On 04/29/06 10:06 Josué Conti said the following:
is that if the agent transfers the call, for another user and this user
takes care of the call, the status of the agent in the "show agents" is
of that it the same continues speaking (talking to zap) with circuit
how are you performing the t
On Friday 28 April 2006 15:32, Eric "ManxPower" Wieling wrote:
> What does the "R" option do?
Indicate 'Ringing' as soon as the called party indicates 'Ringing'.
The 'r' option indicates 'Ringing' as soon as the connection is built, even if
the called party is not yet ringing.
With some SIP Ser
I've been playing around with a new system I'm going to install in
another office. In setting up the Polycom's, I accidently used a new
power supply from a new 601 (24VDC) with an 600. The 600 only
require 12VDC. Now, I get nothing on the screen of the 600 when I
plug in 12 VDC. (At the tim
Jeffery, IAX + gsm is a very good integration. I use and do not have problems.
One remembers to have a good router, good switch, one link of Internet with QoS or same one link frame-relay for this way you will not have problems. I wait to have helped.
Regards
Josué
200
You can use the __Variables They are passed along the IAX2 channel
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Steve Totaro
> Sent: Friday, April 28, 2006 9:24 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Su
hi,all..
i think IAX + GSM codec is good quality for asterisk rgiht ?
-- Jefferyiaxtel Num: 1-700-576-1311fwdnet Num: 728150
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AMEN!!
Any consultant that DOESNT take this into consideration
should stick to installing Windows and calling themselves an IT
"Expert".
You can screw up someone's network, mess up a
workstation, hose their email, but you break someone's telephone service there
will be hell to pay.
Ker
The following information is accurate for a situation where both T1s are
connected to telco switches and the telco is therefore providing the
timing signals. If one of the T1s is point to point (such as a tie line)
then this information may or may not apply depending upon what's on the
remote end.
I’d set your box to DMZ on the
router & see if the problem exists first. If so, you probably forgot to
forward something.
Make sure that you forwarded both TCP
& UDP ports.
Billy P.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aryn Nakaoka
Sent: Friday, Ap
I have 2 Asterisk servers connected via IAX2 connections.
PBX1 is on the internet with a public IP Address
- with PRI
PBX 2 is behind a NAT router with IAX2 Ports forwarded
1-way audio is an issue with incoming and outgoing calls
using the PRI. However whenever 1-way a
Il Neofita wrote:
Hi,
I am pretty sure that you already answer to this question, but I was
not able to find the solution
on the console I have over and over the following msgs
-- Remote UNIX connection disconnected
-- Remote UNIX connection disconnected
-- Remote UNIX connection disc
Hi All.
I am with the following problem. The agents of my Queue when they receive a call originated from the pilot of queue, are with the status of "talking to zap". Until all good, he is correct there, but the fact is that if the agent transfers the call, for another user and this user takes care
thanks you, bill,,,
rtcachefriends=yes is working.
-Jeffery
On 4/29/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
I'm not sure about IAX, but in SIP… you can use rtcachefriends=yes in the general section to accomplish this.
Don't know about #2
Billy P.
From:
[EMAIL PROTECTED
Hi,I am pretty sure that you already answer to this question, but I was not able to find the solutionon the console I have over and over the following msgs -- Remote UNIX connection disconnected -- Remote UNIX connection disconnected
-- Remote UNIX connection disconnected -- Remote UNIX c
I’m not sure about IAX, but in SIP… you can use rtcachefriends=yes
in the general section to accomplish this.
Don’t know about #2
Billy P.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ??
Sent: Friday, April 28, 2006 8:40
PM
To: Asterisk
Users Mailing List -
I am not sure if it works but if you construct your iax.conf to include
a [slave] section there is an option (sendani=yes I believe) but then
your dial statement would not include IP and credentials, just
Dial(IAX2/SLAVE/${EXTEN})
Isn't the ${EXTEN} variable being received on your slave box th
hi, alll,,,
there two asterisk box share one realtime database... and all the client is IAX2.. and registery dynamic...
there have some question need to confirm..
1, when i run iax2 show peers,,,there no show the peers that registed with real time... the same as run iax2 show users..there not
Make sure that sip.conf has externip and localnet are properly
configured. I have many GXP-2000 on different nets as my * box with
no problem.
- Waldo
On Apr 28, 2006, at 11:12 AM, Mimmus wrote:
Hi,
I have a lot of GXP-2000 phones not registering with Asterisk server.
After two days of att
Just saw these new Asterisk compatible cards at Rhino's homepage. Anyone
has experience with these babes?
http://www.rhinoequipment.com/index.html
http://www.rhinoequipment.com/t1card.html
The 4-port FXO looks interesting. Looks like a very clean design.
Leo
__
In our case, it was cpuspeed (a daemon) interfering with the zaptel drivers.
Paul Hales
Technical Manager
AsteriskIT
- Original Message -
From: "Rich Adamson" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, April 27, 2006 10:14 PM
Subjec
Exactly why I chose to go with a PRI for business use. There is something to be said for the stability of a telco, even if it's not SBC (or AT&T now). In some cases, government interference is good. How many businesses can survive a loss of their phone number? I know the ones I deal with cannot
Hi,I try to use my phone that has a SMS capability with asterisk.I am not able to receive SMS, someone can help me out?This what I am able to have but nothing more -- Executing SMS("SIP/503-7d2e", "508|sa") in new stack
-- SMS TX 93 00 6D
___
--Bandwi
Dear Asterisk-Users:
Question:
How do I get asterisk to pass DNID/RDNIS information between
asterisk machines using iax2, in a Dial(IAX2...) command ?
Setup:
=
I have two asterisk boxes, MASTER and SLAVE. MASTER is running
1.2.0 and SLAVE is running 1.2.1. The main box handles
Rich Adamson wrote:
Well... outbound calls via Nufone still function, so I'd guess they
are scrambling to find another cost effective source for DID's (which
will likely not be the same DIDs they had due to probable portability
issues). That would also suggest 800 numbers can't be remapped unt
I'm sorry. That was really vague. I'm tired and need to go home...
Here's some more detail.
I'm calling from a Nokia GSM cell phone ala Cingular.
I'm calling to a zap tdm channel. I have already tried 'relaxdtmf=yes'.
Bob McDowell
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[
When my Asterisk box is called from my company's standard type of phone,
it does not detect normal button presses on the keypad. You can use the
phone's 'touch tones' feature to send data to the box, but that is a bit
cumbersome. My Norstar voicemail system (that is being replaced) does
not have
Hi,
I have configured Asterisk with Fax-to-Email feature. Fax is coming to
Asterisk through DID. What is happening is that sometimes Asterisk receives
Fax in first attempt and sometimes in 2 to 4 attempts. On DID Sip,G711 codec
& T.38 protocol is enabled.
Please advise me how I can make Fax servi
Hi
Someone to integrated Asterisk with pbx Panasonic KX-T336?
Regards
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Hello,
I recently acquired
a Digium TE210P for a faxing solution I'm working on. I would like to use
the card to send faxes from my infrastructure using Asterisk and HylaFAX.
I'm curious if the zaptel driver will recognize in/out faxes to/from the
TE210P. Is this going to work or should I
There are some settings that you should look at. For options setting the hardware echo canceller you would use 'vpmsupport' and 'vpmdtmfsupport
' (there are probably others) Search through the archives here for 'vpmsupport' and 'vpmdtmfsupport' and you will find quite a few discussions on this, lik
List,
Per someone's suggestion (thanks, whoever you were) from this list, I
implemented dnsmasq to prevent the issue of resolving DNS when the net
connection goes down.
This morning the net connection was down, and our * server didn't miss a
beat. I recommend looking into:
http://thekelleys.org.u
So if I switch between a TE411P and TE410P or vice versa, I should not
change anything in my config.
Correct?
-- -- Steven
http://www.glimasoutheast.org
"Rob Lith" <[EMAIL PROTECTED]>
wrote in message news:[EMAIL PROTECTED]...On
28/04/06, Steven <[EMAIL PROTECTED]> wrote:
I
Ok.. I sort of lied.. it's the same CLEC but two different switches..
was told by them since they are different switches I needed primary
timing so in theory it should work if I set it as secondary.. ok
we'll try!
Just out of curiousity.. what happens if I set both as 1 (primary?)
On 4/28/06
On 28/04/06, Steven <[EMAIL PROTECTED]> wrote:
I have seen conflicting references in regards to the Digium Wildcard TE411P echo settings in zapata.conf.Does anyone have the official word on this?Should echo cancel be enabled in zapata.conf if the card has built in EC?
If so, should a particular EC
I have the following in my extensions.conf
[ext-local]
exten => _53XX,1,Wait(2)
exten => _53XX,2,NoOp,Dialing ${EXTEN} from ext-local-custom
exten => _53XX,3,Macro(dialout-trunk,2,${EXTEN},,)
This is used to match inbound caller-id for my legacy PBX.
It works fine for inbound calls, but not for i
Hello...I buyed a Voicetronix OpenPri. I have configurate this with wanroute, for logger App (/logger-1.1.tar.gz), buy i start aplication and i see next error. ./logger 2 VPB Cards Detected!
60 VPB Total Channels Detected!Started!monitor fifo thread startedmonitor_open OK!started m
Matt wrote:
Hi,
With a digium dual PRI card (dual span). Is there any reason I can't
have both PRIs being PRIMARY timing sources? They are both from
different CLECs, and as such I need them both to do their own timing.
No. However, if the two CLECs are close enough in timing, it should
n
Matt wrote:
Hi,
With a digium dual PRI card (dual span). Is there any reason I can't
have both PRIs being PRIMARY timing sources? They are both from
different CLECs, and as such I need them both to do their own timing.
If the two CLECs had different timing, it would mean modem calls betwee
On Friday 28 April 2006 14:07, Matt wrote:
> With a digium dual PRI card (dual span). Is there any reason I can't
> have both PRIs being PRIMARY timing sources? They are both from
> different CLECs, and as such I need them both to do their own timing.
Nope. Digium cards do not support this co
Well... outbound calls via Nufone still function, so I'd guess they are
scrambling to find another cost effective source for DID's (which will
likely not be the same DIDs they had due to probable portability
issues). That would also suggest 800 numbers can't be remapped until
new DIDs are impl
Seems they have gone belly up... oh well... if you can't keep your
contracts in order, and have backups you don't deserve to be in the
game.
On 4/28/06, Tom Vile <[EMAIL PROTECTED]> wrote:
They dont answer their phone either.
On 4/28/06, Matt <[EMAIL PROTECTED]> wrote:
> > And why all these coc
Hi,
With a digium dual PRI card (dual span). Is there any reason I can't
have both PRIs being PRIMARY timing sources? They are both from
different CLECs, and as such I need them both to do their own timing.
___
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Here are the settings that I have for the Mitel and the Asterisk server, as well as logs of errors, etc. We've been chasing this issue for months now and it's getting frustrating. Any help would be appreciated!Thanks!
On 4/27/06, Geoff Manning <[EMAIL PROTECTED]> wrote:
I have a Dell PE SC420 (a no
They dont answer their phone either.
On 4/28/06, Matt <[EMAIL PROTECTED]> wrote:
> And why all these cock robin "the sky is falling" statements. Nothing
> has been interrupted yet and until it happens, no point in starting an
> avalanche from the torrent.
Well services broke. It's down.. DIDs
I have seen conflicting references in regards to the Digium Wildcard TE411P
echo settings in zapata.conf.
Does anyone have the official word on this?
Should echo cancel be enabled in zapata.conf if the card has built in EC?
If so, should a particular EC method be compiled into the zaptel build?
And why all these cock robin "the sky is falling" statements. Nothing
has been interrupted yet and until it happens, no point in starting an
avalanche from the torrent.
Well services broke. It's down.. DIDs ring fast busy.Does anyone
know the details of why nufone did not have backup provid
NENA i2
The NENA i2 architecture was designed to support the interconnection
of Voice over Internet Protocol (VoIP) domains with the existing
Emergency Services Network infrastructure. This overview will
describe the functional elements and call flow of a VoIP 9-1-1 call
over the i2 architecture
Hello,
We've released another update to our Asterisk GUI Client suite: 1.1.11
http://astguiclient.sf.net/
The client suite runs on most modern web browsers on almost any
GUI-capable operating system, and it includes the astGUIclient
client-side web app which extends your phone's functionality a
At 07:07 AM 4/28/2006, you wrote:
Has anyone every worked with these types of phones/intercoms? Or any
suggestions on getting it to work through Asterisk?
I think you'll need to have the device programmed to send the phone
number, a suitable pause and then 1, or check the caller ID to see if
In case most of us are not already busy, phishing with phones now
makes it fun.
http://it.slashdot.org/article.pl?sid=06/04/28/1347259
--
---
Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
[EMAIL PROTECTED] - [EMAIL PROTECTED]
If any of the above are down we have bigger problems than my email
Thczv F. Thczv wrote:
I have an old broken laptop that I am attempting to turn into a mostly
solid state Asterisk box, mainly because I am worried about the fans
or hard drive in my other Asterisk box failing (I already lost one
power supply). The laptop has no screen or hard drive. But it does
what's the use of enabling multiple codecs for a peer?
can asterisk avoid trancoding when both phones are capable of a common codec enabled for them?
e.g. first priority codec = g729, 2nd= ulaw,,, if phone 1 calls another
ip phone capable of g729 to use it and when calling through Zap, can
asterisk
Jonathan k. Creasy wrote:
Just appending the area code variable is not always going to be correct.
You will need to lookup (google local calling guide) the proper NPA for
the NXX you are dialing. For example, in Louisville, Ky if you dial
948-1592 you will actually reach 812-948-1592 instead of 5
In extensions.conf file, in that context, you must have:
include => featuremap
Thats lets you transfer calls.
Regards.
--
José Luis Gómez
Qualis Information Technology
Av. Rivadavia 2553, PB Of. 43 EP
+54-342-4565684 int 102
www.qualis.com.ar
Soporte 0810-8880022
Santa Fe - Argentina
El vie, 28
i used to have this problem,
with me, it appeared that i had to press the feature keys very quickly.
my solution was to set featuredigittimeout higher than the default 500.
also, when i use IAX phones, i had to set dtmf to ouband audio for asterisk to recognize the keys pressed.On 4/28/06, Ronald W
I have an old broken laptop that I am attempting to turn into a mostly
solid state Asterisk box, mainly because I am worried about the fans
or hard drive in my other Asterisk box failing (I already lost one
power supply). The laptop has no screen or hard drive. But it does
have USB and a bootabl
Carlos Alberto Bernat Orozco wrote:
I'm making tests for Asterisk. I've tested with 2 users installing SJphone
and it worked fine, but when I install it over a third user with the
softphone, the phone dial for 2 seconds and a window alert goes out on the
softphone:
Busy
Call rejected: 486 Busy H
That was it thanks...no if I could only get those lines showing busy, when they
are not, I will be a happy camper
> -Original Message-
> From: Gareth Blades [mailto:[EMAIL PROTECTED]
> Sent: Friday, April 28, 2006 7:36 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Su
nction result is '0' Apr 28 07:48:25 VERBOSE[4600] logger.c: --
Executing GotoIf("SIP/103-c10f", "0 > 0?2:4") in new stack Apr 28
07:48:25 VERBOSE[4600] logger.c: -- Goto (macro-record-enable,s,4) Apr
28 07:48:25 VERBOSE[4600] logger.c: -- Executing AGI("SIP/103
I am trying to integrate Asterisk with traditional phone central, this issue
is sometimes tough. After some testing and measuring I think what is
bothering my Asterisk; I need to dial a number digit after digit and not the
whole string, so for example:
1, 2, 3, 4, 5, 6
and not:
123456
How can I
Just appending the area code variable is not always going to be correct.
You will need to lookup (google local calling guide) the proper NPA for
the NXX you are dialing. For example, in Louisville, Ky if you dial
948-1592 you will actually reach 812-948-1592 instead of 502-948-1592
even though your
Bob,
Thanks very much for your followup summary. I think I have the same
problem here with a fresh install of FreePbx, but haven't taken the time
to troubleshoot the lack of audio problem. This system has both a digium
two-port T1 card and a TDM card installed, and had been working just
fine
Hi,
I have a lot of GXP-2000 phones not registering with Asterisk server.
After two days of attempts, it seems that problem is due to the fact that
phones and server are not on the sme network.
Do you know if this is known issue?
--
Domenico Viggiani
_
Ajit wrote:
Hi,
I am trying to use the manager API to originate a SIP call from one
asterisk extension to another. eg extension [EMAIL PROTECTED] calls
extension [EMAIL PROTECTED] this fails with a 482 "Loop
detected".
I beleive this behaviour is incorrect according to rfc3261 :
The UAS proces
[featuremap]
blindxfer => #1; Blind transfer
;disconnect => *0; Disconnect
automon => *1; One Touch Record
atxfer => *2; Attended transfer
extensions.conf of all phones I tried have the dial options: tTwWr
I try to call from one phone to the other and
Something worth checking is are you going through a firewall? I've had
problems similar to this when the firewall (A fortigate 100) had a so called
SIP helper, that somehow cached each session, and when I made changes, kept
the old session open!
Hope that helps
-Original Message-
Fro
You must activate call waiting for those extensions, this way you will get correctly voicemail busy and unavailable.From the sip extension dial *70On 4/28/06,
Johnny Stork <[EMAIL PROTECTED]> wrote:
I have a fairly new, but functional install of [EMAIL PROTECTED] 2.7 with a TDM400 (1 FXS) and T101
Forget the sound card. It isn't related. The subject above should have
read 'TE405P No Voice Problem' or something similar. It appears to be a
zaptel timing issue, but I have found a workaround. For those of you
just tuning in, here is the story:
I have a CentOS/Intel 865 box currently runnin
Joshua Colp wrote:
Please don't do product announcements on asterisk-users, that's what
asterisk-biz is for. Thanks!
Please don't send the whole announcement back to the list just to add a
line tot he bottom. Trim.
--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int: (305)
Your problem could be DTMF-related. Make sure that both your sip peer
has dtmfmode=rfc2833 and the GXP-2000 is configured for RFC2833 as well.
- Waldo
On Apr 28, 2006, at 10:20 AM, Johnny Stork wrote:
I seem to be having a problem with my GXP-2000. No matter how
carefully I type in the mail
Hi,
I am trying to use the manager API to originate a SIP call from one
asterisk extension to another. eg extension [EMAIL PROTECTED] calls
extension [EMAIL PROTECTED] this fails with a 482 "Loop
detected".
I beleive this behaviour is incorrect according to rfc3261 :
The UAS processes the first
Make sure you have the DTMF mode set to RFC.
On Fri, 2006-04-28 at 15:20, Johnny Stork wrote:
> I seem to be having a problem with my GXP-2000. No matter how carefully I
> type in the mailbox number and password when calling the mailbox (*98), it
> keeps complaining that the password is not corr
Hi,
A TERMINATION provider will do just that take your calls over
VoIP, and TERMINATE them to the PSTN.
An ORIGINATOR will provide only INBOUND service to you. (ie someone
dials a number and you get the call)
TERMINATION: YOU-->PSTN
ORIGINATION: PSTN ->YOU
Some companies only do one..
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Aaron Daniel
>
> Are you using Realtime or static sip.conf?
Static (AsteriskAtHome base setup)
Thanks
DV
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I just resolved the issue this morning, but to answer your questions:
1) There are simply no voices of any kind. No parking digits, no
greetings, no voicemail, nothing at all.
2) The condition is the same when tested via SIP and via PSTN.
3) The CLI does not issue anything abnormal, except it sa
I seem to be having a problem with my GXP-2000. No matter how carefully I type
in the mailbox number and password when calling the mailbox (*98), it keeps
complaining that the password is not correct? I can use any other phone to
check the same mailbox and it works fine, just not with the GXP-20
I have a fairly new, but functional install of [EMAIL PROTECTED] 2.7 with a
TDM400 (1 FXS) and T101P (1 FXO) hardware. For some reason the analog phone
connected to the FXS port and one SIP softphone goes straight to the voicemail
indicating "Is On the Phone" although it is NOT off the hook
On Friday 28 April 2006 09:53, Wai Wu wrote:
> I don't realy mind if the algorithm work the same. The true issue here
> is how much CPU load is it going to save me over the software echo
> cancellation.
Depends on how many active channels, the algo used in the software EC, how the
driver's compi
Am Freitag 28 April 2006 15:33, Eric "ManxPower" Wieling schrieb:
> Hans-Peter Straub wrote:
> > Hello all,
> >
> > is it possible to make an outgoing call transferable for the dialing
> > phones like the 't' or "T" option on the Dial-Command does this for
> > incoming calls?
>
> The t and T option
Hi list!I'm making tests for Asterisk. I've tested with 2 users installing SJphone and it worked fine, but when I install it over a third user with the softphone, the phone dial for 2 seconds and a window alert goes out on the softphone:
BusyCall rejected: 486 Busy HereAnd on my Asterisk server th
I have something called an "EnterPhone 2000" intercom system in my complex
which rings the phone when someone dials my buzzer number on the keypad. I can
use any Asterisk extension to anser the call and hit "6" to open the door.
However, I have tried using a "Digitial Assistant" with the message
Title: Messaggio
Hi
Franklin,
I've
downgraded the firmware to 4.5 but that didn't solve. The problem was in
sip.conf in the field "fromuser" which i set to Name Surname. If I set the
fromuser field that way it doesn't transfer or hold. If iI set it to N.Surneme (
without space ) it works.
I don't realy mind if the algorithm work the same. The true issue here
is how much CPU load is it going to save me over the software echo
cancellation.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Don Pobanz
Sent: Friday, April 28, 2006 9:46 AM
To: Ast
Alistair Cunningham wrote:
Integrics is pleased to announce version 2.0 of Enswitch, the most
integrated platform available for offering commercial telephony services
such as ITSP, hosted PBX, calling cards, call shops, number translation
services, and much more.
Enswitch was formerly known a
Are you using Realtime or static sip.conf?
On Fri, 28 Apr 2006, Mimmus wrote:
Hi,
during tests, I configured different SIP accounts on the same phone.
Now I see this 'sip show peers output':
Name/username HostDyn Nat ACL Port Status
259/25910.97.1.19 D 5060 O
Integrics is pleased to announce version 2.0 of Enswitch, the most
integrated platform available for offering commercial telephony services
such as ITSP, hosted PBX, calling cards, call shops, number translation
services, and much more.
Enswitch was formerly known as ITSP in a box, and Enswitc
Wai Wu wrote:
One question thought, does
the hardware echo cancellation work much better than software?
I bought a Digium TE411P hoping the hardware echo canceler would improve
my echo problems over the software echo canceler, but had no performance
improvement. Since then I have heard that
Hans-Peter Straub wrote:
Hello all,
is it possible to make an outgoing call transferable for the dialing phones
like the 't' or "T" option on the Dial-Command does this for incoming calls?
The t and T option works for ANY call using Dial. Incoming or outgoing.
--
Now accepting new clients
What does the "R" option do?
Benoit Panizzon wrote:
After migrating from 1.2.4 to 1.2.5 I noticed that:
show application dial
does not show the 'R' option anymore. Has this become an undocumented feature
or has it gone completely?
--
Now accepting new clients in Birmingham, Atlanta, Huntsvi
Alejandro Vargas wrote:
>2006/4/27, Sarafoleanu Catalin <[EMAIL PROTECTED]>:
>
>
>>Hello folks!
>>
>>I'm trying yo set up a email2fax and fax2email on my asterisk box.
>>The rxfax works fine in my setup.
>>
>>
>
>I had good results using iaxfax + hylafax. I receives the faxes and
>converts t
Hi Klaus,
Please see the following document.
[Diva Server Adapter Installation Guide]
http://www.eicon.com/pubs/20319511.pdf
Page 24 (back-to-back cable pin layout for BRI interface)
David
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Klaus
Darilion
Sen
is it possible to make an outgoing call transferable for the dialing phones
like the 't' or "T" option on the Dial-Command does this for incoming calls?
This is exactly what the option T does.
't' -- allow the called user transfer the calling user by hitting #.
'T' -- allow the calling
Hello all,
is it possible to make an outgoing call transferable for the dialing phones
like the 't' or "T" option on the Dial-Command does this for incoming calls?
Does someone have any idea?
Thanks
Hans-Peter Straub
--
---*
I-NetPartner GmbH
Hans-Peter Straub
S
Hi
After migrating from 1.2.4 to 1.2.5 I noticed that:
show application dial
does not show the 'R' option anymore. Has this become an undocumented feature
or has it gone completely?
Mit freundlichen Grüssen
Benoit Panizzon
--
I m p r o W a r e A G-System Services
__
Remco Barende wrote:
I guess that I'm the only one experiencing this problem is there
any way to debug this problem?
Does anyone know how to debug this particular item in *? (Or should I
open a bug in Mantis?)
Thanks!!
Luki's response is the most likely cause. I would suggest followin
OK, thanks, I will see if I can convince my asterisk to load those
modules -- for some reason which has something to do with my use of
freepbx asteriskwon't load such things.
on Friday 04/28/2006 Steve Feinstein([EMAIL PROTECTED]) wrote
> Once you plug it in, the computer sees it as a speaker, an
2006/4/27, Sarafoleanu Catalin <[EMAIL PROTECTED]>:
> Hello folks!
>
> I'm trying yo set up a email2fax and fax2email on my asterisk box.
> The rxfax works fine in my setup.
I had good results using iaxfax + hylafax. I receives the faxes and
converts this to pdf for sening via e-mail.
--
Alejandr
Why not just create a .call file when the number is busy? The .call
file tries to dial the destination with the retry interval and max
attempts you specify, when the call goes thru, dial that other number.
Nathan Alberti wrote:
8<
Thanks for the pointer Nathan. I slapped something together
Hi all,I have a requirement:Calls arrive my Asterisk server, and are forward to Call queues.Currently i would like that my callers instead of just listenning Music on Hold they could listen an IVR with questions and answers, while they stay in queue.
This way i could have users that find their answ
Once you plug it in, the computer sees it as a speaker, and mic. So if
you can get OSS or ALSA to use it, you can use it just as you'd use any
sound card in asterisk.
Kerry Garrison wrote:
I use a softphone
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