morning,
does anyone know of a (PCI-)card to allow asterisk to have an internal
ISDN bus, ie. being able to utilize ISDN phones as extensions to
Asterisk, like FXS for analog?
--
Best regards
Peer Oliver Schmidt
the internet company
___
Asteris
Hi,
> -Original Message-
> does anyone know of a (PCI-)card to allow asterisk to have an
> internal
> ISDN bus, ie. being able to utilize ISDN phones as extensions to
> Asterisk, like FXS for analog?
The QuadBRI from www.junghanns.net/asterisk should be able to do it I
think...
Flori
You are not going to find much for less than $100/port.
On Mon, 2004-01-05 at 01:38, Asterisk Newbie wrote:
> Does anyone know of any inexpensive alternatives to the four port
> analog module offered by Digium ($305) what work seamlessly with
> asterisk?
>
>
>
> Thanks
--
Go to http://www.dig
That's EuroISDN only, isn't it?
On Mon, 2004-01-05 at 03:10, Florian Overkamp wrote:
> Hi,
>
> > -Original Message-
> > does anyone know of a (PCI-)card to allow asterisk to have an
> > internal
> > ISDN bus, ie. being able to utilize ISDN phones as extensions to
> > Asterisk, like FX
Nicolas Bougues wrote:
On Sun, Jan 04, 2004 at 07:38:16PM +, WipeOut wrote:
Also a failover system would typically only be 2 servers, if there were
a cluster system there could be 10 servers in which case five 9's should
be easy..
Err, no. five 9s is *never* easy.
Does your telco p
Hi,
> -Original Message-
> That's EuroISDN only, isn't it?
Ehm, I'm not sure, but I think I've heard that yes. Kapejod can probably
provide more accurate info.
> > > -Original Message-
> > > does anyone know of a (PCI-)card to allow asterisk to have an
> > > internal
> > > ISD
Hi!
> >>I want to have Asterisk as my gateway to the outside world and use
> >>another PBX to connect my existing phones.
> >>exten => ${OUTSIDEMSN},1,Dial,CAPI/${MSN2NDPBX}:${EXTEN}
> >>How do I transfer the caller Id information initially coming in?
>
> > I have strong doubts that this can be
Thanks Guys.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Monday, January 05, 2004 1:23 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] RE: Inexpensive Analog Ports
You are not going to find much for less than $100/port.
On M
Hi!
> The problem I have is when I put a call on hold, or attempt to transfer
> a call (same issue, I expect). Once the call is put on hold, I can't
> resume it and, after a couple of seconds, I get a "Maximum retries
> exceeded" error (verbose mode).
> ...
> I suspect that the Cisco 7960 confi
I didn't get any response on
that question, so i supose this feature is possible but there isn't an
implementation of it.
I'm ready to sponsor this feature in the
manager interface (i tried the redirect command but it doesn't work) can
somebody help me ?? this feature would make it possbil
hi
I have a passive isdn port configured in modem.conf
in extention.conf i use this two channel (ttyI0 and ttyI1) with the string:
exten => _NX,1,Dial,Modem/g1:${EXTEN}|60|r
how can i hide my msn?
is it possible to activate the clir with the @ before the ${EXTEN}?
thanks
Cristian
Hi!
> If you need to know whether it is coming in for Pres. or Support, use
> CallerID. We have a similar need. We have a timeout on our main line
> that will ring all phones, and our tech support line rings all phones.
> It is acceptable for our programmers to ignore the main line if there
> are
Hi,
> -Original Message-
> I have a passive isdn port configured in modem.conf
> in extention.conf i use this two channel (ttyI0 and ttyI1)
> with the string:
> exten => _NX,1,Dial,Modem/g1:${EXTEN}|60|r
>
> how can i hide my msn?
> is it possible to activate the clir with the @
>
> Does your telco provide you with SLAs that make five 9s reasonable at
> all ?
>
LOL... Our telco services could be down for several hours at a time.
We found than most US Broadband carriers (DSL and Cable) offer a
"best effort" zero SLA service. If you are using broadband as a primary
transpo
Hi Again,
Apart from X-lite client I have also tried eStara, diax phone, iaxcomm and
some others. I have tried different codecs - GSM, aLAW uLAW. They all give
the same result. In the direction PSTN user ---> Softphone user the sound is
crystal clear (also tried on a dial-up connection), in the o
Mike Machado wrote:
I guess that was another thing that was strange. When I talked, I saw no
RTP coming from the handytone to *. Would there be a reason the
handytone would not send RTP until it successfully received a RTP packet
from *, but since its not accepting RTP, it would not send it either
> -Original Message-
> From: Joe Kellman [mailto:[EMAIL PROTECTED]
>
> i am not sure if somebody asked this question in relation to your
problem,
> what do you get when you do a pri debug span X. i know with some
> providers
> (i am using xo) the messages are transmitted as a facility mes
Philipp von Klitzing [EMAIL PROTECTED] wrote:
> >
> > The problem I have is when I put a call on hold, or attempt to transfer
> > a call (same issue, I expect). Once the call is put on hold, I can't
> > resume it and, after a couple of seconds, I get a "Maximum retries
> > exceeded" error (verbose
Hi,
I was just wondering if anyone has seen this:
http://www.vovida.org/applications/downloads/G729A/
and if this could be integrated into *? I think it would be a great
thing for noncommercial applications.
Regards,
Robert
___
Asterisk-Users mail
Hi,
> -Original Message-
> I was just wondering if anyone has seen this:
>
> http://www.vovida.org/applications/downloads/G729A/
> and if this could be integrated into *? I think it would be a great
> thing for noncommercial applications.
Yes, this has been discussed before.
As you ma
> > - Original Message -
> > From: "Sean Cheesman" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>
> > Sent: Sunday, January 04, 2004 7:30 PM
> > Subject: RE: [Asterisk-Users] Newbie - MWI
> >
> >
> > -Original Message-
> > From: Andrew Thompson [mailto:[EMAIL PROTECTED]
> > Sent: S
> > Using another load-balancing box (F5 or whatever) only moves the problem
> > to that box. Duplicating it, moves the problem to another box, until
> > the costs exponentially grow beyond the initial intended value of the
> > solution. The weak points become lots of other boxes and infrastructure
- Original Message -
From: <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, January 04, 2004 7:31 PM
Subject: [Asterisk-Users] Earpiece Connections
> Does anyone know of a piece of hardware that can allow multiple earpices
> to be connected directly to a server running Asterisk.
Try using canreinvite=yes in all three contexts. Would that screw-up ATA, I
do not know, cause I have no Cisco's ?
SW
Message: 5
Date: Mon, 05 Jan 2004 02:29:49 -0500
From: SamW <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Codec Negotiation Does not seem to work as
expected
On Mon, 5 Jan 2004, Philipp von Klitzing wrote:
> Am I correct by stating that as of now none of the VoIP protocols (SIP,
> MGCP, H.323, Skinny) supports such a "silent ring" feature? Would SIMPLE
> solve this? Naturally I wouldn't like those calls to show up in my list
> of unanswered calls. For
Title: Message
I have an
installation which is currenly using 14 DID Trunk Lines. I need to be able
to use Caller ID information and currently it is not available on these
lines.
Is there another way
to access this information?
Thanks
Brent,
I have 3 in my * server with no issues thus far. The box isn't in
serious use as yet but I have tried all three lines active in testing
MeetMe functionality.
Michael
On Sun, 4 Jan 2004 21:31:51 -0500, Brent Franks wrote:
>Has anyone had any success using more than one or two X100P cards
Comments inline
Read the faq, checked the config files... can't find anything about an
echo problem like this. Here's what I've got:
4 channel t1 card, span 2 going to channel bank with both fxs and fxo lines
Polycom IP600 phones on same LAN with asterisk
iax connection to voicepulse (T1 going ou
On Mon, 5 Jan 2004 13:29:06 +0100, "Dawid Mielnik" <[EMAIL PROTECTED]>
wrote:
>
>Hi Again,
>
>Apart from X-lite client I have also tried eStara, diax phone, iaxcomm and
>some others. I have tried different codecs - GSM, aLAW uLAW. They all give
>the same result. In the direction PSTN user ---> Sof
Comments below:
On Mon, 2004-01-05 at 07:10, Andrew Thompson wrote:
> Hmm... I was trying to seperate the mailbox= definition from the context
> specifically named "default". IMHO, the implied(i.e. default) context for a
> mailbox should be the context that's indicated for the device.
>
> Take th
On Mon, 2004-01-05 at 09:28, [EMAIL PROTECTED] wrote:
> I have an installation which is currenly using 14 DID Trunk Lines. I
> need to be able to use Caller ID information and currently it is not
> available on these lines.
> Is there another way to access this information?
Convert to PRI.
Whil
Hi Richard,
>Load balancers have some added value, but those that have had to deal
>with a problem where a single system within the cluster is up but not
>processing data would probably argue their actual value.
I've done quite a lot of work with clustered/ha linux configurations. I
usualy try to
Michael Van Donselaar wrote:
On Mon, 5 Jan 2004 13:29:06 +0100, "Dawid Mielnik" <[EMAIL PROTECTED]>
wrote:
Hi Again,
Apart from X-lite client I have also tried eStara, diax phone, iaxcomm and
some others. I have tried different codecs - GSM, aLAW uLAW. They all give
the same result. In the dir
Hi list,
I have to connect two asterisk box, in this scenario:
[asterisk1]sip[asterisk2]PSTN
I must use sip, cos we'll use cisco rtp header-compression to save
bandwidth.
Could you tell me the best way to send calls from asterisk1 to
asterisk2, since I canno
Title: Message
Hello
all. I know * doesn't directly support recording mp3 files, but I was
wondering if anyone has created an AGI to do it indirectly. Thanks in
advance.
B. J.
On Mon, 2004-01-05 at 09:04, Steven Critchfield wrote:
> On Mon, 2004-01-05 at 09:28, [EMAIL PROTECTED] wrote:
> > I have an installation which is currenly using 14 DID Trunk Lines. I
> > need to be able to use Caller ID information and currently it is not
> > available on these lines.
> > Is ther
Christian has been testing the snom code against on of our Asterisk
systems.
2.03f did address some multiple registration issues but the newest snom
200 version is 2.03g - that has addressed some additional registration
issues.
New versions for the other snom phones are expected today.
Richard
Thanks for all who is helping.
I tried, canreinvite=yes on all contexts but that do not seem to work as
well. But the issue is not related to negotiating between end points,
but for me, asterisk do not have a proper configuration scheme which
works, to the requirement of the user. The Code-Negotia
Michael,
The PSTN cards and the network card are not sharing an interrupt, the PSTN
interface is sharing an interrupt with an audio controller and an smbus, the
network card is not sharing an IRQ with anything though.
Best regards,
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailt
Andrew Thompson wrote:
Hmm... I was trying to seperate the mailbox= definition from the context
specifically named "default". IMHO, the implied(i.e. default) context for a
mailbox should be the context that's indicated for the device.
Take the below grossly simplified sip definition:
[sipphone1]
t
On Mon, 2004-01-05 at 09:24, Eduardo Goncalves wrote:
> I must use sip, cos we'll use cisco rtp header-compression to save
> bandwidth.
>
> Could you tell me the best way to send calls from asterisk1 to
> asterisk2, since I cannot use IAX trunking?
Maybe I'm way off base here, but I
Jared Smith wrote:
Comments below:
On Mon, 2004-01-05 at 07:10, Andrew Thompson wrote:
Hmm... I was trying to seperate the mailbox= definition from the context
specifically named "default". IMHO, the implied(i.e. default) context for a
mailbox should be the context that's indicated for the device
Why not use IAX2 trunking you can accomplish the same results with .. I
tried SIP to SIP with asterisk you must do it without passwords.
bkw
On Mon, 5 Jan 2004, Eduardo Goncalves wrote:
> Hi list,
>
> I have to connect two asterisk box, in this scenario:
>
> [asterisk1]sip[asterisk
I still have the problem, but I have noticed one interesting fact. I
have choppy sound from SIP to PSTN, but the voicemail prompts sound
great (asterisk generated sounds are working well)... I will keep trying
and keep you informed.
On Mon, 2004-01-05 at 13:22, WipeOut wrote:
> Michael Van Donsela
On Mon, 2004-01-05 at 10:35, Jared Smith wrote:
> On Mon, 2004-01-05 at 09:04, Steven Critchfield wrote:
> > On Mon, 2004-01-05 at 09:28, [EMAIL PROTECTED] wrote:
> > > I have an installation which is currenly using 14 DID Trunk Lines. I
> > > need to be able to use Caller ID information and curre
Oh btw.
I am using a P4, 500Mb RAM
regards,
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of WipeOut
Sent: Monday, January 05, 2004 5:22 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] one way choppy sound problem !
Michael Van Donselaar wrote
On Mon, 2004-01-05 at 10:24, Eduardo Goncalves wrote:
> Hi list,
>
> I have to connect two asterisk box, in this scenario:
>
> [asterisk1]sip[asterisk2]PSTN
>
> I must use sip, cos we'll use cisco rtp header-compression to save
> bandwidth.
Will rtp header compression n
I am having the same problem, but only with one specific user, so I
believe it is network related.
Anyone that can point me in the specific direction of what would cause
this?
> -Original Message-
> From: WipeOut [mailto:[EMAIL PROTECTED]
> Sent: Monday, January 05, 2004 10:22 AM
> To: [E
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
PBX wrote:
| I wanted to post the beginings of my latest IVR Project for an automated
| Time Clock software.
Not to alarm you too much, but MCI WorldCom has a patent on this kind of
thing and is suing people that develop/implement/use these kinds of
sys
This is the low cost option for an FCC
certified board with support from a real company. ;-)
Paul Mahler
mail:[EMAIL PROTECTED]
phone: 650.207.9855
fax: 877.408.0105
-Original Message-
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Asterisk Newbie
Sen
On Mon, 2004-01-05 at 10:36, B. J. Bomar wrote:
> Hello all. I know * doesn't directly support recording mp3 files, but
> I was wondering if anyone has created an AGI to do it indirectly.
> Thanks in advance.
That should be fairly trivial depending on what you want to accomplish.
If all you wan
I have observed this also.
Try downgrading the firmware on the Snom to 1.16x. That usually fixes
it for me. Although that's obviously a workaround and not a true fix.
It is the Snom phone sending the 486 message; I just don't know why.
- Matt
Hello All,=0A=0AWhenever I try calling SNOM 200
Wipeout, If you want you can send me an Ethereal trace of the RTP stream
and I can do an analysis of it to determine if there is anything obvious
there. (please use G.711, and try something like counting from 1 to 20).
Regards,
Andres
- Original Message -
From: "WipeOut" <[EMAIL PROTECT
Michael,
We also have the same DID trunk lines and was told by Telco, on this line we
can't get callerID facility. I don't know is there some way to do it.
I would like to ask you a question, what hardware are you use to terminate
all 14 DID trunk line, into asterisk.
I am facing little proble
We have a pre-release candidate at http://snom.com/download/share for snom
200 (2.02h). We found one issue with DNS names for NTP (only IP supported in
this version) and a problem with a call park/pickup (which should not have
any affect with Asterisk). Anyway, I think its a big progress in compar
I agree w/ Michael...if you've got 14 one-way Trunk Lines, you should be
able to get a PRI for the same cost they are paying for the 14 Trunk
lines, and then you will get 23 in/out lines and DID and all the other
bells and whistles for those lines.
It will also save you the cost of a Channelbank s
my biggest concern about defaulting the context to anything at all
besides [default] is that you then have to remember to configure the
voicemail.conf with the corresponding contexts. as it stands, you have
the ability to do just that, but you don't have to. if you have several
hundred extensions
On Mon, 05 Jan 2004 10:19:24 -0700
Jared Smith <[EMAIL PROTECTED]> wrote:
> On Mon, 2004-01-05 at 09:24, Eduardo Goncalves wrote:
> > I must use sip, cos we'll use cisco rtp header-compression to
> > save
> > bandwidth.
> >
> > Could you tell me the best way to send calls from asteri
Title: Re: [Asterisk-Users] Codec Negotiation Does not seem to work as expected ?? Help Please !!
Steve,
My Problem is not a problem, with the codec negotiation between end points. But when asterisk does it with canreinvite=no, * do not do it right. I replied with a lengthy discussion about m
On Mon, 2004-01-05 at 11:20, Olle E. Johansson wrote:
> Jared Smith wrote:
> > Comments below:
> >
> > On Mon, 2004-01-05 at 07:10, Andrew Thompson wrote:
> >
> >>Hmm... I was trying to seperate the mailbox= definition from the context
> >>specifically named "default". IMHO, the implied(i.e. defa
On Mon, 2004-01-05 at 11:20, Olle E. Johansson wrote:
> Jared Smith wrote:
> > Comments below:
> >
> > On Mon, 2004-01-05 at 07:10, Andrew Thompson wrote:
> >
> >>Hmm... I was trying to seperate the mailbox= definition from the context
> >>specifically named "default". IMHO, the implied(i.e. defa
On Mon, 5 Jan 2004 11:20:08 -0600 (CST)
Brian West <[EMAIL PROTECTED]> wrote:
> Why not use IAX2 trunking you can accomplish the same results with ..
> I tried SIP to SIP with asterisk you must do it without passwords.
Because cisco doesn't compress IAX headers, only rtp.
[ ]'s
Eduardo
- Original Message -
From: "Olle E. Johansson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, January 05, 2004 12:20 PM
Subject: Re: [Asterisk-Users] mailbox= wrong context. was: Newbie - MWI
> Jared Smith wrote:
> > Comments below:
> >
> > On Mon, 2004-01-05 at 07:10, Andrew T
This newbie has been trying out Asterisk. It has been both a) surprisingly
painful and b) impressive in terms of helpful support from other users.
Having got two phones to communicate and then got voicemail MWI going
(neither painlessly) I decided the next step was to implement call transfer
as pe
I am considering switching to Asterisk for my ISP.
I currently use a NorTel ICS with a NAM II (standard voicemail and Minuet
ACD software).
I have a technical support center staffed by a max of 5 agents at one time.
I am wondering if it is possible to do these things with a queue:
1) When a cu
Jason A. Pattie wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
PBX wrote:
| I wanted to post the beginings of my latest IVR Project for an
automated
| Time Clock software.
Not to alarm you too much, but MCI WorldCom has a patent on this kind of
thing and is suing people that develop/implem
Send me all your grandstreams.
I don't see anything wrong with em. (Now that I figured out how to set
them up with *) :)
> -Original Message-
> From: Nick Bachmann [mailto:[EMAIL PROTECTED]
> Sent: Friday, December 26, 2003 4:42 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users
Hi All
I have problem trying to receive incoming calls from iaxtel.com. The
error message is " rejected connect from ip address - .
I have set up the iax.conf file as follow:
port=5036
allow=gsm
register=>dkwok:[EMAIL PROTECTED]
[dkwok]
type=friend
context=from_iaxtel
My extensions.conf is as
Hi,
That would be great..
I will do the trace tomorrow, Its 21:00 and I am not really awake
enough.. :)
Later..
TeleSIP wrote:
Wipeout, If you want you can send me an Ethereal trace of the RTP stream
and I can do an analysis of it to determine if there is anything obvious
there. (please us
You also don't need such a complicated perl script, just muxing them
without cutting them is enough.
(Timing was fixed)
zoa.
At 14:41 4/01/2004 -0600, you wrote:
you nolonger need set-timestamp.agi as we have ${TIMESTAMP} in that format
by default now.
bkw
On Sun, 4 Jan 2004, John Baker wrote:
Hi there,
I stumbled on this list mostly by accident. I came across Asterisk * as a
means to help me get a better handle on my soaring telephone costs. Each
month I look at my phone bills and my stomach just turns because I can not
find any competition to Verizon which is the local anointed pho
Title: Echo with polycom phones
I have soundpoing ip 500 phones and the first few seconds of every call has echo, which then goes away. Is there a way to have the echo cancel on at the beginning? It seems like it is testing at the beginning but it would be nice if I could have it start clos
Hi,
I just setup my * with digium. I started testing
voicemail first between atas, and i am not sure why it is not prompting me any
when the call is not answered or if busy. i only get continuous
ringback and the following message:
asterisk*CLI> --
Executing Dial("SIP/6882332-1697",
agents if they're too slow answering the phones. He was kidding. :)
yes I was kidding... And a group of people are working on a new plugable
agents and queues setup. So far from what I have seen/heard its going to
rock. I magine loading a xyz_acd.so for xyz call strategy and it just
becomes aval
On Mon, 2004-01-05 at 12:47, Eduardo Goncalves wrote:
> On Mon, 05 Jan 2004 10:19:24 -0700
> Jared Smith <[EMAIL PROTECTED]> wrote:
>
> > On Mon, 2004-01-05 at 09:24, Eduardo Goncalves wrote:
> > > I must use sip, cos we'll use cisco rtp header-compression to
> > > save
> > > bandwidth.
> > >
I have seen in the list been receiving the HTML encoded part in the list, is
it possible to strip off the HTML part and to keep the text alone, so that
it will be clean and simple to read, both in the list and in the web.
Is there is any other reason to keep the HTML contents in?
Kannaiyan
__
This newbie has been trying out Asterisk. It has been both a) surprisingly
painful and b) impressive in terms of helpful support from other users.
Having got two phones to communicate and then got voicemail MWI going
(neither painlessly) I decided the next step was to implement call transfer
as pe
I've submitted a message twice this evening and it has not appeared. Are
messages censored on this board?
regards
john
-
John A Coll, Director, Connection Software
391 City Road, LONDON, EC1V 1NE, UK
Tel: 020 7713 8000 From outside UK Tel: +
Has anyone tested asterisk with Lindows ? just
curious ?
thanks,
Francisco
Is this setup possible?
ATA -> Asterisk -> SER with RTPproxy -> AnySIPGateway
how can I instruct * to send all unknown extensions to go to SER/RTPProxy? Do
I have to use "exten"? What syntax shld I use?
I cannot find any matching document from wikki. :(
Hi All,
Earlier today I tried using my new * server to host a simple 3 party
conference in a MeetMe conf room. I was on a snom 200, one other party
called inbound from PSTN via X100p, while the third party was a call
that I placed outbound via VoicePulse Connect. I transfered both
external parties
Andrew,
I need a keypad-less phone to listen and talk.
I would like to have up to 10 of these and I would like to have each of
them have their own extension so that calls can be routed directly to
them.
I would then be able to trigger the Pickup and Hangup functions on the
server itself.
Have yo
I don't think this will entirely work, but I'm brainstorming to get
wheels turning,
I don't know if there is a built in $DIALED_FROM_EXTEN, or you might
have to put it into the macro as $ARG2 or something like that.
Something like
;extensions.conf
[macro-transfer]
;
exten => s,1,Answer
exten
Basically I guess I am thinking of the traditional key systems approach
which is to have the CO lines appear on the phone. The problem it
appears with SIP (not really *) and the particular phones, is to have
the reporting. I guess what I was looking for was to have the buttons
not only represent
On Monday 05 January 2004 13:44, John Coll wrote:
> This newbie has been trying out Asterisk. It has been both a)
> surprisingly painful and b) impressive in terms of helpful support
> from other users.
>
> Having got two phones to communicate and then got voicemail MWI
> going (neither painlessly)
Title: This is a test
It appears that my replies aren't getting to the list. Just testing to see what is going on…
Sean
I have soundpoing ip 500 phones and the first few seconds of every call
has echo, which then goes away. Is there a way to have the echo cancel
on at the beginning? It seems like it is testing at the beginning but
it would be nice if I could have it start closer
Thanks
Sean Garland
Siskiyou
hi,
Hi there,
I stumbled on this list mostly by accident. I came across Asterisk * as a
means to help me get a better handle on my soaring telephone costs. Each
month I look at my phone bills and my stomach just turns because I can not
find any competition to Verizon which is the local anoint
I have set canreinvite=no in the sip.conf for each user (well, there are
only two) using a cisco phone. What does this imply?
As for whether the problem is due to the phones or asterisk however,
indications would suggest both, because:
- Voicemail works fine (and is clear)
- I can initiate a call
>
> Hi there,
>
> I stumbled on this list mostly by accident. I came across Asterisk *
> as a means to help me get a better handle on my soaring telephone
> costs. Each month I look at my phone bills and my stomach just turns
> because I can not find any competition to Verizon which is the local
> 1) Am I correct to assume that there is a way to dump Verizon and strictly
> go VOIP in a SOHO situation?
Yes
> 2) Can 1-800 numbers terminate to a VOIP assigned number?
Yes, I currently use Nufone.net for this now.
> 3) With VOIP am I under the assumption that one must also purchase license
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