Matthew T. O'Connor wrote:
Kristian Kielhofner wrote:
Matthew T. O'Connor wrote:
Any Ideas?
Have a look at /etc/asterisk/sip_notify.conf look for:
[polycom-check-cfg]
So, from the CLI:
asterisk -r
sip notify polycom-check-cfg [name]
Isn't sip_notify.conf just an Asterisk 1.2 thing?
folks,
I am doing an install of AMP from the AMP PDF file. I get to the
part where I need to install the addons and I get the folloing error on
a make. I have done a make clean before I did a make. I can
see the errors, common.o is in place as far as I can tell.
format_mp3.so is no where to be
Kristian Kielhofner wrote:
Matthew T. O'Connor wrote:
Isn't sip_notify.conf just an Asterisk 1.2 thing? I'm running
1.0.9. I'm trying to setup a production system for my company, do
you think 1.2 is ready for that?
It sure is! You should be testing it! :) Test it and see, but
1.2
Hi,
I want to do FAX through Asterisk with the following
scenario:
Fax Machine --Nortel PBX --- E1 (euro-isdn)
--- Asterisk - SIP -Asterisk E1 (euro-isdn)-Nortel PBX-- Fax Machine
Is there anyone who can help me to configure the above
scenario
Hi Nahid,
I think youll want a fax on-ramp and off-ramp on your asterisk boxes instead
of trying to send a fax using VoIP (SIP). I believe it is possible but not
recommended. There are technical reasons for this that you can find online
in many places.
Basically asterisk answers the fax and
Hi,
I'm sorry about the false information.
It seems after the crash, the problems is still exist.
Anyone can help me? Could it be IRQ issue?
Here is output from cat /proc/interrupt:
CPU0
0: 92807252 XT-PIC timer
1: 8 XT-PIC i8042
2: 0
hi frank,
i was able to find gmake at /usr/sfw/bin, however, i got this new error :
gmake[1]: Leaving directory `/export/home/fst/ice/cvs/asterisk/stdtime'
cd editline unset CFLAGS LIBS test -f config.h || ./configure
creating cache ./config.cache
checking for gcc... gcc
checking whether the
Dear friends ,
Through the a bit more probeing i found out that
we need to use the username and the password
which we have given in the manager.conf
as the parameters to the perl script file .
even though i get the error messages as follows
[EMAIL PROTECTED] root]# ./a.out -h localhost
Damon Estep wrote:
I did not see an actual error message in your first post, what is the
error message?
Damon,
Well, it is not a 'real' error message, asterisk logs it as a
'warning' , but for me it looks like it is linked to the problem. See
my comments in the logs between [ ].
--
On Sun, 2005-08-28 at 12:45 -0700, Geoff Karl wrote:
If you are using Async and the action ID for some reason the Event:
Newstate doesn't respond with the ActionID, but only a automatically
generated Uniqueid.
When using Async you receive an OriginateSuccess or OriginateFailure
event.
These
Hi,
I'm using a HFC card on my asterisk box. I tried to make a conference
but it doesn't work. I read on internet to use ztdummy but my server has
no uhci (only ohci but it doesn't work) so I cannot use it. I tried
zaprtc but after loading the module (it appears when typing lsmod)
nothing has
Hi all,
Has anybody used this card (Digi QuadMicro) with asterisk or can anybody
tell me the likelyhood of it working out OK?
I need a multiport BRI adapter for use with asterisk in Japan and this card
seems to support INS64 (Japanese BRI standard) and also CAPI 2.0.
here is a link to the
Hi,
you are right!!!
I tried zaprtc but even if it doesn't give me errors and I loaded it as
a module, it is not working: with or without is the same, conference
doesn't work with asterisk and HFC card.
Giorgio
--
GIORGIO
Hi
It looks to me that the intel board is the same as the dialogic board.
Clive
On 29 Aug 2005 at 11:43, Mick Hastings wrote:
Hi All,
I currently run asterisk in our office (in Japan) and use a cisco PRI
gateway for connection to the PSTN. I would like to setup some more systems
for
Hi,
My variant is standard ITU, I tried almost all versions I could put my hand on
to no avail.
I tried also to profile the channel and related libraries to no avail as my
profiling skills on linux are abit lacking.
If anybody with this problem and knows how to profile multithreaded apps on
Hi,
is there any support for include statement in the database when using
realtime configurations? I would like to have as much as possible
configuration in my postgres db but we have different access controls
for different user contexts (allow international, national etc). Today
we have
Hi all,
i've a problem receiving faxes. I'm using AMP and i hope that all work well
without big changes. However i've done some tests on .tif file created by
asterisk and i've noticed that it truncates my fax almost after 5-6 seconds.
As results my pdf are corrupted and i receive a mail with
I have tried everything. to register with this gatekeeper to make and
receive calls
These are the details I received from the voip provider:
protocol H.323
Gatekeeper Address - [EMAIL PROTECTED]
Port - 1719
RAS - 53
Q931 - 80
h245 - 1722
RTP - 1722
Username - H323
I have 2 phone
I want to dial for example 1* to set a different peer
Ie:
;1* gives:
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],,tT)
;2* gives:
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],,tT)
How can I do this?
Regards,
Arne Morten
___
--Bandwidth and Colocation
Hi All,
Just wondering if anyone has managed to get line presence working on the
7 indicator lights on a grandstream gxp-2000 with asterisk? If so, what
is the trick? :)
I have simple presence working with my polycom phones but cant seem to
get it working with the gxp-2000 - is it available
Ok. I figured it out.
exten = _2*X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],,tT) ;
-Opprinnelig melding-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Arne Morten
Johansen
Sendt: 29. august 2005 11:21
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne:
Hi list!
Our CRM app is creating call files for outgoing calls which is working
great I just have one problem.
I am using this as my call file:
Channel: SIP/228(my phone)
MaxRetries: 0
Context: from-internal (the context to dial from)
Extension: 003120531234 (the phone number)
Hi Steve,
Your [general] section looks fine.
In the [register] section remove everything else and leave these lines.
context=incoming-h323-calls
alias=HMA0200.10szxn-
alias=22xx2912
alias=HMA0200.10szxn-
alias=22xx2913
Now all H.323 calls will enter in 'incoming-h323-call' context.
Hi Clive,
Thank you for your response to my posting.
It looks to me that the intel board is the same as the dialogic board
Can you please tell what that means? I haven't worked with any BRI cards
before so I don't know if it's a good thing or a bad thing.
Is / was the dialogic board
Title: Message
I'm assuming no apps/scriptsexist which completes
this?
Can
someone please confirm thatif I use a FQDN in sip.conf for my external IP,
the FQDN is only resolved at the time of loading, therefore if my IP changes
after sip is loaded, I will have to manually reload
Hi,
I changed my phones settings to inband and then
changed and then changed the settings in sip.conf to
dtmfmode=inband. It didnt work. I tried rfc and sip
info method too. I dont think its a problem with the
phone because audio works perfectly when I am leaving
a message, the problem is playing
Hi
I am using a Lucent MAX TNT to terminate 11 PRIs and
using a single Asterisk box to handle all calls
--- Andrew Thrift [EMAIL PROTECTED] wrote:
We have the ability to do this on a large scale, but
want to do it on a
smaller scale for 1 to maybe a maximum of 5 TNT's.
Andrew Thrift
Hi
no i write this application for my custom needs, but
anybody of you can use it or customized it according
to your needs
cheers
--- Matt Riddell [EMAIL PROTECTED] wrote:
Gulzar Hussain wrote:
Hi All
I just completed a custom application for Asterisk
(i
m not a C guru so i just
I'd suggest turning off echotraining on the FXS altogether, and perhaps
even
killing the echocanceller on FXS entirely. (you won't be getting
significant
echo from the FXS, and the FXO should be handling it anyway) --
echocancelwhenbridged might be an interesting thing to play
On Mon, 2005-08-29 at 01:23 +0200, Goran Dj. wrote:
Dialtone detection should be an option in .conf for zap channel, i agree
with that.
Are you trying to play with the case where you have an analog phone
bridged on your fxo line, and detect the lack of dialtone when
someone is using that
The firmware on the phones is version 3.1.3(a). I will try today using the
3.1.4 firmware. The size of the display could be better, but the lack of a
backlight is what really bothers me.
On Sunday 28 August 2005 11:46, John Novack wrote:
I have not experienced that problem, but earlier
I tried changing the gain settings and also the volume settings in the User
tab, Audio Volume section. I didn't notice any change in the microphone
output volume.
On Sunday 28 August 2005 18:20, Rob Lith wrote:
In Admin/Advanced have you tried the Handset Input Gain: settings?
Rob
On
Should it be in half duplex or full duplex?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Riddell
Sent: Sunday, August 28, 2005 11:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IAX2 Softphone
Hi,
Hadi Jadallah wrote:
My variant is standard ITU, I tried almost all versions I could put my hand on
to no avail.
I tried also to profile the channel and related libraries to no avail as my
profiling skills on linux are abit lacking.
If anybody with this problem and knows how to profile
Oh meaning it won't work w/ a Cisco? :-)
-Original Message-
From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
Sent: Thursday, August 25, 2005 11:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] HDLC/Zaptel/Kernel 2.6.11(.9)
Matt Schulte
yes thats one issue the other issue is that sometimes the pstn line is dead due
to some technical problems so people trying to make calls will just listen
silence and they'll never know whats going on...
-- Original Message --
From: Rich Adamson [EMAIL
Adam Robins wrote:
Should it be in half duplex or full duplex?
Full.
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
-- Original Message --
From: bodra [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Date: Sun, 28 Aug 2005 02:35:01 -0700
Hi all
i am developing a client for the asterisk that controls ur phone from an Xp c#
application
what functions in Asterisk that
Anything like this for grandstream phones?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Matthew T. O'Connor
|Sent: Lunes, 29 de Agosto de 2005 12:22 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: [Asterisk-Users]
I doubt my cvs was that current so... It's a clena install then...
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Kevin P. Fleming
|Sent: Domingo, 28 de Agosto de 2005 09:56 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
Matt Riddell wrote:
Adam Robins wrote:
Should it be in half duplex or full duplex?
Full.
AFAIK, depends...
If you have your switches doing autonegotiation, you can't disable
autoneg in the NIC and hardcode it to do 100/Full-duplex, or you WILL
have a duplex mismatch.
This is as
Tobias Wolf ha scritto:
Let us assume that i have a couple of phones which should be able to
receive SMS directly from my * box ( and not from an SMSC from BT or
Deutsche Telekom ), So all these phones have the phone number of the *
as Service Center configured. I recognized that the numbers
Everything is set to autoneg, NICs, switches and router
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julio
Arruda
Sent: Monday, August 29, 2005 8:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IAX2
Hi
two weeks ago I posted a message concerning static noise on our asterisk system
we have made a bunch of tests and these are the results
We use a TDM card revision I and on the card there is a sticker that says
revision G
If we put one fxo modules there is no noise
if we put two fxo
bodra wrote:
yes thats one issue the other issue is that sometimes the pstn line is dead due
to some technical problems so people trying to make calls will just listen
silence and they'll never know whats going on...
Which should be less of a problem, given that the FXO card gives a red
Hi,
I am trying to update Asterisk from cvs as I think it might solve a secondary problem that I am
experiencing (see below). In the /usr/src/asterisk
directory I typed make upgrade. However I get an error:
Makefile:16: ***
missing separator. Stop.
Make[2]L Leaving directory
Hi,
On several occasions one or more of our grandstream Handy tone 486 ATA
would crash. If for some reason that ATA is not rebooted immediately,
asterisk would not disconnect the call, even though the party on the other
end of the call have already hung up the call. The call would continue
You may want to check if the autonegotiation agreed in both sides.
Older nic/drivers/switches would have problems with autonegotiation.
Also, statistics can tell you something about this..
Example, if you have shorts/runts in one port, and late-collisions in
the L1 'peer' port (the other side
Michel
Send me the same output for a dial string that only sends the *31*
Is this an ISDN line? What type of card/signalling/switchtype are you
using?
It looks as if the PSTN switch accepts the *31* and then hangs up so you
can make the NEXT call with the *31* feature enabled. If so I assume
Everything is set to autoneg, NICs, switches and router
To ensure reasonable performance, key devices (eg, routers, servers)
should _always_ have duplex settings statically defined. Speed is
less of an issue as the 10/100 negotiation is hard to get wrong.
Part of the duplex negotiation
For an asterisk server _always_ statically define the duplex setting
on both the switch and the nic card. On sip phones and workstations,
Can you give an example of how to check the duplex setting and statically
define it for, say, RedHat9
On Mon, 29 Aug 2005, Rich Adamson wrote:
Has anyone else got 1.2 compiled from cvs ? I've posted the question
below to the -dev list but got no answers:
1) No-one else is trying beta 1
2) No-one else is having any issues (I must be the idiot)
3) No-one else saw my message :)
I have been trying to compile 1.2 beta 1 on a centos 4 box,
For an asterisk server _always_ statically define the duplex setting
on both the switch and the nic card. On sip phones and workstations,
Can you give an example of how to check the duplex setting and statically
define it for, say, RedHat9
Multiple ways... try 'dmesg | grep duplex' or
Hello,
Thanks for help it's ok with static file
voicemail.conf
However something is wrong with ARA .
app_voicemail search entries in voicemail.conf ?!
I set apps/Makefile for USE_ODBC_STORAGE.
Regards
Harry
//
Connected to Asterisk CVS-HEAD
Julian Lyndon-Smith wrote:
Has anyone else got 1.2 compiled from cvs ? I've posted the question
below to the -dev list but got no answers:
Mine complies fine under Mandrake and a kernel downloaded from
kernel.org, ztdummy won't load, but other then that no issues.
Doug
If nic is loaded using modprobe - you can set options for duplex -
depending on the nic...
See /etc/modules.conf
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Rich Adamson
Sent: Monday, August 29, 2005 11:13 AM
To: Asterisk Users Mailing
Urban wrote:
Hi,
is there any support for include statement in the database when using
realtime configurations? I would like to have as much as possible
configuration in my postgres db but we have different access controls
for different user contexts (allow international, national etc).
Hi:
Is there anybody familliar with www.plainvoip.com voip
provider.
I sent them money through paypal and they didn't add
the money to my account and they didn't respond to my
request to send the money back to paypal. Is there
anything I can do besides disputing the charge with
paypal?
Regards;
If nic is loaded using modprobe - you can set options for duplex -
depending on the nic...
See /etc/modules.conf
I assume you really meant /etc/modprobe.conf ;)
-Original Message-
For an asterisk server _always_ statically define the
duplex setting
on both the
On 8/28/05, Matt Riddell [EMAIL PROTECTED] wrote:
Steve Edwards wrote:
Normally the way I do it is to program the failover into the dialplan
and then
send the call to Local/[EMAIL PROTECTED] to initiate it.
How about a snippet? (Local channels somewhat escape me.)
Ok,
If you had
Hi,
I have asterisk 1.0.7 and spandsp-0.0.2_pre18.
txfax return a non-zero return code only if the
fax file is not found.
Unfortunately I can't get any information, whether
the fax was transmitted completely or not.
Will an update to a newer version change this?
Thanks for telling me your
I am trying to update Asterisk from cvs as I think it might solve a
secondary problem that I am experiencing (see below). In the
/usr/src/asterisk directory I typed “make upgrade”. However I get an error:
Makefile:16: *** missing separator. Stop.
Are you on FreeBSD (or not Linux)? You
On 8/29/05, Doug Lytle [EMAIL PROTECTED] wrote:
Julian Lyndon-Smith wrote:
Has anyone else got 1.2 compiled from cvs ? I've posted the question
below to the -dev list but got no answers:
Mine complies fine under Mandrake and a kernel downloaded from
kernel.org, ztdummy won't load, but
I'm using suse linux.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin
Bockman
Sent: 29 August 2005 16:13
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] FW: cvs update error?
I am trying to update Asterisk
I have a running * with a TDM40B board in it.
I have 3 analog phones that works (rings) perfectly when connected
to a Telco POTS line.
When connected to the Digium TDM40B (with FXS port), I have problems
with 'ringing':
1 phone 'ringes' normally
1 phone 'ringes' a bit cripled (instead of
hello!
i'm looking for a feature to play a sound-file containing a text until the
called party picks up the phone. i've already tried with the 'special'
musiconhold-feature by adding the m-option at the end of DIAL but it is not
exactly what i want. the problem with the m-option is that the
hi,
i have problem with compiling cdr_sqlite
rhel4(gcc3.4.3) + sqlite3 (from fc4 - rebuilded)
any ideas?
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT
-D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS
Hello
i am trying to use this exmple with SER-0.9.3
but still NATED Clients are not working any other
requirement
http://www.voip-info.org/tiki-index.php?page=SER+example+NAThelper
---
# $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $
On Mon, 2005-08-29 at 14:04 +0100, Aisling wrote:
Hi,
I am trying to update Asterisk from cvs as I think it might solve a
secondary problem that I am experiencing (see below). In
the /usr/src/asterisk directory I typed “make upgrade”. However I get
an error:
Makefile:16: ***
Had the same issue, tried to submit the bug and the bug tracker would
not take bugs for versions other than CVS head.
I did a little more research and found a directory
/usr/src/asterisk/asterisk!
I did not create the folder above!
CVS Head compiled on the same machine without issues
There has
Work intrudes again and I will not be able to get to modifying the db
and gui
to support per-conference flags as soon as I expected. So I have
released
an update with what I do have available.
[Location]
http://www.fitawi.com/Asterisk
[Features]
1. Schedule new conferences
a.
Asterisk has this error on compile:
flex ast_expr2.fl
ast_expr2.fl, line 50: unrecognized %option: reentrant
ast_expr2.fl, line 51: unrecognized %option: bison-bridge
ast_expr2.fl, line 52: unrecognized %option: bison-locations
make: *** [ast_expr2f.c] Error 1
2.6.12-1.1447_FC4smp #1 SMP
bison
Hello,
I have attached my makefile. I don't know what I should be looking for
in it but if it is somehow different to everyone elses make file, will
someone please point that out? I never modified it in any way. How would
I get a new copy of the Makefile from CVS?
Many Thanks.
-Original
Hi there,
We are using * with an Option 11C - we tried all of the various
protocols and the only one we could get to work satisfactorily was
5ESS, with the * as CO and the Nortel as remote. The one drawback of
this approach is getting name information for caller ID - because the
Nortel
On Mon, Aug 29, 2005 at 09:54:11AM -0700, Anthony Rodgers wrote:
We are using * with an Option 11C - we tried all of the various
protocols and the only one we could get to work satisfactorily was
5ESS, with the * as CO and the Nortel as remote. The one drawback of
this approach is
I havea couple
SIP phones on a PIII 1Ghz 256MB* server with a TDM01B connected to the
PSTN. Calls between SIP phones are clear. Calls to the PSTN are
quite noisy. The other person does not hear noise but I hear quite a
bit. It is not an annoying sound but definitely much noisier than
Is there a problem at Teliax? I'm looking for a VoIP provider and when I call
them they never answer the phone and the voice mail says it's full.
Chris___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
\I concur. They seem to be always busy.
Chris wrote:
Is there a problem at Teliax? I'm looking for a VoIP provider and when I call
them they never answer the phone and the voice mail says it's full.
Chris
I like the plans they offer, but this doesn't give me much confidence in
their ability.Can anyone recommend someone else?
- Original Message -
From: Joshua Abbott [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent:
Joshua Abbott wrote:
\I concur. They seem to be always busy.
Chris wrote:
Is there a problem at Teliax? I'm looking for a VoIP provider and
when I call them they never answer the phone and the voice mail says
it's full.
Have you tried emailing them or using their online support?
Darrick
Their online support says off line and goes to email.I have emailed
them several times and still haven't got answers to my questions.Everytime
I get a response from them I have to repeat my question and then I never hear
the answer.
Regards,
Chris
- Original Message -
Chris wrote:
Their online support says off line and goes to email.I have emailed
them several times and still haven't got answers to my questions.Everytime
I get a response from them I have to repeat my question and then I never hear
the answer.
Regards,
I'm glad I haven't
Their plans look good, but it just feels like I am being ignored. Some
guy named David emailed me off the Asterisk-biz list from Teliax with his
direct number.I'll give that a try.
Regards,
Chris
- Original Message -
From: Darrick Hartman [EMAIL PROTECTED]
To: Asterisk
Is there anyon here currently in New Zealand that use asterisk, I need to help getting voice and internet services. I will be moving in a week. Any help would be great. Please use the details below to get ahold of me.
Thanks in advance.
James Jones
Signate, LLC
[EMAIL PROTECTED]
using ARTDIO clone IAX2 phone set
connected on the same LAN as Asterisk server
Ring...
when off hook :
- we can hear correctly the caller
- but the caller continue to hear the ring tone
Any idea ?
___
--Bandwidth and Colocation sponsored by
can someone who has a grandstream handytone 488 working with making
outgoing calls through the fxo port please post the parts of their
config that deal with this port? i cant quite seem to get it to make
outgoing calls despite having tried several completely different ways of
making that happen.
I am searching for a way to add a 2 second delay before calling out with
Dial().
Sometimes I get the message you must first dial a 1 to place this call.
I presume the phone company is missing the first digit pulsed out sometimes.
How do I put a 2 second delay after coming offhook and before
Hello,
We've released another update to our Asterisk GUI Client suite: 1.1.6
http://astguiclient.sf.net/
The client suite runs on Windows, UNIX and Mac, includes the
astGUIclient client-side web app which extends your phone's
functionality and the VICIDIAL client-side web app auto-dialer. This
Jerry Geis wrote:
I am searching for a way to add a 2 second delay before calling out
with Dial().
Sometimes I get the message you must first dial a 1 to place this call.
I presume the phone company is missing the first digit pulsed out
sometimes.
How do I put a 2 second delay after coming
Hi All,
Just wondering if anyone has managed to get line presence working on the
7 indicator lights on a grandstream gxp-2000 with asterisk? If so, what
is the trick? :)
last week i asked the grandstream support for this, and got this short answer:
This feature is not supported yet, it
Samy,
Thanks for the suggestion - however I am confused on the wiki the
'w' stands for:
*w*: Allow the /called/ user to start recording after pressing *1 or
what defined in features.conf (Asterisk v1.0.x)
This is not a delay of any kind.
Jerry
[Asterisk-Users] delay before dial on
Chris wrote:
Is there a problem at Teliax? I'm looking for a VoIP provider and when I call
them they never answer the phone and the voice mail says it's full.
I don;t see any network problems, and I monitor Teliax and a few other
providers. Teliax is my main provider and I have never had
Casey Boone escreveu:
can someone who has a grandstream handytone 488 working with making
outgoing calls through the fxo port please post the parts of their
config that deal with this port? i cant quite seem to get it to make
outgoing calls despite having tried several completely different ways
I think Asterisk is sending some signal to my cordless phone that is
causing it to constantly display message: MSG Waiting Off.
The problem is that it is impossible to program anything into the phone
or sometime dial a phone numbers as the when I try to program a number
or dial a number and
Jerry Geis wrote:
Samy,
Thanks for the suggestion - however I am confused on the wiki the
'w' stands for:
*w*: Allow the /called/ user to start recording after pressing *1 or
what defined in features.conf (Asterisk v1.0.x)
This is not a delay of any kind.
Jerry
[Asterisk-Users]
They are always there at the online chat.(during business hours)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason
(Lists)
Sent: Monday, August 29, 2005 11:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
On Mon, 2005-08-29 at 12:20 -0600, Joseph wrote:
I think Asterisk is sending some signal to my cordless phone that is
causing it to constantly display message: MSG Waiting Off.
The problem is that it is impossible to program anything into the phone
or sometime dial a phone numbers as the
Where does the CNAM originate, is it sent to the Nortel from the PSTN
and then passed on to *, or does it originate on the Nortel?
There may be another way, but without more info I do not wan to peak out
of context.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
Samy Kamkar wrote:
Jerry Geis wrote:
Samy,
Thanks for the suggestion - however I am confused on the wiki the
'w' stands for:
*w*: Allow the /called/ user to start recording after pressing *1 or
what defined in features.conf (Asterisk v1.0.x)
This is not a delay of any kind.
Jerry
Title: Message
I have recently
discovered a problem that I cannot dial internal extensions - I either get a
busy tone or directed to voicemail depending on if the extension has
voicemail.
This was working
fine, but not sure what has changed to stop this working.
Today I did delete a
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