2008/10/13 Tilghman Lesher <[EMAIL PROTECTED]>
>
>
> Pray tell, how do you echo cancel in both directions? Wouldn't that
> necessitate cancelling echo before it occurs on the line (sort of a white
> noise/pink noise kind of operation)? Seems like modelling a projectile
> such
> that when it rea
Hi,
These questions are very pertinent and I would be also very curious to read
answers.
I hope these simple lines would encourage readers to respond.
Cheers
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asterisk-users maili
Vieri wrote:
> According to this article:
> http://www.audiodesignline.com/howto/206800151;jsessionid=UJRLVDJCJT2QMQSNDLPSKHSCJUNN2JVN?pgno=1
> I (on the Asterisk system) should take care of the remote user's echo issue,
> ie. either my BRI card or my Asterisk add-on software should correctly ech
Yep, we can probably help you, if you are interested send an email to
[EMAIL PROTECTED] and someone will get back to you to discuss
it.
Tim.
On 13 Oct 2008, at 18:58, Dean Collins wrote:
> Tim Panton from Phone From Here was able to implement this
> functionality when he was at Mexuar so I w
On Mon, Oct 06, 2008 at 08:26:30AM +0100, Roberts Klotins wrote:
> Hi All,
>
> I am getting these events in asterisk message log:
> NOTICE[16647] chan_zap.c: Got event 4 (Alarm)...
> NOTICE[16647] chan_zap.c: Alarm cleared on channel 1
>
> after that asterisk exits silently until I re
On Mon, 13 Oct 2008, Mike wrote:
> On Sun, Oct 12, 2008 at 11:18:40AM +0100, Gordon Henderson wrote:
>> On Thu, 9 Oct 2008, Mike wrote:
>>
>> I'm guessing this lamp is on an ordinary analogue phone you have?
>>
>
> Yeah, this is a bog standard 9 quid analogue phone.
>
>>
>> OK. A bit convoluted th
Hi folks,
I'm working on a solution using the Asterisk voicemail component and
wondered if anyone knew the answer to this question please?
I understand that Asterisk saves voicemail to
/var/spool/asterisk/voicemail///INBOX/ but I
wondered if * creates the file in memory (or tmp/or wherever) and
> Hi folks,
>
> I'm working on a solution using the Asterisk voicemail component and
> wondered if anyone knew the answer to this question please?
>
> I understand that Asterisk saves voicemail to
> /var/spool/asterisk/voicemail///INBOX/ but I
> wondered if * creates the file in memory (or tmp/or
Hi Duncan,
I have tried more times to make the reset phone but is displays always and
only 'upgrading' and MAC address and I cann't access the phone
configuration.
Thanks.
--
Salvatore.
- Original Message -
From: "Duncan Turnbull" <[EMAIL PROTECTED]>
To: "Asterisk Users Mail
On Tue, Oct 14, 2008 at 1:00 PM, Chris Rowson
<[EMAIL PROTECTED]> wrote:
>> Hi folks,
>>
>> I'm working on a solution using the Asterisk voicemail component and
>> wondered if anyone knew the answer to this question please?
>>
>> I understand that Asterisk saves voicemail to
>> /var/spool/asterisk/
> >With ISDN, the conversion is done in your phone
>
> Exactly. Or in the case of Asterisk, it is a 4 wire digital right into =
> the switch--no degradation. Even converting back and forth between =
> analog and digital multiple times compromises quality. Try doing a =
> dial-up modem across suc
Am Montag, den 13.10.2008, 22:54 -0500 schrieb Jorge Mendoza:
> Asterisk is more featured than Panasonic, but you must to know Asterisk
> to convince your executives ;-)
Yes, that's what I need help for.
...
> > when a menu answers, and I dial over, the menu dialed keys works only 20% of
> > a
> On Mon, 13 Oct 2008, Steve Totaro wrote:
> > I have done this. Why BRIs exist in the US is beyond me. If you can,
> > don't go with BRI.
>
> Why didn't BRI catch-on in the US?
Stupidity.
Okay, well, many reasons.
It was targetted as a business service, and the pricing models (at least
loca
On Tuesday 14 October 2008 05:00:45 Chris Rowson wrote:
> > I'm working on a solution using the Asterisk voicemail component and
> > wondered if anyone knew the answer to this question please?
> >
> > I understand that Asterisk saves voicemail to
> > /var/spool/asterisk/voicemail///INBOX/ but I
>
On Monday 13 October 2008 18:57:41 Lee, John (Sydney) wrote:
> By the way, did you see anything wrong with my config files?
>
> /etc/asterisk/res_mysql.conf
> [general]
> dbhost = localhost
> dbname = db1
> dbuser = user
> dbpass = password
> dbport = 3306
> dbsock = /var/lib/mysql/mysql.sock
You
Hi All,
Have created a system that involves using call files in the outgoing
spool folder. On some occasions it retries which is fine is there
any way to view calls waiting retries from the CLI? Using 1.4 btw.
Have googled to no avail (although it is near the end of the day so I
might
Steve Totaro wrote:
>
> My only wish is that Linux had a facility like XP to bridge NICs
> without running all sorts of commands for brctl. Just a GUI like XP.
> Last time I setup a bridge in Linux, I had to change many kernel
> options and rebuild the entire kernel to get bridging working
>
Thanks for the reply.
asterisk-1.4.21.2
zaptel-1.4.12.1
I have posted a more detailed error log in this thread.
Roberts
On Tue, 2008-10-14 at 11:54 +0200, Tzafrir Cohen wrote:
> On Mon, Oct 06, 2008 at 08:26:30AM +0100, Roberts Klotins wrote:
> > Hi All,
> >
> > I am getting these events in a
Brent Davidson wrote:
Babcock, Michael Alex wrote:
hey;
i'm at best western and am curious is there a way i could find out if
our best western, with out asking, is using asterisk?
oh and petsmart i think is using asterisk they have alason voice for
there main voicem enu.
mike
thanks fo
Edgar Guadamuz wrote:
> Hello,
>
> I followed the steps by Russell_
> http://www.venturevoip.com/news.php?rssid=1980_
> and I got it working for publish_event only. As soon as I add
> subscribe_event, Asterisk doesn't start and I just get the following
> message:
>
> *Oct 11 6:38:04.340485 [C
>Why didn't BRI catch-on in the US?
I a word--greed. It arrived shortly after divestiture when there was a lot of
competition in the market and a dozen independent regional telcos. Apparently
they saw a huge cash cow for this data service and yet another competitive
advantage to proprietary im
>> > I'm working on a solution using the Asterisk voicemail component and
>> > wondered if anyone knew the answer to this question please?
>> >
>> > I understand that Asterisk saves voicemail to
>> > /var/spool/asterisk/voicemail///INBOX/ but I
>> > wondered if * creates the file in memory (or tmp
Steven Howes schrieb:
> Have created a system that involves using call files in the outgoing
> spool folder. On some occasions it retries which is fine is there
> any way to view calls waiting retries from the CLI? Using 1.4 btw.
> Have googled to no avail (although it is near the end of
I'm trying to test out Speex for our branch to branch connections, but
am running in to a problem. I downloaded the Speex source code for
1.2rc1, did a ./configure, make and make install then went to my
asterisk folder did a ./configure, make clean make menuconfig verified
that speex is enable
Steve Murphy <[EMAIL PROTECTED]> writes:
> Other than the above, we could invent a slightly different syntax for
> pcre type expressions; and you'd have to invent some sort of
> disambiguation
> for when multiple extensions might be matched, to choose the 'best' one.
I'd just use strict ordering
>>> Hi folks,
>>>
>>> I'm working on a solution using the Asterisk voicemail component and
>>> wondered if anyone knew the answer to this question please?
>>>
>>> I understand that Asterisk saves voicemail to
>>> /var/spool/asterisk/voicemail///INBOX/ but I
>>> wondered if * creates the file in me
Brent Davidson wrote:
> I'm trying to test out Speex for our branch to branch connections, but
> am running in to a problem. I downloaded the Speex source code for
> 1.2rc1, did a ./configure, make and make install then went to my
> asterisk folder did a ./configure, make clean make menuconfig
C. Savinovich wrote:
> Can somebody please give a pointer to a complete neophyte (like me) on
> text messaging, what product can I use to send and automatic text message to
> a cell phone from within the asterisk dialplan? (the part of the dialplan I
> have down, the part of the text message no)
I like to determine that the called user pressed ** to disconnect the call (h
option for Dial CMD ), and not just hang up the phone.
Is there a way to get that information?
The context file where the call file connects the call person is included (it
is simplified ).
First thing in the co
I have to eat crow here guys. I was completely wrong about the use of
dialplan wildcards and non numerics such as *,# and +.
My test was invalid and I drew the wrong conclusion. So to summarize:
A single dialplan extension that matches
'3129842314' or
'*989' or
'+13129842314'
BUT NOT
'i' nor
'h
Hi Salvatore
Do you have a TFTP server that serves the phone configuration files?
This is very separate to the phone, i.e. on a server/pc somewhere, and
will log all the file requests it receives. You can check this
irrespective of the phone
Have you checked whether tftp requests are being mad
On 14 Oct 2008, at 11:00, Chris Rowson wrote:
>> Hi folks,
>>
>> I'm working on a solution using the Asterisk voicemail component and
>> wondered if anyone knew the answer to this question please?
>>
>> I understand that Asterisk saves voicemail to
>> /var/spool/asterisk/voicemail///INBOX/ but I
>
On Mon, Oct 13, 2008 at 11:53:33PM +0200, Hans Witvliet wrote:
> On Mon, 2008-10-13 at 17:37 -0400, Steve Totaro wrote:
> > I have done this. Why BRIs exist in the US is beyond me.
> Much of the idea's behind ISDN are hopelesly outdated, except for one:
BRI is actually ISDN BRI. PRI is ISDN PRI.
On Tuesday 14 October 2008 02:10:39 Olivier wrote:
> 2008/10/13 Tilghman Lesher <[EMAIL PROTECTED]>
>
> >
> >
> > Pray tell, how do you echo cancel in both directions? Wouldn't that
> > necessitate cancelling echo before it occurs on the line (sort of a white
> > noise/pink noise kind of operatio
Hi John,
On Tue, Oct 14, 2008 at 3:36 AM, Lee, John (Sydney)
<[EMAIL PROTECTED]> wrote:
>> if you have applied everything correctly - queue_log file shoudln't
>> have any more lines (except init when restarting asterisk).
>
> Thanks Atis.
> I see what you are saying. In the patch for logger.c,
>
On Mon, 13 Oct 2008, Steve Totaro wrote:
> I have done this. Why BRIs exist in the US is beyond me. If you can,
> don't go with BRI.
Why didn't BRI catch-on in the US? It's been in-use in the UK and Europe
for a long time (especially Germany AIUI). I have several sites with
ISDN2e (BRI) in t
>There's a HFC-S winbond card. How does that card show on lspci?
Network controller: Dynalink IS64PH [0675:1702] ISDN Adapter
kernel modules: hisax
Does that tell you anything useful? Do you want more details? Would you
like to borrow one for a while (I have two)?
Wilton
_
L PROTECTED]
> >> twitter: http://twitter.com/creepyblindy
> >>
> >>
> > What does your sip.conf look like?The only way I
> could see this
> > happening would be if the IP's or Identities were
> somehow getting
> > crossed up. Do your phon
Hi,
I'm using heartbeat as a failover for my asterisk server.
on the active server 1 i have
10.10.10.1 eth0
10.10.10.3 secondary eth0
asterisk listens to the secondary ip, so that if server 1 fails, server
2 will then get that IP.
so if server 1 fails, server 2 will have the IP
10.10.10.2 et
>>> Hi folks,
>>>
>>> I'm working on a solution using the Asterisk voicemail component and
>>> wondered if anyone knew the answer to this question please?
>>>
>>> I understand that Asterisk saves voicemail to
>>> /var/spool/asterisk/voicemail///INBOX/ but I
>>> wondered if * creates the file in me
Hi Dave,
I don't view nothing in tftp server because the phone is stopped on start
screen with displayed 'upgrading' and MAC address..I don't understand what
happened after the reset. phone
Regards.
--
Salvatore.
- Original Message -
From: "David Gibbons" <[EMAIL PROTECTED]>
Never really tried just a dns entry.
I'm wanting to give the phone the ability to sink up to which ever of my
three servers are online.
I've got all my Linksys gear doing this. Just wondering if I can get Aastra
equipment to do this too.
Tom
-Original Message-
From: [EMAIL PROTECTED]
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Cliff Sifton
> Sent: Tuesday, October 14, 2008 4:26 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Looking for a mentor
>
> Looking for a mentor...
>
> Having some issues with
On Mon, Oct 13, 2008 at 7:52 PM, Tzafrir Cohen <[EMAIL PROTECTED]>wrote:
> On Mon, Oct 13, 2008 at 04:55:01PM -0500, Michael Graves wrote:
> > I had converations with both Pika and Xorcom wherein the thought that
> > it should be possible using their interface hardware. There might be
> > some sof
Welcome,
On Tue, Oct 14, 2008 at 01:26:04PM -0700, Cliff Sifton wrote:
> Looking for a mentor...
>
> Having some issues with Asterisk 1.4.22 install. I am
> new to both Linux and Asterisk, however have 20+ years
> programming experience.
>
> First off I hate asking questions I could answer
> my
On Tue, Oct 14, 2008 at 4:59 PM, Steve Totaro <
[EMAIL PROTECTED]> wrote:
>
>
> On Mon, Oct 13, 2008 at 7:52 PM, Tzafrir Cohen <[EMAIL PROTECTED]>wrote:
>
>> On Mon, Oct 13, 2008 at 04:55:01PM -0500, Michael Graves wrote:
>> > I had converations with both Pika and Xorcom wherein the thought that
>
On Mon, Oct 13, 2008 at 09:44:30PM -0600, Wilton Helm wrote:
> The card I have has no name but is based on the Winbond W6692CF
> chip and ships with RVS, which I think is for Windows and of no
> use to me. I'm not sure about whether it is supported by DAHDI or not.
There's a HFC-S winbond card.
Or they just insert your BTN.
I had an issue when sending less than ten digits with Global Crossing. It
would take the digits supplied and fill in the remaining digits with what
was left of my BTN.
Thanks,
Steve Totaro
On Thu, Sep 18, 2008 at 6:32 AM, Jorge Nunes <[EMAIL PROTECTED]> wrote:
> I
Hi Salvatore
You need to look at the logs of the tftp server, not the phone.
Hopefully you can see the ip address of the phone asking for files
If there is nothing at all being requested from the tftp server then you
probably want to reset the phone to defaults again.
Usually it stalls when yo
You need to download a patch for zaptel, thats why your server is crushing.
Search through the forum, there is a known problem or reverse to a version
of Asterisk that is compatible with you zaptel.
On Tue, Oct 14, 2008 at 2:19 AM, Roberts Klotins <[EMAIL PROTECTED]> wrote:
> Hello there,
>
> Wit
Hello,
I'm using Asterisk with an ISDN30e PRI line (only 16 channels active).
Every now and then I get a CONGESTION error even-though there are only
2 channels in use out of the 16.
When this happens, the user just needs to re-dial and the call goes
through OK.
[2008-10-14 15:41:40] -- Execut
This sounds like a likely source of the problem. I changed the ports on
two of the phones and the problem seems to have gone away. Thank you,
Trevor, and others who responded.
--Paul
Trevor Peirce wrote:
> I have seen this with Polycom phones. In my case the problem turned out
> to be because
--- On Tue, 10/14/08, Kevin P. Fleming <[EMAIL PROTECTED]> wrote:
> > According to this article:
> >
> http://www.audiodesignline.com/howto/206800151;jsessionid=UJRLVDJCJT2QMQSNDLPSKHSCJUNN2JVN?pgno=1
> > I (on the Asterisk system) should take care of the
> remote user's echo issue, ie. either my
On Tuesday 14 October 2008 08:29:18 Rodolfo Alcazar Portillo wrote:
> 1xx -- office extensions
> 92xx -- local city calls, which must be free
> 90800xx. -- national 0800 calls, which must be free
> 4___#9xxx. other calls, which must be paid: cell, national,
> international. the three "_
Hello everyone,
I'm getting DIALSTATUS=CHANUNAVAIL when a call is trying to get one of my
queue interfaces, despite the fact it is free at that time, can you give
help?
1. I see many sip channels from that extension:
[EMAIL PROTECTED] asterisk -rx "*sip show channels*" |grep 648
Peer
Setting 'nat=yes' in your sip.conf for each phone will fix this. When
set, Asterisk will ignore the ports defined in the SIP packet (always
5060 with the internal NAT IP) and instead use the IP and port the
packet arrived on post-NAT.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMA
I am trying to get updateconfig working.
I found an example of updating configuration files here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Upd
ateConfig
When I tried it the conf file was updated but the new entry was not added.
action:updateconfig
reload:no
srcfi
Dear list
I'm sorry for not trimming my last post, I know
better.
My ./configure problem has been solved already by
another list member, thanks MartinS! When extracting
files on another computer(a wintel box) and then
copying the files to /usr/src it appears some files
were not flagged as execute
"Karl Fife" <[EMAIL PROTECTED]> writes:
> is in fact simply something like:
>
> exten => _[0-9*#+]X.,1,NoOp(*** match ***)
As long as you're happy to match *9foo and not match **123, then yes,
that will work.
/Benny
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-- Bandwidth and Colocation Pr
On Wed, 15 Oct 2008 01:54:40 +0200, "Benny Amorsen"
<[EMAIL PROTECTED]> said:
> "Karl Fife" <[EMAIL PROTECTED]> writes:
>
> > is in fact simply something like:
> >
> > exten => _[0-9*#+]X.,1,NoOp(*** match ***)
>
> As long as you're happy to match *9foo and not match **123, then yes,
> that will
Thanks, excellent point. Furthermore, a google search on fastsms.conf
yielded the existence of a couple of 'Asterisk SMS gateways'..wow
CS
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson
Sent: Tuesday, October 14, 2008 2:22 PM
To: Asteris
The most flexible way but will require a bit of work and scales SMS modem
per SMS per second.
Install kannel and configure it to work with your SMS modem (many cell
phones work just fine for sending and receiving). It does not have to go on
the asterisk box, just a box you can hit with HTTP or HT
> You might want to double check the socket path. Some distributions
use
> /var/run/mysqld/mysqld.sock as the socket file.
Thanks for the suggestion Tilghman.
I am using Redhat and the socket file is indeed
/var/run/mysql/mysqld.sock.
Actually, if you specify the wrong socket file, you will see
Hi Atis,
> queue_log => mysql,asteriskcdrdb,queue_log
> that is ,,
> If it's wrong, you should see some warnings when asterisk is starting
up.
>
Thanks for the suggestion. I did not put in queue_log for and
it has just taken the default which is queue_log.
In the console startup, you can see be
i don`t want anymore this messages.thanks
_
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