[ replying to myself ]
> freebsd just upgraded to 1.01, thanks sobomax and team!
> with the upgrade, on inbound from an spa3k pstn call,
> i started getting the classic
> "Failed to authenticate user "
> when the authenticating client should have been the
> spa3k/pstn/userid
> i can get around
Amen.
Oops, did I just top post? How terribly lazy of me.
But I digress.
Few will disagree that the careful application of netiquette will be a
benefit to any newsgroup/mailing list/board; and top posting is
something that should be used sparingly. Nevertheless, top posting is
not the horrid cri
Hello, been digging through the archive and the Wiki and it looks like this
phone I bought just can't be configured to work behind a NAT.
Just wanted to check one last time before sending it back if anyone has had
any luck with this.
It's pretty useless to me if it can't work behind a NAT.
Than
Rather than use AgentCallbackLogin, try AddQueueMember and
RemoveQueueMember.
On Oct 21, 2004, at 1:22 PM, Jolan Luff wrote:
Hi,
I am trying to setup two extensions by which agents can automatically
login and logoff with asterisk-1.0.0. My extensions.conf looks like
this:
exten => *904,1,AgentCa
On 07:24 PM 10/22/2004, Mark Phillips wrote:
>Firstly, I have a PRI which will be connected to the card. How does *
>know that a call for 732 111 3714 should get routed to extn 3714?
Simple enough, your PRI provider will send some portion of the number to
you, many times it's the last 4 digits, so
Mike Boger Jr [EMAIL PROTECTED] lazily top-posted:
>
> There seem to be alot of folks who "lazily top post" and therefore don't
> meet your criteria of acceptable asterisk users mailing list posters.
> Maybe all the $0.02 donations could be used to fund a manual on what the
> acceptable Kevin Wals
Maxim Litnitsky wrote:
Thanks you very much, I'll try to do that.
Any more advises before I start my way? :)
Yeah... based on your questions, you have a lot of learning ahead of you
:-) Please make sure you spend as much time as possible using the
mailing list archives, the Wiki and Google searc
freebsd just upgraded to 1.01, thanks sobomax and team!
with the upgrade, on inbound from an spa3k pstn call,
i started getting the classic
"Failed to authenticate user "
when the authenticating client should have been the
spa3k/pstn/userid
i can get around this by setting the spa3k
PST
Are there any changes to chan_sip since 09/16/04 in the
stable branch that could affect the way Asterisk issues an ACK?
The reason I ask…I have a product by INGATE called the
Siparator which assists in NAT traversal. It worked great until I upgraded to
Asterisk v1.0. After comparing th
Flamebait,
There seem to be alot of folks who "lazily top post" and therefore don't
meet your criteria of acceptable asterisk users mailing list posters. Maybe
all the $0.02 donations could be used to fund a manual on what the
acceptable Kevin Walsh method of replying to asterisk user messages i
Does anyone ever tried to connect Asterisk to the public network using GSM
mobile phones connected through the USB (or Serial) port on the PC?
Is it possible to connect Asterisk to the PSTN with a GSM/GPRS card (PCI or
PCMCIA) connected to the PC?
In case it's possible, would I also be able to ma
Hi Folks,
I'm about to install my first quad T1 card. I don't foresee any issues
with compiling the software but I'm looking for advice on how to handle
a few things.
Firstly, I have a PRI which will be connected to the card. How does *
know that a call for 732 111 3714 should get routed to extn
I am trying to connect asterisk to a H323 gateway , the call rings and
connects perfectly , but there seems to be no audio. It gives out
this error:
Invalid format of RTP addresses.
Please let me know, what may go wrong here.
Thanks.
Ehsanul Karim
__
Hey Jay
All the stuff you've described is possible. I've done some playing with the
XML services on a Cisco 7960 to give ACD queue stats and system uptime info.
The phone has a mini web browser built in so it's pretty easy to knock up
some glue scripts in the back end to do what you want to do. I
Okay, it turns out I was being stupid. ZapRAS does not automatically
answer the channel, so, the PRI_NET side below should have read:
PRI_NET side
--
extensions.conf:
[incoming]
exten => 6999,1,Answer
exten =>
6999,2,ZapRas(debug|64000|noauth|multilink|netmask|255.255.255.0|192.
Linux and Windows compiles of today's CVS are posted on
http://iaxclient.sf.net/iaxcomm/index.html
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Many ATA's have a build in transfer feature. Most of them are probably
more complicated than the ## transfer however.
I know on my Sipura, I can hit flash, then hit *98 + number to do a
blind transfer, and it seems to work fine. Not exactly intuitive
however.
Brian
On Fri, 22 Oct 2004 14:12:19
Any tips, tricks or treats out there? I'm building a new system and
would like to move away from my SQL based call rating solution...
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Brian West wrote:
www.bkw.org/load.txt
thank pfn for that one.
bkw
Brian,
Thanks for the link, and if it weren't for your and Josh's presentation
at Astricon res_config wouldn't have even crossed my mind. It's always
nice for an "authority" to answer your posts!
Works perfectly!
--
Kristian Ki
On Fri, 22 Oct 2004 16:07:08 -0600, [EMAIL PROTECTED]
<[EMAIL PROTECTED]> wrote:
>
> Do you have a list of those providers that use IAX?
check the Wiki ...
http://www.voip-info.org/tiki-index.php?page=VOIP%20Service%20Providers
in addition to those entries which mention IAX there are also some
[EMAIL PROTECTED] wrote:
Do you have a list of those providers that use IAX?
http://www.voip-info.org/tiki-index.php?page=VOIP+Service+Providers
is a good starting point..
Try a search on google, you would be surprised on how many of these will
pop...
-Original Message-
From: [EMAIL PROT
[EMAIL PROTECTED] wrote:
> Do you have a list of those providers that use IAX?
Check the: Asterisk to/from PSTN services section on the wiki page:
http://www.voip-info.org/wiki-VOIP+Service+Providers
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf
I've been working on the asterisk manager for a few days, today I
started on a little prototype click to talk on the web via php thing. I
have it working properly except for one thing. I do a SetVar:
CTTN= over the socket and when the call gets to the socket
its empty. Is this intended? Is it su
Do you have a list of those providers that use IAX?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benjamin
on Asterisk Mailing Lists
Sent: Friday, October 22, 2004 4:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asteris
tks
I went through the debug, and what was happening was that when the called
party did not answer I was getting a ACK sent from the caller (for some
reason...i will try to work out why Xlite is doing this), anyhow in my
routing logic I had no match for the ACK, however if I enter that
parameter
On Fri, 22 Oct 2004 23:39:56 +0200, Stewart Nelson <[EMAIL PROTECTED]> wrote:
> I would appreciate your opinions on the feasibility of these
> techniques, and also about any other methods that have been
> tried to achieve direct SIP connectivity.
If you are that desperate to use Vonage, then why d
Brian West wrote:
> http://bugs.digium.com/bug_view_page.php?bug_id=0002639
>
> There use that.. then pounce on kram to get it in CVS!!
>
> bkw
Yeah.. This works like a charm... Thanks to anthm :)
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htt
http://bugs.digium.com/bug_view_page.php?bug_id=0002639
There use that.. then pounce on kram to get it in CVS!!
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Ryan Courtnage
> Sent: Friday, October 22, 2004 3:24 PM
> To: Asteri
On Fri, 22 Oct 2004 17:07:11 -0300, Damian Rosales <[EMAIL PROTECTED]> wrote:
> The best explanation I could find whas that a standard modem is half-duplex
> and the connection needs to be full-duplex. Is that all? I wouldn't mind
> having to talk in a half-duplex mode if that's the big issue.
Do
www.bkw.org/load.txt
thank pfn for that one.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Kristian Kielhofner
> Sent: Friday, October 22, 2004 4:03 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [
JOAO CARLOS MOURA wrote:
Thank you Michael,
I tried to use RFC2838 without success. Which another type?
Which endpoints (SIP Phones, ATA, ???) are we talking about ?
You need to match the configuration on the end-point, it may seem
obvious, but if you leave your IP phone doing dtmf inband, while
I presently have a small VoIP network using H.323 and gnugk,
and would like to upgrade it to an Asterisk-based system,
primarily to take advantage of low cost unlimited calling
plans offered by SIP providers such as Vonage. However, the
carriers with good reputations for reliability and quality
se
Look in /usr/src/asterisk/configs/sip.conf.sample for a list of all legal
options.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of JOAO CARLOS MOURA
> Sent: Friday, October 22, 2004 1:24 PM
> To: Asterisk Users Mailing List - Non-Comm
- Original Message -
From: "Kevin Walsh" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]>
Sent: Friday, October 22, 2004 3:52 PM
Subject: RE: [Asterisk-Users] G.729 licensing/patent?
| Benjamin on Asterisk Mailing Lists [EMAIL PROTECT
On Fri, 22 Oct 2004 20:52:40 +0100, Kevin Walsh <[EMAIL PROTECTED]> wrote:
> Your views are biased upon the ridiculous legal system used in the US
Well, I have had no exposure to the legal system in the US, since I
never lived there. I did live in the UK though and I did have exposure
to the legal
Benjamin on Asterisk Mailing Lists wrote:
On Fri, 22 Oct 2004 13:10:38 -0600, Joseph <[EMAIL PROTECTED]> wrote:
How could I test if my ports 1 to 2 are open on my firewall?
nmap is your friend
Or alternatively (if you don't have an outside machine to test from) you
could go to http://www.a
Thank you Michael,
I tried to use RFC2838 without success. Which another type?
Thank you
jmoura
- Original Message -
From: "Michael Bielicki" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Friday, October 22, 2004 6:14 PM
Subjec
Okay, I know there have been previous posts about this, and there is the
patch 110 on mantis that was never added to CVS.
If you have a mixed environment of SIP based phones and ATA adapters, how
can you still allow the ATA style phones to use the transfer function, yet
allow all dtmf to be pa
Hello,
I am just getting started with res_config and ODBC. I have MySQL all
setup and am filling it with my data. Everything seems very straight
forward. One thing catches me so far:
1) How are register lines in sip.conf and iax.conf represented?
i.e. register=> username:[EMAIL PROTEC
Hello all,
we have been random freezes in two asterisk systems more or less once a
day last mounth.
In both system we are using a T100P card and a channel bank for local
extensions (fxs) and h323 or sip (we have tried both with same results)
voip gateways for FXO connectivity.
We used to have so
i had this problem last night. sometimes it would work find and then i
would get errors or timeouts???
- hcir
On Oct 22, 2004, at 9:07 AM, pixelFiend wrote:
Hello,
I'm having problems connecting to other * boxes through IAXTel. I've
seen this addressed in the list archives, and other places on th
Thank you everyone for your responses. Pretty much we're just looking
for a way to bring in some lines with room for expansion to an aserisk
server. We thought a T1 with a wildcard would be a good route since we
could just have channels turned on as necessary. From everyone's
comments it's soun
On Fri, 2004-22-10 at 16:05 -0400, Kanwar Ranbir Sandhu wrote:
> On Fri, 2004-10-22 at 05:56, Manfred Petz wrote:
> [snip]
> > Is there a way to force MusicOnHold() to be restarted from the beginning for
> > each call which has been answered?
> [snip]
>
> Why? What would be the point?
off the t
Inband dtmf only works on alaw/ulaw. Use any other mode and it should work
On Fri, 22 Oct 2004 17:02:00 -0200, JOAO CARLOS MOURA
<[EMAIL PROTECTED]> wrote:
>
> I just installed G729a my Asterisk. I am
> facing some problems on DTMFMODE=INBAND. I just can t transfer my calls. Is
> there anybody
Going back to your first comment "...Since I couldn't find any cheap modem
that would help me, ..."
For what I've been reading it is not possible to use a standard modem (US
Robotics or anything like that) to connect * to an analog line. But can
anyone explain why? Still don't get it.
Isn't there
Neil Cherry wrote:
I never could get my time server to work with the GS.
The following is an excerpt from my web page. I also use phones on a
private subnet. Linux is RedHat. This works for me:
The time service must be configured to allow the phones to request the
time from your server, which mu
- Original Message -
From: "Daniel Daley" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, October 21, 2004 5:49 PM
Subject: [Asterisk-Users] Digium Wildcard T1 Compatibility
| Hi,
|
| I have a quick question about the T100P. I've used the card before in a
| PRI setup and it
Well,
Kevin I agree .. but we really had this topic not too long ago ...
--snip--
Nevertheless, it's worth to visit
http://www.nosoftwarepatents.com/
and to think about the (mostly bad) consequences of software patents ...
And see this example
http://webshop.ffii.org/
it will open your eyes ...
I t
On Fri, 2004-10-22 at 05:56, Manfred Petz wrote:
[snip]
> Is there a way to force MusicOnHold() to be restarted from the beginning for
> each call which has been answered?
[snip]
Why? What would be the point?
Ranbir
--
Ranbir
Systems Aligned Inc.
www.systemsaligned.com
__
I just installed G729a my Asterisk. I am facing some problems on
DTMFMODE=INBAND. I just can t transfer my calls. Is there anybody out there
who could give me WHICH DTMFMODE to use?P.S -> I already tried
DTMFMODE = RFC2833 and did not work!Thank you.Jmoura
__
Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] wrote:
> > Now, if someone had designed a chip that infringed the patents
> > then the registered patents would be enforceable. This is the real
> > reason for using "an apparatus" in the claims. In countries that
> > don't allow software and m
Jay Milk [EMAIL PROTECTED] lazily top-posted:
> Maybe you should leave the DIGIUM SPONSORED list then? And will you
> stop wasting everyone's time with your pseudo-political ramblings and
> infantile signature, while you're at it? Sheesh!
>
If you feel the need to dispute the clarification I made
To helpful link!. It answer lot of my question.
Thank you very much "Asterisk in Yahoo"...
Nahuel Ramos.
On Fri, 22 Oct 2004 11:46:46 -0700 (PDT), Asterisk .
<[EMAIL PROTECTED]> wrote:
> --- Nahuel Alejandro Ramos <[EMAIL PROTECTED]> wrote:
>
> > Performace: CPU & MEM per Call.
> > I
On Fri, 22 Oct 2004 14:12:19 -0500, Kristian Kielhofner <[EMAIL PROTECTED]> wrote:
>
> Kent,
>
> Where is this documented?
There is a provisioning document available from Sipura. Now that I
have looked on their web site that document does not appear to be
available unless you have a supp
On Fri, 22 Oct 2004 13:10:38 -0600, Joseph <[EMAIL PROTECTED]> wrote:
> How could I test if my ports 1 to 2 are open on my firewall?
nmap is your friend
rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.
NB: Spam filters in place. Messages u
On Fri, 22 Oct 2004 18:19:33 +0100, Kevin Walsh <[EMAIL PROTECTED]> wrote:
> Basically, a patent could be granted in the USA and, by way of
> treaty, be registered in all signatory countries. The patent may
> not actually be valid in all of the countries mentioned in the patent,
> depending upon t
Kent wrote:
On Fri, 01 Oct 2004 23:16:45 -0500, Kristian Kielhofner <[EMAIL PROTECTED]> wrote:
Kristian Kielhofner wrote:
They do, but the format of that file has to be generated with Sipura's
proprietary config tool. Currently NOT available for anyone that Sipura
doesn't want to have it.
Actuall
How could I test if my ports 1 to 2 are open on my firewall?
--
#Joseph
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On Fri, 01 Oct 2004 23:16:45 -0500, Kristian Kielhofner <[EMAIL PROTECTED]> wrote:
> Kristian Kielhofner wrote:
>
> They do, but the format of that file has to be generated with Sipura's
> proprietary config tool. Currently NOT available for anyone that Sipura
> doesn't want to have it.
>
Actual
Thanks you very much, I'll try to do that.
Any more advises before I start my way? :)
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usedcanon wrote:
>>Hi All,
>>
>>Is it possible to detect an incomming fax just as it is possible with
Answer
>>on a Zap channel. If not do others find the possibility of this
enhancement
>>useful too?
>>
>>
>Detecting that an incoming is a FAX has been present in * since its
>early days.
>
>Regard
--- Nahuel Alejandro Ramos <[EMAIL PROTECTED]> wrote:
> Performace: CPU & MEM per Call.
> I will use SER to route call but I want to know if it is better to
> register SIP clients on SER or Asterisk?
SER. You will get all the functionalities of a SIP Proxy there.
This link might be helpful: http:
On Fri, 22 Oct 2004 18:06:27 +0200, Yves-Marie CRABBE <[EMAIL PROTECTED]> wrote:
> I tried to define a disconnect tone description this way :
> [EMAIL PROTECTED],[EMAIL PROTECTED];2(.5/.5/1+2)
> I'm located in France.
Try the localisation wizard on Voxilla.com. France should be on the list.
rgds
Maxim Litnitsky wrote:
Ok, clear about PPP.
Why did u say about 20 channels? I have E1 and 30 channels.
I would like to have 2 ip links on 20 timeslots and 10 ZAP channels.
Is it possible?
Sure, if your provider will provision two separate IP links for you, 10
channels each, then the Zaptel and ke
Ok, clear about PPP.
Why did u say about 20 channels? I have E1 and 30 channels.
I would like to have 2 ip links on 20 timeslots and 10 ZAP channels.
Is it possible?
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Maybe you should leave the DIGIUM SPONSORED list then? And will you
stop wasting everyone's time with your pseudo-political ramblings and
infantile signature, while you're at it? Sheesh!
> -Original Message-
> From: Kevin Walsh [mailto:[EMAIL PROTECTED]
> Sent: Friday, October 22, 2004
Performace: CPU & MEM per Call.
I will use SER to route call but I want to know if it is better to
register SIP clients on SER or Asterisk?
Thanks.
On Fri, 22 Oct 2004 10:24:17 -0700 (PDT), Asterisk .
<[EMAIL PROTECTED]> wrote:
>
> --- Nahuel Alejandro Ramos <[EMAIL PROTECTED]> wrote:
> > Can I
Since this is for my home network, I suspect I can just assign an IP to the
one or two GS phones to error on the safe side.
Of the GS users, is there any difference between the GS 102 and 102D?
-Dave
-Original Message-
From: Neil Cherry [mailto:[EMAIL PROTECTED]
Sent: Friday, October 22
Maxim Litnitsky wrote:
Hi all, my provider offers me Internet and CO lines within one E1
channel. Can I split time slots so that 10 can be used as ZAP channels
for inbound/outbound calls, 10 for internet office needs (can it be
10 multilink PPP???), 10 time slots for different internet channels
f
David Ishmael wrote:
I think my Netgear router will try to lease the same DHCP address to a
device based on MAC automatically each time the device queries for an
address (but I'm not 100% sure about that, never really watched it). So the
problem is with the address changing?
I can't infer that fro
Title: [Asterisk-Users] answer on # key?
Did you see the code I posted a day or two
ago, it should do exactly what you want by running a macro before the calls are
joined. However I tested it on Zap so I have no idea if it works between other
mediums, please check it out and let me know.
I think my Netgear router will try to lease the same DHCP address to a
device based on MAC automatically each time the device queries for an
address (but I'm not 100% sure about that, never really watched it). So the
problem is with the address changing?
-Dave
-Original Message-
From: Ne
Hi all, my provider offers me Internet and CO lines within one E1
channel. Can I split time slots so that 10 can be used as ZAP channels
for inbound/outbound calls, 10 for internet office needs (can it be
10 multilink PPP???), 10 time slots for different internet channels
for VoIP. Gurus please a
David Ishmael wrote:
I was considering a GS phone (102 or 102D), what version of the GS are you
using? Do all GS phones have issues with DHCP? I use DHCP on my network so
I want to make sure I understand potential issues before making any
purchases.
I have my GS101 working with DHCP, I setup my d
options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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>
>This message was checked by NOD32 antivirus system.
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>
>
>
>
>
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Hi all,
I am trying to set up a way to have an incoming SIP call being
transfered to another sip number.
However.. when someone calls the first sip number, it will ring the
other sip extension, but no sound is passed through. It's
almost like the incoming call is put on hold, the other number is
Steve Kann [EMAIL PROTECTED] wrote:
> Stefan de Konink wrote:
> >
> > Isn't this an opportunity for Digium to offer encoded G729 files for a
> > fixed price directly encoded from the original wav files?
> >
> I think this is an opportunity for people to use unencumbered codecs..
>
> If even just
Amen
On Oct 22, 2004, at 1:26 PM, Kevin Walsh wrote:
Kanuri, Seshu (Company IT) [EMAIL PROTECTED] lazily
top-posted:
Just my $0.02 Cents
I propose that an Asterisk development fund be set up to hold all of
these $0.02 donations. People who are not quite as cheap could donate
a little bit more.
--
Kanuri, Seshu (Company IT) [EMAIL PROTECTED] lazily top-posted:
> Just my $0.02 Cents
>
I propose that an Asterisk development fund be set up to hold all of
these $0.02 donations. People who are not quite as cheap could donate
a little bit more.
--
_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_
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--- Nahuel Alejandro Ramos <[EMAIL PROTECTED]> wrote:
> Can I post this again? :)
> Where do I register the SIP clients for more performance? (Asterisk or SER)
> Thank you very much.
Performance? If you mean scalability, then SER as it does'nt handle media.
Regards, Girish
Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] wrote:
> The facts are this:
>
> A patent for which a letter of patent has been issued is legally in
> force. Period.
>
> A patent issued in the US may directly or indirectly be in force in
> other countries depending on various treaties or bil
On Oct 22, 2004, at 9:15 AM, [EMAIL PROTECTED] wrote:
Hello all. I am new to the list and after doing some research on
"Asterisk" this
week I would like to get started testing.
I currently have an unused ISDN (BRI) line that I was thinking about
cancelling until I
learned of Asterisk. I thought
Hello,
I'm having problems connecting to other * boxes through IAXTel. I've
seen this addressed in the list archives, and other places on the web,
but haven't seen that anyone has come up with a solution. I'm dialing
in to my Asterisk server using DISA, authenticating OK, then
attempting to dial o
Crap, I already ordered the X100P.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Crown
Sent: Friday, October 22, 2004 12:35 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Hardware Recommendations
D
David,
I would suggest you consider a TDM400 Card with 1 FXO and 1 FXS module.
This will handle all of your requirements with a single card, and allow you
to add two more ports later.
http://www.thevoipconnection.com/store/catalog/product_16185_Digium_TDM400P_
FXO_FXS_Interface_Card.html
It actu
Doh haha you already figured that out. :P
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Robert Jackson
> Sent: Friday, October 22, 2004 10:01 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Aste
Don't put .pub on the inkey/outkey just key
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Robert Jackson
> Sent: Thursday, October 21, 2004 11:39 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [
I know the compatibility of various IP phones (Cisco, Snom, Polycomm...)
has been discussed at length. What I'd like to know is how useful the
"pixel-based" screens on some of those phones are. Can I create
interactive applications, informative screens, feed still pictures (even
B/W), etc...? I
Can I post this again? :)
Where do I register the SIP clients for more performance? (Asterisk or SER)
Thank you very much.
Nahuel Ramos.
On Fri, 22 Oct 2004 02:53:36 -0700 (PDT), Asterisk .
<[EMAIL PROTECTED]> wrote:
> Hi,
>
> --- Iqbal <[EMAIL PROTECTED]> wrote:
> > asterisk would only
Ronald Wiplinger wrote:
I would like to setup Asterisk PBX for following purposes:
1. dial-in from Europe via sipgate (Germany & UK)
Easy, I think.
2. dial-in from Canada via primus
Not quite so easy, you might want to look at some other options than
Primus unless you have a specific reason
On Fri, 2004-10-22 at 11:47, Ben Merrills wrote:
> Is anyone planning on patching chan_agent.c to reflect the new transfer
> method (using the patch linked below)?
>
> I had a stab at it, but my c skills are next to none :)
>
> Cheers,
>
> Ben Merrills
I would love to see that added.
Will it
Hello everybody,
I'm a new user of * and I just bought a Sipura SPA-3000 to make a home voip
installation.
I actually have a problem when a PSTN user calls and hangs up. The
disconnect tone is not
detected by the SPA, the the call continues and, for example, leaves an
empty message on
the voicemail
I was considering a GS phone (102 or 102D), what version of the GS are you
using? Do all GS phones have issues with DHCP? I use DHCP on my network so
I want to make sure I understand potential issues before making any
purchases.
Also, does anyone know of any wireless SIP phones?
Asterisk/VoIP N
Todd Routhier - Lightwave Technologies, LLC. wrote:
OK, this is a different flashing issue than the one that's being talked
about.
I have a few of these phones (GrandStream 101) and when a voicemail is
received the light on the LED starts blinking and the dial tine stutters,
this is cool. BUT.
Is anyone planning on patching chan_agent.c to reflect the new transfer
method (using the patch linked below)?
I had a stab at it, but my c skills are next to none :)
Cheers,
Ben Merrills
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Jackson
S
Matthew Crocker wrote:
The info sheet at Verisign says their SIP-7 product does the MGCP with
my AS5400 and SIP with a sip server (Asterisk). An inbound call would
generate a SS7 ISUP message to Verisign. They would send a SIP message
to Asterisk. Asterisk would respond with a SIP message bac
> -Original Message-
> From: Joseph [mailto:[EMAIL PROTECTED]
> Sent: Friday, October 22, 2004 11:22 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Queue / Agent Problem
>
>
> But if the agent does a consultive transfer, the queue system
> thinks the agent still has the call a
You will be able to make only one connection (in or out) over a POTS
line with the X100P. This is true with the TDM card as well keeping in
mind that every FXO card provides connectivity to ones POTS line.
This brings up the quesiton of whether to buy the TDM card with a FXO
and FXS module.
One at a time, as X100P is to be connected to a single PSTN phone line
with a RJ-11.
christophe de coninck wrote:
Hey,
I knew that info already but the question i ment to ask was: how many
calls will I be able to make to the outside from my asterisk server with
one X100P card, only one at a tim
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