Hi, all. I'm glad I put asterisk and hylafax
togetherjust like PSTN-Asterisk-Hylafax-Email.And the
fax2email functionworks well.
But I also find some bugs about CID number.
I use TE405P as gateway and Eicon PRI card as fax
card.
When I receive the caller number from PSTN, I
Hello,
from what I see, I guess they're only ways to insert a piece of speech
without recording it; you could easily record the phrases yourself and add
Playback()s instead.
BTW, I'd like to thank Tim for sharing his recipe with us. Anybody else's
got a recipe to share? :-)
l.
In data Sat,
This question will be better addressed on the aah forums.
I would suggest:
1) have you setup a DID?
2) take a look in the log file
tail -f /var/log/asterisk/full
3) see the numerous threads on the aah forums about how to configure FWD
and Teliax (and other providers)
I personally have both FWD
On Sat, 30 Apr 2005, Ma Zhiyong wrote:
I use TE405P as gateway and Eicon PRI card as fax card.
When I receive the caller number from PSTN, I found it was 51863500. While I
dial the FAX trunk, FaxGetty get the caller number 051863500.
-- Executing NoOp(Zap/124-1, 51863500) in new stack
Me wrote:
Hi all,
I'm trying to use spandsp and asterisk to send faxes. To do so I am
creating tiffs with Ghostscript. When I use Ghostscript 6.50 it seems
to work fine, but when I create the tiff using Ghostscript 8.51 (or
7.06) txfax garbles the tiff and it comes through all messed up.
First of
Hi,
Citeren Sander [EMAIL PROTECTED]:
Please can anyone help me with my quadbri card
--
Modprobe zaptel
Insmod qozap.ko
Ztcfg
The it complains it can't find
ZT_SPANCONFIG failed
Sendmail isn't really that hard to configure for simple stuff like this.
Most Linux distros have /etc/mail/sendmail.mc, so set your smart relay host
and the appropriate masquerading options - the options for these are spelt
out in the sendmail.mc file. If you want to receive bounces then also set
Sounds like a wonderful idea!
I can tell you from personal experience that the performance of Asterisk and
its stability are in a one-to-one relation to the hardware that you're
using. We've been using mostly Intel boards for Asterisk, mainly the
ClearWater (XEON) and TorryPine (P4) boards for
Anybody has some command line examples on how to run sipp against asterisk?
..
I tried using sipp -sn uac 127.0.0.1 and I get
Apr 30 02:14:14 NOTICE[3619]: chan_sip.c:8361 handle_request_invite: Failed
to authenticate user sipp sip:[EMAIL PROTECTED]:5061;tag=62
Question: how can I block someone from calling us?
Sometimes we get crank calls into our office. We'd like to build a list
of callers to be blocked. When they call, they should hear busy and
then we hang up. We have about 100 DIDs routed to different contexts
and I wouldn't want to have to
From what I've read, this is a H.323 phone only. Only the 4602 has SIP
images. Has anyone gotten a 4610 H.323 working with *?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Hi,
I am looking for a way to dynamicly put phones in a group so if someone
calls an extentions everyone's phone who's member of the group will
ring.
Queues are not an options because as soon a call comes in to a queue
there is no getting out.
I want to let the phones ring and after a period of
On Fri, April 29, 2005 4:58 pm, GEOFFREY SACHS said:
I would also be interested in alternatives to the Tdm400p. I have had endless
problems with a tdm400p card not being able to get the zttest numbers above
99.975 and as a result not being able eliminate an intermitent but consistent
echo.
I
On Fri, April 29, 2005 4:58 pm, GEOFFREY SACHS said:
I would also be interested in alternatives to the Tdm400p. I have had
endless
problems with a tdm400p card not being able to get the zttest numbers above
99.975 and as a result not being able eliminate an intermitent but
consistent
It looks like it registers:
asterisk1*CLI iax2 show registry
Host UsernamePerceived Refresh State
208.139.204.228:4569 memy.external.ip:154660
Registered
65.39.205.121:4569me my.external.ip:154660 Registered
But the
Hello,
They are being rejected because the extensions (your DIDs) do not exist in
the context from-pstn. How did I know? I read the error ;)
- Joshua Colp.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Patrick M.
Gray, Jr.
Sent: Saturday, April 30,
On Sat, April 30, 2005 10:52 am, Rich Adamson said:
On Fri, April 29, 2005 4:58 pm, GEOFFREY SACHS said:
I would also be interested in alternatives to the Tdm400p. I have
had endless problems with a tdm400p card not being able to get
the zttest numbers above 99.975 and as a result not being
D-oh!!! I told you I was missing one simple piece of the puzzle. Light
finally dawns on marble head.
Thanks for your help (with the obvious).
Pat
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joshua Colp
Sent: Saturday, 30 April, 2005 10:05
To:
Inline
It looks like it registers:
asterisk1*CLI iax2 show registry
Host UsernamePerceived Refresh State
208.139.204.228:4569 me my.external.ip:154660
Registered
65.39.205.121:4569me my.external.ip:154660
Joris Vandalon wrote:
Hi,
I am looking for a way to dynamicly put phones in a group so if someone
calls an extentions everyone's phone who's member of the group will
ring.
Queues are not an options because as soon a call comes in to a queue
there is no getting out.
I want to let the phones ring
I would think what you would need to look at is how to do this with the *
Data Base. I haven't done this, but it would seem that there is a way to
make it work with that.
Joel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Robert P.
McKenzie
Sent:
If it references anything that is not in the default asterisk - then it
came from asterisk at home. I just looked again - it uses the festival
text to speech engine to say the words - record those messages and then
use playback(filename) instead of AGI(festival-script.pl|words to say)
for
Hows does this look?
Opened pseudo zap interface, measuring accuracy...
8192 samples in 8192 sample intervals 100.00%
8192 samples in 8193 sample intervals 99.987793%
8192 samples in 8193 sample intervals 99.987793%
8192 samples in 8193 sample intervals 99.987793%
8192 samples in 8193 sample
Polycom IP500 Forward problem codec issue
All,
Im running the Polycom IP500 phones at several sites. My * server is
at a collocation site and I have complete control of the T1s running to
the remote sites with the IP500 phones. Connectivity to the PSTN is
through a Cisco 2600 with a PRI
Andy
How did the 7910 worked with skinny under *? Did all the keys on the phone
worked? Ive seen sometimes the forward key or something does not fully do
what you would excpect.
What are the drawbacks from using skinny vs sip under *?
|-Original Message-
|From: [EMAIL PROTECTED]
webmin works good for configuring sendmail too if you are not that
familiar with the sendmail 'mc' files...
http://www.webmin.com
On Sat, 2005-04-30 at 00:37, Craig Guy wrote:
Sendmail isn't really that hard to configure for simple stuff like this.
Most Linux distros have
On Sat, 30 Apr 2005, Joris Vandalon wrote:
I am looking for a way to dynamicly put phones in a group so if someone
calls an extentions everyone's phone who's member of the group will
ring.
One way is to place the logic in an agi script. It can then dial all the
current members of the group
The way that zttest is written makes it a little difficult to
interpret, but it essentially means that zttest tried to read
8192 bytes from the TDM card, and it took more then 1 second to
do it (the objective is exactly 1.0 seconds, or 100%).
The 99.987 numbers says it took something like
Mmhh let me try that. Thx!
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Kevin P. Fleming
|Sent: Jueves, 28 de Abril de 2005 11:02 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Queues
Weird..
I also have joinwhenempty=no and user can still go into the queue without
any agents logged in.
Any ideas? Im using cvs head
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Kevin P. Fleming
|Sent: Jueves, 28 de Abril de 2005 11:02 a.m.
Citeren Robert P. McKenzie [EMAIL PROTECTED]:
It may not be exactly what you are after but I do something like this:
extensions.conf
HOUSEPHONES=SIP/somepcSIP/anotherpcIAX2/desktopIAX2/someotherdesktopSIP/sipuraline1SIP/sipuraline2
; London Number - SIP Inbound provider
exten =
Im using RH9 and celerom 1.7 with 256 Mb RAM
Can you give me the detailed math on your calculations?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Rich Adamson
|Sent: Sábado, 30 de Abril de 2005 11:07 a.m.
|To: Asterisk Users Mailing List -
Look at the ms = statements in the code. I'm trying to rewrite
the code right now to provide something more useful.
Im using RH9 and celerom 1.7 with 256 Mb RAM
Can you give me the detailed math on your calculations?
|-Original Message-
|From: [EMAIL
Version 0.111 - 30. April 2005.
* Security added, you can now specify what the user of IPS is allowed to do
such as start different programs, hang-up calls etc.
* Many bug fixes
Download: http://ipswitchboard.thorben.dk
I just had a very successful installation of Asterisk and have a
question. On my 7910's using the Skinny protocol, the user does not
hear ringing when they make another call. I found a patch that makes
the ringing work, but something is still wrong with it. If I use the
7910 to make
What type of phone SIP or analog? What is your DTMF type set for?
It's a system phone, via PBX to a PRI to Operator to my SIP Provider to my
Asterisk box.
Sip.conf is
[general]
context=default
port=5060
bindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=alaw
allow=gsm
;allow=ilbc
realm=asterisk
what is the url for the version of the framework it wants now ?
the dot net auto installer is busted ?
- Original Message -
From: Thorben Jensen [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Saturday, April 30, 2005
On Fri, Apr 29, 2005 at 01:44:24AM +0200, Sebastian Voitzsch wrote:
I can?t get chan_capi to work with any version of asterisk. I tried several
versions, all with the same effect: the phone rings, as soon as the call gets
answerd, asterisk crashes.
Certainly it is chan_capi 0.3.5, but which
You will find the URL on my download page
thorben
TC [EMAIL PROTECTED] skrev i en meddelelse
news:[EMAIL PROTECTED]
what is the url for the version of the framework it wants now ?
the dot net auto installer is busted ?
- Original Message -
From: Thorben Jensen [EMAIL PROTECTED]
To:
Hi,
Is this a known problem with Grandstream Budgetone 100, I could see
several people complaining about this but no answers.
Details
--
1) I am simply trying to go from one SIP extension to another, so the
zapata.conf and zaptel.conf entries are irrelavant.
2) I added a
Hi,
I was looking for solutions for simple FXO cards, and came across the
two modem channels in the asterisk channels/ dir, i assume they are
there becuase someone made these two types of modems work as FXO (or are
they there for other purpose ?),
does anyone have any info on these channels ?
Forgive me if this has been asked before, I wasn't able to find any
clear answers in the archives.
Will the Intel 536EP function as a FXO? And if so, do I need to use a
different version of the Zaptel driver?
Any assistance would be great.
PS - that's 536EP, not 537EP.
Thanks!
I was wondering if there was a way to have incoming
calls to my PSTN line be transferred to a voip line?
I would like to make it so that as soon as the pstn
call is recieved it will switch the call to the voip
line, thus freeing up the pstn line to get more calls.
Kind of like roaming.
Tom
I was wondering if there was a way to have incoming
calls to my PSTN line be transferred to a voip line?
I would like to make it so that as soon as the pstn
call is recieved it will switch the call to the voip
line, thus freeing up the pstn line to get more calls.
Kind of like roaming.
If
Im running Asterisk 1.0.7. Ive checked
out using cvscheckoutasterisk-addons.
When I make install I get the following
errors:
app_addon_sql_mysql.c:162:36: macro
AST_LIST_REMOVE requires 4 arguments, but only 3 given
Im using the default FC3 mysql:
mysql-server-3.23.58-16.FC3.1
Hi, Is there a script in amp for adding the extensions? And can it be
modified? When adding a new extension it rewrites all of the
information it the context blowing out my additions.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hi, Is there a script in amp for adding the extensions? And can it be
modified? When adding a new extension it rewrites all of the
information it the context blowing out my additions.
You my want to try the AMP forum. Since they are the producers of AMP,
they may have a little better info.
I was wondering if there was a way to have incoming
calls to my PSTN line be transferred to a voip line?
I would like to make it so that as soon as the pstn
call is recieved it will switch the call to the voip
line, thus freeing up the pstn line to get more calls.
Kind of like roaming.
Anton:
I'll be able to get back to you Sunday night about specifics; the
phone is not where I am right now. Using chan_sccp, (I think November
2004 or so CVS Head) I know I can receive calls, place calls, etc. It
is a rather low volume phone, so I don't know off hand about specific
keys; I'll
in app_voicemail.c in the function vm_exec set the tmp[256] to be
tmp[4096]
Chris Stinson wrote:
I have one with 33. but I can't get the voicemail to copy to more than
20 mailboxes.
Eric Wieling aka ManxPower wrote:
Has anyone set up Group/Broadcast Voicemail for 50 or more mailboxes?
--
On Sat, 2005-04-30 at 13:23 -0400, Time Bandit wrote:
I was wondering if there was a way to have incoming
calls to my PSTN line be transferred to a voip line?
I would like to make it so that as soon as the pstn
call is recieved it will switch the call to the voip
line, thus freeing up
Well for some reason, you decided to use the stable version of asterisk but
also decided not to use the stable version of addons. Hmm...interesting
decisions.
rm -rf asterisk-addons/
cvs co addons -r v1.0.7
Then it will work.
-Matthew
From: forums [EMAIL PROTECTED]
Reply-To: Asterisk Users
Asterisk-java 0.1 a Java control for the Asterisk PBX has been released.
The Asterisk-java package consists of a set of Java classes that allow
you to easily build Java applications that interact with an Asterisk
PBX Server. Asterisk-java supports both interfaces that Asterisk
provides for this
Joseph wrote:
On Sat, 2005-04-30 at 13:23 -0400, Time Bandit wrote:
I was wondering if there was a way to have incoming
calls to my PSTN line be transferred to a voip line?
I would like to make it so that as soon as the pstn
call is recieved it will switch the call to the voip
line, thus freeing
Amit Sharma wrote:
Hi,
Is this a known problem with Grandstream Budgetone 100, I could see
several people complaining about this but no answers.
Details
--
1) I am simply trying to go from one SIP extension to another, so the
zapata.conf and zaptel.conf entries are irrelavant.
2) I
Hi All,
For some time now I've had issues with ringing voltages on my TDM400P. Numerous
folks have told me that using modprobe wcfxs boostringer=1 when loading the
module will force the driver to use boosted ring voltage. For some strange
reason this has never worked for me. Today I got
Hi All, i´m new in this list.
I have an Asterisk behind a firewall with forwarded ports
and my SIP clients is X-Lite.
In local connection we dont have problem, and the
same with VPN connection. All work fine.
But when I try to connect to * from Internet or from others
LANs, the
I have a problem. The average person is too freaking stupid to use a VOIP
phone. My experience has so far been that if it doesn't have 20 buttons with
little red LED's on it, the user cannot comprehend call parking, attended
transfer, blind transfer, DND, and navigating through a voicemail
Jason Brown wrote:
So seriously does anyone have a recommendation for a good receptionist phone? I tried the Snom today and I can't get the programmable buttons to do this, even by following the manual. So please, any suggestions would be great, before I get fired at my dayjob for everyone else's
On 4/30/05, Jason Brown [EMAIL PROTECTED] wrote:
I have a problem. The average person is too freaking stupid to use a VOIP
phone. My experience has so far been that if it doesn't have 20 buttons with
little red LED's on it, the user cannot comprehend call parking, attended
transfer, blind
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Jason Brown wrote:
I have a problem. The average person is too freaking stupid to use a VOIP
phone. My experience has so far been that if it doesn't have 20 buttons with
little red LED's on it, the user cannot comprehend call parking, attended
Jason Brown wrote:
I have a problem. The average person is too freaking stupid to use a VOIP
phone. My experience has so far been that if it doesn't have 20 buttons with
little red LED's on it, the user cannot comprehend call parking, attended
transfer, blind transfer, DND, and navigating
So seriously does anyone have a recommendation for a good receptionist phone? I tried the Snom today and I can't get the programmable buttons to do this, even by following the manual. So please, any suggestions would be great, before I get fired at my dayjob for everyone else's idiocy.
How many
Hi,
Citeren Jason Brown [EMAIL PROTECTED]:
I need a good receptionist phone that works with Asterisk. It basically needs
to act like an avaya partner phone, I don't need 20 buttons with little red
LED's...what I do need is for the phone to register multiple extensions to my
asterisk server
Hi,
Citeren Michael Welter [EMAIL PROTECTED]:
In a multi-tenant environment, is there a way to display, on the phone,
which DID (which tenant) is being called?
We use the callerID name for that purpose.
Florian
___
Asterisk-Users mailing list
Not sure what you mean exactly... Can you give me a hint?
Private Label Wholesale Internet Access!
http://www.YourOwnISP.com
- Original Message -
From: Michael D Schelin [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent:
Someone to know how can I send a DTMF after the channels are bridged?
I need something like the D option of the Dial application, but this
option sends the DTMF before the channels are bridged. In fact I want the
caller and the callee to receive the DTMF. Please help :)
Thank you for the detailed description Andy.
Please let me know how about the specs when you can.
My client has legacy 7610 but I am trying to suggest swithcing to native sip
phones like grandstream or better in order to make everything 100% asterisk
compliant.
Plus, Cisco charging for the sip
:-) ok... so I feel foolish... Mathew thanks a lot, worked like a charm.
Jim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm
Sent: Saturday, April 30, 2005 11:16 AM
To: Asterisk Users
Subject: Re: [Asterisk-Users] help with compiling
Hi All,
I am replacing Cisco Call Manager with Asterisk. As you know CCM
is on a MCS 7835 Server which comes with a custom version of
Windows. Does any one know how to install Linux on that H/W. My
guess is that someone must have tried the same thing before. I
know how to install Linux however I
Guys.
Anybody knows of asterisk compliant cdr software for Hotel that will let you
enter diff. rates, checkin and out that will create the extension and setup
voicemail for the room, etc?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
If a person parks a call, the call hits the
timeout exten for that context after the park expires.. Is there any way to
make it ring back to the person who parked the call instead of using the
timeout?
___
Asterisk-Users mailing list
Can you show us an example of using the callerID for this purpose?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Florian
Overkamp
Sent: Saturday, April 30, 2005 3:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Like this:
[dids]
Exten = 2145550001,1,dial(SIP/6001)
Exten = 2145550002,1,dial(SIP/6002)
Exten = 2145550003,1,dial(SIP/6003)
Include = default-did
[default-did]
Exten = _.,1,dial(SIP/6000)
Seems pretty simple. I used this method of least/highest cost routing
to choose my
Take a look at the Polycom IP 600
I just added one to my desk as a test unit, I can't image you would need
anything more. We have Mitel 4015/4025/Superset for the office pbx I will be
replacing with *, and the Polycom 600 is a much better unit by far.
Chris Mason
www.anguillaguide.com
Kerry Garrison from The Geek Gazette (http://geekgazette.com) will be
interviewed tonight on Mick Mick Williams' Cyber Line radio program at
9:00PM PST. The show is broadcast on the USA Radio network. If you do not
have a channel in your area, you can listen listen live online
Hi,
I am a new user of Linux and Asterisk. I bought Digium TDM400P card and now
want to setup my dial plan. With some help from the suggestions given online
I have been able to configure the two SIP phones to interact with each
other.
I want to use this to call on to a Telecom line(PSTN) and
I have installed Asterisk on a CentOS4 box and then installed Asterisk from
CVS.
I installed a Sangoma A101 and connected it to a Adtran 600 using a T1
Crossover cable. The 600 has 12 x FXS, 12 x FXO interfaces.
I ran through the wanpipe install instructions and configured it, now I can
run
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Michael Welter
Sent: Saturday, April 30, 2005 12:53 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] A good SIP receptionist phone
I have two asterisk boxes. I'm running an IVR script in one of them and
I have agents registered on the second box.
I wish to create an extension on the * box where the agents are
registered, so that when dialed, it will connect the agent to the IVR
script on the other * box. However, I'd like
I was reading on the wiki about the supported kernels and I __THINK__
the main issues with the kernel versions have more to do with Zaptel
driver and not necessarily Asterisk itself. Is this correct?
Thanks,
Daniel
___
Asterisk-Users mailing list
Dear all,
I'm trying to get asterisk to register to budgetphone.nl. In several threads
I saw people who got this to work:
http://www.voip-info.org/wiki-Talkin2ya
http://lists.digium.com/pipermail/asterisk-users/2005-March/092850.html
But I've spent a whole saturday on it now and didn't get any
On 01:11, Sun 01 May 05, Bert Haverkamp wrote:
Dear all,
I'm trying to get asterisk to register to budgetphone.nl. In several threads
I saw people who got this to work:
http://www.voip-info.org/wiki-Talkin2ya
Hi,
The directions on the page there are working like a charm.
The one who made
I believe it is Zaptel only that becomes a problem. I'm running asterisk on a
2.6 kernel... the only concession I had to make was to use make linux26 when
I compiled Zaptel.
Thanks,
Ian
[EMAIL PROTECTED] 30/04/2005 19:10
I was reading on the wiki about the supported kernels and I __THINK__
Guys.
I have some dialing rules defined for my internal extensions but I am now
defning a call forward option that allow an extension to be forwarded to an
outside number, right now Im using Dial cmds but I was wondering if ther is
a way to do this but using the dialing rules that I have also
I understand and I guess I know how to do that within a single box.
If I have the following:
Asterisk Box 1 (no agents)
extensions.conf
[test-ivr]
exten = s,1,AGI(play_ivr)
exten = s,2,Hangup
Asterisk Box 2 (agents register)
extensions.conf
[agents-context]
exten = 1234,1,Dial(?)
exten =
Guys.
I just programed a feature that allows any extension to be forwarded to any
outside number, for example, forward your extension 201 to any number
outside (via zap) so that if somebody calls your extension either from
inside out outside (using another zap we have) it gets directed.
Problem
Asterisk Box 2 (agents register)
extensions.conf
[agents-context]
exten = 1234,1,Dial(SIP/[EMAIL PROTECTED])
exten = 1234,2,Hangup
Asterisk Box 1
Sip.conf
[ab1]
type=friend
host=ip of ab2
context=incoming
canreinvite=yes
qualify=yes
extension.conf
[incoming]
Exten = 1234etc...
-Original
On April 30, 2005 02:56 pm, Ian Pattison wrote:
For some time now I've had issues with ringing voltages on my TDM400P.
Numerous folks have told me that using modprobe wcfxs boostringer=1 when
loading the module will force the driver to use boosted ring voltage. For
some strange reason this
On April 30, 2005 10:23 am, Kim Culhan wrote:
If so, what do see if you run 'vmstat 1' and let it run for about
twenty seconds? Do you see the cpu utilization going to about 100%
every five or six seconds?
Negative:
That's interesting; so that can potentially narrow the problematic code
Shady wrote:
Someone to know how can I send a DTMF after the channels are bridged?
I need something like the D option of the Dial application, but this
option sends the DTMF before the channels are bridged. In fact I want the
caller and the callee to receive the DTMF. Please help :)
If using a
Anton Krall wrote:
Problem I have is that if somebody using a cel phone calls in and gets
directed to my extension which in turn is directed to my cel phone, the call
comes thru but after 2 seconds, the call gets all garbled and with a sound
like b and the caller cant be heard anymore.
Jejejejeje I didnt know how to put the sound it does... Its like an
intermitent sound like when you are to lose a cel phone connection.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Matt Riddell
|Sent: Sábado, 30 de Abril de 2005 07:15 p.m.
|To:
Jason Brown wrote:
I have a problem. The average person is too freaking stupid to use a VOIP
phone. My experience has so far been that if it doesn't have 20 buttons with
little red LED's on it, the user cannot comprehend call parking, attended
transfer, blind transfer, DND, and navigating
Anton Krall wrote:
Jejejejeje I didnt know how to put the sound it does... Its like an
intermitent sound like when you are to lose a cel phone connection.
Ah...and the cellphone has range?
Normally I would say that seeing as you normally hear in on a cell phone
it would likely be that end, but
Hello Sean,
I thought the Polycom's had some kind of BLF Feature don't they?
I am thinking of getting two of them, so it would be nice to know,
otherwise I would get 2 more 7960's. (which are great phones)
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
I know I saw something about not using GSM codecs when on cell phones, could
this be the case?
The 2 second delay, well unfortunately all cell's have about a .5 second delay
on their own, so that may be what you are hearing. You just need to learn how
to talk like you are on an international
I just made an extension 390 that calls my cell, so people can hold,
then send to 390 and hangup.
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Saturday, April 30, 2005 7:47 PM
To: 'Asterisk Users Mailing List - Non-Commercial
Curious,
How did you do the forward? Was it a script or programming in C?
Any output from debug?
Race The Tyrant Vanderdecken
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Saturday, April 30, 2005 8:00 PM
To: 'Asterisk Users Mailing
:)
No problem dialing another cell phone from asterisk or incoming from cel
phone, etc.
Console says nothing.
The forwarded call is been directed using zap (x100)
So nothing looks wrong... But still...cant figure out why forwarding the
call to a cel phone via zap gets those weird sounds after 2
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