James Harper wrote:
I believe the OP wants to use GSM handsets as extensions, like running
your own localized GSM network. That's not the same as using a GSM
terminal to connect Asterisk to the cellular network.
Correct!
IP Access makes such products.
It seems the proxy address is added to all incoming calls to the Cisco phone.
On 3/16/06, Tim Connolly [EMAIL PROTECTED] wrote:
I'm not sure this is the issue. Every call seem to get the proxy
address added whether it's the main proxy or the backup. What has to
match to make the phone
Hi Brent
Anyone ever seen MeetMe cause * to crash? Specifically, it happens
consistantly if someone begins to enter a conference and then decides to
hangup while Allison is introducing them - like playing back
conf-onlyperson. This has been seen with the MeetMe participant
connecting via IAX
On Sat, Mar 18, 2006 at 04:49:53PM +1100, James Harper wrote:
I believe the OP wants to use GSM handsets as extensions, like running
your own localized GSM network. That's not the same as using a GSM
terminal to connect Asterisk to the cellular network.
Correct!
IP Access makes such
I believe the OP wants to use GSM handsets as extensions, like
running
your own localized GSM network. That's not the same as using a GSM
terminal to connect Asterisk to the cellular network.
Correct!
IP Access makes such products.
http://www.ipaccess.com/products/nanoBTS.htm
Subject: Re: [Asterisk-Users] Fake Ring Tone/Compile AddonTo: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.comMessage-ID: [EMAIL PROTECTED]Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Kenige Ho wrote: Dear All, I am currently have this
On Sat, Mar 18, 2006 at 10:16:27PM +1100, James Harper wrote:
[snip]
Ah. More complicated than I'd hoped but not more than I suspected :)
So the product that can accept gsm phone registrations and calls and
trunk them to asterisk via E1/TDMoE/TDMoIP/SIP/IAX is still wishware? Oh
well. I guess
On 3/18/06, Watkins, Bradley [EMAIL PROTECTED] wrote:
cluster (or clusters, in the case of one site). So there is no NAT, and it
is an Asterisk-only solution (at least insofar as telephony software is
concerned).
I'm just barging in.. This all looks 'very' promising stuff, I'm
looking forward
Douglas Garstang wrote:
Trying to get SIP realtime working here...
I'm connected to the database...
*CLI realtime mysql status
Connected to [EMAIL PROTECTED], port 3306 with username voxadmin for 6 seconds.
I can get information for the extension in question...
*CLI realtime load sipusers
On Friday 17 March 2006 23:23, James Harper wrote:
Care to give me any more clues? Google only wants to tell me about
articles about the use of picocells in aircraft and how much better the
world will be when it happens :) Maybe I'm using the wrong search terms.
I apologize; When I was
Is there a list or matrix somewhere that shows what codec can be
transcoded? I am playing with different allowed codecs between my
asterisk box and some of my providers testing voice quality and
bandwidth usage on my cable connection, and I occassionally run into an
issue where asterisk cannot
Robert Webb wrote:
Is there a list or matrix somewhere that shows what codec can be
transcoded? I am playing with different allowed codecs between my
asterisk box and some of my providers testing voice quality and
bandwidth usage on my cable connection, and I occassionally run into an
issue
Thanks Jonathan.
In this case, how do you actually mute everybody but the admins?
Imagine giving a training to 100 people, and not wanting anybody to say
anything except the trainer...
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
Thanks for the link. The ultimate solution was to change from fxs_ls to
fxs_ks. Now it works great!
Thanks,
James
Dr. Michael J. Chudobiak wrote:
[EMAIL PROTECTED] wrote:
If so, is there a way to detect the hangup?
Check out
In article [EMAIL PROTECTED],
Michael Gaudette [EMAIL PROTECTED] wrote:
Thanks Jonathan.
In this case, how do you actually mute everybody but the admins?
Imagine giving a training to 100 people, and not wanting anybody to say
anything except the trainer...
Here's an idea.
Have the leader
I would like to make a suggestion and recommend that you put your Asterisk
box on the outside and let it also pull duty as your firewall/nat router. The
iptables overhead will be minimal on the system and you'll save yourself a lot
of headaches in the long run.
The biggest problem being
Can Comedian Mail handle more than just an away and busy message? I've got
a client that would like even more of them.
I can write an app to replace messages externally, but I was wondering of
comedian could handle it internally.
As far as I know, no.
But, what I did for a customer
I have three Sipura 3000 FXO untis for incoming PSTN lines on a small
pbx. There is an IVR to select the extension. The DTMF tones are not
being sensed so the IVR does not work and incoming calls are not being
answered. I have listed my sip.conf entries.
Is there any solution to this?
Yusuf,
No I don't have the switch statement in extensions.conf. I'm not trying to do
realtime extensions. I'm trying to do realtime SIP. They're different.
Doug.
-Original Message-
From: yusuf [mailto:[EMAIL PROTECTED]
Sent: Sat 3/18/2006 6:49 AM
Check for :
dtmfmode=outband
Good luck !
Francois BERGERET,
France
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Chris Mason
(Lists)
Envoyé : samedi 18 mars 2006 17:43
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet :
Try with dtmfmode=auto and DTMF Tx Method: InBand+INFO, this was the
best configuration for me, although still not 100% guarantee. If the
dtmf tones are sent very fast without a 1 sec delay, in most of the
cases asterisk won't detect half of them. There are a couple of patches
for the trunk
hello
i want to spy on a chennel listen the voice conversation between two person.
i also want talk to one of them but others will not listen my voice.
how can i configure this using ChanSpy?
thanks
atik
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On Mar 16, 2006, at 12:36 PM, Martin Joseph wrote:
So, I am answering my own post (bad form I know)...
I am trying to get my DTMF to use RFC 2833 rather then inband, so that
I can utilize lower bandwidth codecs through my FXO.
Ok, I have given up on this. There seems to be some kind of
Just spent hours dicking around with SIP Realtime.
Every time a phone came up and sent a registration to Asterisk, Asterisk would
simply NOT query the database. I had sipusers in extconfig, but added sippeers
as well. NOW I can see Asterisk doing a 'SELECT * FROM sippeers WHERE name =
We have two Linksys 942 phones which sound great when they call each other directly through Asterisk. But when they both dial in to a meetme conference room, the sound is very jittery. Other phones like Polycom 501 and Snom 360 sound fine when using meetme.
Both Linksys phones are set to use
Oh heck. It really looks like realtime has been seriously screwed up.
When a call comes in to Asterisk, I can see asterisk executing these queries.
SELECT * FROM ast_sip_peers WHERE host = '2XX.YYY.142.205'
SELECT * FROM ast_sip_peers WHERE name = '2944093'
SELECT * FROM ast_sip_peers WHERE name
This is an age old question. Unless something has changed, it is
possible but not included functionality. A group of people paid to have
this functionality developed but since they paid they decided not to
release it back into the asterisk community. I am not sure if it for
sale or not or even
Hello Fernando,
I have checked this card with and without hardware echocan : the hardware
echocan module does the job better than the zaptel software can do it. I
recommand this module without any doubt.
But, the echocan algorithms in zaptel are better and better and the CPUs
power grows
mailing list
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http://www.eset.com
I was also thinking a list for newbies...
PaulH
- Original Message -
From: Robert La Ferla [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Saturday, March 18, 2006 2:33 PM
Subject: [Asterisk-Users] Asterisk Users Mailing List Traffic
The volume/traffic on this list has
On 18 Mar 2006, at 19:21, Douglas Garstang wrote:
Oh heck. It really looks like realtime has been seriously screwed up.
When a call comes in to Asterisk, I can see asterisk executing
these queries.
SELECT * FROM ast_sip_peers WHERE host = '2XX.YYY.142.205'
SELECT * FROM ast_sip_peers WHERE
Hi Yrving.
I dont use [EMAIL PROTECTED] but if you ever use
plain ol' asterisk, I might be able to give you a hand.
drop me aline when you do.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of yrving
rivasSent: Tuesday, March 07, 2006 6:18 AMTo: Asterisk
Users Mailing List
Man! I love polycoms.. They are good phones and highly configurable.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|[EMAIL PROTECTED]
|Sent: Tuesday, March 07, 2006 7:41 AM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject:
Has anyone have integrated Panasonic PBX QSIG with asterisk servers
using E1 interfase?
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I have not used the Linksys 942 phones yet, but I have a couple of Sipura
841's. Check to see what your RTP payload encoding frame length is, ie.
20ms or 30ms. Also check to see if there is a setting to surpress or
transmit silence. If so you want to transmit silence.
Rana Dutt [EMAIL
On Mar 18, 2006, at 2:05 PM, [EMAIL PROTECTED] wrote:
I was also thinking a list for newbies...
As a newb I think that is a bad idea. First of all, the heavy hitters
will all want to avoid it( a newb list). Secondly I have learned a LOT
just by reading other peoples (non newbs) problems
Hi all,
First post, etc, etc. :)
Robert La Ferla wrote:
Perhaps, we could split the list into two:
As the list uses Mailman, it is possible to specify topics on the
subject prefix and then set your Mailman preferences to select which
topics you like.
The admin could, for a simple
HI, all
This is a test. By some reason I stopped received e-mails from the list.
Rudolf
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On Friday March 17 2006 8:07 am, Can2002 wrote:
I'd planning on install Asterisk on a hosted Linux box we're setting up.
The hosting provider that seems to offer the best deal can install
either Debian 3.1 or SUSE 9.x or 10.0 in either 32 or 64 bit editions
(running on AMD 64 bit).
My
What are your zttest results? zttest can be run from
/usr/src/zaptel/ directory (run ./zttest from there). Do you have
Digium hardware or ztdummy?
Pedro
http://www.TRACI.netOn 3/18/06, Rana Dutt [EMAIL PROTECTED] wrote:
We have two Linksys 942 phones which
sound great when they call each other
FYI for anyone using the dual ethernet ports on a Cisco 7960.
I'm using a Cisco 7960 connected to an HP2524 10/100 switch, which has
an asterisk box connected directly to it. No VLANs defined or in use.
Measured bandwidth:
PC - HP Switch - Asterisk : actual throughput measured at 94.1 mbps.
Seems to me that it's more logical for the phones to know what their
SRV records are than the server. You shouldn't rely on the dns to
ensure that your system is redundant.
Aaron
On Mar 16, 2006, at 1:03 PM, Douglas Garstang wrote:
I know someone who's at VON this week. Apparently Mark
FYI for anyone using the dual ethernet ports on a Grandstream BT102.
I'm using a BT102 connected to an HP2524 10/100 switch, which has an
asterisk box connected directly to it. No VLANs defined or in use.
Measured bandwidth:
PC - HP Switch - Asterisk : actual throughput measured at 94.1
FYI for anyone using the dual ethernet ports on a Polycom IP600.
I'm using a Polycom IP600 connected to an HP2524 10/100 switch, which
has an asterisk box connected directly to it. No VLANs defined or in use.
Measured bandwidth:
PC - HP Switch - Asterisk : actual throughput measured at 94.1
Splitting the list by type of request may be a good idea, but
splitting based on skill level is just a bad idea... I'm pretty sure
that regardless of a newbie's status, they'll still just go to the
other lists as the newbie list likely won't do much good.
In short, I agree with different
This same issue has been discussed many times over the last two years.
Not likely its going to change now.
Aaron Daniel wrote:
Splitting the list by type of request may be a good idea, but splitting
based on skill level is just a bad idea... I'm pretty sure that
regardless of a newbie's
Have a strange problem...
When a C7960 calls the Polycom ip600, the ip600's first line button
blinks, the ip600 display shows the proper callerid, but the phone does
not ring at all.
If I call the same ip600 from a bt102, the ip600 rings properly.
If I call the same ip600 from another
Does anyone have a guesstimate of how many active Asterisk
installations there are? Sorry this is off topic, need it for a
customer proposal and they need comfort on stability. A count of the
downloads from Digium would be a good start but I couldn't find this
anywhere with Google.
Have a look in the Polycom phone directory - see if the number of the first
7960 is defined in there with a ring type of 0 (silent).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Sunday, 19 March 2006 1:51 PM
To: Asterisk Users-List
Huh? Phones do a NAPTR/SRV lookup in a specified domain to get a list of SRV
records to use. The phones don't query the DNS server every time they make a
call... they have a cache. You also run primary and a secondary (or two
primary) dns servers. It's a simple scalable solution. It's a shame
Of course, but if newbies are separated and together only without any
expert, who can explain them anything ?
I am actualy a subscriber for all the Digium lists. If more lists will be,
more subscribtions I will get and I will receive the same quantity of
messages ;-)
Francois BERGERET,
France.
Hi, we have set up a small project in a school the following way:
SITE_A(4 port analog to ip
g729)--ADSL_ISP1---ISP2Asterisk-PSTN
Site A has 1 Megabit of bandwith (up 512kilobit down 1 megabit)
The asterisk box gets internet service via a wireless antenna. 1 Mbit
of up/down
Hello--
In the interest of Symmetry, does anyone else in the world see any need
for a device like the IAXy (or the SIP ones from other manufacturers,
like the ATA186), but one that presents an FXO interface instead, so it
can be connected not to phones, but the PSTN?
murf
Does anyone know how much was paid ? We would be willing to part-fund
this and to release it as part of the distribution.
Julian.
Steven Totaro wrote:
This is an age old question. Unless something has changed, it is
possible but not included functionality. A group of people paid to have
On Mar 18, 2006, at 11:31 PM, Steve Murphy wrote:
Hello--
In the interest of Symmetry, does anyone else in the world see any need
for a device like the IAXy (or the SIP ones from other manufacturers,
like the ATA186), but one that presents an FXO interface instead, so it
can be connected not
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