One last thing:
Playback sounds now like MickeyMouse, much to slow
One sanity check: try zttest . See that it gives values ov 100% or very
close to that.
--- Results after 141 passes ---
Best: 99.499512 -- Worst: 83.691406 -- Average: 95.715467
Hi.
I'm trying to install and configure a TDM01B -1 FXO card.
I'm getting the following errors when starting up asterisk:
Jul 25 08:48:40 WARNING[1775]:
chan_zap.c:923 zt_open: Unable to specify channel 1: No such device
Jul 25 08:48:40 ERROR[1775]: chan_zap.c:6879 mkintf: Unable to open
No rpms, all compiled from source. I cheat though, AMP/freepbx for the init
script. GUI is a godsend when you just need to add a simple phone on a box
that sits across the country and all you have on your mom´s old computer is
a browser. https looks real nice right about then. :P
-Original
Pleasehow can I enable the 3way-conference on a sip
gateway like (addpac) by
dialing an extension example pressing
(*) since the gateway do not have this feature ,I want to make it on server level or if
you know the concept of how call-conference work .
Regards
Hi all,
someone ca suggest me an ata device that can send an hook-flash to fxo
port from voip?
tank's in advance
Regards
Accursio Avona
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Hi,
I'm having problems with the ip-adress Asterisk sends my peer. My
Asterisk server has two ips, one public (212.xxx.xxx.xxx) and one
internal (192.168.1.1)
I want Asterisk to communication with my sip-provider on my public
address, which works as it should. However, my sip-peer connects over
Hello,
sorry if I'm having a hard time understanding your question, but I
doubt that being purely on the phone, it's not possible to know what
userID belongs to what callerID. I don't think from a phone, as an
admin, you can list all the attendees of the conference and have it
read out in a way
Sebastian Reitenbach ha scritto:
I found the same indentical problem, the trouble was the switchtipe, i
am using national and i switched to unknown.
is unknown allowed for switchtype?
when I take a look here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf
then
Hi.
I'm trying to install and configure a TDM01B -1 FXO card.
I'm getting the following errors when starting up asterisk:
Jul 25 08:48:40 WARNING[1775]:
chan_zap.c:923 zt_open: Unable to specify channel 1: No such device
Jul 25 08:48:40 ERROR[1775]: chan_zap.c:6879 mkintf: Unable to open
Pleasehow can I enable the 3way-conference on a sip
gateway like (addpac) by
dialing an extension example pressing
(*) since the gateway do not have this feature ,I want to make it on server level or if
you know the concept of how call-conference work .
Regards
Hi all,
I cannot exactly reproduce your problems, but I can tell you, what problem
we have on this topic:
a calles b.
b takes the call and can speak to a.
b sets up a attendend transfer (via the softkey configured in asterisk) to
c and hears ringing.
a hears music on hold.
b hears ringing
if c
Hi.
Typing in ztcf -vv gives the following:
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
...
My /etc/zaptel.conf looks as follows:
fxsks=1
...
The relevant stuff in my /etc/asterisk/zapata.conf looks as
follows:
signalling=fxs_ks
--
Hello,
I have a problem with the ringing on a Asterisk 1.2.x and a Digium TE410 and
TE411P.
if i Dial without any dial parameter through a Zaptel channel i hear the
ringing from the telco and the ringing generated from asterisk.
If i Dial through the Zaptel with Parameter r i get the
Robin Ericsson wrote:
Hi,
I'm having problems with the ip-adress Asterisk sends my peer. My
Asterisk server has two ips, one public (212.xxx.xxx.xxx) and one
internal (192.168.1.1)
I want Asterisk to communication with my sip-provider on my public
address, which works as it should. However,
Accursio Avona wrote:
Hi all,
someone ca suggest me an ata device that can send an hook-flash to fxo
port from voip?
The sipura spa3000 can do it. The user (on the fxs port) must
double-flash to make the pstn (fxo) port flash.
I don't know of a way for a sip device (or any other non-fxs
Anyone know if it possible to create binary/obfuscated/ human
unreadable extensions.conf/sip.conf etc.? We would like to deploy a
system in an environment where not giving out root is still not enough.
We want to hide the contents of these normally plain text files.
Hi,
i tried deny and permit in the peer definition.
It works fine for registration purpose.
But if the peer is dialing through Asterisk these settings are ignored. Only
username and password are used for authentification.
Is there anythink additional what I can use to prevent that the phone
Don't really know if this is possible but the way I think it works it
should be doable.
Have the configfiles encrypted and decrypt when asterisk is
starting/reloading and then encrypt again.
Marcus
Eric Bishop skrev:
Anyone know if it possible to create binary/obfuscated/ human
unreadable
2006/7/24, Steve Langstaff [EMAIL PROTECTED]:
I couldn't find an open source phone, but NCH Express Talk appears to be a
free download that supports G726-32.
Hi
I installed NCH Express but not support G726-32 like as says the
company's website.
Thanks
roberto
--
Ing. Roberto Pereyra
Dovid Bender wrote:
I am sure you prob. know this but in your configs it shows secret
commented out. Also it with a softphone if it dosent work then, then its
your configs. Also did you remember to reload asterisk ?
- Original Message - From: James Fromm [EMAIL PROTECTED]
To:
J == Jones [EMAIL PROTECTED] writes:
J Hi Folks, We're migrating from a conventional KSU/PBX to Asterisk
J and I'm trying to determine the best way to allow our receptionist
J to answer certain executives telephone lines.
J It seems there are probably two routes, but I'm not sure of the
J
show dialplan or other commands from cli renders this unnecessary.
the only way to make those things unreadable, IMHO is an
sophisticated,komplex dialplan/extension.conf which is unreadable at all.
or an other way may me using as much agi as you can,
and an binary exe file which is encrypted.
Hi,
I am wrong... sorry :-) pridialplan not switchtype!
Ok... this is:
the number on the PRI is always:
XXXYYY where is the radical and YYY is the local number
this is wath is working for me:
switchtype = national
pridialplan = unknown
prilocaldialplan = unknown
I disagree with you entirely which is why I've been such a strong
supporter for [EMAIL PROTECTED] and now of Trixbox.
There is no reason why this technology shouldn't be installed directly
by an end user.
It's no more complicated than messenger with LCS etc, it's just MS does
a better job of
Rich Adamson ha scritto:
Accursio Avona wrote:
Hi all,
someone ca suggest me an ata device that can send an hook-flash to
fxo port from voip?
The sipura spa3000 can do it. The user (on the fxs port) must
double-flash to make the pstn (fxo) port flash.
I don't know of a way for a sip
A bad product is in no way the only reason to encrypt your configuration files. The configuration of a given set is your IP... Most people don't just give that stuff away.Alex
On 7/25/06, Kai Ober [EMAIL PROTECTED] wrote:
show dialplan or other commands from cli renders this unnecessary.the only
so, whats the idea of open source?
files. The configuration of a given set is your IP... Most people don't
just give that stuff away.
okay... dont feed me, the troll, i will stop answering this thread.
regards
KAI
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Hi all,
Has any of you got hold of an IAX ATA with FXO? If yes, pls provide
some info and on the ams with a review of it.
Thanks in advance.
Dan
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Yes but no offense to [EMAIL PROTECTED] it dosent have everything. It seems simple to set
up but if there is a problem since the end user dosent know what configs [EMAIL PROTECTED]
is creating he has no real way of trouble shooting. When I started in
asterisk first I was looking all over for
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Come again?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Carlos
Alberto Bernat Orozco
Sent: 25 July 2006 14:59
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Voice with echo
___
Asterisk 1.2.4
Various hardware on the PBXs P4 3ghz, Celerons, etc
Polycom phones running 1.6.5 or other ATA hardware
Talking to Cisco 3660 for PRI access as well as a Cisco 3660
for long distance
Ulaw or g729
AMP or FreePBX as the GUI to control
Randomly, and this is very hard to
Verify if you have Microsoft DirectX activated in your SJphone, i've
had problems in the past in a few desktops that were solved disabling
it.
On 7/25/06, Steve Langstaff [EMAIL PROTECTED] wrote:
Come again?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf
I'll probably get blasted for this. I hope I'm wrong, and then a little
blasting is ok. It appears that Asterisk may have let us down again as a
'carrier grade' solution.
1. User A calls User B. The call is bridged.
2. User B wants to transfer User A to user C. When this happens, User B's phone
Hi all,
I've been wondering if do i need G729 license to accomplish bridging
two calls to PSTN through a VoIP service Provider that allows me G729.
Do I need to get G729 license?
I might need chan_g729.so but only to negotiate with my VoIP provider,
the truth is that i will be only passthrough.
I am having a problem with my VegaStream 50 10 FXO. The unit does not
recognize DTMF sent from asterisk on outgoing calls. I have been trying to
resolve this with VegaStream support but they have not been very helpful so
far. On the last test we ran we used Eathereal to capture traffic and it
I stumbled across a draft rfc for IAX2 at:
http://www.eguy.org/presentations/2006/draft-guy-iax-01.html#sec.new
It describes all the IAX2 'information elements' sent when a new call is
established.
Here's the list.
http://www.eguy.org/presentations/2006/draft-guy-iax-01.html#sec.new
When I
Pleasehow can I enable the 3way-conference on a sip
gateway like (addpac) by
dialing an extension example pressing
(*) since the gateway do not have this feature ,I want to make it on server level or if
you know the concept of how call-conference work .
Regards
Hi
Please how can I get the user register ip address and put it at cdr ,its too
important
Thanks
*
No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail
I agree with a bad phone or config problem. I use Polycom extensively
and have had very few problems. Can you give us more info on the config
so we can try and help?
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How do you mean it does not recognize them? By the routing not
working properly?
Or by not outdialing properly?
No need for ethereal, just turn on sip debugging and it will display
the messages for you, just like * will.
On Jul 25, 2006, at 9:43 AM, Issac Simchayof wrote:
I am having
The Mitel 3300 natively supports SIP as of version 7.0.21.4 (also
referred to as 7.0 UR2) which was recently released. We currently have
several Asterisk systems integrated with Mitel 3300's via SIP. You will
require SIP trunking licenses.
Contact me off list if you need any assistance with
Thanks Richard, somebody pointed me to the CDP setting, and to a bad hub
(which, even if I said it wasnt the case, seemed to have been my second
problem)
Everything works now, thank you to this group for giving me such quick help.
Mike
-Original Message-
From: [EMAIL PROTECTED]
On Tue, 2006-07-25 at 20:12 +1000, Eric Bishop wrote:
Anyone know if it possible to create binary/obfuscated/ human
unreadable extensions.conf/sip.conf etc.? We would like to deploy a
system in an environment where not giving out root is still not
enough. We want to hide the contents of these
You were already given an answer to this:
Check out this page:
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sippeer
I believe that some variation of that command and the Set(CDR(cdrfield)=whatever) will do it for you.
Perhaps there is an easier way, but I don't know it.
Good luck,
Hi groupThanks Marty for your colaboration. I tried the my voice call with 2 extensions and SJphone as softphone as you know. For the test I used a normal mic plug into the mic port from a laptop and made the call to another pc wich has second extension. At first time I believed what you told me
Hi:
I'm setting up a branch office, but I don't want to trunk from the main
office because I don't want to introduce any more latency. Also, the
office will have only a single extension, so I can't justify the expense
of a second Asterisk server for it.
SIP is a pain when going through
Hi:
If I connect two offices through an IPsec tunnel, what is the impact on
latency, and does it noticeably affect calls?
Has anyone out there tried this? What were the effects?
Cheers,
-Stephen-
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Hi,
Would it be possible to use Asterisk to retrieve podcasts and make them
accessible via a softphone like Ekiga ?
Thanks.
--
kael
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This morning I found this message on my Asterisk Console. Does it
mean I should be concerned about the security of my system?
-- Remote UNIX connection
== Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger Restarted
-- Remote UNIX connection disconnected
The Mitel 3200 was Mitels first
effort at an IP PBX and ran on Windows NT. It has been long discontinued
(many years ago) and was replaced by the Mitel 3300.
There is SIP firmware available for the
5212, 5224, 5215, 5220 and 5235 phones on Mitels sip firmware site at
sipdnld.mitel.com.
I'm not sure, but
can asterisk-BE do something like that?
regards
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It seems you didn't post any thing about you [general] sip.conf
neither allowed codecs
On 7/25/06, Carlos Alberto Bernat Orozco [EMAIL PROTECTED] wrote:
Hi group
Thanks Marty for your colaboration. I tried the my voice call with 2
extensions and SJphone as softphone as you know. For the test I
my mistake you post it! could you pos it in file.conf format?
On 7/25/06, Marco Mouta [EMAIL PROTECTED] wrote:
It seems you didn't post any thing about you [general] sip.conf
neither allowed codecs
On 7/25/06, Carlos Alberto Bernat Orozco [EMAIL PROTECTED] wrote:
Hi group
Thanks Marty for
Asterisk is sending the DTMF as we can see in ethereal but the Vega is not
sending them out. We did try the debug before ethereal but the tech at
VegaStream insisted we will need ethereal to troubleshoot this.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Make sure you have enough CPU bandwidth on both sides IPsec has to
encrypt every little packet.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen
Bosch
Sent: Tuesday, July 25, 2006 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial
GABcast has IVR to allow users access podcast from Asterisk
On 7/25/06, kael [EMAIL PROTECTED] wrote:
Hi,
Would it be possible to use Asterisk to retrieve podcasts and make them
accessible via a softphone like Ekiga ?
Thanks.
--
kael
___
Subject: [asterisk-users] Re: Operator Console(s)/Shared Call Appearances
Hi Folks, We're migrating from a conventional KSU/PBX to Asterisk
and I'm trying to determine the best way to allow our receptionist
to answer certain executives telephone lines.
It seems there are probably two
Stephen wrote:
If I connect two offices through an IPsec tunnel, what is the impact
on
latency, and does it noticeably affect calls?
That would depend a lot on the equipment that services the IPSEC tunnel
endpoints.
Has anyone out there tried this? What were the effects?
I've run small to mid
Someone connected to the Asterisk console using asterisk -r then typed
logger reload then exited the session.
Ira wrote:
This morning I found this message on my Asterisk Console. Does it mean I
should be concerned about the security of my system?
-- Remote UNIX connection
== Parsing
On Tue, 25 Jul 2006, Stephen Bosch wrote:
Hi:
If I connect two offices through an IPsec tunnel, what is the impact on
latency, and does it noticeably affect calls?
There should be no change compared to a non IPsec tunnel.
Has anyone out there tried this? What were the effects?
We have
Should be doable, but it would take a bit of scripting. You would have to get a program that subscribes to the feeds in Linux (bashpodder does this) and downloads the files to a given directory. You would then have to run something to convert those mp3s into something Asterisk can use, then move
On Tuesday 25 July 2006 06:12, Eric Bishop wrote:
Anyone know if it possible to create binary/obfuscated/ human unreadable
extensions.conf/sip.conf etc.? We would like to deploy a system in an
environment where not giving out root is still not enough. We want to hide
the contents of these
so the vega is pulling dialtone on the proper co line but not dialing
anything?
Can you post the appropriate profiles?
Are you sending inband or out of band?
On Jul 25, 2006, at 10:47 AM, Issac Simchayof wrote:
Asterisk is sending the DTMF as we can see in ethereal but the Vega
is not
On Monday 24 July 2006 19:59, Mike wrote:
Thanks Eric, you found it. I just turned off the CDP setting and at first
glance, everything works. Thanks. No need to change hub (Im on a small
home network, which is why I can afford having a hub).
A small 5-to-10-port switch is not going to break
Do you have a cron job running asterisk -rx logger rotate ? That is all that the SLI is showing is that a connection was opened to the CLI and the logs were rotated.On 7/25/06,
Ira [EMAIL PROTECTED] wrote:
This morning I found this message on my Asterisk Console. Does itmean I should be concerned
If you're doing a lot of IPsec traffic, you should invest in hardware devices to do it if it's mission-critical (Cisco ASA 5500, something from Checkpoint, etc).-brandonOn 7/25/06,
Alexander Lopez [EMAIL PROTECTED] wrote:
Make sure you have enough CPU bandwidth on both sides IPsec has toencrypt
On Monday 24 July 2006 18:33, Steve Underwood wrote:
This statement is very very wrong. The timing matters enormously. If the
timing doesn't match, there will be frame slips, and things like modems
will not work. The snag is, right now neither Asterisk or the cards it
uses have the ability to
Does anybody knows how to transfer calls from Sipura SPA 1001 configured
as asterisk internal ?
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That's simply the remaining rationalization that is left in the absence of the bridged line appearances.
On 7/25/06, Matthew Warren [EMAIL PROTECTED] wrote:
Subject: [asterisk-users] Re: Operator Console(s)/Shared Call Appearances Hi Folks, We're migrating from a conventional KSU/PBX to Asterisk
Couldn't this has been done from any GUI installed?like AMP or freepbx
On 7/25/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Someone connected to the Asterisk console using asterisk -r then typed
logger reload then exited the session.
Ira wrote:
This morning I found this message on my
Ive done it with a tunnel set with OpenVPN, and works quite good,
there is a slight increase of lattency but not noticeable to humans.
that is doing it via UDP tunnel, we also tried via a TCP tunnel and
results weren't good, lattency increased more than desired and voice
quality was
We have two offices - one in Oklahoma and the other in Vancouver, BC -
connected via an OpenVPN connection. We have big pipes at each site
(15Mb and 10Mb), and it works great. We average about 70ms latency
through the tunnel. We have about 5-6 conference calls per day with up
to 20 users, and
I have been experiencing clicks and pops during music on hold playback. Any
ideas of what usually causes this? It seems to be some timing problem but I
am using a Digium TDM04B.
Best,
Francisco
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Hi Issac,
If I recall correctly, out of band DTMF didn't seem to work for us on
our Vega 50 (atleast not when using the Vega with Asterisk). We had to
tell Asterisk to use dtmfmode=inband in our sip.conf. It didn't seem
like we had to change any settings on the Vega, because it was sending
both
I think is a problem in the reload routine of unicall. Note that I
have not the newest version, and im not able to reload, it does not
give me the same message, but still i cannot reload and the unicall
channels are no longer available after executing reload. I think you
should avoid using
Hello All , Is there a command or set of commands that will give the
same data resources as 'iax2 show netstats' for sip ?
Tia , JimL
--
+--+
| James W. Laferriere | SystemTechniques
Andrew Kohlsmith wrote:
What I was trying to state was that if you have two data streams that are
solidly clocked but out of phase, you will not encounter any of these issues.
If the clock period of either (or both) drifts then yes, you will run into
trouble.
So it sounds like Asterisk
Well, I was asleep when it happened and no one else has access to the
machine. Does that mean someone logged in from outside and I should
be worried about the security of my machine?
Ira
Someone connected to the Asterisk console using asterisk -r then
typed logger reload then exited the
Thanks Pete,
I did try dtmfmode=inband and it did not work for us.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter Doyle
Sent: Tuesday, July 25, 2006 12:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
Hi all,
Here is the situation:
A call comes in to an Alcatel PBX and it sends it to an E1 on * , this
* either sends the call to a VoIP extension or needs to forward it to an
extension back on the Alcatel, but WITHOUT using another slot of the
E1 (no tromboning or hairpinning).
I've
Check your cron jobs, especially since it happened while you were asleep, mine runs at 4:00 am evey day.On 7/25/06, Ira
[EMAIL PROTECTED] wrote:Well, I was asleep when it happened and no one else has access to the
machine. Does that mean someone logged in from outside and I shouldbe worried about
On Jul 25, 2006, at 12:53 PM, Manrique Feoli wrote:
Hi all,
Here is the situation:
A call comes in to an Alcatel PBX and it sends it to an E1 on * ,
this * either sends the call to a VoIP extension or needs to forward
it to an extension back on the Alcatel, but WITHOUT using another
slot
It will cause issues if you are using fax/modems on the channel bank and trying to send out via the PRI. We had a great deal of problems with timing sync between 2 spans on a Sangoma A104D until the latest beta drivers were released.
On 7/25/06, Shaw Terwilliger [EMAIL PROTECTED] wrote:
Andrew
Hi Matt, thanks for your answer,
I guess it is still as you said a while back that you did it using 5ESS
Can you share how you did in 5ESS? (a sample of the extensions.conf )
and what kind of switch you were connected to?
I'm not sure if the Alcatel 4400 and the Nortel Meridian 11
Bruce Reeves wrote:
It will cause issues if you are using fax/modems on the channel bank and
trying to send out via the PRI. We had a great deal of problems with
timing sync between 2 spans on a Sangoma A104D until the latest beta
drivers were released.
No faxes here. After reading dozens of
I have
three phones here with extensions 2944093, 3254103 and
9220371.
2944093 calls 3254103. 3254103 transfers 2944093 to 9220371. We want the
caller id of 2944093 to be presented on the display of
9220371.
However, the caller id of the transferer, 3254103, is appearing. This
doesn't
Carlos Alberto Bernat Orozco wrote:
Hi group
Thanks Marty for your colaboration. I tried the my voice call with 2
extensions and SJphone as softphone as you know. For the test I used a
normal mic plug into the mic port from a laptop and made the call to
another pc wich has second extension.
- Original Message -
From: Douglas Garstang
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Tue, 25 Jul 2006 15:25:15 -0300
Subject: [asterisk-users] Caller ID on
Transfers
I have three phones here with extensions
-Original Message-
From: Joshua Colp [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 25, 2006 8:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID on Transfers
- Original Message -
From: Douglas Garstang
On 25 Jul 2006, at 16:23, Stephen Bosch wrote:
Hi:
I'm setting up a branch office, but I don't want to trunk from the
main
office because I don't want to introduce any more latency. Also, the
office will have only a single extension, so I can't justify the
expense
of a second Asterisk
- Original Message -
From: Douglas Garstang
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Tue, 25 Jul 2006 15:37:10 -0300
Subject: RE: [asterisk-users] Caller ID on
Transfers
What type of transfer? blind or
I don't want this to go Jihad.
End users have EVERY right to have a phone that is easy to use. That is all
I am saying. If it is a nightmare to configure but easy to use that is
fine. The original post suggested it is neither easy to configure nor use.
-Original Message-
From:
I have an Asterisk host connected to a T1 facility, and another Asterisk
host connected via an IAX trunk in another location. I have Ring groups
defined to ring a number of extensions at once. Intermittently when one
of these ring groups is triggered, everyone that is on a phone call in
the
On 7/25/06, Stephen Bosch [EMAIL PROTECTED] wrote:
Hi:
I'm setting up a branch office, but I don't want to trunk from the main
office because I don't want to introduce any more latency. Also, the
office will have only a single extension, so I can't justify the expense
of a second Asterisk
On Tuesday 25 July 2006 14:37, Douglas Garstang wrote:
What type of transfer? blind or attended?
Does it matter? Both...
Yes it does matter. On any KSU or PBX I have used, attended transfers show
the name/extension of the transferer (presumably because it is THEM you are
talking to).
Douglas Garstang wrote:
I'll probably get blasted for this. I hope I'm wrong, and then a little
blasting is ok. It appears that Asterisk may have let us down again as a
'carrier grade' solution.
Did the list software screw up, or did you post this exact same mail
yesterday?
B.
Hi Stephen,
+99 ms via IPSec FreeSWan
But good protection and no NAT issue.
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Stephen Bosch
Envoyé : mardi 25 juillet 2006 17:25
À : Asterisk Users Mailing List - Non-Commercial
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 25, 2006 12:48 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Caller ID on Transfers
On Tuesday 25 July 2006 14:37, Douglas Garstang wrote:
What type of transfer?
Alex Robar wrote:
Should be doable, but it would take a bit of scripting. You would have
to get a program that subscribes to the feeds in Linux (bashpodder does
this) and downloads the files to a given directory. You would then have
to run something to convert those mp3s into something
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