Oh dear. You can either manually enter in the missing line or apply the
attached patch as before (you need a clean res_parking.c which you can get
by deleting the file and then doing cvs co asterisk again). This patch
works on my system updated to the latest cvs.
Iain
--On Wednesday, March
At 05:44 AM 2/18/2004, you wrote:
2. can Two SIP phones login
to * at the same time with the same
username/pwd ? how to prevent
this?
I also want to know if its possible to prevent multiple
logins..
Regards,
Antonio Rabena
Paul,
Are you able to get the music on hold working with the Mepis asterisk
package? I dont have a Zaptel board and I guess the solution is that I
would have to modify the source?
Thanks,
Kevin
-Original Message-
From: Paul Mahler [mailto:[EMAIL PROTECTED]
Sent: Sunday, January 18,
Hi,
At my company we have a large CCM-installation, is it possible to / how to
connect between asterisk and CCM.
I'm quit shure that the CCM only use Skinny.
Any idea of the hardware-size for 1000 users ?
/Hans-Henrik Andresen
___
Asterisk-Users ma
On Wed, 2004-03-10 at 20:30, Steven M. Sokol wrote:
Or, you could apply my patch, that I've been upgrading on the asterisk bug
site.
Check http://bugs.digium.com/bug_view_page.php?bug_id=752
Steve (and anybody else who may know about this code),
I have the code for your privacy enhanceme
I am having a similar problem... I get the same message, but inbound
calls can go through This is only Cisco phones that are behind NAT.
I have tried your recommendations from below, but still no luck.. User
can make outbound calls, just can't receive any. Any ideas would be
greatly appreciate
Is there any rule of thumb that should be used for bandwidth required per
user?
Ben
- Original Message -
From: "Stephen Varga" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, March 10, 2004 7:11 PM
Subject: Re: [Asterisk-Users] IAX Connection-Latency/Bandwidth Question
Hi List,
I am trying to send calls to the PSTN via a Quintum CMS using H.323.
The calls are failing due to the absence of the ANI or Calling Party
Numer/CallerID
The Quintum is trying to read this info using H.245 which apparently is not
present.
I have set H245 tunnelling ON, set the callerid
Or, you could apply my patch, that I've been upgrading on the asterisk bug
site.
Check http://bugs.digium.com/bug_view_page.php?bug_id=752
Steve (and anybody else who may know about this code),
I have the code for your privacy enhancements compiled and installed. My
one question is, how do I
On Thursday 11 March 2004 02:39 am, admin wrote:
> Ping times (latency) and bandwidth are really not related unless you are
> filling the pipe. Your ping times are too high. My understanding is that
> anything over 100ms is not good. Your problem probably lies with too many
> hops or slow or ove
admin wrote:
From a windows box run tracert and you will get a better idea where the
problem lies.
Ew! Why on earth would you want to use winblows for traceroute? From a
*nix box, run traceroute . Now, that's better.
John
___
Asterisk-Users ma
Better program to use is Neotrace just googled up this link for a
downloadable copy
http://www.networkingfiles.com/PingFinger/Neotraceexpress.htm
Not sure of the differences between this version and the pro version I
use but shows node by node or world views etc.
Also just noticed that Mcafee ha
Ping times (latency) and bandwidth are really not related unless you are
filling the pipe. Your ping times are too high. My understanding is that
anything over 100ms is not good. Your problem probably lies with too many
hops or slow or overburdened router along the path.
>From a windows box run
> >Can you test this with an extension that goes into VoiceMailMain(). My
> >7960 and 7960G phones both get the first couple letters of "Commedian
> >Mail" cut off (usually "...median Mail").
For my first two or three months of using Asterisk I had this problem
(with a Cisco 1750 and Cisco FXO an
On Wed, Mar 10, 2004 at 04:40:38PM -0500, John Fraizer wrote:
> Steve Kennedy wrote:
> >It't not quite that simple, DSL in the UK is PPPoATM, so you need to
> >take into account who IP is encoded at the ATM layer etc. If you're
> >using a reasonable bit rate codec, you can't really expect to get m
Steve Creel wrote:
On Wed, 10 Mar 2004, John Fraizer wrote:
For what it's worth, I don't have any delay between answer and audio with my
asterisk server and 7960G either originating or answering. It doesn't
matter if it's a call to/from another SIP/IAX device or to/from PSTN. It's
pretty muc
Jeremy, I am really not interested in rehashing this again.
You know my views on the matter, I know yours. We disagree.
On Wed, Mar 10, 2004 at 08:04:52PM -0500, Jeremy McNamara wrote:
>
> I seem to recall your http://www.gnutel.com publicly discussing a fork
> of Asterisk.
This is no secret (t
Hi Gus,
Thanks for your help. But I am still struggling to get it to work.
How do you put the data into 'provider' to allow a call which works in
asterisk (extensions.conf)
exten => s,1,Dial(IAX2/Username:[EMAIL PROTECTED]/${EXTEN})
Where ${EXTEN} is a dialed number
Thanks in advance.
Alex.
--
Adam-
Thanks for the info- there are not too many out there, and before I start
placing these in the field, I'm looking for any early bugs.
Look forward to your other email..
regards
Scott
Scott M. Stingel
Emerging Voice Technology Inc.
Palo Alto, California and London, England
Email:
Ok so Benjamin, can I take it from your comment about beefing up
customer sales that you are going to run an off shore call centre and
the majority of your calls are going to be coming externally from the
pstn into your asterisk then out your dsl connection to the overseas
site through their dsl to
William Waites wrote:
On Wed, Mar 10, 2004 at 05:01:38PM -0500, Jeremy McNamara wrote:
That fact is not the problem. It the fact that there is no FORK of
Asterisk that Digium secretly maintains. This is how rumors get
started.
If memory serves, you were the one who started that rumour.
Dean,
I'm a newbie butYes, we are going to route all of our call from the
remote office through our local office and then to the PSTN.
We want to beef up our customer service so they will all be talking at the
same time.
Ben
- Original Message -
From: "Dean Collins" <[EMAIL PROTECTE
Ben,
Are you going to be routing all of your calls from the remote office
back through your pabx onto the PSTN or are you assuming that they have
no need for local calls?
If so how many calls do you expect over the network at any one time?
Cheers,
Dean
-Original Message-
From: [EMAIL PR
Hello!
I am new to * here and have a question
We are installing * currently in the US. We have a small remote office
overseas. We would like to enlarge that office to about 20 people using IP
phones.
My questions are:
1. I pinged the website of the ISP twice from the dsl line I am on wh
Alex:
In 'call' table stores call details.
'card' stores user & pin (10 digits in original version)
'country' associates a short description with a long description of
destination.
'countryprefix' associates prefix (i.e. 1305) with short description (of
'country' table) and type of destination (fi
On Thu, 2004-03-11 at 03:11, Scott Stingel wrote:
> Hello-
>
> I'm considering some TE405P's for a customer of mine. This is the 5 volt
> version of the TE410P.
>
> Digium is now shipping these - does anyone have production experience with
> these cards in the field?
yes, was there something
I have looked at matt's asterisk gui client at sourceforge. I am not a
programmer by trade. The documentation there seems to be a bit lacking.
Has anyone have the experience in installing the gui client and may
perhaps have a how-to document available for sharing.
--
David Kwok
Tel: 612 9929208
Hello all,
When I dial an international number, Asterisk dials
the incomplete number before I put in all digits.
For example, when I dial to 01135321XXX, it
pickups and dials 01135321 before I complete the dial.
If I press the redial, it will dial with all digits
that I put in b
On Wednesday 10 March 2004 17:00, Stephen R. Besch wrote:
> I have noticed that the voicemail app always keeps filenames in in
> a strict numerical sequence, obviously renaming files whenever a
> message is deleted. I assume that message processing depends upon
> this sequencing. Does the file age
Hi guys,
Has anyone played around/got it to work app_prepaid.c?
(http://www.voip-info.org/wiki-Asterisk+callingcard)
With what data do you populate the database with cards, providers,
tariffs, tariffrates etc.. (format) to make it work. What is the
meaning/purpose of each table/field?
I am gett
Hi guys,
Has anyone played around/got it to work app_prepaid.c?
(http://www.voip-info.org/wiki-Asterisk+callingcard)
With what data do you populate the database with cards, providers,
tariffs, tariffrates etc.. (format) to make it work. What is the
meaning/purpose of each table/field?
I am gettin
hi
we need a predictive dialer which can be used with asterisk software. Is it possible?
Bye
Owais Bin Zuber
Do you Yahoo!?
Yahoo! Search - Find what youre looking for faster.
I have noticed that the voicemail app always keeps filenames in in a
strict numerical sequence, obviously renaming files whenever a message
is deleted. I assume that message processing depends upon this
sequencing. Does the file age make any difference in determining the
numerical ordering (i.e
> >This is why disclaimers are important for those who
> >contribute patches. If there isn't a disclaimer, Digium can not include
> >it in the proprietary version of asterisk. If they can not include it in
> >the proprietary version, they tend to not allow it in their version of
> >the GPL release
On Wed, 2004-03-10 at 15:53, Doug Harris wrote:
> Thanks Steven
>
> >> Dough
>
> >>Fairly appropriate sig.
>
> Didn't you notice my email handle :)
Yes, I understood what it was for. A former boss used dough for his
email when his doug address was over run with spam. It took a few of us
pointin
On Wed, Mar 10, 2004 at 05:01:38PM -0500, Jeremy McNamara wrote:
>
> That fact is not the problem. It the fact that there is no FORK of
> Asterisk that Digium secretly maintains. This is how rumors get
> started.
If memory serves, you were the one who started that rumour.
I remember you claim
Jeremy McNamara wrote:
started.Didja hear about RedHat buying out Digium ? Or was it
Microsoft?
Jeremy McNamara
You've got it all wrong, man! Microsoft bought SCO who bought RedHat who
bought Digium. ;)
John
___
Asterisk-Users mailing list
[E
Thanks Steven
>> Dough
>>Fairly appropriate sig.
Didn't you notice my email handle :)
Well I will wait for "dbruce" for my specific questions on how to use it.
Thanks
D.Harris (new sig.)
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lis
This is more of a feature request. I've seen that there's no configuration
that allows you to remap voicemail keys(it being compiled into the source and
all) but I'd like to request the ability to do so for replacement of existing
PBX systems. It makes the integration so much easier.
Also,
On Wed, 2004-03-10 at 15:42, Jeremy McNamara wrote:
> Steven Critchfield wrote:
>
> >Maybe you should read and understand the comments on licensing. Maybe a
> >going over the licensing threads here would also be needed.
> >
> >For the short story, Digium dual licenses asterisk. There is a GPL
> >
Hey,
I am looking for corporate level solution for a conference phone to use with
an Asterisk system.
Any ideas?
Current System:
Asterisk CVS-02/25/04-18:06:52
Red Hat 9.0
3 X100P cards
3 PSTN lines
Ryan R. Fligg
Secured Digital Storage, Inc.
104 SW 4th St.
Des Moines,
[no name provided] wrote:
You're right. It's symmetric so it only takes 83Kbits/sec for u-law.
IPTraf is confusing me :-)
IPTraf is a neat tool but, information it gives should be taken with a grain
of salt. If you're looking at "general" statistics, it is going to show you
the combined IN+OU
Derek Samford wrote:
I believe that's what Steven was saying. That if this code was to ever
be included in the asterisk source then there would need to be a
disclaimer from Derek Bruce, as well as the Freeradius dev team. He
pointed out that this keeps there from being any need for a fork, and
mak
It is included with Asterisk.
John
hank wrote:
- Original Message -
From: hank
To: [EMAIL PROTECTED]
Sent: Wednesday, March 10, 2004 1:07 PM
Subject: where can I get Commedian mail at?
hello where can I get Commedian
Mail at?
thanks
hank
__
Anthony,
Asterisk by default allows pass through. You shouldn't need a
license. It's only when you need to do transcoding (I.E. you need to
decompress the voice, whether it be for Codec translation, to dial out a
Zap channel, which would be ULAW or ALAW, Voicemail (still technically
codec t
You're right. It's symmetric so it only takes 83Kbits/sec for u-law.
IPTraf is confusing me :-)
- Original Message -
From: "John Fraizer" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, March 10, 2004 11:41 AM
Subject: Re: [Asterisk-Users] Asterisk Codecs [G.729]
>
> You n
Here's my patch results:
[EMAIL PROTECTED] asterisk]# patch -p0 < ./Parking.patch
patching file res/res_parking.c
Hunk #1 FAILED at 25.
Hunk #2 succeeded at 228 (offset 13 lines).
Hunk #3 succeeded at 288 (offset 12 lines).
Hunk #4 succeeded at 408 (offset 13 lines).
1 out of 4 hunks FAILED -- sav
I believe that's what Steven was saying. That if this code was to ever
be included in the asterisk source then there would need to be a
disclaimer from Derek Bruce, as well as the Freeradius dev team. He
pointed out that this keeps there from being any need for a fork, and
makes things much less co
- Original Message -
From: hank
To: [EMAIL PROTECTED]
Sent: Wednesday, March 10, 2004 1:07 PM
Subject: where can I get Commedian mail at?
hello where can I get Commedian> Mail
at?
thanks
hank
Steven Critchfield wrote:
Maybe you should read and understand the comments on licensing. Maybe a
going over the licensing threads here would also be needed.
For the short story, Digium dual licenses asterisk. There is a GPL
license for those of us that don't need proprietary support, and then
t
Steve Kennedy wrote:
It't not quite that simple, DSL in the UK is PPPoATM, so you need to
take into account who IP is encoded at the ATM layer etc. If you're
using a reasonable bit rate codec, you can't really expect to get much
more than 1 voice channel out of an ADSL service (assume something lik
Try the attached patch. Go to your asterisk root directory and type:
patch -p0 < path_to_patch/Parking.patch
.. then rebuild asterisk.
Iain
--On Wednesday, March 10, 2004 7:43 am -0500 John Congdon <[EMAIL PROTECTED]>
wrote:
I have applied the patch and restarted Asterisk.
But it still
On Wed, Mar 10, 2004 at 08:41:00PM +, Michael T Farnworth wrote:
> Mark Phillips wrote:
> >What codecs are you using? You should be able to get quite a few speex
> >channels or even a load of 729 or gsm channels down your 256K.
> You always have to remember that a UK ADSL line has a contention
Dear Michael,
Does your H323 driver run T38 Fax? Also, does your H323 driver have the
capability of just proxying signal, and NOT proxying signal and media, just
like the canrevite=yes in the sip scenario? Thanks
TC
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Beh
On Wed, 2004-03-10 at 14:13, Doug Harris wrote:
> Hi,
>
> http://bugs.digium.com/bug_view_page.php?bug_id=0001193
>
> First of all thanks for doing this. Now we can play with any VoIP g/w
> in the same level field. Being a new user always wondered why there is
> no radius support in asterisk.
>
Doug Harris wrote:
First of all thanks for doing this. Now we can play with any VoIP g/w
in the same level field. Being a new user always wondered why there is
no radius support in asterisk.
RADIUS is absolutely not necessary... We have countless gateways (5300s,
Qunitum, etc) running perfect
On Wed, Mar 10, 2004 at 11:50:55AM -0800, Lee Howard said:
> On 2004.03.10 10:18 Jim Sneeringer wrote:
>
> >Do you know if there is other fax software that supports ECM?
>
>
> Modems that I know of that support ECM in some fashion in Class
> 2/2.0/2.1 are the MultiTech 5634 V92 family (MT5634Z
Steve Creel wrote:
Can you test this with an extension that goes into VoiceMailMain(). My
7960 and 7960G phones both get the first couple letters of "Commedian
Mail" cut off (usually "...median Mail").
Just trying to quantify the delay we're talking about...
Steve
exten => 8500,1,Answer
exten =
Mark Phillips wrote:
What codecs are you using? You should be able to get quite a few speex
channels or even a load of 729 or gsm channels down your 256K.
Mark
You always have to remember that a UK ADSL line has a contention ratio
of 20:1 if you have business ADSL or 50:1 if you have the consumer
Sorry for my ignorance but what is the difference between using the G.729
codec and using G.729 pass thru. In my scenario below does it consider to be
using the G.729 or using it as pass through?? Do I still need licence for
the G.729?
SIP(if using g.729) --->asterisk->h323
softswitch(g729)--
Same behavior here. IP500 and 7960G phones cutoff first part of VoiceMailMain.
-sb
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve Creel
Sent: Wednesday, March 10, 2004 3:18 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco 7960 and short
I have experienced this behavior on the 7960 as well.
- Chris
- Original Message -
From: "Steve Creel" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, March 10, 2004 3:18 PM
Subject: Re: [Asterisk-Users] Cisco 7960 and short delay before voice starts
after ring.
> On Wed,
On Wed, 10 Mar 2004, John Fraizer wrote:
>
>For what it's worth, I don't have any delay between answer and audio with my
> asterisk server and 7960G either originating or answering. It doesn't
>matter if it's a call to/from another SIP/IAX device or to/from PSTN. It's
>pretty much instant (not
Hi,
http://bugs.digium.com/bug_view_page.php?bug_id=0001193
First of all thanks
for doing this. Now we can play with any VoIP g/w in the same level field. Being
a new user always wondered why there is no radius support in
asterisk.
Sorry for the stupid
question; Why is this in bug note
On Wednesday, March 10, 2004 12:19 PM, Jim Sneeringer [SMTP:[EMAIL PROTECTED]
wrote:
> Darren,
>
>
> Also, why should Asterisk introduce a lot of noise? (Faxes work fine
without
> Asterisk or with the old phone system.) I'm using a 2.4GHz Pentium 4
> with two X100P's and two TDM400P's. Faxes are
On 2004.03.10 10:18 Jim Sneeringer wrote:
Do you know if there is other fax software that supports ECM?
Please forgive me for butting in on this thread, but I can't resist
plugging HylaFAX.
Almost all fax software, including your WinFax 7.0, that supports Fax
Class 2, Class 2.0, or Class 2.1 wi
Bisker, Scott (7805) wrote:
> What versions of Zaptel, Asterisk, and libpri?
>
>
I downloaded them all at the same time from CVS. I really couldn't tell you
though off the top of my head.
John
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://l
You need to decide if you're going to measure both sides of the call or not.
ITU standard is 64Kbits/s. That is correct. It is a standard DS0. But,
guess what. That DS0 goes both directions so, "measured bandwidth per call"
is 128Kbits/s using your logic.
Only "consumer" grade DSL/Cable ba
I believe your gatekeeper or your gateway is refusing the call. This can be
a authorization problem in the gatekeeper or codec problem in the gateway.
You need to see where your call is failing. Try to do the following:
1 - Turn on the oh323 trace in the oh323.conf file adding these lines to
your
What versions of Zaptel, Asterisk, and libpri?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Fraizer
Sent: Wednesday, March 10, 2004 2:08 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco 7960 and short delay before voice
starts after ring
All the fax modems I have seen do send CNG tones, but I have seen some very
old standalone fax machines that do not. They used to be quite common, but
in the last few years I have only seen one fax machine that does not support
CNG.
One workaround is to send the call to the fax machine on a timeou
Absolutely agree,
ITU standard is 64Kbits/sec.
VoIP traffic with U-law per channel is 83Kbits/sec.
VoIP traffic with U-law per call is 166Kbits/sec [measuring bandwidth per
call]
- Original Message -
From: "Nicolas Bougues" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, M
For what it's worth, I don't have any delay between answer and audio with my
asterisk server and 7960G either originating or answering. It doesn't
matter if it's a call to/from another SIP/IAX device or to/from PSTN. It's
pretty much instant (not detectable by humans at least). So, there may
Hi,
Thanks for your answer, but my asterisk is working as a H.323 - SIP gateway and
calls between SIP clients (phone and soft clients) are working all right. The
only problem I have, is like I have said in my mail is between sip phones and
PBX.
Best Regards,
Mireia
PS: Someone have other ideas?
I've been playing around with an AGI script that pops up a callerID window
on a user's computer and I have it working with most Asterisk calls except
for calls that a queue agent receives from a queue. Does anyone know how to
run an AGI script at the time that a call is transferred from the queue t
On Wed, 2004-03-10 at 12:25, Michiel Betel wrote:
> Eric Wieling wrote:
>
> >First of all Asterisk does not support ${VARIABLES} as part of the
> >extension number. i.e. exten => ${BLAH},1,NoOp is not valid, but exten
> >=> 1234,1,NoOp(${BLAH}) is valid.
> >
> >
> Huh !?! Astreisk might not sup
Eric Wieling wrote:
First of all Asterisk does not support ${VARIABLES} as part of the
extension number. i.e. exten => ${BLAH},1,NoOp is not valid, but exten
=> 1234,1,NoOp(${BLAH}) is valid.
Huh !?! Astreisk might not support it but it seems to work fine in
my setup, show dialplan expand
Darren,
Thanks so much for your help.
Do you know if there is other fax software that supports ECM?
Also, why should Asterisk introduce a lot of noise? (Faxes work fine without
Asterisk or with the old phone system.) I'm using a 2.4GHz Pentium 4 with
two X100P's and two TDM400P's. Faxes are garb
Adam,
Does Polycom license the SIP stuff from Cisco? If not, then Asterisk may be the
culprit, because all of my Polycom IP500s exhibit the same behavior. I'm running
asterisk 0.7.1, Zaptel CVS and libpri CVS both from a few days ago, but I don't recall
having this problem a few months ago wh
First of all Asterisk does not support ${VARIABLES} as part of the
extension number. i.e. exten => ${BLAH},1,NoOp is not valid, but exten
=> 1234,1,NoOp(${BLAH}) is valid.
Also Asterisk NEEDS the sending fax machine to send standard fax machine
tones (CNG, I think) for it to be detected. When yo
Well I just took a look at the TAC case and things dont look good, seems the TAC are
now blaming Asterisk for the problem but I will go through there debugs and push back,
will let you know.
-Original Message-
From: James Sizemore [mailto:[EMAIL PROTECTED]
Sent: 08 March 2004 22:09
To: [
Jim,
For the sake of argument, let's assume something about your Asterisk setup
is introducing acoustic artifacts like crackles, pops, echoes etc into the
fax transmission. The best way to overcome something like that is to make
sure your fax transmissions use ECM error correction whenever possibl
On Wed, Mar 10, 2004 at 09:03:57AM -0800, wrote:
> Rich,
>
> In real world, using real tool, getting real number. You don't expect to
> either talk only mode or listen only mode. Per call must have Rx & Tx for
> inbound & outbound.
>
[...]
>
> Engineering rate is per channel but to calculat
It can also be due to the fact that the GS phone can't get in touch with
the time server.
Make sure you have no voicemail. Check the sip debug to make sure
message waiting is set to no.
Then its probably the time server.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL
I think this is specific to WinFax. I have my setup similar where I have
X100 -> TDM400P -> faxmodem(Intel 144) -> Win2000 Server running its
FaxService.
I have the faxmodem plugged into a el cheapo faxmachine(for sending only,
crappy for receiving) which is plugged into the TDM400P. I have the
What codecs are you using? You should be able to get quite a few speex
channels or even a load of 729 or gsm channels down your 256K.
Mark
WipeOut said:
> Simon Coles wrote:
>
>>
>>
>> --On Tuesday, March 2, 2004 9:49 am + Steve Kennedy
>> <[EMAIL PROTECTED]> wrote:
>>
>>> That's the crunch (
On Tuesday 09 March 2004 18:14, Derek Bruce wrote:
> Well, after thinking about it some more... try this:
>
> exten => s,1,ChanIsAvail(SIP/2001&IAX2/[EMAIL PROTECTED])
> exten => s,2,cut,ToDial=${AVAILCHAN},1
> exten => s,3,Dial(${ToDial},20)
The correct syntax for Cut is:
exten => s,2,Cut(ToDial
Rich,
In real world, using real tool, getting real number. You don't expect to
either talk only mode or listen only mode. Per call must have Rx & Tx for
inbound & outbound.
The numbers look more like 83Kbits/sec (thanks Andrew Gillham) in the wiki
page or the cisco bandwidth consumsion (thanks
Simon Chappell wrote:
I am no Asterisk mystro (a newbie really)
but here is my pennys worth..
I have GS budgettones..
extracts from conf files..
sip.conf
[2000]
type=friend
username=2000
host=dynamic
dtmfmode=info
[EMAIL PROTECTED]
context=sip
callerid=2000
secret=password
canreinvite=no
extensions
I'm having the same symptoms using X100P's, TDM400P's and WinFax, and have
and have had no luck correcting it. I don't have a standalone fax machine to
test with.
Does anyone know if this a problem whenever faxes are sent and received with
a modem, or is it specifically WinFax? Is there any other
I am no Asterisk mystro (a newbie really)
but here is my pennys worth..
I have GS budgettones..
extracts from conf files..
sip.conf
[2000]
type=friend
username=2000
host=dynamic
dtmfmode=info
[EMAIL PROTECTED]
context=sip
callerid=2000
secret=password
canreinvite=no
extensions.conf
[sip]
;local ext
Apologies if this is a dupe. I haven't seen my post in archives or echoed
back to me.
I'm trying to set up my first phone, Windows Messenger running on an
attached PC.
It's talking to the server, but the server never responds. Instead, it
gives this error:
Mar 10 06:26:20 WARNING[163851]: cha
Hi Mieria,
Mireia Munoz de jesus wrote:
Hi!
When I try to call from a SIP phone to a PBX phone I get this error:
chan_oh323.c [1004] Couldn`t call 483377839
and if I get the messages from SIP debug, I have a 403 message. The
configuration of my system is:
SIP Phone ASterisk Gatekeepe
On Wed, 2004-03-10 at 17:02, Chris Lee wrote:
> My backlight is flashing and I have the stutter tone, only I dont have
> any mail waiting, why can I not get the phones to stop doing this?
Check that
/var/spool/asterisk/voicemail/default/MAILBOXNUMBER/INBOX
does not contain a msg.txt file.
do you need a spicial phone line to run capi cards?
- Original Message -
From: "Diego Ercolani" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, March 10, 2004 5:47 AM
Subject: [Asterisk-Users] Fax support and 'f' DTMF tone extension
> Hello,
> probably is a feature what I'm a
what is this vb 206/?
- Original Message -
From: "Abdul Hakeem" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, March 10, 2004 5:27 AM
Subject: RE: [Asterisk-Users] From 0 to PBX in 2 hours
> Hi,
> Yes, and many thanks for the offer.
> Can you email or arrange for FTP ?
> C
Hello-
I'm considering some TE405P's for a customer of mine. This is the 5 volt
version of the TE410P.
Digium is now shipping these - does anyone have production experience with
these cards in the field?
Many thanks in advance.
Scott M. Stingel
Emerging Voice Technology Inc.
Palo Alto,
don't think so but I wish it did
- Original Message -
From: "Abdul Hakeem" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, March 10, 2004 12:53 AM
Subject: RE: [Asterisk-Users] From 0 to PBX in 2 hours
> Hello,
> Does anyone know if a GUI for Asterisk exists ?
> Regards,
> A
On Wed, Mar 10, 2004 at 07:37:03AM -0600, Jim Sneeringer wrote:
> I have been told that Voicetronics cards are not supported by Asterisk, but
> I don't know for sure.
I used to use the VPB4 cards.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
Tim Sailer wrote:
On Mon, Mar 08, 2004 at 03:12:52PM -0600, [EMAIL PROTECTED] wrote:
Simon,
Do the GS phones support stutter tone as-well-as
the message light?
I'm not Simon, but yes, they do. At least my -100 does. The display
backlight flashes, and you get the stutter dialtone.
Tim
My backli
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