On Thu, 24 Aug 2006, Jeremy McNamara wrote:
Rushowr wrote:
Hey all, I have an interesting issue that just recently started when I
grabbed a copy of the trunk about a week ago and compiled it. Ever since
that compile, if I start Asterisk (disconnected terminal, using
safe_asterisk to launch)
Hello everyone,
I have released AstLinux 0.4.3:
http://sourceforge.net/projects/astlinux/
For all of those that have been waiting to switch to 0.4.x, this is
your chance. The few remaining problems with uclibc have been fixed
(i.e. voicemail timezones and voicemail - email via
On Wed, Aug 23, 2006 at 11:03:23PM -0700, Steve Edwards wrote:
On Thu, 24 Aug 2006, Jeremy McNamara wrote:
Rushowr wrote:
Hey all, I have an interesting issue that just recently started when I
grabbed a copy of the trunk about a week ago and compiled it. Ever since
that compile, if I start
Hello,
If a call is connected to an AGI application, the application terminates if
the call is hang-up.
I have an AGI application in which I want to do a certain operation while
call hangs up. Is it possible for my application to know when a call is
hang-up? Is there any way to monitor this
Does anyone have any other tips.
use mISDN ;)
or are you bound to bristuff because you need speciall features of this?
KAi
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I'll run some more tests but it's not very different from the posting?
Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent: 22 August 2006 18:22
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Hi,
I have set some variable in a call.
Set(testmode=1)
For some reason, such as forward the call, the follow command called.
Dial(Local/1234567)
It will go through the dial plan again but the value of variable
testmode is nothing instead of 1.
How can I maintain the value of the variable in
or are you bound to bristuff because you need speciall features of this?Well you are right. Bristuff has more features than mISDN.After loading the florz patch messages in kernlog turn into this
Aug 24 10:18:08 asterisk kernel: zaphfc[0]: received d channel frame with bad CRC.Aug 24 10:20:45
In article [EMAIL PROTECTED],
unplug [EMAIL PROTECTED] wrote:
Hi,
I have set some variable in a call.
Set(testmode=1)
For some reason, such as forward the call, the follow command called.
Dial(Local/1234567)
It will go through the dial plan again but the value of variable
testmode
On 23 Aug 2006, at 23:07, Javier Lara Sanchez wrote:
Dear All,
I need to buid an IVR that could make a request to a data base
(oracle) in a remote host.
The idea is that an user dial a extension with 2 options and one
of them ask for a data (in the case a date). This data is the
H == Haspers [EMAIL PROTECTED] writes:
H We are using some E61 and E70's with asterisk. Only problem we have
H at this moment is that we are unable to use a password for the
H authentication. I haven't found out yet why this isn't working.
H They are working good, but I would like to see some
Hi Benny,The E61 handles this just fine. With SIP as the default channel to dial and no WiFi coverage, you get a message asking if you'd like to dial by cellular. Works nicely other than a few stability issues.
SimonOn 24 Aug 2006 11:23:39 +0200, Benny Amorsen [EMAIL PROTECTED] wrote:
H ==
Sorry for the duplicate post.
H == Haspers [EMAIL PROTECTED] writes:
H We are using some E61 and E70's with asterisk. Only problem we have
H at this moment is that we are unable to use a password for the
H authentication. I haven't found out yet why this isn't working.
H They are working good,
Do you mean it is a global variable instead of channel variable?
Try using Set(__testmode=1), but still refer to it as ${testmode}.
The __ tells Asterisk to propagate the variable to created channels.
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play:
Hi
I wish to setup asterisk for training purposes so that I am able to
listen in to an extension while a call is going on?
Has anyone done this?
Thanks
SP
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Strange,
What settings do you use? I followed this link
http://www.newlc.com/Using-SIP-with-Nokia-Series60-and.html but without any
luck.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Benny Amorsen
Sent: donderdag 24 augustus 2006 11:24
To:
Simon Woodhead wrote:
Hi Benny,
The E61 handles this just fine. With SIP as the default channel to
dial and no WiFi coverage, you get a message asking if you'd like to
dial by cellular. Works nicely other than a few stability issues.
The E60 appears to handle WPA2 fine, with roaming across
Hi Haspers,Makes sure you have created an 'Internet tel' profile. It doesn't appear to do anything but was vital in getting it working for me. The other settings in the how to look sensible.Simon
On 8/24/06, Haspers [EMAIL PROTECTED] wrote:
Strange,What settings do you use? I followed this
[EMAIL PROTECTED] wrote:
Hi all,
Just having a strange situation with no clues how to solve.
I have an Asterisk/TRIXBOX located in US and an IAX extn running on
PA168V ATA in another country. All my configs seems to be on 4569 but
i see my extn connected at a different port like 13569.
Hi everybody,
I'm trying to send a user event from the dialplan like this:
exten = s,n,UserEvent(EventName|var1:value1^var2:value2)
The event is sent just fine, but the body is not split in two lines as
it should be according to
I wish to setup asterisk for training purposes so that I am able to
listen in to an extension while a call is going on?
http://www.voip-info.org/wiki-Asterisk+cmd+ZapBarge
and
http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy
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Hi all,Could anyone provide me some usefull link or some idea, how to configure quintum as calling card purpose with Asterisk.Already i created AGI script which working with SIPURA well. But i do not have the idea about quintum how to configure so quintum will dial our asterisk calling card
Abdul, it doesnt sound like you
need to do anything to the Quintum. I would recommend making your dial plan execute
the AGI script of your choice no matter what number is dialed from the context
where the quantum users land.
-Jonathan
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Dear All,
I am currently very stumped on the subject of Active Directory listing,
as I am unable to find any documents regarding this feature thus I am unable
to configure it or know how to use it. Does anyone have any useful info
or documents regarding this feature in terms of how to or guides
Mohamed A. Gombolaty wrote:
Dear All,
I am currently very stumped on the subject of Active Directory
listing, as I am unable to find any documents regarding this feature
thus I am
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Directory
Doug
--
Ben Franklin quote:
Those who
Anyone here use the Nokia E61 ? I am looking to
invest in a wifi phone and I want to get the best. Is it good as far as
reception ? That is of most importance to me. Thanks.
Dovid
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Anyone here use the Nokia E61 ? I am looking to invest in a
wifi phone and I want to get the best. Is it good as far as
reception ? That is of most importance to me. Thanks.
I've tried it in the last couple of days. The biggest issue for
me ist that it HAS to be on the same side of a NAT as
I have not found any solution to the problem I am talking about in
the archives Steve. I have reinstalled and downgraded from pre26 to
pre21 of SpanDSP 0.0.2 and still to no avail. I have followed all
the instructions and ways of fixing this problem and have found none
to be a solution.
Hi Andreas,That is incorrect. It works just fine through NAT providing:- The server is proxying RTP as it has no support for STUN etc.- The NAT is the basic domestic router style, not a full blown firewall requiring port mappings
SimonOn 8/24/06, Andreas Sikkema [EMAIL PROTECTED] wrote:
Anyone
I also have this phone, and have stumbled in to the same problem.
I just think that it isn't in nokia's interest to change this, as it forces
consumers to have some sort of local hardware, that (possibly) only the telecom
provider can give them. This forces the users away from using cheaper
I've got them all. It registers correctly with Asterisk,
and get incoming calls, but it complaints about outgoing calls (Connection
Error). SIP Debug is giving me: SIP/2.0 407
Proxy Authentication Required
But
those settings are the same (Proxy Server/Registrar Server). So what could be
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Simon,
That is incorrect. It works just fine through NAT providing:
- The server is proxying RTP as it has no support for STUN etc.
- The NAT is the basic domestic router style, not a full
blown firewall requiring port mappings
Strange, then you must have some other firmware, because I
Hi Andreas,I'm on 1.0610.04.04 19-04-06 RM-89WOn 8/24/06, Andreas Sikkema [EMAIL PROTECTED]
wrote:Simon, That is incorrect. It works just fine through NAT providing:
- The server is proxying RTP as it has no support for STUN etc. - The NAT is the basic domestic router style, not a full blown
On Thu, 2006-08-24 at 09:29 -0400, Christian Jensen wrote:
I have not found any solution to the problem I am talking about in
the archives Steve. I have reinstalled and downgraded from pre26 to
pre21 of SpanDSP 0.0.2 and still to no avail. I have followed all
the instructions and ways
Thank you so much. After fighting with a large/extensive QOS policy from Cisco's SDM tool, I used your sample and tweaked it for my needs and everything started working fine.Bruce
On 8/23/06, Rich Adamson [EMAIL PROTECTED] wrote:
Bruce Reeves wrote: I'm needing some pointers from anyone who has
Hi,Yes, the proxy and registrar settings are identical in my set-up. Getting registered is the hard part but you've done that. I'd be looking at the Asterisk end of things rather than the E61 to see why that authentication issue is arising.
WOn 8/24/06, Haspers [EMAIL PROTECTED] wrote:
I've
Hi!
I can't find the link anymore where it was a statement from the nokia
support that they are working on a STUN implementation.
A firmwareupdate (with STUN support) will be available in fall 2006.
tom
Andreas Sikkema wrote:
Anyone here use the Nokia E61 ? I am looking to invest in a
wifi
Are you asking about the LDAP stuff or voicemail directory?
On 8/24/06, Mohamed A. Gombolaty [EMAIL PROTECTED] wrote:
Dear All,
I am currently very stumped on the subject of Active Directory listing, as I
am unable to find any documents regarding this feature thus I am unable to
configure
here is the link (german):
http://www.my-s60.com/de/news/news/newswann_kann_die_e_serie_nat_traversal/back/105/cHash/bac3645f5e/index.html
STUN will come in fall 2006
TURN and ICE in 2007
Thomas Artner wrote:
Hi!
I can't find the link anymore where it was a statement from the nokia
The majority of the sample qos policies seem to be based on either five
or seven qos queues, and most folks don't need all of that. What I've
shown as a sample only has three queues; one for voip, one for my
outbound web traffic, and the default queue that everything else falls
into.
You can
Christian Jensen wrote:
I have not found any solution to the problem I am talking about in
the archives Steve. I have reinstalled and downgraded from pre26 to
pre21 of SpanDSP 0.0.2 and still to no avail. I have followed all
the instructions and ways of fixing this problem and have found
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tzafrir Cohen
Sent: Thursday, August 24, 2006 2:32 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] SSH connection hangs on
Can anyone give me some insight as to what this message means from
/var/log/asterisk/full
Aug 24 10:32:02 DEBUG[3016] chan_zap.c: Write returned -1 (Resource
temporarily unavailable) on channel 1
Aug 24 10:32:02 DEBUG[3016] chan_zap.c: Write returned -1 (Resource
temporarily unavailable) on
Doug I have
Exten = 10,hint,SIP/11010
and in mysql I have
exten = 10,1,Dial(SIP/11010)
- Original Message -
From: Douglas Garstang [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, August 21, 2006 3:37 PM
Maybe he is tryin to make it work. This is much better than the old Doug.
Also if he needs features that currently dont exist maybe some one will
create it and then we will all benefit from it :) .
- Original Message -
From: Brian Capouch [EMAIL PROTECTED]
To: Asterisk Users Mailing
I was wondering if anyone was able to execute custom commands on a
channel once a caller connects to an agent after being in a queue. The
reason I ask, is because I would like to use SendText to send a message
to the agent receiving the call to let the agent know how many calls are
waiting in the
Hello,
is possible execute a dialplan before make a call from Zap/g0/dnis? How can
i do that?
Thank you
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I want that each call from PSTN goes to Asterisk to the context for this
line. Within this context can be a menu or a dial command, ...
As more I read, as more I get confused, ... and each try is not working!
My sip.conf:
[WG88621001]
type=friend
defaultip=192.168.250.244
insecure=very
However, reinstalled the box from ground up and installed 1.2.10 and now
CLID isn't working at all. The PSTN line is still transmitting it, as
I've plugged in my Uniden cordless with CLID and it shows up fine on
there, but getting absolutely nothing inside the ${CALLERIDNUM} and
${CALLERIDNAME}
I know this isn't answering your question, but what I did for queue
notification was use softkeys on the phones that call a PHP script on the *
box that'll output XML for the phone to parse and display the queue stats on
demand. Of course your phone would need to have an XML parser or some
In article [EMAIL PROTECTED],
unplug [EMAIL PROTECTED] wrote:
Tony Mountifield wrote:
Try using Set(__testmode=1), but still refer to it as ${testmode}.
The __ tells Asterisk to propagate the variable to created channels.
Do you mean it is a global variable instead of channel variable?
But... you need _both_ in your dialplan.
My extensions.conf has:
exten = 2944054,hint, SIP/2944054
exten = 2944054,1, Dial(SIP/2944054)
ie two lines for the hint.
-Original Message-
From: Dovid Bender [mailto:[EMAIL PROTECTED]
Sent: Thursday, August 24,
That's what he was gettin at. Take the second line out, and put the
first priority in the database.
On Thu, 2006-08-24 at 10:17 -0600, Douglas Garstang wrote:
But... you need _both_ in your dialplan.
My extensions.conf has:
exten = 2944054,hint, SIP/2944054
exten =
List members,
When I dial to a PSTN number, the A2Billing script does all the tasks,
until it shutdown without make the dailout by sip trunk set.
Lasts outputs fro the a2billing.php debug are:
a2billing.php|2: RESFINDRATE:: 0
a2billing.php|2: UPDATE cc_card SET inuse=inuse-1 WHERE
Using STUN isn't a solution to NAT either, as it won't work with symmetrical
NAT, which is very common (or for at least to partially use symmetrical).
I'll be interested to see how the Paragon Wifi phone fares out when it starts
making an appearance in the US.
-Brodie
On Thursday 24 August
I don't see how that helps. If you have a portion of the hint still in
extensions.conf, then what use is the database?
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Thursday, August 24, 2006 10:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
HiCan anyone confirm a working asterisk 1.2 from bristuff with 1 port PCI, hfc-s based ISDN card (zaphfc driver). If so, could you send your configuration. I mean OS (linux distribution) type, kernel version.
Thanks in advanceCheersAndrew
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Hi,
I've installed Asterisk t38passthrough branch and I'm using one
Grandstream ATA to connect Asterisk to a Fax machine. Every time I send
a fax, it gets sent using codec G711, and never T.38. I added the
following parameters in the [general] section as well as in device
configurations:
I have a sangoma 104d that is our main pbx now( legacy system died ). I
have replaced every phone in the building and things are going very well.
We have fax working well and calls are routing properly... All is well...
Except for our support modems... we have support people that dial out with
Quick question maybe someone can point me in the right direction...
Caller -- Receptionist -- ParksCall
Receptionist makes announcement for individual to pick up parked call.
No one picks up so it rings back to receptionist within a minute and a
half. Is there any way to change the ringer for
Just gotta check, I've never seen a complete day with no posts
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All,
I am looking at taking on a project for a hotel that is
using Teledex systems. I see that they have a SIP based phone and the information
says that there is some CMS server part that appears to be the brains behind
the device. My questions are; has anyone out there used this type
Sean Cook wrote:
I have a sangoma 104d that is our main pbx now( legacy system died ). I
have replaced every phone in the building and things are going very well.
We have fax working well and calls are routing properly... All is well...
Except for our support modems... we have support people
What do you mean?
I'm not looking for someone elses work, I'm developing an application from scratch.
Michael
2006/8/24, Andrew Kirch [EMAIL PROTECTED]:
Umm… Flash operator panel?
Andrew
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]
On Behalf Of MirSent: Thursday, August 24,
Michael,
you might take a look at this, as it does most of that + more:
http://micpc.com/eventmonitor/
earl
On Thursday 24 August 2006 14:17, Mir wrote:
Hi
I'm working on a project, where I need the status of every telephone on the
system. (Idle,ringing,busy)
If a phone is busy,
At 15:12 23/08/2006 -0700, Ron Wellsted wrote:
Here are mine (with UK regional settings/A-law).
Thanks a bunch :-) After more search, it turns out that if you don't need
the router feature of the 3102 (I already have a router), the unit must be
connected to the LAN through its... WAN plug.
On Tue, Aug 22, 2006 at 11:56:26 +0200,
Tomislav Parčina [EMAIL PROTECTED] wrote:
The thing is that I can't rely on yum update for asterisk installation. I
would like something that will work like this: I install FC5 from CD/DVD,
install RPM's that I need from my ftp server or from CD,
I guess what Andrew was saying is what are
you trying to do specifically that Flash Operator Panel doesnt already give
you
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mir
Sent: Thursday, 24 August 2006
2:48 PM
To: Asterisk Users Mailing
H == Haspers [EMAIL PROTECTED] writes:
H I've got them all. It registers correctly with Asterisk, and get
H incoming calls, but it complaints about outgoing calls (Connection
H Error). SIP Debug is giving me: SIP/2.0 407 Proxy Authentication
H Required
H But those settings are the same (Proxy
Aaron Daniel wrote:
Not sure about that Doug. It should read:
exten = a,1,VoicemailMan([EMAIL PROTECTED])
You are correct.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
I turned off hyper threading and it did
not help, I also tried telling it to record as gsm files but it still makes an
awful noise. Anyone have any other ideas?
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Don
Sent: Friday, August 18, 2006
12:04 PM
To: Asterisk
Howdy,
I have a Debian box using Debian's Asterisk package. People can leave
voicemail for the extensions that are setup in the configuration, and
asterisk e-mail's the user a .wav file (voicemail.conf). This works
perfect.
However, I want to have VoicemailMain sit on an extension so people
can
On Friday 25 August 2006 08:39, existx wrote:
The error from the CLI is:
Aug 24 16:13:49 NOTICE[23174]: chan_iax2.c:7241 socket_read: Rejected
connect attempt from 192.168.0.23, request '[EMAIL PROTECTED]' does not
exist
It looks like you have created 2699 in a different context than your
Perhaps a stupid suggestion... but did you make sure that the ATA had the T38 selected in the GUI?
bp
On 8/24/06, Ricardo Carvalho [EMAIL PROTECTED] wrote:
Hi,I've installed Asterisk t38passthrough branch and I'm using oneGrandstream ATA to connect Asterisk to a Fax machine. Every time I send
a
Cristian,
The only other line in extensions.conf that references VoicemailMain is this:
exten = a,1,VoicemailMain(${ARG1})
The error from the CLI is:
Aug 24 16:13:49 NOTICE[23174]: chan_iax2.c:7241 socket_read: Rejected
connect attempt from 192.168.0.23, request '[EMAIL PROTECTED]' does not
existx wrote:
Cristian,
The only other line in extensions.conf that references VoicemailMain
is this:
exten = a,1,VoicemailMain(${ARG1})
This should read:
exten = a,1,VoicemailMain([EMAIL PROTECTED])
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a
Ok you have two optionsthe iax extension is created under default
context???
The VoceMilMain could be configured with the options of wich context use
like this:
extensions.conf:
exten = 2999,1,Answer
exten = 2999,2,Wait,2
exten = 2999,3,Voicemailmain(@test)
Hi:
First it at all check if you have a different extension for voicemailmain.?
Then use VoiceMailMain syntax.
And send me the CLI log when you try to connect to VoiceMailMain.
regards.
Cristian.
From: existx [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Not sure about that Doug. It should read:
exten = a,1,VoicemailMan([EMAIL PROTECTED])
If you put it in the brackets, it becomes part of the variable name
instead of part of the argument.
On Thu, 2006-08-24 at 16:57 -0400, Doug Lytle wrote:
existx wrote:
Cristian,
The only other line in
Howdy guys,
Thanks for your help, it works fine without editing the default line of:
exten = a,1,VoicemailMain(${ARG1})
The issue was that I had specified VoicemailMain by the default line,
which was way above the rest of my extensions (out of context).
Hopefully this will help someone in the
Title: Message
I have
no where else to turn to so if anyone has an answer please send
it.
I am running sip version 1.6.on a Polycom 601on
Asterisk and am unable to get the microbroser to work. The phone returns
a 406 error for both idle and services.
I can see the file being requested
Rich Adamson [EMAIL PROTECTED] writes:
Sean Cook wrote:
I have a sangoma 104d that is our main pbx now( legacy system died ). I
have replaced every phone in the building and things are going very well.
We have fax working well and calls are routing properly... All is well...
Except for
Hi All
I have had a problem with a few Snom 320's on several sites locking up
after a few days. I am running application ver 6.2.2 with the latest
jffs2 ver and tried the latest 5.x ver with similar results. Is this also
experienced with other Snom users?
I know some posts say it could be the
As a complete newcomer to Asterisk, Digium and PBX, I have several questions.
But I'll start with this.
To setup a simple system with only a couple of POTS lines, I gather I will need
a TDM400 board with FXO and/or FXS modules.
So, a TDM400 card will support up to two analog (POTS) lines?
FXO is coming from the PSTN, FXS is what devices connect to (like a analog phone).If you are using VOIP phone then you dont need the FXS modules, just FXO.On 8/24/06,
joea, j4computers [EMAIL PROTECTED] wrote:
As a complete newcomer to Asterisk, Digium and PBX, I have several questions.But I'll
This is my first attempt to setup intercom and paging for some Grandview
sip phones per instructions from Grandview.
I put the lines below in extensions.conf and did the CLI reload command.
When I issue
**1 or **2 from a phone I get a 404 error.
Shouldn't that be ringing the 3 phones on my
joea, j4computers wrote:
As a complete newcomer to Asterisk, Digium and PBX, I have several questions.
But I'll start with this.
To setup a simple system with only a couple of POTS lines, I gather I will
need a TDM400 board with FXO and/or FXS modules.
So, a TDM400 card will support up
I have had a problem with a few Snom 320's on several sites locking up
after a few days. I am running application ver 6.2.2 with the latest
jffs2 ver and tried the latest 5.x ver with similar results. Is
this also
experienced with other Snom users?
not sure if this will help you
a TDM400 card will hold up to four modules, which can be FXO or FXS.
depending on the modules purchased you can connect up to four phones
(using FXS modules) or four incoming phone lines (using FXO modules), or
any combination thereof.
joea, j4computers wrote:
As a complete newcomer to
You will need a TDM400 with an FXO module for each line you want. A TDM400
supports up to four lines or analog stations. For two lines, you should get a
TDM04B.
-Original message-
From: joea, j4computers [EMAIL PROTECTED]
Date: Thu, 24 Aug 2006 14:58:21 -0700
To:
Also, make sure you have
a T.38 enabled device at the other end
Edgar
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of William Piper
Sent: quinta-feira, 24 de Agosto
de 2006 21:09
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re:
Title: Message
We had
a similar problem. Eventuallywe gave up and just used apache. We found
that _exactly_ the same content would not work with IIS, but WOULD work with
Apache.
-Original Message-From: Phil Menico
[mailto:[EMAIL PROTECTED]Sent: Thursday, August 24, 2006 3:06
Look over there : http://bugs.digium.com/view.php?id=6953
David
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de J. Oquendo
Envoyé : 24 août 2006 13:54
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : [asterisk-users] Call Parking Ring
Well I got it resolved. Thanks steve for telling me. I looked all
around the filesystem and found some files I did not delete from
spandsp0.0.3. I saw that solution before and tried it and it didn't
seem to work and I wondered why but now i know. Thank you.
Chris
On Aug 24, 2006, at 10:54
I have the hint priority defined for a few SIP
phones.
When I make a call OUT from one of the phones I see that the
show hints picks up a status change from 0 to 1 for the extension,
but when I call IN to that extension the hint status is still 0.
This is on a server built back in
Luciano Moreira wrote:
List members,
When I dial to a PSTN number, the A2Billing script does all the tasks,
until it shutdown without make the dailout by sip trunk set.
Lasts outputs fro the a2billing.php debug are:
a2billing.php|2: RESFINDRATE:: 0
a2billing.php|2: UPDATE cc_card SET
I would suggest buying a very lowprice FXO to begin with which would probably be x100p PCI card at ebay for about $10 +shipping.
On 8/24/06, Adam Collard [EMAIL PROTECTED] wrote:
You will need a TDM400 with an FXO module for each line you want. A TDM400 supports up to four lines or analog
So how about inventing a car? The auto industry is much more profitable.
The point; there is no point in reinventing the wheel, why are you
writing this from scratch?
On 8/24/06, Mir [EMAIL PROTECTED] wrote:
What do you mean?
I'm not looking for someone elses work, I'm developing an
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