Does the same apply to GSM channels?
- Original Message -
From: Andrew Kohlsmith [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 19, 2003 9:34 PM
Subject: Re: [Asterisk-Users] Unable to detect process 256 frames
WARNING[1232119104]: File dsp.c, Line 1198
Title: Chan_h323 docs
Jeremy,
In some posting in the mailing lists, you mentioned that docs for h323 had been submitted but hadn't made it into distribution.
Could you post those docs in your download directory?
I'm trying to understand the nuances of your driver, gnugk, and
Hi,
I am trying to run ZTMonitor to get debug info from my E100P board but I
got the following message:
-bash-2.05b# ./ztmonitor 1
Unable to open /dev/dsp: No such file or directory
Cannot open audio ...
-bash-2.05b#
Thanks,
Daniel
___
We included presence in the latest builds. This is necessary to enable
auto-redial (call completion), at least in the eyes of the SIP hot-shots. In
the latest image (2.03c, see http://snom.com/download/share) we included a
flag where this feature can be turned off.
But as long as it's not causing
Hi,
I am trying to figure out if * can register as a client on a remote MGCP
service. Just like SIP and other protocols
Do. Anyone tried this?
Ta
SJ
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[EMAIL PROTECTED]
Ok, I've managed to get inbound and outbound calling to work with chan_h323
and gnugk.
A few questions:
1) if I do a reload in *, chan_h323 loses its registration with gnugk, and
will no longer pass calls to it. A second reload will crash *. Is this
supposed to be?
2) For a configuration in
Hi People,
Can anyone help-me here with a simple question.
I wanna buy a Sip Phone, but what is the best and cheap one ?
I see alot of messages about, grandstream , snow etc etc.
So for use with my * system, what sip phone is the best ??
Can him be used behind a nat system etc ?
Or with a
Can someone point me to some reasonable example / starting point to implement
a basic IVR menu? Looking for something rather simple like the press 1 for
sales, 2 for tech support, and probably an option to list the voicemail
directory kind of thing. Nothing elaborate needed, just basic menu.
Rich-
Chapter 4 of the (so-called) draft handbook, details what you need to know
pretty well.
Here's the link: http://www.digium.com/handbook-draft.pdf
Regards,
Scott
Scott M. Stingel
Emerging Voice Technology Inc.
Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
URL:
This was a typo--it already is:
exten = 70,1,Dial(IAX/mike,30,tr)
and I still get a busy...
Thoughts?
Dan wrote:
Hi Michael,
- Original Message -
From: Michael Welter [EMAIL PROTECTED]
Subject: [Asterisk-Users] DIAX phone busy
I've configured the DIAX phone. It registers with
exten = _X., Goto(ivrmenu,s,1)
[ivrmenu]
exten = s,1,Ringing
exten = s,2,DigitTimeout,30
exten = s,3,Background(something) ; press 1 for sales
exten = t,3,Goto(business,0,1)
include = business
;map yourl extens here...
[business]
exten = 0,1,...
exten = 1,1, ..
From: Rich
I didn't repeat this question. I read the responses and amended my
request. I was simply asking for any suggestions as how to diagnose
this problem including the ways to provide additional information to
this list to help solve the problem. After reinstalling Linux and
Asterisk from scratch
Hi guys
i signed up to iconnect a few hours ago to try do some cool stuff. but im having a few
problems. Im running asterisk and xp/xlite softphone.. both xp box and asterisk box
are on public ips.
The problem is that when i ring anyone in the world it'll ring they'll pickup and i
can hear
Hi,
Did you registered using IAX or IAX2?
Check if in Registering page you have selected IAX2 or not.
Did you check this one too?
BR,
Dan
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Hi,
try to put in both users setup,
canreinvite = no
and what is this codec allow=g721.1 ... :)))
Lubo
vocalvoip wrote:
Hi guys
i signed up to iconnect a few hours ago to try do some cool stuff. but im having a few
problems. Im running asterisk and xp/xlite softphone.. both xp box and
Rich,
This is in response to your question about and IVR Menu. Below is the
dial-plan from the * at SAI. If you dial 1-800-747-9111 and the
Extension 2998, you'll be able to hear this one in action.
The key to creating it is to use Extension 205 (defined below) to record
your menu prompts.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of vocalvoip
Sent: 20 December 2003 16:14
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] iconnect 480 unavailable msgs
Hi guys
i signed up to iconnect a few hours ago to try do some cool stuff. but im
having a
[iconnect]
type=friend
username=
secret=
host=sipauth.deltathree.com
qualify=1000
callerid=178197026
allow=g721.1
Hello...looks like you have your codecs messed up. If you have enough
bandwidth I recommend you modify it to:
[iconnect]
type=friend
username=
On Friday 19 December 2003 08:12, David Gomillion wrote:
How do I make x100P does not answer incoming calls ?
The only thing that springs to mind is that you create an incoming
context, and have an extension like:
Exten = s,1,Wait(1000)
Dunno if it will work or not, but
On Sat, 20 Dec 2003, Carlos Arnt wrote:
I wanna buy a Sip Phone, but what is the best and cheap one ?
I see alot of messages about, grandstream , snow etc etc.
We have bought around 30 Grandstream phones, both BT101 and BT102. In
general the phone is reasonable, but it does have
Using DIAX softphone which seems to be working OK can get to VM/echotest etc
in the demo context
Am trying to setup FWD but get the following problems
Can hear it ringing when dialing FWD no 612 for time. Connects but no sound
from remote end.
Does anyone have any suggestions.
Softphone on
Exten = s,1,Wait(1000)
This will make * not answering the call, but still you would see notices
coming on your screen and also an entry in CDR.
immediate=no
Will try this out
SW
From: Tilghman Lesher [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] x100P
Once the connection is made between your * and FWD, your * passes it off
so its really just a connection between your soft-phone and FWD. So,
your soft-phone needs to be properly NATed, and the appropriate ports TO
your soft-phone need to be opened. Each SIP phone will attempt to use
different
Hi
I'm also interested in this. I currently have to press # to stop
asterisk when I pick up the phone. Even if I answer before asterisk,
it still answers.
Cheers
Rob
On Sun, Dec 14, 2003 at 11:00:40PM -0500, Jim Flagg wrote:
If Asterisk is configured as a simple answering machine
On Sat, 2003-12-20 at 03:22, Senad Jordanovic wrote:
Hi,
I am trying to figure out if * can register as a client on a remote MGCP
service. Just like SIP and other protocols
Do. Anyone tried this?
No I don't believe it can. The MGCP implementation in Asterisk is a
CallAgent not a
John,
I spoke with Level(3) last week regarding SIP termination. They
quoted $0.01/minute, with an 11 Million Minute / Month minimum.
Ugh!
-dg
--
Darnell Gadberry
President
binaryMedia
darnell AT binmedia DOT com
Date:
On Saturday 20 December 2003 02:13 pm, Robert Murray wrote:
Hi
I'm also interested in this. I currently have to press # to stop
asterisk when I pick up the phone. Even if I answer before asterisk,
it still answers.
Cheers
Rob
Hello Rob.
That's interesting. My system is/was like
Rich Adamson wrote:
Can someone point me to some reasonable example / starting point to implement
a basic IVR menu? Looking for something rather simple like the press 1 for
sales, 2 for tech support, and probably an option to list the voicemail
directory kind of thing. Nothing elaborate needed,
Joe Dennick wrote:
Rich,
This is in response to your question about and IVR Menu. Below is the
dial-plan from the * at SAI. If you dial 1-800-747-9111 and the
Extension 2998, you'll be able to hear this one in action.
Now documented on
I thought I'd share my Asterisk experience, which hasn't exactly been as
pleasant as I would like but now seems usable in most ways and more then
I expected in other ways. I wanted a home PBX system, that would let me
treat different callers different ways depending on CID.
I initially bought
Hi All,
Can i install * on a beowulf cluster or Is *
compatible to clusters. I am planning to install a 4 node beowulf cluster using
few cheap hardwares. If no one had tried before i can spend some time on
installing and configuring * on this cluster. Let me know.
thanks,
-Balaji
Do you
Yes, IAX2 is checked.
Thanks
Dan wrote:
Hi,
Did you registered using IAX or IAX2?
Check if in Registering page you have selected IAX2 or not.
Did you check this one too?
BR,
Dan
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Asterisk-Users mailing list
[EMAIL PROTECTED]
immediate=no
did not work either, so at least to get * not write to CDR, I will have to
use command noCDR().
My extension context is as follows
[x100pincoming]
exten = s,1,Ringing
exten = s,2,Wait(40)
exten = s,3,NoCDR()
It still write the CDR in MySQL, any idea how to get rid of that ?
SW
On Sat, 2003-12-20 at 14:54, Balaji NJL wrote:
Hi All,
Can i install * on a beowulf cluster or Is * compatible to clusters. I
am planning to install a 4 node beowulf cluster using few cheap
hardwares. If no one had tried before i can spend some time on
installing and configuring * on this
Hi,
Yes, IAX2 is checked.
Then change the line to:
exten = 70,1,Dial(IAX2/mike,30,tr)
BR,
Dan
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http://lists.digium.com/mailman/listinfo/asterisk-users
- Original Message -
From: Michael T Farnworth [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, December 21, 2003 4:31 AM
Subject: Re: [Asterisk-Users] Best SIP PHones to buy ?
We have bought around 30 Grandstream phones, both BT101 and BT102. In
general the phone is
On Sat, Dec 20, 2003 at 03:01:37PM -0500, Martin wrote:
Hello Rob.
That's interesting. My system is/was like that. (I can still simulate it
with the emergency phone that is paralleled/split at the incoming point).
Pressing # certainly doesn't stop asterisk on my system.
It only works
That's interesting. My system is/was like that. (I can still simulate it
with the emergency phone that is paralleled/split at the incoming point).
Pressing # certainly doesn't stop asterisk on my system.
It only works if you don't have a # extension, and no i extension.
Another
I'm in the process of reworking my dialplan to include an ivr and
other items. I've seen several examples over the last several months
that mention the s, h, t (and probably others) extensions, but
I don't fully understand what they are used for. Can someone either
give a short definition of each
Hi !
We have Cisco 7912 phones, and the doc says that I can create up to four speed
dial buttons on my phone using the Cisco CallManager.
Does anyone knows which protocol is used to configure speed dials (Is it
documented somewhere) ?
Did someone tried to reverse engineer the protocol ?
It would
Yes, I've tried that as well. When I dial 70 from another extension,
I hear ringing but the DIAX doesn't ring.
Dan wrote:
Hi,
Yes, IAX2 is checked.
Then change the line to:
exten = 70,1,Dial(IAX2/mike,30,tr)
BR,
Dan
___
Asterisk-Users mailing list
What I mean, is that if I pick up the analogue phone connected in
parallel with the x100p and blow into the handset, The x100p detects
it as ringing and rings the phone connected to the s100u.
Also, if I phone asterisk from another phone, and hang up when it is
ringing, the s100u phone keeps
On Saturday 20 December 2003 05:54 pm, Rich Adamson wrote:
That's interesting. My system is/was like that. (I can still simulate
it
with the emergency phone that is paralleled/split at the incoming
point).
Pressing # certainly doesn't stop asterisk on my system.
It only
I'm in the process of reworking my dialplan to include an ivr and
other items. I've seen several examples over the last several months
that mention the s, h, t (and probably others) extensions, but
I don't fully understand what they are used for. Can someone either
give a short definition of
On Sat, 2003-12-20 at 16:57, Rich Adamson wrote:
I'm in the process of reworking my dialplan to include an ivr and
other items. I've seen several examples over the last several months
that mention the s, h, t (and probably others) extensions, but
I don't fully understand what they are used
On a side note.. you can't use exten = h, if you have any hope of getting
accurate billing info. Its wise to call ResetCDR(w) in your exten = h,
or not use it at all.
bkw
On Sat, 20 Dec 2003, Steven Critchfield wrote:
On Sat, 2003-12-20 at 16:57, Rich Adamson wrote:
I'm in the process of
ztmonitor 1 -v
On Sat, 20 Dec 2003, Daniel Bichara wrote:
Hi,
I am trying to run ZTMonitor to get debug info from my E100P board but I
got the following message:
-bash-2.05b# ./ztmonitor 1
Unable to open /dev/dsp: No such file or directory
Cannot open audio ...
-bash-2.05b#
Thanks,
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
Another problem I have is that the x100p detects loud noise on the
line as ringing.
Use callprogress=no in the zapata.conf
That's unclear as you left my text as well as Roberts text.
Are you talking about Another problem I have is that the x100p detects loud
noise on the line
--- Jon Creasey [EMAIL PROTECTED] wrote:
Using DIAX softphone which seems to be working OK can get to
VM/echotest etc
in the demo context
Am trying to setup FWD but get the following problems
Can hear it ringing when dialing FWD no 612 for time. Connects but
no sound
from remote end.
- Original Message -
From: Darnell Gadberry [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, December 20, 2003 2:53 PM
Subject: [Asterisk-Users] Level(3) SIP termination services
John,
I spoke with Level(3) last week regarding SIP termination. They
quoted $0.01/minute, with
Hey I need another pair of eyes on this!
I would like to add phones numbers to the blacklist from any handset so I
did this:
exten = _*66XX,1,StripMSD,3
exten = _XX,2,DBPut,blacklist/BYEXTENSION/1
exten = _XX,3,Hangup
However what I get in the database is:
Walt Davis wrote:
Hey I need another pair of eyes on this!
I would like to add phones numbers to the blacklist from any handset so I
did this:
exten = _*66XX,1,StripMSD,3
exten = _XX,2,DBPut,blacklist/BYEXTENSION/1
exten = _XX,3,Hangup
However what I get in
Don't use BYEXTENSION use ${EXTEN}
bkw
On Sat, 20 Dec 2003, Walt Davis wrote:
Hey I need another pair of eyes on this!
I would like to add phones numbers to the blacklist from any handset so I
did this:
exten = _*66XX,1,StripMSD,3
exten =
I'm testing an ivr implementation (first time) using:
exten = 620,1,Wait,1
exten = 620,2,Answer
exten = 620,3,DigitTimeout,5
exten = 620,4,ResponseTimeout,10
exten = 620,5,Background(npi-greeting) ; Thanks for calling press 1 for
exten = 1,1,Goto(npi-directory,s,1)
For initial testing, I've
On Sat, 2003-12-20 at 21:06, Rich Adamson wrote:
I'm testing an ivr implementation (first time) using:
exten = 620,1,Wait,1
exten = 620,2,Answer
exten = 620,3,DigitTimeout,5
exten = 620,4,ResponseTimeout,10
exten = 620,5,Background(npi-greeting) ; Thanks for calling press 1 for
exten =
Care to expound a bit on that topic for the wiki, with some details as to why?
JT
At 6:32 PM -0600 12/20/03, Brian West wrote:
On a side note.. you can't use exten = h, if you have any hope of getting
accurate billing info. Its wise to call ResetCDR(w) in your exten = h,
or not use it at all.
Do you have the context with exten 3000 included in the same place as
exten 620 is? If not then it can in no way work.
bkw
On Sat, 20 Dec 2003, Rich Adamson wrote:
Hi Steve,
On Sat, 2003-12-20 at 21:06, Rich Adamson wrote:
I'm testing an ivr implementation (first time) using:
exten =
[crossposted to isp-clec and asterisk-users]
Date: Fri, 19 Dec 2003 21:12:22 -0500
To: [EMAIL PROTECTED]
From: John Todd [EMAIL PROTECTED]
Subject: [Asterisk-Users] Level(3) SIP termination services?
Reply-To: [EMAIL PROTECTED]
Anyone investigated the new service offerings from Level(3) in the
Hi sergio
On Fri, 19 Dec 2003 14:49:15 +0100
Sergio Serrano Revuelto [EMAIL PROTECTED] wrote:
Hi all,
I have tested RxFAX application through X100P card. When Fax
arrive i obtain the next trace:
snip
5 (0.01679,-0.16590) - 0.02781
6 ( -0.04451,
Folks,
I can't seem to get DTMF signaling working properly using SJphone connecting
to Asterisk via a SIP connection. Here's an example of a voicemail session
where I entered 1234 for both the username and the password:
-- Incorrect password '11223344' for user '11223f344' (context = any)
Kudos John on an excellent set of questions! Not to mention the pointer
to isp-clec.
Thanks!
John Todd wrote:
[crossposted to isp-clec and asterisk-users]
That's not such a great price at 11 million minutes, in my opinion.
Did you ask them if they would speak to Asterisk via SIP? We have
On Sunday 21 December 2003 00:29, Darren Nickerson wrote:
Folks,
I can't seem to get DTMF signaling working properly using SJphone
connecting to Asterisk via a SIP connection. Here's an example of a
voicemail session where I entered 1234 for both the username and the
password:
--
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