Hello,
I am using voipuser.org service, and am trying to make a SIP call.
Everything seems to work fine, except I can't hear anything on my end.
When I make a SIP call, the other party can hear me, but I can't hear
anything. I am using asterisk + Digium TDM board with an FXO port
where
i had a similar problem a while ago. I solved it by defining
externip=xxx.xxx.xxx.xxx in sip.conf. It tells the remote SIP
client where you are.
-chuks.
Original Message Subject:
[Asterisk-Users] Can't hear the callerFrom: Lane
[EMAIL PROTECTED]Date: Mon, March 21, 2005 11:53
Hello,
I just started using asterisk, and have a question. I have setup two
asterisk servers, A and B. A has a Digium TDM400 11B card (1 FXO and 1
FSX modules) and is connected to the PSTN. B has same, but is NOT
connected to PSTN. I want to configure B to call A via iaxtel, and
connect to the
Hello,
I bought a TDM400P, and intended to use it with my analog
phone, which is RJ11 ofcourse. So, the question now, how do I plug in
my RJ11 phone to the TDM400P card, which has an RJ45 interface? Also,
since it's an 11B card, I also intend to bring in an analog line into
the RJ45, so i am still
Hello,
two questions:
1: How can I open/enable network connection to
B?
scenerio:
I have 2 Asterisk servers, A and B, running Fedora Core1
on my local network.B refuses any network connection attempts from
A, i.e. I can't even telnet or FTPto B from A, but I canto A
from B. This makes B refuse
Hello,
two questions:
1: How can I open/enable network connection to
B?
scenerio:
I have 2 Asterisk servers, A and B, running Fedora
Core1 on my local network.B refuses any network connection
attempts from A, i.e. I can't even telnet or FTPto B from A, but
I canto A from B. This makes B refuse
Hello,
can someone tell me what's wrong with this? I can't make toll
free calls via iaxtel. Here's the definition in my extensions.conf
[iaxtel-trunks]
;
;outbound 1-700 and toll free calls go via iaxtel
;be sure to include the iaxtel-trunks context in dialing
context
;add function here to
Hello,
can someone tell me what's wrong with this? I can't make toll
free calls via iaxtel. Here's the definition in my extensions.conf
[iaxtel-trunks]
;
;outbound 1-700 and toll free calls go via iaxtel
;be sure to include the iaxtel-trunks context in dialing context
;add function here to
could you help me out with this? I have a posting on this list, bu
nobody has replied yet. Titled "why can't I make IAX calls between 2
asrterisk servers"? I'd appreciate.
-chuks.
Original Message Subject: Re:
[Asterisk-Users] bridging iaxtel calls to PSTNFrom: "Michael
Graves"
Hello,
actually I did, but nobody responded to that. So, here it is
one more time:
___
Hello,
can someone tell me what's wrong with this? I can't make toll
free calls via iaxtel. Here's the definition in my extensions.conf
[iaxtel-trunks]
;
;outbound 1-700 and toll free calls go
Hello,
Can anyone help with this please?
thx,
chuks
Original Message Subject:
[Asterisk-Users] Why can't I make inter IAX calls between2 Asterisk
serversFrom: [EMAIL PROTECTED]Date: Mon, February 21, 2005
11:04 amTo: asterisk-users@lists.digium.com
Hello,
two questions:
1: How
ok, thanks for pointing that out...
Original Message Subject: RE:
[Asterisk-Users] bridging iaxtel calls to PSTNFrom: "Rich Adamson"
[EMAIL PROTECTED]Date: Mon, February 21, 2005 4:00
pmTo: "Asterisk Users Mailing List - Non-Commercial
ok, thaks for pointing that out...how can I turn off the HTML tags? I am
using a web based email client.
BTW, sorry if this has been annoying, it's not been on purpose.
Original Message
Subject: Re: [Asterisk-Users] bridging iaxtel calls to PSTN
From: Jens Vagelpohl [EMAIL
Hi!
I've been using asterisk for 1 year now, however since yesterday
something odd happened. From my office i dial extension 125 and it
wont work, it sounds as a busy tone, and in the x-lite gives
me call failed 404 not found, however if i dial from my house to
that same extension then it
Hi!Steve,
Glad to receive your message. Could you tell me if the card can work
properly in Asterisk? and what is the different bettween the Tormenta ISA E1
card and the ISA T1 card (I refer to the hardware). Thanks.
john
Steve Underwood writes:
[EMAIL PROTECTED] wrote:
pullable characteristics to get reliable
results. If you downlod the info from Dallas for the framer chip they
give some recommendations for off the shelf crystals which will work.
Regards,
Steve
[EMAIL PROTECTED] wrote:
Hi!Steve,
Thank you for your reply. I think the change for Tor ISA card from
If I comment all lines of zaptel.conf,after I run the ztcfg,the warning in
the zttool is OK, that is to say,no warning.
If I leave the zaptel.conf's content as:
span=1,0,0,ccs,hdb3
span=2,0,0,ccs,hdb3
then the RED warning will appear.
I I connect a loop back line into span1's RJ45
If you have an FXS port and would like to attach a PSTN
analog line to it this device would do the job by converting the FXS port to
FXO. Its a small external
device. Works well with VOIP FXS and other FXS interfaces.
Interface: 2 RJ11 Jacks (one for the FXS
port and one for the PSTN
If you have an FXS port and would like to attach a PSTN
analog line to it this device would do the job by converting the FXS port to
FXO. Its a small external
device. Works well with VOIP FXS interfaces and other FXS
interfaces.
Interface: 2 RJ11 Jacks (one for the FXS
port and one for
Visit www.aislecom.com for FXS to FXO
converters.
Product: ACOM300
]
CSeq: 1 INVITE
Max-Forwards: 70
Contact: sip:[EMAIL PROTECTED]:5060;line=v8ppcao5
P-Key-Flags: keys=3
User-Agent: snom190-3.56i
Accept-Language: en
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported
protection.
Info: http://copilotconsulting.com/sig
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Hello every one,
I have got a
H323 gatekeeper for testing. The informations are something like
this:
account code: test01
gk ip address:192.168.10.12
I don't know how to set it in the h323.conf or
oh323.conf, I have tried it for almost one day but I always got the error. Help
me please.
Hi,Lubo,
Thank you very much for your reply. I want to use the gatekeeper for
outbound call, but I really don't know how to use it in the extensions.conf
,I think there are something diffrence between the chan_h323 channel and the
chan_oh323 channel. A little example of extensions.conf would
Thank's Lubomir and Jeremy! It's working now. That's to say,I could dial
long distance call from MSN or NetMeeting now.
Regards.
frank
- Original Message -
From: Jeremy McNamara [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 03, 2003
Hello,everyone,
I encoutered some difficult with IAX when I run
the asterisk.
internet -- asterisk + NAT -- DIAX
my * box and NAT are at the same linux box which connecting to the internet
using ADSL. The box has two network cards and two IP address,such as
public
Question2:
If I dial the IAX2 user registed to my * inside my NAT,it will
success,but
if I dial other IAX2 user registed to my * in the internet (not inside
my NAT),I alway get the result:
== Everyone is busy at this time
Take care that there is an issue with DIAX and IAX2... after some
Yes,I often get the same result, but not always.
- Original Message -
From: Michael Welter [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, December 20, 2003 3:40 PM
Subject: Re: [Asterisk-Users] DIAX phone busy
Yes, I've tried that as well. When I dial 70 from another
Interesting! Surely it would be another greate project.
Happy christmas!
- Original Message -
From: Bob Knight [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 24, 2003 11:30 AM
Subject: [Asterisk-Users] time to build an open phone?
Open software seems to work.
Why
My kernel is a 2.4.27 and I think that mISDN is available only for a 2.6.x
But I can't use a 2.4.26 for some security reasons...
Alain
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Armin
Schindler
Envoyé : mercredi 5 avril 2006 18:05
À : Asterisk
OOps
The correct answer is
My kernel is a 2.4.27 and I think that mISDN is available only for a 2.6.x
But I can't use a 2.6.x for some security reasons...
Alain
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de
[EMAIL PROTECTED]
Envoyé : mercredi 5
HI
I am trying to establish a connection between ASTERISK and ALEPO but I can
not,
since you have reached to make them communicate can you help me with the
changes made to asterisk, in this way I will be able to check if the
problem is the same with my ALEPO .
I would appreciate every help
OOPPS! Looks like someone just broke voipjet's tos
gw at adcomcorp.com gw at adcomcorp.com wrote on
Sat Nov 5 11:36:46 CST 2005
I tend to agree with you, my experience with Teliax has been decent,
and
Title: Message
Hello:
Hoping someone might
know how to resolve this (probably an easy one). I have one Asterisk PBX with a
single NIC and an FXO card with PSTN line attached, and one IP phone (Budge Tone
100) on the LAN.Via the phone Iget no dial tone, and dialing 9,
number doesn't allow
Title: Message
Thanks to Greg Hill
for pointing me to the 'sip debug on' cmd that helped me resolve the sip
connection problem!
The other issue I'm
trying to resolve is configuring outgoing calls. I need to configure outgoing
calls to use the FXO card in the PBX (zaptel device) via sip
Title: Message
Nevermind. Figured this out. I needed the following in
extensions.conf to enable outbound dial.
exten
= _9.,1,Dial(Zap/2/${EXTEN:1},70,Tt)
Thanks
-Original Message-From: Info
[mailto:[EMAIL PROTECTED] Sent: Tuesday, August 17, 2004 9:27
AMTo: '[EMAIL
Bedankt voor uw bericht.
Online4You is sinds 1 augustus niet meer operationeel. Per e-mail hebben wij u
geinformeerd over de omstandigheden en uw opties.
Helaas kunnen wij u niet meer helpen, uw mail wordt niet doorgestuurd en/of
beantwoord.
Indien uw abonnement is overgenomen door KovoKs,
Bedankt voor uw bericht.
Online4You is sinds 1 augustus niet meer operationeel. Per e-mail hebben wij u
geinformeerd over de omstandigheden en uw opties.
Helaas kunnen wij u niet meer helpen, uw mail wordt niet doorgestuurd en/of
beantwoord.
Indien uw abonnement is overgenomen door KovoKs,
Bedankt voor uw bericht.
Online4You is sinds 1 augustus niet meer operationeel. Per e-mail hebben wij u
geinformeerd over de omstandigheden en uw opties.
Helaas kunnen wij u niet meer helpen, uw mail wordt niet doorgestuurd en/of
beantwoord.
Indien uw abonnement is overgenomen door KovoKs,
Bedankt voor uw bericht.
Online4You is sinds 1 augustus niet meer operationeel. Per e-mail hebben wij u
geinformeerd over de omstandigheden en uw opties.
Helaas kunnen wij u niet meer helpen, uw mail wordt niet doorgestuurd en/of
beantwoord.
Indien uw abonnement is overgenomen door KovoKs,
John Todd said:
...
Ideas welcome for more text; I may have another timeslot with Allison
early next week in which there will be some leftover room for
additional words. Short phrases and meaningful sets of words for
existing applications are desired; please don't give me words for
apps
John Todd said:
The freenum.org project wants to use your trunks! The freenum.org project
is an ENUM parallel tree, which has as an eventual goal the distribution
of ENUM numbering in nations or areas which due to political or other
issues are not able to get secure, inexpensive, or
Top posting(sorry) then imbedding the answers to your questions. Otherwise
doesn't make sense.
Thanks for your reply. Sorry it took a while to get the answers. I'm in
Germany and your email came last night just as I was headed to the rack.
Robert
John Todd said:
my sip.conf contains:
Looks like the list server is really lagging tonight. I found out some
more info so will just post it in a new email with the same subject.
I added: search = freenum.org to enum.conf and got a match (SIP
system) when doing the lookup Maybe I overlooked that in the
original instructions
be accessible from the dialplan when an EnumLookup is
returned.
Anyone want to take a swing at it? Otmar? :-)
JT
John,
Thanks for the info. I'll leave the code commented out in the dialplan.
If I put in the NAT SIP patch then will reenable it. Is an interesting
concept for some long snowy night
John Todd said:
United States:* +1-800-...
+1-888-...
+1-877-...
+1-866-...
via: Telesthetic/Local Exchange Carriers of Michigan
JOhn, Good idea on leaving the code in. I'll do that. Since IAXtel has
8xx dialing
Info based on how I do it is imbedded below.
Robert
Larry Keyes said:
I've got two Grandstream phones talking to * and a X100P card going, so
that
I can make inbound and outbound calls via the PSTN, and calls from one
extension to another.
1. Is there an equivalent to the more command
Kannaiyan Natesan said:
Have anyone tried to interface BT's Broadband Voice with asterisk?
Kannaiyan
___
No, and not sure of their rates but http://www.telappliant.com/ has good
rates, voice quality and is easy to interface to Asterisk.
Robert
Kannaiyan Natesan said:
Do they offers, free evening and weekend calls? I get from BT.
You can get a free 0870 number from http://www.speak2world.com but they
charge for it.
Kannaiyan
Don't think so but sometimes free isn't free. Depending on calling
patterns it might actually be lower cost
John Todd said:
Time to dump the Netgear router. That's an unacceptable answer for a
router vendor to say Oh, well, for this MAJOR protocol we're going
to simply corrupt those packets so they're unusable. What!?
JT
__
OR get an older one from
Mike Nash said:
Hi
I'm trying to configure my Asterisk box to provide a simple sample
configuration. It's a mandrake 9.1 box, no cards except a sound card.
The
config I am trying to achieve is simply one server, with two SIP clients.
Two issues are cropping up - the first, when I start
of your termination into Brazil. We
have several Brazilian expatriates here in Germany that might be
interested in your service. Partially would be Asterisk using IAX2 and
others using SIP Phones. Can you please pass along additional info?
Regards,
Robert
Friedrichshafen, Germany
I have compiled the zaptel library and zaprtc on a system that gives the
following from uname -a:
Linux fxx76.mydomain.de 2.4.18-64GB-SMP #1 SMP Wed Mar 27 13:58:12 UTC
2002 i686 unknown
Makefile for zaptel had the following line uncommented:
#
KFLAGS+=-D__SMP__
When doing the make load for
Rob Fugina said:
On Fri, Jan 30, 2004 at 10:48:35PM -0500, John Todd wrote:
In the mean time, I've seen references to bug #'s, here on the list and
in the CVS logs. I've yet to stumble across the bug tracking system,
though -- can you give me a nudge in the right direction?
Thanx,
Rob
Martin said:
Hello.
I vaughely remember someone talking about an asterisk implementation at a
University in germany some months back.
Any other information ?
Regards...Martin
--
http://graphics.cs.uni-sb.de/VoIP/en/index.html
Some of those folks and also from the Uni Stuttgart hang out
Real Player is required. Excellent video/slide presentation.
http://graphics.cs.uni-sb.de/VCORE/recordings.html
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Andy,
I would be interested in your Cepstral engine code.
Regards,
Robert
Friedrichshafen, Germany
Andy Powell said:
lo,
Is there a single central location for code and applications other than
CVS? I'm talking about code that can't/wont be included in CVS for various
reasons? Does the wiki
Feedback for the list. I compiled Andy's code. Installation went well
(except for me misspellng something in the dialplan) with no problems.
The Application works great. Will run down Brian's and give it a try too.
Robert
___
Asterisk-Users mailing
Tim Sailer said:
I've looked, poked, and hoped, but I can't seem to make * understand
the difference between a SIP channel being busy or not being there.
Both come up as 'busy'. I would expect the unregistered SIP to be seen
as unavailable. Am I just missing something obvious, again?
Tim
^
Christian,
Where is a good place to purchase your phones in Germany? I found a
distributor in the UK but maybe just am not looking in the right place for
Germany.
Thanks,
Robert
American Expatriate in Friedrichshafen (Grund oder Entschuldigung für die
englisch)
Christian Stredicke said:
Sorry,
[EMAIL PROTECTED] said:
I like using whisper tones...
recored the file companyname_whisper.gsm and put it in
/var/lib/asterisk/sounds
Then add the lines to extensions.conf
exten = 0031,1,Dial(SIP/Recp|20|A(companyname_whisper.gsm)r)
In my implementation of this the file extension had
Jim Sneeringer said:
Whenever an outside number is dialed, Asterisk says We're sorry. Your
call
did can not be completed as dialed. Please check the number and dial again
or call your attendant to help you. I have tried many configurations,
but
let me give the simplest: It fails when a
John,
You are now advertising your EMEA company in your signature block. Maybe
I missed an email that explains the EMEA pricing and availability. Could
you please give an update via the list as to the status of your product
availablity, pricing and delivery times in Europe? The ordering
Soren Rathje said:
- Original Message -
From: Olle E. Johansson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, February 22, 2004 8:52 PM
Subject: Re: [Asterisk-Users] SIP extension busy when not available ??
Although the current logic does not require a sip phone to
, yes please...
Well, I'm about three weeks into my very first * installation (that sort
of
works), so basically any info/tips/tricks/word of advice is accepted
with
appreciation...
-- Soren
I use a macro to define the extensions. In this way I only have to enter 1
line per actual extension
Comments are inline.
Robert
Jeroen Rikhof said:
Hello,
Can somebody give me some information about:
1. How stable Asterisk is?
My experience and from what I have read on the list is that it is very
stable if run on stable hardware and you don't mess with the program code.
If you mess with
Paul Mahler said:
With record:
; Record voice file to /tmp directory
exten = 9000,1,Record(/tmp/asterisk-recording:gsm)
exten = 9000,2,Hangup
Is there a way to stop recording other than hanging up?
Thanks!
Press the # key.
Below is from my extensions.conf. It plays the
Angel Gabriel said:
I have 5 BT phone lines coming into my office. We use four for
international calls, and one for local/mobile calls. We have just obtained
another call carrier, and now we would like to be able to make calls from
any phone to any carrier, without having to remember what
Tim,
It looks interesting.. Are you willing to release the source code?
Robert
Tim Sailer said:
On Fri, Mar 05, 2004 at 01:29:38PM -0500, Tim Sailer wrote:
Since there's not too much out there, I decided to take about 2 hrs and
pound something into shape for a simple status for my * server.
I have made no recent changes to the IAX2 config on my system. Today I
tried a 1800 call and got the below error. Not sure when this started
since only use 800 once in a while. Does anyone know if IAXTEL is
experiencing problems connecting to the 8xx gateway?
7 16:14:54 WARNING[147466]:
Philipp,
Good document, my comments are inline with the parts to which they apply.
(and yes, this was a top post, otherwise it wouldn't make sense.)
Robert
Hi there,
mostly based upon list postings I compiled a couple of administrative
suggestions on the Wiki page below. I'd be glad to have
Philipp,
Good document, my comments are inline with the parts to which they apply.
(and yes, this was a top post, otherwise it wouldn't make sense.)
Robert
Hi there,
mostly based upon list postings I compiled a couple of administrative
suggestions on the Wiki page below. I'd be glad to have
It looks like Mark and others have addressed the development/CVS issues.
We should let their plan be put into effect and give it a chance to work.
Regarding the email list: A number of people have suggested creating more
email lists. I think this is not a good idea because there will be even
Hi,
Do the callers in USA dialling from USA Telco lines always have to
prefix the CITY/AREA code with 1 in order
To successfully make a call to other USA destinations?
I have not been to USA (yet) :)
Ta
SJ
For comprehensive info by area code (and as pointed out it does differ
from
Philipp von Klitzing said:
oHi!
Ladies and Gentlemen, can anyone please help and let me know what is
the way to start Asterisk automatically using a cronjob, thanks
http://www.voip-info.org/wiki-Asterisk+administration
Philipp
Guess maybe I don't leave my system running long enough for
admin said:
I work for an interconnect that sells 3com and NEC. When I made this
project my own and followed through to show my boss, he said, this is
going
to ruin our industry
If that is the case then so be it. Same with mp3s and the music industry.
Had they embraced the technology,
,
or call +1 505 830 1200 and please do leave good
information (name, phone number, what you ordered)
we don't always receive enough info to respond back
(missing phone numbers or complete names are common)
If you have any issue you can call my direct number at
+1 505
Chandra said:
i also had the same problem temporarily i solved my problem with both
outside NAT. u can also do it if both inside NAT. * outside NAT and
Budgetone behind NAT simply doesn't seem to work. if u ever solve this
problem please let me know too.
thanks
cm
I am able to use my
Chris Albertson said:
I'm looking for a service that will accept VOIP calls and
send them to the PSTN. Or, I should say _another_ service
that will do this. I don't need the other direction
Currently I'm using IconnectHere and it works, but I get
complaints of poor audio quality from the
Distribution channels aren´t being made to the public yet, other then direct from digium. Looks like they will be waiting 2 weeks before we hear anything else. (After VON) The questions you are asking, I dont believe have yet been confirmed by asterisk or digium. Though I am sure it is on there
I am trying to compile * on SuSE 8.2. When doing the make install in
/usr/src/zaptel I get the following error.
**
/usr/src/linux/include/asm/system.h:189: warning: dereferencing
type-punned pointer will break strict-aliasing rules
freeIn
Hello All,
Thanks to those that responded to my problem of compiling on SUSE 8.2. I
was not able to get the compile done so decided to put RedHat 9 on this
system. After getting a RedHat supported NIC and RedHat installed,
Asterisk compiled cleanly, one SIP phone is connected and voice mail
I am getting the following messages that seem to be coming from Asterisk.
In the system there are no ZAPTEL cards installed. I did uncomment ztdummy
in the Makefile in /usr/src/zaptel before running make install. Any
ideas on how to get rid of this message. I looked through all the config
files
is same as extension
Authenticate ID: 2000
Authenticate password: 9overthruster7
Send DTMF: Via SIP info (in order for the dtmf to be recognized by
voicemail)
Hi People,
Ok i've tried everything I can think of but cant get this to work.
Can someone please give me an example
as extension
Authenticate ID: 2000
Authenticate password: 9overthruster7
Send DTMF: Via SIP info (in order for the dtmf to be recognized by
voicemail)
Hi People,
Ok i've tried everything I can think of but cant get this to work.
Can someone please give me an example of their sip.conf settings
Do you have a 100 or 101? You have indicated different models in your
postings. Were you able to get Call Transfer and Call Waiting working
with your Asterisk system and other phones? Which version of the
Grandstream firmware do you use? There most recent on their website this
weekend was at
I only have 1 but the absolutly only time it has to be rebooted is when I
change a parameter or upgrade the firmware. It has run for weeks without
any problem. Another poster mentioned the 10 vs. 100 Ethernet speed.
Maybe Grandstream can upgrade the interface in future hardware. I don't
imagine
Michael,
That would work for me too. If the volume can be reduced (maybe to zero or
almost zero) then my request for the ability to disable it is not needed.
Since the volume of the speaker and handset can be controlled maybe the GS
folks can include a patch in the next release of the firmware
The only thing that is wrong is that there seems to be some expectation of
Digium that they have to tell things... The source code is available. If
someone isn't happy with the Digium methods then they should find a
solution and post it to the list and/or one of the several Asterisk Wiki's
that
Andrew,
I am running it rather well on a original Pentium 100 Mhz, 32 MB RAM, no
USB adapter. I agree with you this would not be an ideal setup for a
business but in a home it will work rather well. I think it'll handle 2 CO
analog lines fine.
Yes, my wife thinks its overkill. Probably is, but
Chris,
Good point. As I understand it, the Asterisk software requirement was to
be a PBX between normal telephone lines and VoIP. Maybe even it was just
to replace the expensive PBXs. As such seems to me that it clearly met
and exceeded its design requirements since it utilizes the hardware
Yes, I am a newbie too. I am having a problem with meetme. From what I
have seen it will work without a Digium card but with audio problems. My
goal is just to see how it works not the quality of the audio.
When I dial into the conference room the following message is played:
That is not a valid
On Wed, 2003-10-15 at 14:16, rnc Info Lists wrote:
Yes, I am a newbie too. I am having a problem with meetme. From what I
have seen it will work without a Digium card but with audio problems. My
goal is just to see how it works not the quality of the audio.
When I dial into the conference
Did you modprobe ztdummy before running asterisk ? I have meetme
running in one * box without zaptel harware.
I just tried that.
The following messages are given:
/lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o: init_module: No such
device
Hint: insmod errors can be caused by
Asterisk...
Linux...
You get what you pay for. And it's free
:P
Thats true but free (cost) doesn't have to mean cheap (quality). Maybe
what we need is to collect business requirements and build a configuration
for a typical system. (hardware spec. and actual config files) What Dave
has
look at the rtc driver then. you do have a rtc chip already on the
system.
I looked back in the list and looks like the message that mentioned who
wrote ztrtc I deleted. Can someone please let me know where to obtain
ztrtc? I did a google on it and came up empty.
Thanks,
Robert
Seems you used my abreviation. It is really known by zaptelrtc. It seems
to be written by Klaus-Peter Junghanns [EMAIL PROTECTED] and is
distributed at http://www.junghanns.net/asterisk/.
Thanks for the info Steve. I got it but the make didn't work. Will work
on it over the weekend
I have my sip phones going into the context [from-sip] and would like to
play an introduction message and then have the caller enter the extension.
The message (dir-info was picked just to have something) doesn't play.
Maybe I misunderstood the s extension. According to what I read
The s extension is used when there is no known called number. In
other words, if you are dialing 2000, the dialplan will always prefer
the priority list for 2000 instead of going to 's', so that is why
your current system doesn't work.
John,
Thanks for the details. Actually what I want
7 - Ringer volume control
4 - plug in module of user programmable buttons for frequently called
numbers. Not everyone would need this so being able to add as an
optional module would keep the base phone cost effective.
9 - ability to switch back and forth between speakerphone and handset
7
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