2009/7/7 Doug Lytle supp...@drdos.info
Olivier wrote:
Please, allow me to ask what is this Transmitting Station ID ?
Google is you friend:
http://encyclopedia.thefreedictionary.com/Transmitting+Subscriber+Identification
Thanks !
I still have to improve my googling !
Doug
Hi,
Reading this thread, is this correct to say CallerName is widely used in the
US ?
Here in France, this service is optional but I don't think many companies
are subscribing to it and I'm not aware of any non-Telco CNAM providers.
I would curious to know how the situation is elsewhere.
2009/7/7 Brent Davidson br...@texascountrytitle.com
Is there any possibility of DAHDI supporting Automatic gain control on
TDM ports? I'm having issues at a couple of offices where calls made to
local numbers are fine but a when a calls from or goes to a large
percentage of long-distance or
2009/7/7 Jared Smith jsm...@digium.com
On Tue, 2009-07-07 at 15:42 +0200, Klaus Darilion wrote:
I am searching for the description of the available dialstrin options
for the DAHDI channel (and also other channel types).
I am not looking for outdated voip-info links, but for the
2009/7/6 Jonathan Thurman jthurma...@gmail.com
We are currently moving away from a wide-spread Cisco CallManager
deployment to Asterisk. For many of our small sites we have the routers
configured for what Cisco calls SRST so if we have a WAN failure, the router
acts as a SCCP registrar. We
This is not currently possible. Work in progress.
--
Sent from mobile device
On Jul 8, 2009, at 1:31 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
Hello All,
can anybody tell me how can i integrate asterisk and skype users
so that skype users can dial my asterisk number or dial
Jared Smith schrieb:
On Tue, 2009-07-07 at 15:42 +0200, Klaus Darilion wrote:
I am searching for the description of the available dialstrin options
for the DAHDI channel (and also other channel types).
I am not looking for outdated voip-info links, but for the authoritative
source, e.g.
17. Automatic Gain Control (Brent Davidson)
Is there any possibility of DAHDI supporting Automatic gain control on
TDM ports?
Have a look at asterisk-1.6.1 and module func_speex.so, which provides
AGC function. This function can be applied to any channel.
Documentation:
DHAVAL INDRODIYA wrote:
Hello All,
can anybody tell me how can i integrate asterisk and skype users
so that skype users can dial my asterisk number or dial internal
dialplan form skype
regars
Dhaval
Chan_celiax can apparently interface with a copy of the skype client
running on the
That's exactly the way I do it as well :D
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian
Lyndon-Smith
Sent: 06 July 2009 11:16
To: Asterisk Users Mailing List -
On 8/7/09 8:52 PM, Andrew Thomas wrote:
That's exactly the way I do it as well :D
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian
Lyndon-Smith
Sent: 06 July 2009
This is my jabber.conf :
[general]
debug=yes ;;Turn on debugging by default.
;autoprune=no ;;Auto remove users from buddy
list.
;autoregister=yes ;;Auto register users from buddy
list.
[asterisk]
when using sisky you could integrate an ivr menu
Alex Balashov schreef:
This is not currently possible. Work in progress.
--
Sent from mobile device
On Jul 8, 2009, at 1:31 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
Hello All,
can anybody tell me how can i integrate
Thanks Miguel Molina :)
I was bit curious about that as I am using few asterisk boxes connected to
a mysql server. And that mysql server sometimes gets lots of connections
from other sides ( other than asterisk boxes) . So if asterisk-mysql holds
dedicated persistant connection , it means cdr are
On Wed, Jul 8, 2009 at 2:14 AM, Olivieroza-4...@myamail.com wrote:
Hi,
Reading this thread, is this correct to say CallerName is widely used in the
US ?
Here in France, this service is optional but I don't think many companies
are subscribing to it and I'm not aware of any non-Telco CNAM
Hello, I heard that since asterisk 1.6.0.6, now you can send faxes with t.38
through asterisk to a PST gateway that supports t.38 too. Is that true ? If
so, what elements you need to make it work beside asterisk and the PSTN
trunk ?
Thanks all.-
___
--
On 7/8/09, Steve Totaro stot...@first-notification.com wrote:
On Wed, Jul 8, 2009 at 2:14 AM, Olivieroza-4...@myamail.com wrote:
Hi,
Reading this thread, is this correct to say CallerName is widely used in
the
US ?
Here in France, this service is optional but I don't think
To calculate the monthly data traffic that is generated by VoIP-calls,
is it as simpel as
80kbps (G.711 SIP) x 6s (1000 minutes) = 480 kilobits / month =
585.9375 MB traffic / month
???
Jonas.
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Title: asterisk-users@lists.digium.com
Wed, 8 Jul 2009 03:34:15 +0100
Thanks for the reply.
1. The extensions in the Queues are setup as Agent members, defined in
Agents.conf, then within the definition of the queue in queues.conf
they are made members of the queue.
2. As for the recording my diaplan is as follows:
[main-line]
exten = s,1,NoOp()
exten =
If you are using a large number of DAHDI channels, you could designate a
chunk of them as non-local since you can control RXGAIN on each channel.
You would have to work out something with your TELCO since your'e a dead
duck control-wise once you answer the call.
-Original Message-
From:
The sort of trunk does matter; I don't know about ISDN, but I get different
behavior on DAHDI vs SIP, so that's one verification that you are dealing
with a necessarily fixed set of values.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
You should initiate a second call or send a voicemail. You don't want to
mess too much with what is working.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Tuesday, July 07, 2009 1:32 PM
To: Asterisk Users
CALLERID(name) is a TELCO specific field. In the long run, you will be best
served using your own lookup of a database using CALLERID(num), since
CID(name) is unreliable and in some cases costly. IMO, you would be well
served with an app (AGI?) that recorded valid names into the database and
let
Hose wrote:
Hi,
Hi All,
Hope you all are fine and good, Today i have found that Mine all PRI Channels
are restating after every interval of one hour, and i have search and psot on
fourms and everyone said that this is a normal behaviour.
If this is a normal behaviour is there is any way to stop it { i
Hose wrote:
I have a feeling that the issue is between transcoding of ulaw to g.722
and it's too loud during the transcoding - anyway to adjust the levels?
There was a flaw in Asterisk's G.722 transcoder module that was fixed
recently (on May 15, 2009), so any release made after that date
add resetinterval=never in your zaptel.conf, or chan_dahdi.conf
depending on what you are running. zaptel or dahdi.
On Wed, Jul 8, 2009 at 10:35 AM, Aman Dhallyaman.dha...@live.com wrote:
Hi All,
Hope you all are fine and good, Today i have found that Mine all PRI
Channels are restating
On Mon, 06 Jul 2009 10:31:18 -0500
Brent Davidson br...@texascountrytitle.com wrote:
Botond Botyanszki wrote:
Hi,
I have an x100p zaptel card with asterisk 1.4. I'm using the system for
outgoing calls.
My problem is that Answer() is falsely returning while the call is still
ringing
Can anyone here have experience using G.722 on the Grandstream GXP-1200?
It's supposed to support the codec, but I wonder if the handset does it
justice?
The older BT-200 also supported the codec, but the handset was not good
enough. You could only hear the improved call quality using a headset.
snip
Audiocodes supports SRST on their mediapack analog gateways.
This might be a viable option. I haven't used any Audiocodes devices
before. Are people pleased with them?
snip
Deploy a lot of small asterisk based appliances...
This way you can completely decentralise your setup and give
Kevin P. Fleming wrote:
Hose wrote:
I have a feeling that the issue is between transcoding of ulaw to g.722
and it's too loud during the transcoding - anyway to adjust the levels?
There was a flaw in Asterisk's G.722 transcoder module that was fixed
recently (on May 15, 2009), so any
The two logs that I have been able to find are messages on the
asterisk server in debug. Unfortunately, Nokia does not have any kind of
logging (sucks). What I can see is that it is definitely a phone issue,
just stuck on where to go from here.
First, this if from asterisk in debug
1.
Well, Teliax says they have no access to the PSTN's database, but I'm
suggesting they check out TargusInfo as mentioned above. One of their
suggestions, is to contact the local ILEC to get the number published in
their white pages. Will that accomplish the same thing (I doubt it).
On Wed, Jul 8,
Un-topposting...
On Tue, Jul 7, 2009 at 7:08 PM, Miguel Molina mmol...@millenium.com.co
wrote:
Darrin Henshaw escribió:
2. The issue does seem to be limited to MixMonitor and the Queue
application, as in testing I setup mixmonitor on my extension dialed it from
outside the
Danny Nicholas wrote:
If you are using a large number of DAHDI channels, you could designate a
chunk of them as non-local since you can control RXGAIN on each channel.
You would have to work out something with your TELCO since your'e a dead
duck control-wise once you answer the call.
hi, i
@asterisk
- svn-ed asterisk from digium 1.6
- make install
its running and i can access the CLI
@gui
then i
-svned asterisk-gui from digium
- installed
- repointes apache /var/www/1234 /var/lib/asterisk/static_html
now, i see the login box, but i dont have any credentials. tutorials are
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
- tom tomabr...@gmail.com wrote:
hi, i
@asterisk
- svn-ed asterisk from digium 1.6
- make install
its running and i can access the CLI
@gui
then i
-svned asterisk-gui from digium
- installed
- repointes apache /var/www/1234 /var/lib/asterisk/static_html
now, i see
- tom tomabr...@gmail.com wrote:
*MY* browser must be experiencing problems. I thought you posted a message but
it appears blank. /sarcasm
I'm a huge fan of elinks. It's cross platform and works great.
--Tim
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shame on me...yes i had several different installations of asterisk, just to
try it out. but i deleted everything before i went on installing a different
version or vendor.
so, make samples did the trick! i now have the missing files. thx (i didnt
do it before coz somehow samples + freepbx) screwd
On Wed, 2009-07-08 at 14:49 -0400, tom wrote:
- repointes apache /var/www/1234 /var/lib/asterisk/static_html
The Asterisk GUI uses the web server built into Asterisk, so what you're
attempting to do here isn't going to work. I suggest you follow the
instructions at
Since /etc/asterisk is empty, you have either relocated your conf files or
put them in a database. Assuming neither, just create manager.conf in
/etc/asterisk with this setup
[general]
Enabled = yes
Port = 5038
Webenabled=yes
Bindaddr = 1.2.3.4
[loginname]
Secret=secret
And restart
On Wed, 8 Jul 2009, tom wrote:
None. I'm a command line weenie.
) GUIs don't let you annotate your changes -- who did what (or what they
thought they were doing), when, and why.
) GUIs don't support any sort of versioning.
) GUIs don't support any sort of configuration rollback.
All of
/etc/manager.conf:
[admin]
secret = test
read = system,call,log,verbose,command,agent,user,config
write = system,call,log,verbose,command,agent,user,config
- doenst let me log in ;-(
- i tried chown /static_http/config
this is in my apache-logs:
[Wed Jul 08 15:36:23 2009] [error] [client
Do you have the [general] section with enabled, webenabled, port and
ipaddress?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of tom
Sent: Wednesday, July 08, 2009 2:39 PM
To: Asterisk Users Mailing List - Non-Commercial
thx, but still struggeling:
http://blabla:8088/asterisk/static/docs/index.html
NO GO
---
;
; Asterisk Builtin mini-HTTP server
;
;
; Note about Asterisk documentation:
; If Asterisk was installed from a tarball, then the HTML documentation
should
; be installed in the
stupid me, i had a ; in front of the [general] line.
thx so far
im logged inand now?
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In http.conf make bindaddr be the address of your asterisk server.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of tom
Sent: Wednesday, July 08, 2009 3:01 PM
To: jsm...@digium.com; Asterisk Users Mailing List -
Because DEVSTATE is for custom hints - and have you tried to set one
every time a phone rings/is answered? This was thought about - but the
logic in the dialplan would be a nightmare.
Anyway, doing it the way I do it works for me (and others) as my
dialplan contains nothing but 'include' and
yeah thx i did that. now if i log in (
:8088/asterisk/static/ajamdemo.html)
, i see the
Asterisk™ AJAM Demo. but thats it:
i tries the urls givin by : http show status, but none of them gives me a
real webinterface to administrate the whole asterisk etc
i thought asterisk-gui gives me the
You're confusing the manager interface with the gui interface. The gui
interface would be 8088/asterisk/static/config/index.html
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of tom
Sent: Wednesday, July 08, 2009 3:19 PM
:8088/asterisk/static/config/index.html
wes my missing link
thx 2 all for ur help
-- Forwarded message --
From: tom tomabr...@gmail.com
Date: Wed, Jul 8, 2009 at 4:19 PM
Subject: Re: [asterisk-users] q: install asterisk + asteris-gui
To: Asterisk Users Mailing List -
thx again,
one last question: as i mentioned, i used freepbx before. now i facing only
the section:
- users
my goal right now is to use that asterisk instance just to have intenral
extensions to talk to each other...whats the quickest setup here? i mean i
dont need trunks, dialplans etc, right?
If you're just going to use Asterisk as an internal system, you just need a
simple users.conf, sip.conf and about a 5 line dialplan.
Sip.conf
[general]
srvlookup=yes ;allows DNS lookups of server names
naxexpirey=180
defaultexpirey=160
context=default ; Default context for incoming calls
Hi all!
I want to autopause my queue member when they are not answering within
20 seconds, and the autopause
should affect all queues they are member of, not only the queue where
the call was not answered.
Is there a way to do that?
The members gets dynamically added. I'm using asterisk
thx danny,
(sorry, bad day today)
one more question: deviceandusers
i had this distinction with freepbx, though i dont know whether this is a
freepbx-thing or an asterisk-setting...
thx
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That's a new one on me, but check out this link
http://forums.digium.com/viewtopic.php?t=3689
http://forums.digium.com/viewtopic.php?t=3689highlight=sid=acbc25fd45bae1
ecc42b0d7ca66fe88c highlight=sid=acbc25fd45bae1ecc42b0d7ca66fe88c
As I read it, you want to be able to dial 1001 and get
Barry D. Hassler wrote:
Well, Teliax says they have no access to the PSTN's database, but
I'm suggesting they check out TargusInfo as mentioned above. One of
their suggestions, is to contact the local ILEC to get the number
published in their white pages. Will that accomplish the same thing
Thank you for the info
Does anyone know if the cdc-modem interface which is available on mobile phones
can actually potentially be used to initiate, or register for receiving a voice
call?
If so, I suppose USB 3G dongles could even be used as a voip-air interface!
Would be interesting to find
Christian Gansberger escribió:
Hi all!
I want to autopause my queue member when they are not answering within
20 seconds, and the autopause
should affect all queues they are member of, not only the queue where
the call was not answered.
Is there a way to do that?
The members gets
What you say...Dave Fullerton (dfullertaster...@shorelinecontainer.com):
Kevin P. Fleming wrote:
Hose wrote:
I have a feeling that the issue is between transcoding of ulaw to g.722
and it's too loud during the transcoding - anyway to adjust the levels?
There was a flaw in
It does which is why it was not included in a release code set. The patch
could be changed to do an OR type compare for the bridge class. I have
changed my implementation to use only user events for everything that I now
need so I did not pursue this patch.
--
Jim Dickenson
Hose wrote:
What you say...Dave Fullerton (dfullertaster...@shorelinecontainer.com):
Kevin P. Fleming wrote:
Hose wrote:
I have a feeling that the issue is between transcoding of ulaw to g.722
and it's too loud during the transcoding - anyway to adjust the levels?
[Jul 8 21:23:49] WARNING[4358]: chan_sip.c:10458 check_auth: username
mismatch, have 6001, digest has 1160
[Jul 8 21:23:49] NOTICE[4358]: chan_sip.c:18529 handle_request_register:
Registration from 'sip:6...@192.168.1.4 sip%3a6...@192.168.1.4' failed
for '192.168.1.3' - Username/auth name
hi,
checking my freshly installed astersik-gui, i can see a menu entry called
Users. clicking on that one gives me the pages labeled (on orange) User
Extensions on PBX. if i do make an entry here, it ends up in the user.conf.
file.
so i created a new entry in the sip.conf, reloaded asterisk
Hi,
My Dial() is set to the following, but always stops about 30 seconds into
the call even when I set it to try for 60 seconds.
exten = dialnumber,1,Dial(${DialInfo},60)
I am running on 1.6.1-r199820.
Is there some other setting that is overriding mine? Or an issue with this
release?
On 9/7/09 12:11 AM, jonas kellens wrote:
To calculate the monthly data traffic that is generated by VoIP-calls,
is it as simpel as
80kbps (G.711 SIP) x 6s (1000 minutes) = 480 kilobits / month =
585.9375 MB traffic / month
http://www.asteriskguru.com/tools/bandwidth_calculator.php
On 9/7/09 1:39 PM, tom wrote:
hi,
checking my freshly installed astersik-gui, i can see a menu entry
called Users. clicking on that one gives me the pages labeled (on
orange) User Extensions on PBX. if i do make an entry here, it ends up
in the user.conf. file.
so i created a new entry in
On 9/7/09 2:06 PM, John Regal wrote:
Hi,
My Dial() is set to the following, but always stops about 30 seconds
into the call even when I set it to try for 60 seconds.
exten = dialnumber,1,Dial(${DialInfo},60)
I am running on 1.6.1-r199820.
Is there some other setting that is overriding
Hi - yes, you are correct in that I am using AMI. I thought I could override
inline in the dialplan. I will modify the AMI call. Thanks for the quick
response - truly appreciated.
john
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi All,
I never saw a reply to this question. Is anyone able to assist?
Regards
David.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Klaverstyn
Sent: Friday, 19 June 2009 2:28 PM
To: 'Asterisk Users Mailing List -
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