On 10/4/07, Markus Zielonka <[EMAIL PROTECTED]> wrote:
> Hello list
>
> I am new in this list.
> Before I wrote this email, i search with "google" and in the list
> arichves for the question.
> I look for a possibility to install FXO ports not over RJ11 Ports. I
> will install the Ports by LSA+ Pat
Hello list
I am new in this list.
Before I wrote this email, i search with "google" and in the list
arichves for the question.
I look for a possibility to install FXO ports not over RJ11 Ports. I
will install the Ports by LSA+ Patch panel. Someone an idea ore link?
Thanks for help.
Bye MZ
PS: P
Hi List,
Could someone tell me if there are Digium cards (or cards from other
providers) that support the following features for use with Asterisk:
- UDI (Unrestricted Digital Information) over ISDN [ETS 300 108]
- Rate Adaptation V110
Any suggestions are highly appreciated,
Brian
__
Hi all,
How can I update a field as null in table using realtime function?
i.e. : Set(REALTIME(tbl,name,1234,prop)=null)
Above will update the field as a null word in table but I want to
update it as null. Can we do that using realtime fuction?
Ango
___
: [asterisk-users] Question on the Monitor command on AMI
Try MixMonitor()
l.
In data Thu, 09 Aug 2007 00:24:47 +0200, Wai Wu <[EMAIL PROTECTED]> ha
scritto:
> Hi all,
>
> Is there a way to have this command to record a mixed file instead of
> one for in and one for out? I have set
Try MixMonitor()
l.
In data Thu, 09 Aug 2007 00:24:47 +0200, Wai Wu <[EMAIL PROTECTED]> ha
scritto:
> Hi all,
>
> Is there a way to have this command to record a mixed file instead of
> one for in and one for out? I have set the Mix parameter to 1, but it is
> still generating two files. I wo
Hi all,
Is there a way to have this command to record a mixed file instead of
one for in and one for out? I have set the Mix parameter to 1, but it is
still generating two files. I would prefer it to have the in and out
files mixed. Thnx.
___
--Bandwidt
Are you sure calls were dropped with change in IP ?? I think it should let
current calls run and use new IP for new connections . However if
destination serv drops calls then it's a different story .
On 03/07/07, Henry J. Cobb <[EMAIL PROTECTED]> wrote:
Asterisk 1.4.5 full log:
[Jul 2 09:31:16
Asterisk 1.4.5 full log:
[Jul 2 09:31:16] VERBOSE[2682] logger.c: == Refreshing DNS lookups.
[Jul 2 09:31:16] NOTICE[2682] dnsmgr.c: host 'outbound1.vitelity.net'
changed from 64.2.142.17 to 64.2.142.29
[Jul 2 09:31:23] DEBUG[2711] jitterbuf.c: Attempting to exceed
Jitterbuf max 600 timeslots
[Jul 2 09:31:16] VERBOSE[2682] logger.c: == Refreshing DNS lookups.
[Jul 2 09:31:16] NOTICE[2682] dnsmgr.c: host 'outbound1.vitelity.net'
changed from 64.2.142.17 to 64.2.142.29
[Jul 2 09:31:23] DEBUG[2711] jitterbuf.c: Attempting to exceed
Jitterbuf max 600 timeslots
And the calls are droppe
Jerry Geis wrote:
> Can one server (like AMD 6000+ X2) with 2 GIG ram
> running asterisk 1.4 handle having 2100 wireless phones connected.
> All phones will not be talking at the same time only a couple will be.
>
> There may be 1 T1 card in the box.
>
> Will this work? If not how does one handle
Jerry Geis wrote:
Can one server (like AMD 6000+ X2) with 2 GIG ram
running asterisk 1.4 handle having 2100 wireless phones connected.
All phones will not be talking at the same time only a couple will be.
There may be 1 T1 card in the box.
Will this work? If not how does one handle this situat
Can one server (like AMD 6000+ X2) with 2 GIG ram
running asterisk 1.4 handle having 2100 wireless phones connected.
All phones will not be talking at the same time only a couple will be.
There may be 1 T1 card in the box.
Will this work? If not how does one handle this situation.
Thanks,
Jerr
Vietnhi Phuvan wrote:
> Hello Folks,
>
> I am testing Asterisk 1.4.2 running on Fedora Core 5 (version 2.6-17). I
> have loaded the app_meet.so module in order to activate the MeetMe,
> MeetMeCount and MeetMeAdmin applications. While I have been successful
> in loading the app_meet.so module, I am
Hello Folks,
I am testing Asterisk 1.4.2 running on Fedora Core 5 (version 2.6-17). I
have loaded the app_meet.so module in order to activate the MeetMe,
MeetMeCount and MeetMeAdmin applications. While I have been successful
in loading the app_meet.so module, I am experiencing an immediate ker
bkruse wrote:
Good question shawn,
The callfile does get deleted once the call has been finished (I believe
its FINISHED, not processed)
No, they are not executed sequentially..exactly. Well, from your
point of view, you can drop tons of them in there and all of the calls
will fire up.
shawn,
you can set an archive variable in the .call file to 'yes' and it will
save it in ./outgoing_done... if there is now outbound line availible,
the .call file is updated (appended to) as per the status... * will keep
trying till it completes the calls or the number of retries is reached.
Good question shawn,
The callfile does get deleted once the call has been finished (I believe
its FINISHED, not processed)
No, they are not executed sequentially..exactly. Well, from your
point of view, you can drop tons of them in there and all of the calls
will fire up. I have dropped
hello there all,
if i have a script that writes drop files into /var/spool/asterisk/outgoing
asterisk picks up the file and initiates the call just fine.
however, what is supposed to happen if more than one gets dropped in there
within like a second. Will it wait till the first is complete to init
I have a server that only handles the inbound and outbound call and passes
everything to the second server using IAX.
Sometimes it so happen that a call comes in on the First machine, this machine
forward to the second machine as an inbound call using IAX, now the second
machine decides that thi
On Sun, 2007-04-01 at 10:49 +0200, Philipp Kempgen wrote:
> Priority jumping is deprecated anyways. Better use something
> like Goto(s-${DIALSTATUS},1). See extensions.conf for examples.
>
> Regards,
> Philipp
>
I totally agree! While you can get what you want with +101
jumping, I highly sug
Thank you - Got it now. Makes everything look a lot cleaner :)
On Sun, 01 Apr 2007 11:58:20 +0200
Philipp Kempgen <[EMAIL PROTECTED]> wrote:
> --[ UxBoD ]-- wrote:
>
> > [inbound-sip]
> > exten => uxbod(u1),1,Dial(sip/1001,20,t)
> > exten => uxbod,n,PlayBack(uxbod)
> > exten => uxbod,n,Hangup()
--[ UxBoD ]-- wrote:
> [inbound-sip]
> exten => uxbod(u1),1,Dial(sip/1001,20,t)
> exten => uxbod,n,PlayBack(uxbod)
> exten => uxbod,n,Hangup()
> exten => uxbod,u1+101,PlayBack(uxbod)
> exten => uxbod,n,VoiceMail([EMAIL PROTECTED],s)
> exten => uxbod,n,Hangup()
>
> but when I do a extensions reloa
Okay, I have changed it too :-
[inbound-sip]
exten => uxbod(u1),1,Dial(sip/1001,20,t)
exten => uxbod,n,PlayBack(uxbod)
exten => uxbod,n,Hangup()
exten => uxbod,u1+101,PlayBack(uxbod)
exten => uxbod,n,VoiceMail([EMAIL PROTECTED],s)
exten => uxbod,n,Hangup()
but when I do a extensions reload I get
Cool. That is nice and clean :) Many thanks.
On Sat, 31 Mar 2007 23:32:45 -0700
"Yuan LIU" <[EMAIL PROTECTED]> wrote:
> >From: "Rizwan Hisham" <[EMAIL PROTECTED]>
> >Date: Sat, 31 Mar 2007 17:01:51 +0500
> >
> >[inbound-sip]
> >exten => uxbod,1,Dial(sip/1001,20,jt)
> >exten => uxbod,n,Hangup
> >
Yuan LIU wrote:
>> From: "Rizwan Hisham" <[EMAIL PROTECTED]>
>> Date: Sat, 31 Mar 2007 17:01:51 +0500
>>
>> [inbound-sip]
>> exten => uxbod,1,Dial(sip/1001,20,jt)
>> exten => uxbod,n,Hangup
>>
>> exten => uxbod,102,PlayBack(uxbod)
>> exten => uxbod,103,VoiceMail([EMAIL PROTECTED],s)
>> exten => ux
From: "Rizwan Hisham" <[EMAIL PROTECTED]>
Date: Sat, 31 Mar 2007 17:01:51 +0500
[inbound-sip]
exten => uxbod,1,Dial(sip/1001,20,jt)
exten => uxbod,n,Hangup
exten => uxbod,102,PlayBack(uxbod)
exten => uxbod,103,VoiceMail([EMAIL PROTECTED],s)
exten => uxbod,104,Hangup()
here if dial fails then n+
also only priorities are added incase of priority jumping, not extensions.
On 3/31/07, --[ UxBoD ]-- <[EMAIL PROTECTED]> wrote:
Hi,
I am attempting to change my dialplan to use 'n' priorities and labels
for easier reading, and less re-numbering :) but how do you handle the
plus 101 ? In my ext
[inbound-sip]
exten => uxbod,1,Dial(sip/1001,20,jt)
exten => uxbod,n,Hangup
exten => uxbod,102,PlayBack(uxbod)
exten => uxbod,103,VoiceMail([EMAIL PROTECTED],s)
exten => uxbod,104,Hangup()
here if dial fails then n+101 =102 extension will get executed unless you
use j option in dial application
Hi,
I am attempting to change my dialplan to use 'n' priorities and labels
for easier reading, and less re-numbering :) but how do you handle the
plus 101 ? In my extensions.conf I have a simple plan for testing :-
[inbound-sip]
exten => uxbod,1,Dial(sip/1001,20,t)
exten => uxbod,n,PlayBack(uxbod
Hi Steve -
Just my personal experience, but I do not find IAX to be very reliable.
Is there any particular reason you are not using SIP?
I'm curious as to your negative experiences with IAX. I generally use
it for multi-office installations, and have had good expereinces with
it. What reliab
A. Levy wrote:
Hello.
I have a TE405P Digium Card (4 E1's) with ISDN protocol and I need to
find out if there is any limitation about DSP capabilities, I mean, I
am not sure how many phone calls my Digium card supports,
simultaneously. The calling flow goes from IAX <-> ISDN.
I am running thi
Hello.
I have a TE405P Digium Card (4 E1's) with ISDN protocol and I need to find
out if there is any limitation about DSP capabilities, I mean, I am not sure
how many phone calls my Digium card supports, simultaneously. The calling
flow goes from IAX <-> ISDN.
I am running this card into CPU li
Matthew Rubenstein wrote:
> caller in a multi-caller (eg, 2 people or more) or even an app. So if
> both people in a call are sending G.729 encoded data, and your app
> decodes the *mixed* G.729 into ulaw (or slinear or any other decoded
> format it outputs) requiring a single instance of the decod
From: "Rilawich Ango" <[EMAIL PROTECTED]>
Date: Tue, 13 Feb 2007 17:43:05 +0800
Hi, I have tried the regex function below with MACRO_EXTEN=5000*.
However, both of them return 0 instead 1 to me. How can I search the
character in the end of line?
${REGEX("[*]$" ${MACRO_EXTEN})
returns 0
You
Hi, I have tried the regex function below with MACRO_EXTEN=5000*.
However, both of them return 0 instead 1 to me. How can I search the
character in the end of line?
${REGEX("[*]$" ${MACRO_EXTEN})
returns 0
${REGEX("*$" ${MACRO_EXTEN})
returns 0 with error
ango
On Mon, 2007-02-05 at 12:00 -0700,
[EMAIL PROTECTED] wrote:
> Date: Mon, 5 Feb 2007 11:36:28 -0500
> From: Andy Davidson <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] Question on G.729 (was: H.264 *Not
> Patented*)
> To: Asterisk Users Mailing List - Non-
On 1 Feb 2007, at 14:14, Lacy Moore - Aspendora wrote:
> On 2/1/07, Andy Davidson <[EMAIL PROTECTED]> wrote:
> > What I would expect to happen, is that Asterisk would transcode
> > between the ulaw/alaw party, and me, wanting to listen via
g729. Is
> > this what *should* happen ? Worth noti
On 2/1/07, Andy Davidson <[EMAIL PROTECTED]> wrote:
What I would expect to happen, is that Asterisk would transcode
between the ulaw/alaw party, and me, wanting to listen via g729. Is
this what *should* happen ? Worth noting that my provider does not
support G.729. Is what is happening a bug
Hi,
I asked some questions here about G.729 earlier in the week, and it
looks like it would fit the bill for compressing audio between my *
server in colocation and sip phone at home.
This is what I want my setup to look like.
(Wont make sense unless you are using a fixed width font)
Hi!
I have several SPA3000 devices (older versions of SPA3102) and they are
working OK, sound quality is good. It is very configurable to the slightest
details. I use it whenever I need just one or two FXO ports, like for small
scale PSTN integration, or for connecting some other equipment that r
Hello, I intend to buy a FXO/FXS device from Linksys. I'm thinking
about SPA3102.
What you guys think about it. Is ok, is working with asterisk, can i use it
like voip peer. Thank you for your advice.
Jonson.
___
--Bandwidth and Colocation provided by E
Scott Walde wrote:
This page:
http://voip-info.org/wiki/view/480i+Busy+lamp+field+%27BLF%27+support
Sorry, I sent the wrong link... That page was just for regular BLF. The
Metermaid bit is on this page, about half way down:
http://voip-info.org/wiki/view/Asterisk+and+Aastra+Phones
ttyl
Christopher Aloi wrote:
I would like to use a visual park function if possible, here's how I
see it working
What you want is called BLF with Metermaid. There was a question about
it on this list less than a week ago, but with no response.
Metermaid was a patch for the 1.2.x series, bu
Hello List!
I am hoping someone may be able to assist with the following feature I am
looking to implement:
I would like to use a visual park function if possible, here's how I see it
working
-> Call comes into caller A
-> Caller A places the call in orbit (or park) by dialing 700
-> A me
Trevor Peirce wrote:
Lee Jenkins wrote:
Does Asterisk strip off the quotes when storing the value?
You could do a 5 minute test to figure that out...
I am not at my house tonight where I have access to a box and was
wondering about it on the car ride over to where I am.
Of course, you hav
Lee Jenkins wrote:
Does Asterisk strip off the quotes when storing the value?
You could do a 5 minute test to figure that out...
blah.agi:
SET VARIABLE testme "I have quotes!"
dialplan.txt:
exten => s,1,AGI(blah.agi)
exten => s,n,Set(regular=no quotes)
exten => s,n,NoOp(regular is ${regular})
Hi all,
I just finished writing the bulk of an AGI interface to FirebirdSQL
databases and I noticed that when assigning a variable through AGI (I
assume this also applies within the dialplan), you have to enclose it in
quotes if there are any space.
Does Asterisk strip off the quotes when s
Hi
On 12/27/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:
Sounds great. What's the mechanism by which Asterisk servers communicate the
mwi status between them?
With new IAX commands. The client can ask the server how many messages
are waiting.
I've started to port the modification on 1.4,
ercial Discussion
> Subject: Re: [asterisk-users] Question about MWI in Asterisk 1.4.0
>
>
> Hi
>
> On 12/26/06, Kevin P. Fleming <[EMAIL PROTECTED]> wrote:
> > No, Asterisk 1.4 does not include any functionality for multi-server
> > MWI. The SIP functionality impr
Hi
On 12/26/06, Kevin P. Fleming <[EMAIL PROTECTED]> wrote:
No, Asterisk 1.4 does not include any functionality for multi-server
MWI. The SIP functionality improvements are just better support for the
'pull' model of SIP MWI, in addition to the 'push' model Asterisk has
used in the past.
If I
Jean-Yves Avenard wrote:
> Does Asterisk 1.4 introduce new capabilities allowing to remotely
> check the MWI on a remote machine like what my patches are doing? In
> the list of changes for 1.4.0 I read "SIP MWI subscription support",
> what exactly is this ?
No, Asterisk 1.4 does not include any
Hello
I am running the following setup in order to make VoIP calls at home.
Home Phone <-> SPA3000 <-> Asterisk Home <- IAX2 over Internet <->
Asterisk Office
The voice mail for Home Phone is hosted on the Asterisk Office
machine. I wanted to have a way to check the status of my voicemail on
my
I noticed that asterisk will keep the phone record in astdb when the
phone (especially hardphone) unplugged.
After unplug the phone, I still get the phone information in astdb:
database showkey SIP/Registry/1234
/SIP/Registry/1234 :
10.14.43.31:40876:60:1234:sip:[EMAIL
Thanks. I got it.
On 12/21/06, Anselm Martin Hoffmeister <[EMAIL PROTECTED]> wrote:
Am Donnerstag, den 21.12.2006, 11:28 +0800 schrieb Rilawich Ango:
> How about:
> exten => _X.,1,Answer
>
> Does it include all numbers and characters?
As of the docs, no. It should only match 0123456789
See
htt
Am Donnerstag, den 21.12.2006, 11:28 +0800 schrieb Rilawich Ango:
> How about:
> exten => _X.,1,Answer
>
> Does it include all numbers and characters?
As of the docs, no. It should only match 0123456789
See
http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns
BR
Anselm
__
How about:
exten => _X.,1,Answer
Does it include all numbers and characters?
On 12/21/06, David Thomas <[EMAIL PROTECTED]> wrote:
On 12/20/06, Rilawich Ango <[EMAIL PROTECTED]> wrote:
> I have 2 sip accounts with name 1234 and abcd respectively. Account
> abcd can make call to 1234 but not vis
On 12/20/06, Rilawich Ango <[EMAIL PROTECTED]> wrote:
I have 2 sip accounts with name 1234 and abcd respectively. Account
abcd can make call to 1234 but not visa versa. When I change account
abcd to 1abcd, both of them can make call to each others. In the
case, the format of sip account should
I have 2 sip accounts with name 1234 and abcd respectively. Account
abcd can make call to 1234 but not visa versa. When I change account
abcd to 1abcd, both of them can make call to each others. In the
case, the format of sip account should be start with number. I wonder
whether we can use a s
For my home phone system I have an old P-II, which is working perfectly fine
for last more than a year now. I had a P-III before that, but one day it
died. This P-II is still working and we have no problems with our phone
system. I even had conference calls on it with 6 simultaneous users. For the
Speaking of the X100P, I am going to setup an asterisk server next
week for a friend's business to replace his aging system. He
currently has two voice lines and another line for the fax machine.
I was looking at the Sangoma A20200D but that's pretty expensive...
We're going to use Grand
The card will let you interface with a regular telephone line instead
of VoIP. If you want to use a regular phone instead of the computer
softphones, look into the Grandstream handytone devices - they'll
make it so your regular telephones can talk to Asterisk. You can
make the system work
: "Michael Sullivan" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, December 13, 2006 2:51 PM
Subject: Re: [asterisk-users] Question about hardware
> On Wed, 2006-12-13 at 08:29 -0600, jason wrote:
> > cheapy PC (t
nope, just a regular old phone cord. with that card and a PC, you can
receive calls, dial out, terminate SIP, IAX, create an answering
machine, run voicemail, talk to jabber servers, all kinds of fun stuff!
Asterisk is almost as good as Legos and a lot easier on bare feet at 2am!
Michael Sul
Don't forget that IF you have NO card, you need to roll ZTDUMMY into the
compile. With no card though, you will not be able to read the incoming CLID
Also, IF you ever want to progress beyond the X100 card, The Digium
cards ( beyond your present budget ( are really intolerant of older PCI
buse
On Wed, 2006-12-13 at 08:29 -0600, jason wrote:
> cheapy PC (throw away PII is fine) and if you want to use the PSTN, a
> X100P FXO card. These can be had on ebay for 11 bucks, but I understand
> that even that pushes the bank some days. You don't need the card, you
> only need it if you want t
cheapy PC (throw away PII is fine) and if you want to use the PSTN, a
X100P FXO card. These can be had on ebay for 11 bucks, but I understand
that even that pushes the bank some days. You don't need the card, you
only need it if you want to receive or place calls on the PSTN. You can
use aste
IF I wanted to do the whole "sophisticated telephony VoIP stuff"
asterisk, what hardware would I need? I have a feeling that my fax
modem is probably not going to work out. My wife and I have an income
of $650 a month. After the first-of-the-month bills are payed, we're
lucky if we have $100 lef
Great, exactly what I was looking for. Thanks so much!
Shabbat shalom
Jez
--- Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> On Fri, Dec 08, 2006 at 08:23:36AM -0800, je .
> wrote:
> > I understand this function (line 832 in
> > app_voicemail.c) is used to retrieve a voice
> message.
> > What I
On Fri, Dec 08, 2006 at 08:23:36AM -0800, je . wrote:
> I understand this function (line 832 in
> app_voicemail.c) is used to retrieve a voice message.
> What I don't understand however is why ".txt" is
> appended to the end of the filename. Could someone
> shed some light on this for me?
This
I understand this function (line 832 in
app_voicemail.c) is used to retrieve a voice message.
What I don't understand however is why ".txt" is
appended to the end of the filename. Could someone
shed some light on this for me?
Thanks,
Jez
if (msgnum > -1)
make_file(fn, sizeof(fn), dir, msgnum
mailto:[EMAIL PROTECTED] On Behalf Of Tielin Xu
Sent: Tuesday, December 05, 2006 5:23 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Question about Realtime static table
Hi All:
I'd like to use Realtime Static in terms of the performance concern
about dynamic realtime. Assu
Hi All:
I'd like to use Realtime Static in terms of the performance concern
about dynamic realtime. Assume that I create a table:
as following:
CREATE TABLE `extensions_table` (
`id` int(11) NOT NULL auto_increment,
`context` varchar(20) NOT NULL default '',
`exten` varchar(20) NOT NULL defa
Guh! :)
Thanks!
Matt G
On 05/12/06, Humberto Figuera <[EMAIL PROTECTED]> wrote:
http://soft-switch.org/downloads/snapshots/spandsp/
;p
--
Humberto Figuera - Using Linux 2.6.17
Usuario GNU/Linux 369709
Caracas - Venezuela
GPG Key Fingerprint = 5AAC DF0C 00F4 2834 28BA 37AD 3364 01D1 74CA 06
http://soft-switch.org/downloads/snapshots/spandsp/
;p
--
Humberto Figuera - Using Linux 2.6.17
Usuario GNU/Linux 369709
Caracas - Venezuela
GPG Key Fingerprint = 5AAC DF0C 00F4 2834 28BA 37AD 3364 01D1 74CA 0603
___
--Bandwidth and Colocation provide
Hi Jerry,
Where did you find the 1.4 versions of this software? I don't see
anything on the official spandsp downloads site, just pre2 and pre3
releases, no 20061130.tar.gz :)
Thanks,
Matt G
On 05/12/06, Jerry Geis <[EMAIL PROTECTED]> wrote:
I downloaded these 4 files:
app_rxfax.c app_txfax.
lf Of Jerry Geis
Sent: 05 December 2006 14:41
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] question on tx_fax install for asterisk 1.4
I downloaded these 4 files:
app_rxfax.c app_txfax.c asterisk.patch spandsp-20061130.tar.gz
for use with asterisk 1.4 (these are the 1.4 file
I downloaded these 4 files:
app_rxfax.c app_txfax.c asterisk.patch spandsp-20061130.tar.gz
for use with asterisk 1.4 (these are the 1.4 files)
I installed spandsp, copied app_rxfax and app_txfax into
/asterisk-1.4beta3/apps
my question is what do I do with asterisk.patch?
I tried to put it
The CDR could be used by billing software not all billing soultions do
there account that way.
he have only one structure of data or they have multi structure with
more information
logged ? sample: cdr simple and cdr_extended
I am not sure what you are asking. You can log just about anything
I am not sure if i understood what you mean but yes asterisk cdr's can be
used for billing with some modifications of your own. Asterisk can make cdr
in csv,mysql,postgresql with complete call
info which can be used for billing system's .
On 19/11/06, Noc Phibee <[EMAIL PROTECTED]> wrote:
Hi
I
Hi
I have a small question on CDR Database:
It's used by billing software no ?
he have only one structure of data or they have multi structure with
more information
logged ? sample: cdr simple and cdr_extended
thanks bye
___
--Bandwidth and Coloc
- Original Message -
From: "Edwin Lam" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, November 15, 2006 9:22 PM
Subject: Re: [asterisk-users] Question about TFTPD server
> Christian wrote:
>
>> I hav
ay, November 15, 2006 9:22 PM
Subject: Re: [asterisk-users] Question about TFTPD server
Christian wrote:
I have installed this package onto my Debian and placed the files i want
the Cisco 7960 phone to get from the tftpdboot directory. But it doesn't
seem to work. Are ther any special sett
On Wed, Nov 15, 2006 at 03:22:54PM -0800, Edwin Lam wrote:
> Christian wrote:
>
> >I have installed this package onto my Debian and placed the files i want
> >the Cisco 7960 phone to get from the tftpdboot directory. But it doesn't
> >seem to work.
Have you followed the procedure I provided ea
Christian wrote:
I have installed this package onto my Debian and placed the files i want the
Cisco 7960 phone to get from the tftpdboot directory. But it doesn't seem to
work. Are ther any special settings I should do to this server?
Many thanks for all your help,
which tftp server package
ED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Christian
> Sent: Wednesday, November 15, 2006 12:12 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Question about TFTPD server
>
> Hi all,
> I have installed this package onto my Debian and pl
Hi all,
I have installed this package onto my Debian and placed the files i want the
Cisco 7960 phone to get from the tftpdboot directory. But it doesn't seem to
work. Are ther any special settings I should do to this server?
Many thanks for all your help,
Christian
I have a query that query's my database based on the read input for an ID number.exten => s,4,MYSQL(Query resultid ${connid} SELECT\ `FirstName`\ `HomePhone`\ FROM\ `contacts`\ WHERE\ `ContactID`= \'${ID}\')
exten => s,5,MYSQL(Fetch foundRow ${resultid} var1 var2) ; fetch rowProblem is is that when
Hi,
I havent tested this yet, but I am just wondering what are the advantages of
using this GUI?
Does it help you with creating extensions or what?
Many thanks,
Christian
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asterisk-users mailin
Hi all,
Does anyone know if the Mitel phone features a webintreface for configuring the
phone?
Many thanks,
Christian
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From: Matt [mailto:[EMAIL PROTECTED]
Sent: Wednesday, November 08, 2006 6:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question on Aastra phones and Astrisk
It only happens when you go from IAX/SIP --> asterisk box --> aastra phon
PROTECTED]
Sent: Wednesday, November 08, 2006 6:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question on Aastra phones and Astrisk
It only happens when you go from IAX/SIP --> asterisk box --> aastra phone.
Doesn't happen PSTN -->
rcial Discussion
Subject: Re: [asterisk-users] Question on Aastra phones and Astrisk
*bump* Anyone?
On 11/6/06, Curt Shaffer <[EMAIL PROTECTED]> wrote:
> I wanted to add what we have both seen on traffic captures.
>
> You see Caller 1's RTP stream. Call 2 comes in and you see t
ge-
> From: Curt Shaffer [mailto:[EMAIL PROTECTED]
> Sent: Monday, November 06, 2006 6:58 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [asterisk-users] Question on Aastra phones and Astrisk
>
> I'm the friend mentioned here.
>
&g
mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Matt
> Sent: 07 November, 2006 8:31 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Question on Aastra phones and Astrisk
>
> *bump* Anyone?
>
> On 11/6/06, Curt Shaffer <[E
7;Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Question on Aastra phones and Astrisk
I'm the friend mentioned here.
I am using the Aastra 480i CT. It is SIP to my PBX and IAX termination from
the PBX to my provider. My issue has a slight twist to i
l Discussion
Subject: [asterisk-users] Question on Aastra phones and Astrisk
Hi,
Some odd behaviour here. A friend and I were talking tonight,
and it seems we have both seen the same problem. We are both using
aastra phones (I am using 9113is).We have a connection to and from
provi
f the router is on the
same network as the * box.
Thanks!
Curt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Monday, November 06, 2006 6:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Question on A
Hi,
Some odd behaviour here. A friend and I were talking tonight,
and it seems we have both seen the same problem. We are both using
aastra phones (I am using 9113is).We have a connection to and from
providers via SIP and IAX.When I place a call on the local hold of
the phone, and
I am having a problem getting the following logic to work,
in a macro.
Basically, if the caller ID matches, set the outbond trunk
to a Zap channel, otherwise use a SIP provider.
exten => s,n,Set(TRUNK=${IF($[${CALLERIDNUM} = 1234567890]?Zap/g1:SIP/LDPROVIDER)})
; use PRI instead of
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