On Wed, 2004-05-05 at 00:05, Ronald R. McDaniel wrote:
> I am really struggling getting Asterisk up and running utilizing these two
> cards. Has anyone had success with this combination? If so, would you be
> ever so kind to submit sample vpb.conf,zaptel.conf,zapata.conf and
> extension.conf fil
try to ask in english you may get an answer a whole lot faster
Regards,
Marc
- Original Message -
From: "Administrator" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, May 06, 2004 6:11 PM
Subject: [Asterisk-Users] Fehler beim starten...
Hallo,
nachdem mir bis jetzt n
have:
[101]
type=friend
secret=123456
auth=md5
nat=yes
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=inband
callerid="SIP Phone" <101>
mailbox=101
disallow=all
;allow=gsm
allow=alaw
allow=ulaw
context=home
- Original Message -
From: "Eric Wieling" <[EMAIL PROTECTED]>
To
[Literal translation from Google]
Hello,
after me up to now still nobody answered again my asks:
If I asterisk start get I the following error message:
[app_capiCD.so]May 6 00:38:23 WARNING[16384]: loader.c:240
ast_load_resource: /usr/lib/asterisk/modules/app_capiCD.so: undefined
symbol: ast_
hello,
i get an error when i try to loading astersik:
[app_capiCD.so]May 6 00:38:23 WARNING[16384]: loader.c:240
ast_load_resource: /usr/lib/asterisk/modules/app_capiCD.so: undefined
symbol: ast_capi_MessageNumber
May 6 00:38:23 WARNING[16384]: loader.c:408 load_modules: Loading module
app_capiC
Hello all,
This new version (0.6.1) of asterisk-oh323 fixes the "one-way audio"
problem of the previous release.
Download from the usual location:
http://www.inaccessnetworks.com/projects/asterisk-oh323
Regards,
Michael.
___
Asterisk-Users mailing list
[
>
> Sorry, 'mx' isn't yet supported - it will default to 'en'
>
> What quick fix would you like?
> 'mx' going to 'es'
> or
> 'mx' being a separate set of functions with the same syntax as 'es'
>
> It is easy for me to knock-up a patch for either scenario & post it
> to the Bugtracker. If it wor
And it actually is.the only problem is that the downloads on the
Cisco site are actually CallManager updates. So you'd need a CM server
to extract the image file (which you could then toss on whatever tftp
server you want).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PRO
- Original Message -
From: "Andy Farnsworth" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, May 06, 2004 6:02 PM
Subject: [Asterisk-Users] Playing GSM files in Windows
> For the archives...
> In trying to play GSM files in Windows (Windows XP for me, but in
> general) I fo
Fran Boon wrote:
Carlos Chavez wrote:
http://bugs.digium.com/bug_view_page.php?bug_id=0001097
This bug has been fixed in current CVS HEAD (not by using the patch
in this BugID, though).
I updated from CVS as you suggested but somehow things are worse
now. Now ALL sounds are in english. I ch
Markus--
Pardon, mein Deutsch ist, uber zeit, nicht so gut gegangen!
Es siet, als etwas mit app_capiCD.so is los gegangen. Es ist nicht installiert.
Vielleicht, mochten Sie checken, und finden sie das Grund, warum diese module
nicht installiert ist.
murf
On Thu, 2004-05-06 at 11:41, [EMAIL
Hi,
After a few attempts, I've managed to grab the files from CVS and build it
on a rh redora box I've setup especially for Asterisk. Firstly, we're new
to the asterisk scene, so please excuse any "lame" questions which may
follow..
We're a new voiptalk.org customer. We have purchased the voip
Re-flash the device with the latest firmwareI was able to turn my
old att DVG-1120M into a DVG-1120S
Mark Rizzo wrote:
Apologies, my first post had HTML formatting enabled…
I friend gave me his DVG-1120s after he realized that AT&Ts callVantage
stuff would not work for
Allow ULAW or ALAW, not both, at least for trying to solve a problem.
On Thu, 2004-05-06 at 11:36, Kyle Hagan wrote:
> have:
>
> [101]
> type=friend
> secret=123456
> auth=md5
> nat=yes
> host=dynamic
> reinvite=no
> canreinvite=no
> qualify=1000
> dtmfmode=inband
> callerid="SIP Phone" <101>
> m
Title: Message
Ist ja
lustig, jetzt sprechen sie auf einmal alle Deutsch...
Ich
wuerde asterisk komplett neu compilieren:
cd
/usr/src/asterisk
make
clean
make
install
und
dann noch einmal versuchen.
Und
als Deutscher magst Du sicherlich auch das kostenlose
sipgate.de.
Gruesse,
-
Carlos Chavez wrote:
Here in Mexico we use the same tones as in the US. In indications.conf I
simply copied the [us] section and labeled it [mx]. In the general section I
put country=mx.
Since we do not share the same tones as Spain I thought it would be
better to use a different settin
Not really i6comp.exe I think its called can extract the installshield files
and you can get the .bin files out and put them on your tftp.. not that I
have done this because I hate Call Mangler and despise running a phone
system on windows.
bkw
> -Original Message-
> From: [EMAIL PROTECTE
Hi,
> Himm am I right to understand that this box is not doing a gateway
> service, meaning no other machines are behind this machine.
This box is doing gateway services, but * is on it too.
> Or do you have a DSL/router infront of this machine and the router is
> doing the masquerading
No ther
Still one-way audio problems with version V0.6.1.
Hi Michael,
using asterisk as ISDN2H323-Gateway. Call from ISDN - Asterisk - H323 is now
ok, but in the other direction there is still only one-way audio. I hear
nothing on the H323 side.
The 2. thing is after cleareing the 1. Call i try again t
Thanks for this. It was a typing error, but makes no matter to the problem
:(
nico
Karl Brose wrote:
>
> It is iptel.ORG not DE
>
>
> IPTEL
> ===
>
> register=##username##:[EMAIL PROTECTED]/##extension#
>
>
> [iptel]
> host=iptel.org
> type=friend
> username=##username##
> secr
Yes, sounds good, thanks !!
On Thu, May 06, 2004 at 06:52:20PM +0300, Apollon Koutlides wrote:
> Christoph Adomeit wrote:
>
> >Asterisk just starts to dialout when I have entered 4 numbers, I cannot
> >enter
> >more numbers
> >
> >Overlaped dialing is configured but stops at the first match in
On Thu, 2004-05-06 at 11:02, Andy Farnsworth wrote:
> For the archives...
> In trying to play GSM files in Windows (Windows XP for me, but in
> general) I found no help on Google, so when I figured it out I thought I
> would post it here.
>
> Q: How do I play GSM Files in Windows?
> A: Use Quick
Ah and the best:
if i trace the sip-phone (snom 200) i can see traffic if i connect the
gui (it is a web interface).
But no SIP traffic.
Thats what i call "Das Grauen" or "Wudu".
nicolas
Togan Muftuoglu wrote:
> * nicolas; <[EMAIL PROTECTED]> on 06 May, 2004 wrote:
>>Thanks for answer here the
* nicolas; <[EMAIL PROTECTED]> on 06 May, 2004 wrote:
Hi,
Himm am I right to understand that this box is not doing a gateway
service, meaning no other machines are behind this machine.
This box is doing gateway services, but * is on it too.
OK
* nicolas; <[EMAIL PROTECTED]> on 06 May, 2004 wrot
Sorry and now this:
I started kphone (sip-soft-phone) on my workststion behind the gateway.
And i can register and trace traffic, so it is only not running on the
gateway.
nicolas
nicolas wrote:
> Hi,
>
>> Himm am I right to understand that this box is not doing a gateway
>> service, meaning
I don't know about 2 years ago, but it was there over 1 year ago. I
read the terms of service on April 2nd, 2003 and the disconnect fee WAS
there. I almost didn't sign up because of it, but decided the savings
would still be worth it. If only they had said all you had to do was
send the devic
Solved !
Is was so:
All configs was right but there was a little error in the network
configuration (do not know why or if it is an error, should make no
matter).
The DNS-Server entry for the internal network-card has the external
dns-server ip, have set it on the internal (forwarding).
now is
> Don't know how far you've tried to take the 1204 in terms of functions,
> but we did the same thing over a two month period and found:
>
There is also an acknowledged bug that is a showstopper for us: configuration
over DHCP fails, because the vendor code for outbound proxy is not recognised
b
On Thu, 06 May 2004 18:20:31 +0100, Fran Boon wrote
> Fran Boon wrote:
> Actually, looking at this again, 'mx' should still play digits from
> 'digits/mx' although the syntax followed would be the default 'en'
> syntax. I tested this & all seems to work ok on my system.
>
> What is your director
On Thu, 06 May 2004 18:45:15 +0100, Fran Boon wrote
> Carlos Chavez wrote:
>
> ok, we have an 'es' syntax for saynumber() but it doesn't seem to
> support "ciento uno" as yet.
> Is this the only number that changes?
> What about 102? 110? 1001?
>
All numbers like 10,20,30,40,50,60,70,80,90
Hallo!
Yes i have try my english *g*
Thanks for your answer but i don´t know i have donwnload the module and then have
installed it withe make install there is no error only wehn i try to start astersik...
Markus Dohnal
Message: 14
From: "Marc Storck" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED
James Sizemore wrote:
I am going to change my Digest realm to match my DNS SVR record.
I dug through the code in chan_sip.c and on line 2748 I found it hard
coded :
snprintf(tmp, sizeof(tmp), "Digest realm=\"asterisk\", nonce=\"%s\"",
r\anddata);
I'm going to change this to :
snprintf(tmp, si
Hello all,
Just to inform you all, next version released, please try it and let me
know about any bugs you find (or any further features). This release now
includes
1/ Inbound calls
2/ Call origination
3/ Call dialling from phone detected
4/ Call origination using contexts
5/ Can set the caller I
Olle E. Johansson wrote:
James Sizemore wrote:
Has anyone else changed the "Digest realm"? Did you have any odd
problems?
In the chan_sip2 module, I've a setting called "realm=" in sip.conf
Time to port that over to chan_sip.
No, it doesn't cause any harm. On the contrary, the RFC states that th
I recently had a bear of a time getting a Polycom Soundpoint 500IP up
and registered.. Now that its registered I ran into a problem w/ the
dialplan.
Needing to dial x101 I'd dial 10 - then get a fast buzy.. Also making a
local call - dialing 95551212- would give me a fast busy after the 7th
Carlos Chavez wrote:
My sounds live in:
/var/lib/asterisk/sounds/mx
/var/lib/asterisk/sounds/digits/mx
Until I upgraded yesterday to the latest CVS I got most sounds from the
"mx" directories. I only had the problem with some digits. Since the upgrade
all sounds play as "en".
I am still
Carlos Chavez wrote:
ok, we have an 'es' syntax for saynumber() but it doesn't seem to
support "ciento uno" as yet.
Is this the only number that changes?
What about 102? 110? 1001?
All numbers like 10,20,30,40,50,60,70,80,90 and 100 have this problem.
They all change when you have another nu
I tried this lastnight verson 2 and it wouldn't work... hrm I guess i'll try
again.
bkw
- Original Message -
From: "Nick Knight" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, May 06, 2004 2:31 PM
Subject: [Asterisk-Users] Asttapi
> Hello all,
>
> Just to inform you all, ne
Hi,
Does anyone know where you can get a SMARTnet
contract from in the UK to get SIP images for Cisco Phones ?
I urgently need the SIP Image for a Cisco 7905G,
but I can't get hold of a contract.
I e-mailed a company is America and they said they
supplied the contracts, but only if the
On Thu, 2004-05-06 at 18:15, Carlos Chavez wrote:
> On Thu, 06 May 2004 18:45:15 +0100, Fran Boon wrote
> > Carlos Chavez wrote:
> >
> All numbers like 10,20,30,40,50,60,70,80,90 and 100 have this problem.
> They all change when you have another number after. The only exception is the
> 100
Doesn't show up in outlook at all ... oh well guess I'll try again later.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Nick Knight
> Sent: Thursday, May 06, 2004 3:32 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Astt
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi
~ Some one know where i can find some documentation about how to
programm some
modules for asterisk.
~ Becouse i want to program a call limit per user.
- --
Alvaro Ivan Parres Peredo
Director de IT
[EMAIL PROTECTED]
Tel: (33) 36301294
~ (33)
HAHA got it but asterisk seg_faulted off to debug
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of brian
> Sent: Thursday, May 06, 2004 5:05 PM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] Asttapi
>
> Doesn't show up i
app_groupcount.c (this is in cvs-head)
exten => 999,1,SetGroup(moh)
exten => 999,2,CheckGroup(1)
exten => 999,3,Answer
exten => 999,4,MusicOnHold(default)
exten => 999,103,Busy
See?
You can limit that to just 1 user at a time or what ever you wish :
bkw
> -Original Message-
> From: [EM
Hi Nick,
When I click on dialling options and then line properties asterisk
appears 3 times? Is this normal, does it matter which one I choose?
If I try to use this outlook replies with the service is busy on another
call with nothing actually appearing in the asterisk console.
Also one other q
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Yes but i want to limit like user of ext 120 only have 20 min of calls
other one have 60 min of calls at moth thinks like that
brian wrote:
| app_groupcount.c (this is in cvs-head)
|
| exten => 999,1,SetGroup(moh) exten => 999,2,CheckGroup(1) exten =>
Show application SetAbsoluteTimeout
bkw
- Original Message -
From: "Alvaro Parres" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, May 06, 2004 4:45 PM
Subject: Re: [Asterisk-Users] HOW TO PROGRAM NEW MODULES
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Yes but i
Alvaro Parres wrote:
Yes but i want to limit like user of ext 120 only have 20 min of calls
other one have 60 min of calls at moth thinks like that
Then you build extension logic around SetGroup/CheckGroup that does that.
Jeremy McNamara
___
Aste
We find that mpg123 0.59r works best. mpg123
0.59s-mh4 = the devil.
What versions does everyone use without
problems.
0.59r is PERFECT
bkw
| On Thu, 2004-05-06 at 11:02, Andy Farnsworth wrote:
| > For the archives...
| > In trying to play GSM files in Windows (Windows XP for me, but in
| > general) I found no help on Google, so when I figured it out I thought I
| > would post it here.
| >
| > Q: How do I play GSM Files in Windows?
|
Shouldn't the endpoint be MGCP/aaln/[EMAIL PROTECTED] instead of
[EMAIL PROTECTED]? AFAIK, the MGCP RFC recommends aaln/# for
analog lines.
Can you audit the endpoint to check if you got the name right?
Hope this helps.
Juan J. Sierralta P. wrote:
On Wed, 2004-05-05 at 13:45, Brad White wrote
This may have been in the archives but I didn't see it. I have a new * setup
with the demo loaded and ready to go. when I dial in, I get "extension (my *
number) in context default from (calling number) does not exist" and it
denies the call. this is trying to use the "s" extension (default demo
ex
brian wrote:
Not really i6comp.exe I think its called can extract the installshield files
and you can get the .bin files out and put them on your tftp.. not that I
have done this because I hate Call Mangler and despise running a phone
system on windows.
I ordered a desktop charger cradle for the ph
Hi please contact me offlist I may be able to assist [EMAIL PROTECTED]
Kind Regards
Chris Cornish
Cornish Business Solutions 08 9490 1795
Ezy As Internet Services 08 9398 9077
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Altus Snyman
Sent: Friday
Here's our config:
cisco 7960's running 6.3 sip code
latest cvs of *
t100p zaptel card
adit 600 channel bank
7 pots lines and 2 fax machines on the adit 600
dialing out from the cisco phones gets sent out via the zap channels, but
I'm having some serious echo problems. I currently have the adit
Hi
I have recently updates to the latest cvs of asterisk, openh323 and pwlib as
recommended.
The OPenh323 and pwlib compile fine.
When compiling the Asterisk-oh323 I get the following errors, I have checked that the
envorinment variables are set correctlty as below.
PWLIBDIR=/usr/src/pwlib
OP
apply the openh323 patch (it's in the root of ast-oh323), recompile
openh323 and it should work fine
David Hindmarsh wrote:
Hi
I have recently updates to the latest cvs of asterisk, openh323 and pwlib as recommended.
The OPenh323 and pwlib compile fine.
When compiling the Asterisk-oh323 I get
Just had another thought, about replacing the cb and zaptel card with a
sip<>analog gateway... Can anyone recommend one? (in case I can't get
this straightened out)
> Here's our config:
>
> cisco 7960's running 6.3 sip code
> latest cvs of *
> t100p zaptel card
> adit 600 channel bank
> 7 pots l
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