X100P is 3.3v not 5v, at least the one I have. Works fine in a 4801.
John Simon wrote:
Is anyone using a net4801 and an analog only setup? I am looking for a
modem that is PCI 3.3V, apparently the X100P is 5.0V PCI only so it
won't work with the net4801.
I try to change something in ASTCC, but I am now totally blind,
I hang on one line now. I changed:
vpbx:/var/lib/asterisk/agi-bin # diff astcc-original.agi astcc.agi
22c22
< # exten => _00X,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN})
---
> # exten =>
_00X,1,DeadAGI(astcc.agi,
On Sun, 2005-03-20 at 23:12 -0700, Tom wrote:
> Anyway, I know this isn't a supported setup, so if thats your answer don't
> bother replying, I'm know this will be a kludge/hack to get working (if I can
> get it working at all). I'm just trying to do something that will be
> convienient for me and
However, I don't know the specific requirements for the T1 line or how to split the data and voice.
The point of VoIP is to consolidate data and voice onto one network.
Combining both allows for economies of scale:
* you don't have to use sangoma or digium card, this is the VoIP
provider's t
I'm getting the same problem running Asterisk CVS-HEAD-03/21/05-03:24:01
built by [EMAIL PROTECTED] on a i686 running Linux which is the
code from yesterday.
Robert Goodyear wrote:
FWIW I get the same exact error at the end of every VM session as
well, thus:
-- Playing 'vm-intro' (language
Hi,
I'm running CVS-HEAD-03/19/05-11:15:15 on Fedora Core 3 with Digium
TE410P card.
Calling into meeting rooms that have been configured with the p option
works fine.
From ZAP extensions the # key does not work to exit, however from SIP
extensions the # key works fine. This makes me believe t
On Mon, 2005-03-21 at 12:36 +0400, Jean-Michel Hiver wrote:
> >However, I don't know the specific requirements for the T1 line or how to
> >split the data and voice.
> >
> The point of VoIP is to consolidate data and voice onto one network.
> Combining both allows for economies of scale:
[SNIP]
Ummm, in an office of x people, where x is some arbitrary integer > 1,
is 1 PSTN line for emergency services sufficient ?? Personally, I would
think it isn't, but haven't quite determined what number is sufficient.
Consider, an office of 50 people, and a small fire breaks out, how many
people will
On Sun, 2005-03-20 at 23:40, Joseph wrote:
> How to compile additional module to asterisk?
>
> I have app_nv_backgrounddetect.c file and followed instructions below,
> but "make" did not generate app_nv_backgrounddetect.so or
> app_nv_backgrounddetect.o
>
> (1) Drop the code in your /usr/src/aste
On Mon, 2005-03-21 at 13:51 +0400, Jean-Michel Hiver wrote:
> >Ummm, in an office of x people, where x is some arbitrary integer > 1,
> >is 1 PSTN line for emergency services sufficient ?? Personally, I would
> >think it isn't, but haven't quite determined what number is sufficient.
> >
> >Consider
On Mon, 21 Mar 2005, Walter Klomp wrote:
> I'm running CVS-HEAD-03/19/05-11:15:15 on Fedora Core 3 with Digium
> TE410P card.
>
> Calling into meeting rooms that have been configured with the p option
> works fine.
>
> From ZAP extensions the # key does not work to exit, however from SIP
> e
On Sun, Mar 20, 2005 at 11:12:22PM -0700, Tom wrote:
>
> I have a quick question.
> I know that running X on an asterisk server is not officially "supported",
Generally it shouldn't cause "errors", but will probably degregate
performance, as an X server is probably as close as Asterisk is to the
Asterisk Ready.
*CLI> -- Executing route("SIP/7408-02e3", "370263") in new stack
-- odbcquery: query=370263
> Query = 370263 : SQLcmd = select routing, ring_timer from ddi_pool
where ddi_inbound = '370263'
Urgent handler
> app_route: Query Successful!
-- Varname= 55
i have been using the asterisk for some three
weeks. Previously i was using the softphone iax-phone and now i have to shift to
the sip phone xlite.
The problem is that there's always unbearable noice
in sip to sip calls. Is there any way to get rid of this
Kindest
MM Luqman
dear All
i signed up with an Aussie provider who gives me a DID in Aust...
when I call my number I get the following on the console
Mar 21 05:54:15 NOTICE[68071]: chan_iax2.c:6123 socket_read: Rejected
connect at
tempt from 203.13.163.245, who was trying to reach 's@'
the s part i can understand
So far, nobody has been able to tell us what this error means.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of MF Hulber
Sent: Lunes, 21 de Marzo de 2005 02:54 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
My objective is to estimate the performances of *
How much the trancoded can influence the
performances?
Thanks,
Ale
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* Alessandra Grasso ([EMAIL PROTECTED]) ha scritto:
> My objective is to estimate the performances of *
> How much the trancoded can influence the performances?
take a look at translate.c file to see how transcoding costs are
calculated. use the command 'show translation' at the CLI to see
inter
Really?? What if you happen to be the 4th person calling, and need to
inform them that you are trapped in the stationary cupboard, and luckily
you were carrying the cordless handset (but not your mobile phone)??
At the end of the day, I wouldn't expect any office to have a 1:1 ratio
between users:
Could you email login information
anonymous login isn’t allowed.
Sean
From:
Asterisk [mailto:[EMAIL PROTECTED]
Sent: 18 March 2005 16:20
To: Anil Kumar K; Giovanni Powell
Cc:
asterisk-users@lists.digium.com
Subject: Re: Re: [Asterisk-Users]
Meetme2 compilation problem
* Jer ([EMAIL PROTECTED]) ha scritto:
> my iax.conf looks like this
dunno if it may help, try adding
context=default
in your iax.conf 'general' section
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Does anyone have an example for using a live mp3 shoutcast stream with
mpg123 for hold music?
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> Is it possible to initiate/receive calls from a url (that is without
> having to install and configure a PC soft phone) using asterisk?
> If yes, may I please get some sites, pointers, HOWTOs on how its done?
You can also try the Flash Operator Panel, http://www.asternic.org. It
supports click-t
Hello list,
i configured correctly the codes in features.conf, loaded successfully
res_features, but while in a call (any type of call including zaptel to
zaptel, zaptel to sip, sip to sip) both sides hear DTMFs and nothing
happens...
i'm i missing something?
Thanks,
Calin.
1. create a directory inside /var/lib/asterisk or whatever you have
configured for that, i.e. /var/lib/asterisk/mohmp3-radio, then
2. create /var/lib/asterisk/mohmp3-radio/dummy.mp3
3. then add
live =>mp3:/var/lib/asterisk/mohmp3-radio,http://www.yourfavradio.com:port/
into your /etc/asterisk/mus
Hi list,
Is this supposed to be a joke? It doesn't sound very professional.
comedian
n 1: a professional performer who tells jokes and performs
comical acts
Moby Thesaurus words for "comedian":
banana, buffoon, burlesquer, card, caricaturist, choreographer,
clown, comedienne, c
There was a short discussion thread on this on / about 8/31/2004 but
no real answer was ever given.
Some people supposed that since a competing voicemail sounded the same
(Meridian) that there might have been some correlation between the two
but who knows.
I'm sure you're more than welcome to hir
Hi
I am using cvs and updating it every couple of days Unfortunately I am still
getting a 20 second timeout on sip calls placed to various providers, can
anyone see anything wrong from sip debugs? Or have any ideas what the
problem might be?
Cheers
Walt
sip debug peer of a provider:
http://www
joerg hanke wrote:
hi
i wonder why my outbound calls via asterisk->sipgate->german telecom have such
high delay rates (about 500 or mor ms) while inbound signals are quite
ok (max ca 200ms). any idea?
joerg
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Try username=guest pass=emailaddr.Andre- Oorspronkelijk Bericht -Onderwerp:ÂRE: Re: [Asterisk-Users] Meetme2 compilation problemAfzender: ÂSean Lowry <[EMAIL PROTECTED]>Aan:Â"Asterisk" <[EMAIL PROTECTED]>,"Asterisk Users Mailing List - Non-Commercial Discussion" Datum:Â21-03-2005 13:12
Where is my MoH class? I understand what's being done here... but I
don't see where that is.. like for meetme conferences, and being
placed on hold and such... which file?
On Mon, 21 Mar 2005 07:13:16 -0600, Henry Devito <[EMAIL PROTECTED]> wrote:
> 1. create a directory inside /var/lib/asteris
Hi,
Has anyone else experienced problems as of the last couple of months
when outbound calling through Freshtel?
I've started getting a "No authority found" error. I've tried
contacting them, and they seem to have some serious communication
issues with their IT team, infact I think they have seri
IPSwitchBoard Version 0.67 Release notes:
CRM integration, can call a web page with callerid when there's an incoming
call. You can specify the min. and max. length of the callerid.
Drop any active call.
Help file integrated in IPSwitchBoard.
Play button for sound files.
Bug fixes - thank you fo
Rich thanks, this makes it a little clearer. My servers are using NAT
behind a Cisco PIX. I only needed the simple patch (see below).
I configured sip.conf from these instructions:
http://www.voip-info.org/wiki-Asterisk+settings+Broadvoice
Hope this helps somebody. Sorry I wasn't clear about us
I am pretty sure that there are
no IAX providers that offer CallerID name but wanted to double check with the
list in case something has changed recently. Is anyone aware of an
IAX provider that offers incoming CallerID name?
Is there a technical limitation
within IAX which is preventing
> I am pretty sure that there are no IAX providers that offer
> CallerID name but wanted to double check with the list in
> case something has changed recently. Is anyone aware of an
> IAX provider that offers incoming CallerID name?
Xetricom Networks, who only have Toronto DIDs, do provide inco
Peter Loron wrote:
Greetings. I did some digging with Google, the wiki, and on the
archives, but didn't find a recent conclusive answer. If this is
answered in the wiki or archives somewhere, please point me to it.
I'm in the process of setting up an Asterisk box for home use. I've
got a X100P
Nabeel Jafferali wrote:
Xetricom Networks, who only have Toronto DIDs, do provide incoming
CallerID name using IAX.
We also provide Calling Name delivery for our DIDs. It's not an issue of
the VOIP protocol in use (it can be done over SIP or IAX), it's an issue
of what sort of PSTN connectivity t
I guess I should supply my current sip.conf file for net2phone.
[general]
;useragent = X-Lite release 1103m
useragent = "Cisco ATA186"
register => :@sip.net2phone.com
[net2phone]
type = peer
host = sip.net2phone.com
username =
secret =
fromuser =
fromdomain = net2phone.com
insecure = very
canrei
On Sat, Mar 19, 2005 at 09:47:49PM -0700, Tim Pushor wrote:
> Hi,
>
> Is this a silly question? I am trying to come up with an elegant way of
> joining a few small * servers in a peer to peer arrangement, and I am
> just curious as to what algorithm * uses to determine which channel (and
> ther
Hi all,
I am wondering if chan_oh323 or chan_h323 supports NAT traversal the
following setup:
H323 phone -> Asterisk ---> NAT router -> H323 gateway -> PSTN
I am trying to register a H323 gateway through a NAT to Asterisk for
outgoing calls to PSTN.
How can I achieve the above?
More of a case that in many cases the voip carrier would have to do
lookups for CNAM from either their telco or an external CNAM service.
These tend to carry an extra cost so that's why it's not wide spread
on dids via VOIP.
-- William
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This is possible. But success depends also on whether the router can do port
forwarding and whether the H323 Gateway supports NAT.
This is possible with Quintum for instance with some port forwarding rules on
router level.
Selon VoIP Newbie <[EMAIL PROTECTED]>:
> Hi all,
>
> I am wondering
I have 2 X100p clones that do not work in the net4801. The 4801 will not
even power up with them installed. Both cards work fine a a standard
desktop PC.
On Sun, 2005-03-20 at 21:58 -1000, John Breeden wrote:
> X100P is 3.3v not 5v, at least the one I have. Works fine in a 4801.
>
> John Simon w
It never dials the other number and instead goes straight into voicemail.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Sunday, March 20, 2005 8:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-U
Hello,
Can anyone point me to any documentation with regards to using
sip_friends on astcc. astcc already working on our test * server but im
trying to figure out how to sql-ize sip user config. I have thought of
using Asterisk Realtime but is not yet available on stable release.
Appreciate any po
Hi
I believe this is due to the way US phone systems work, however I'm going to
ask anyway. In the UK there are several providers who provide national rate
PSTN => Voip gateways which are free to receive calls on, (for the
recipient), the caller pays the cost of calling. E.g 0844 0870 etc.
I a
Personal opinion here, but in an office of 50 users we just built, here is
what we did. First, the PBX was equipped with 4 analog lines that were setup
as a failover in case the T1 for voice data failed. Secondly, another 4
analog lines were dropped in a central location of the office with analog
h
Thorben,
Please check the behaviour of a Park button.
If you do a vertical resize of a window (application) Park button gets
dislocated.
Ivan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Thorben Jensen
Sent: Monday, March 21, 2005 3:05 PM
To: 'Asterisk U
Matt wrote:
Where is my MoH class? I understand what's being done here... but I
don't see where that is.. like for meetme conferences, and being
placed on hold and such... which file?
"musiconhold.conf"
But once you get it going, it doesn't work anyway.
Would love to have someone prove me wrong.
voicepulse? We get free inbound on them. However, every once in a while
the service degrades for quite some time and they blame it on their
upstream provider; the issue just "goes away" without any real resolution.
Mark Charlton wrote:
Hi
I believe this is due to the way US phone systems work, h
You can roll your own -- for US numbers, my cid_rewrite agi-script does
this nicely:
http://muware.com/asterisk
-Original Message-
From: Keith O'Brien [mailto:[EMAIL PROTECTED]
Sent: Monday, March 21, 2005 8:29 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] CallerID Nam
> Is this supposed to be a joke?
Probably.
> It doesn't sound very professional.
Then change it -- all you need to do is re-record the greeting, or have
Allison do it.
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Someone may correct me, but in the US, the concept of "Calling Party Pays"
went away with early cell phones. Many companies opted for the "Calling
Parry Pays" plans so their clients could call them at will. However, callers
were unaware of the charge which created quite a fuss until the cell
provid
Get a "local" number for $3-$10/month with unlimited incoming minutes,
and the caller will surely pay the cost of the call (unless they're
"local"). If you have fewer than 100 minutes, go with a metered DID and
pay $1-$2/month plus around 1c/minute. Or go with Stanaphone or IPKall
and get a "free
This is the same experiance I had with my net4801 and X100P. Do you know
of any 3.3V PCI modems that will work with Asterisk?
--- Matt Ryanczak <[EMAIL PROTECTED]> wrote:
> I have 2 X100p clones that do not work in the
> net4801. The 4801 will not
> even power up with them installed. Both cards wo
The intel v90 56k pci modem, with the MD3200 chipset.
I'm using it.
John Simon wrote:
This is the same experiance I had with my net4801 and X100P. Do you know
of any 3.3V PCI modems that will work with Asterisk?
--- Matt Ryanczak <[EMAIL PROTECTED]> wrote:
I have 2 X100p clones that do not work in
Its a take-off from Nortel's 'Meridian Mail'. Personally, I think its very
funny, and its only your users who hear it, outside callers don't hear
anything except the greetings you record.
Jay Milk ([EMAIL PROTECTED]) wrote:
>
> > Is this supposed to be a joke?
>
> Probably.
>
> > It doesn't soun
Quoting Roger Gulbranson <[EMAIL PROTECTED]>:
> On Mon, 2005-03-21 at 19:03 +1100, Adam Goryachev wrote:
>
> > In case you need it, there are X servers available for MS Windows
> > platforms as well. Used to be one called exceed, but that was about 10
> > years ago, I just use linux on my desktop
Brian McCrary wrote:
Hello,
I'm wondering if anyone has benn able to successfully get g726-16
passthrouhg to work? I am wanting to use this codec instead of g729 as
I'm running out of DSPs using a high complexity codec on the Ciscos. I
would think it would work just as g729 does, which has been w
Hi all
Has anyone managed to send an outgoing call using
asterisk-h323 and successfully sent the H323_id ?
Sam
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Plus if you send your users to VoicemailMain(${CALLERIDNUM}) they don't hear
it at all.
They just get "enter password".
My 2c
Mark
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick
Sent: 21 March 2005 16:05
To: Asterisk Users Mailing List -Non-
Hi Ken, This has worked fine for me for about 6 months, maybe I just
didn't notice a problem. As far as I know there has been music playing when
people are being put on hold every time.
- Original Message -
From: "Ken Godee" <[EMAIL PROTECTED]>
To: "Matt" <[EMAIL PROTECTED]>; "Asteris
On Sun, 20 Mar 2005 16:36:08 -0800, Kerry Garrison
<[EMAIL PROTECTED]> wrote:
> I am trying to implement a follow-me script
> (http://www.voip-info.org/wiki-Asterisk+Tips+follow+me) but I am having a
> brain fart as I haven't a clue where to get started with what to do with
> this.
Kerry,
I'm mo
> -Original Message-
> From: Mark Charlton [mailto:[EMAIL PROTECTED]
> Plus if you send your users to VoicemailMain(${CALLERIDNUM})
> they don't hear
> it at all.
> They just get "enter password".
Yup. If you do that, the only time they hear it is during the initial setup
call (if you
On Mon, 2005-03-21 at 08:57 -0700, Tom wrote:
> We don't want to have to spend an extra 3 grand for another
> server just to take up more space when we have this box that is sitting here
> idle 99% of the time, and as it has worked spectacularly well with the wctdm
> cards, I don't see why it can'
Yep :)
Use a grandstream and [EMAIL PROTECTED] and you only need to push a single
button and go straight through to messages.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Charlton
Sent: Monday, March 21, 2005 11:18 AM
To: 'Asterisk
On Monday 21 March 2005 16:09, Tom wrote:
> Quoting Roger Gulbranson <[EMAIL PROTECTED]>:
> > On Mon, 2005-03-21 at 19:03 +1100, Adam Goryachev wrote:
> > > In case you need it, there are X servers available for MS Windows
> > > platforms as well. Used to be one called exceed, but that was about 10
David Brodbeck wrote:
-Original Message-
From: Mark Charlton [mailto:[EMAIL PROTECTED]
Plus if you send your users to VoicemailMain(${CALLERIDNUM})
they don't hear
it at all.
They just get "enter password".
Yup. If you do that, the only time they hear it is during the initial setup
cal
Argh. I should have known better.
Sorry,
Tim
Martijn van Oosterhout wrote:
On Sat, Mar 19, 2005 at 09:47:49PM -0700, Tim Pushor wrote:
Hi,
Is this a silly question? I am trying to come up with an elegant way of
joining a few small * servers in a peer to peer arrangement, and I am
just curious
I tried using the iLBC codec, and whlie I like it, I ran into a
strange issue. I did a few searches on Google and haven't found anyone
with the same issue as this.
Anyhow, I was using a Plantronics analog headset and box plugged into
a Digium TDM card, dialed out over my VoIP provider's IAX channe
I can't seem to get the message on hold class to work for anything but
default.. it works if I specify default but if I specify anything else
it hangs up on me:
== Spawn extension (from-internal, 9472, 3) exited non-zero on 'SIP/200-9f2c'
-- Executing Macro("SIP/200-9f2c", "hangupcall") in n
Thanks. Is there any native solution that is also cheap? I need it for
my small office with only a few staff. My H323 gateway is not even a
cisco one but costs only $200.
Thanks.
On Mon, 21 Mar 2005 15:57:13 +0100, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
>
> This is possible. But success de
THANKS! I had heard of that but couldn't find it. I love that whisper
feature.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Roy
Sent: Monday, March 21, 2005 8:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Henry Devito wrote:
Hi Ken, This has worked fine for me for about 6 months, maybe I just
didn't notice a problem. As far as I know there has been music playing
when people are being put on hold every time.
Ah but you might want to take a closer look.
If you can, watch the active connections on
It means the caller hung up in the middle of the voicemail app.
Anton Krall wrote:
So far, nobody has been able to tell us what this error means.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of MF Hulber
Sent: Lunes, 21 de Marzo de 2005 02:54 a.m.
To: Ast
Quoting Roger Gulbranson <[EMAIL PROTECTED]>:
> On Mon, 2005-03-21 at 08:57 -0700, Tom wrote:
>
> > We don't want to have to spend an extra 3 grand for another
> > server just to take up more space when we have this box that is sitting
> here
> > idle 99% of the time, and as it has worked spectacu
Yeah I got it to work... but I can't get the set command to work..
like when I try to set the hold music class it just does nothing and
then when I do musiconhold() it hangsup!
On Mon, 21 Mar 2005 10:19:13 -0600, Henry Devito <[EMAIL PROTECTED]> wrote:
> Hi Ken, This has worked fine for me for a
Is it possible to modify the caller id on the phone during a call (session) ?
If not does anybody know with which SIP request this could be handled ?
I'm know investigating RFC3311 which seems to offer an answer but if somebody
already has an answer ...
Michael
Michael Devenijn wrote:
Is it possible to modify the caller id on the phone during a call (session) ?
If not does anybody know with which SIP request this could be handled ?
Do you mean what is displayed on the phone's display? If so, yes, with
some phones this is possible, by performing a re-INVI
Strange;
It works for me. The x100p (Digium 100 buck model) I have is slotted for
3.3v and works fine.
I'm running gentoo with udev and the 2.6.11 kernel with soekris patches
(udev is cool, coldplug automagically loads the drivers).
The 4801 is flashed with whatever the latest bios is from Sori
[EMAIL PROTECTED] wrote:
> Strange;
>
> It works for me. The x100p (Digium 100 buck model) I have is slotted
> for
> 3.3v and works fine.
>
> I'm running gentoo with udev and the 2.6.11 kernel with soekris
> patches (udev is cool, coldplug automagically loads the drivers).
>
> The 4801 is flash
After 4 hours of debugging codecs, changing config files, etc. as a result of
not being able to capture voice from a Cisco 7960, I eventually found that the
mic in the handset appears to be dead.
Does anyone know where I can get a new handset (just the part you hold to your
head, everything else o
Dear Sirs
We are doing some tests in our lab with Digium/Asterisk boards and we
have some doubts regarding Asterisk´s SIP server configuration, could you help
us please?
See attached our topology.
Thanks in advance.
Best regards
Marcia
<>
Conf
On Mon, 21 Mar 2005, Roger Gulbranson wrote:
> On Mon, 2005-03-21 at 08:57 -0700, Tom wrote:
>
> > We don't want to have to spend an extra 3 grand for another
> > server just to take up more space when we have this box that is sitting here
> > idle 99% of the time, and as it has worked spectacula
Really? I just tried it and WHEN it's working.. it is streaming..
and even when I hang up it keeps mpg123 up and running in the
background.
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On Mon, 21 Mar 2005 08:27:17 -0500, Steve Clark <[EMAIL PROTECTED]> wrote:
> It doesn't sound very professional.
>
> comedian
> n 1: a professional performer
What's not professional about that? :)
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Would anyone know whether a jabber module would be in development for
asterisk? What I'm looking for is something like the SER module that's
out there, with capabilities to send SMS messages from jabber to a phone
connected to the system.
Aaron Daniel
SHSU Computer Services
[EMAIL PROTECTED]
_
Alessandra Grasso wrote:
My objective is to estimate the performances of *
How much the trancoded can influence the performances?
Thanks,
show translation recalc 30
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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A
I am setting up a new asterisk based call center. I just read:
http://www.voip-info.org/wiki-IAX+versus+SIP
After reading this and other google results for "IAX vs SIP" is there
any reason why i should use SIP anywhere !!
t
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Sys Admin wrote:
After reading this and other google results for "IAX vs SIP" is there
any reason why i should use SIP anywhere !!
Well, let's see.. 99.99% of the available VOIP hardware only support
SIP, MGCP and H.323, but not IAX2. Is that a good reason?
IAX2 calls between servers carry the si
On Mon, 21 Mar 2005 10:01:04 -0800, Sys Admin <[EMAIL PROTECTED]> wrote:
> I am setting up a new asterisk based call center. I just read:
> http://www.voip-info.org/wiki-IAX+versus+SIP
>
> After reading this and other google results for "IAX vs SIP" is there
> any reason why i should use SIP anywh
On Mon, 2005-03-21 at 08:57 -0700, Tom wrote:
> This box never was primarily an * box, it is a server that people have used
> VNC
> from windows desktops to run a couple of apps that are X11 only that we need
> in
> house. We just have been trying to get off of our old PBX, and onto * as our
> p
Sys Admin wrote:
I am setting up a new asterisk based call center. I just read:
http://www.voip-info.org/wiki-IAX+versus+SIP
After reading this and other google results for "IAX vs SIP" is there
any reason why i should use SIP anywhere !!
Because most equipment doesn't support IAX
--
Always do righ
well that was my another question I was thinking of starting one more
thread for that:
"Hard phone vs soft phone"
Forget older years but in 2005 do hard phones really add any value
over softphones.
The call center agents already have p4 2.4ghz with 512 MB ram Win2K
why not just get them a nice U
an update: since it might help others
I did the same make on another machine and it worked fine. So it seems
to be a problem with my tool-chain
t
On Sun, 20 Mar 2005 22:18:50 -0800, Sys Admin <[EMAIL PROTECTED]> wrote:
> I am trying to install asterisk on fedora core 3 these are the steps i too
On March 21, 2005 01:07 pm, Kevin P. Fleming wrote:
> Well, let's see.. 99.99% of the available VOIP hardware only support
> SIP, MGCP and H.323, but not IAX2. Is that a good reason?
No. 95% of the marketplaces uses Windows. Drive the marketplace to use
better protocols.
> IAX2 calls between s
Friday and Saturday I was wrestling with a VoIP system that was having very
strange problems.
It was playing the outgoing IVR audio at 2-5x faster than it should have been.
I found that if I stopped asterisk, removed the wcfxs driver and installed the
ztdummy driver, the audio would play fine.
I
Make sure you are feeding your 4801 enough "juice" to support the pci
card, I'm currently feeding 15v @ 1A.
I got the x100p directly from Digium a few years ago, at the time it was
a component of Digium's "developer" package.
lspci identifies the card as a "Tiger Jet Network Inc. Tiger3XX
Mode
can any body refere me please to a callingcard
application that has user's manual or some clear
documentation.
i have installed areskicc and i have been struggling
to make it work. i want to try something else.
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