Oh yeah,
And:
Turn OFF "Filter Packets from Registrar".
Turn ON "Support Broken Registrar".
This may or may not be a security risk, but for testing, it will help to see if
toggling these make a difference.
Shanon
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] O
Hello Matt,
very interesting setup! are you using asteriak queues for inbound or not
at all?
Bye
l.
In data Thu, 22 Sep 2005 06:25:44 +0200, Matt Florell <[EMAIL PROTECTED]>
ha scritto:
We wrote VICIDIAL(part of the GPL astGUIclient suite
http://astguiclient.sf.net) for our call center ope
> > > On a related note, I wanted our phones to display "city, st" for the
> > > caller-ID name in the event that none was provided.
> >
> > Interesting code. What sort of memory does * take up when you load up
> > all those CLID values?
>
> I am a little late to this thread, but the answer is W
SIP Message Reference:
# Reboot Phone which is 2000 monitoring 2001s state:
UA--->>> SUBSCRIBE --->>>Asterisk
UA<<<--- 200 OK <<<---Asterisk
# Asterisk saves subscription:
# Wait for a call:
# Call Comes to 2001:
# Asterisk should realize somehow that it needs to NOTIFY 2000 about the call.
UA
On Tue, 20 Sep 2005 16:25:47 -0700, Jonathan Feally wrote:
> I believe it comes with sox. Both my sox and normalize are in
> /usr/bin.
>> Which package comes this "normalize" from?
http://www1.cs.columbia.edu/~cvaill/normalize/
Has debian, redhat and sparc packages. (For debian do: apt-get inst
On 16:46, Thu 22 Sep 05, Matt Riddell wrote:
> Michiel van Baak wrote:
> > On 02:57, Thu 22 Sep 05, Matt Riddell wrote:
> >>Tomasz Chmielewski wrote:
> >>
> >>>How can I manipulate the incoming callerID number (and add 0 before it)?
> >>
> >>exten => s,1,Answer()
> >>exten => s,2,SetCIDNum(0${CALLE
On 17:34, Wed 21 Sep 05, Alchaemist wrote:
> Hi,
>
> Has anybody seen a non commercial, or freeware, or GPL, or even
> CHEAP... POP/IMAP to Text-to-speech?
>
> I have a working version for POP3 using festival. It DOES
> work... it even cleans the email contents to get th
On Wed, 2005-09-21 at 21:44 -0700, Pradeepa Ramamurthy wrote:
> I am just thinking to develop this using Java and Jsp How to implement
> this?...Need help for the same
This is all being reported by CDR tracking. We log CDR into a pgsql
database, then this database can be queried by whatever appli
Michiel van Baak wrote:
> On 02:57, Thu 22 Sep 05, Matt Riddell wrote:
>>Tomasz Chmielewski wrote:
>>
>>>How can I manipulate the incoming callerID number (and add 0 before it)?
>>
>>exten => s,1,Answer()
>>exten => s,2,SetCIDNum(0${CALLERIDNUM})
>>exten => s,3,...
>
>
> And when using CVS head t
I have installed Asterisk and i have configured with two SJPhones; i am able to make calls between these two phones.I am planning to develop a application basically web based application from which the administrator able to trace the call logs or call summary, i mean from which user agent to user a
Hi,
Some way of VPN seems to be the only solution.
But, you should try something really silly first.
Try to setup your asterisk to listen in one of the open ports (ie
21, 22) with SIP you will require two connections, thus two open ports,
instead with IAX2, one port will
How about Mexico City? :)
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Paul Hales
|Sent: Miércoles, 21 de Septiembre de 2005 11:09 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] AstriCon 2006 Loca
On 9/21/05, izo <[EMAIL PROTECTED]> wrote:
On 9/21/05, Matt Florell <[EMAIL PROTECTED]> wrote:> We have sevaral call centers as well, and we just restrict a single server> to 50 recordings at once and then we would pass the next recording as an
> IAX2 channel to another recording server. It's a sc
Hi Lan,
SPA 3000 <> NAT <-> Internet < NAT> Asterisk
That is two NATs... so, as it is, it will NEVER work, so you have only one
way to go.
This is the recipe:
1- Asterisk side, MUST have SIP/RTP ports forwarded in your router
2- RTP ports must be a fixed range in rtp.conf
[general]
Melbourne, Australia would work for me.
PaulH
- Original Message -
From: "Wayne Gemmell" <[EMAIL PROTECTED]>
To:
Sent: Tuesday, September 20, 2005 7:05 PM
Subject: Re: [Asterisk-Users] AstriCon 2006 Location
> How about someplace central like South Africa?
>
>
> --
> Regards
>
> Wayn
On 9/21/2005, "Min Qiu" <[EMAIL PROTECTED]> wrote:
> Ok, I tried xlite (SIP softphone) and I could get into the
> voicemail now. However, I got busy signal when I called
> any Idefisk softphone from xlite. From Idefisk calling xlite
> seems fine.
>
> Min
Min - make sure your DTMF is working o
On 9/21/05, Matt Florell <[EMAIL PROTECTED]> wrote:
> We have sevaral call centers as well, and we just restrict a single server
> to 50 recordings at once and then we would pass the next recording as an
> IAX2 channel to another recording server. It's a scalable system for us that
> is relatively
Adam Robins wrote:
> I have two Asterisk boxes that I thought were trunked, but based on not
> seeing the (T) in iax2 show peers, now I'm not sure.
Make sure you have some form of Zaptel timing (i.e. Digium Cards/ZTDummy)
--
Cheers,
Matt Riddell
___
Dear Richard and supporters!
I see that you guys could be able to setup the SPA 3000 to connect to the
asterisk thru the NAT. I don't know how would to do this. As my understand
is that the SPA 3000 is just able to configure with the SIP that not NAT
aware in Asterisk.
I am trying to configure th
Guys.
I was going to give spandsp-0.0.3pre1 a try under asterisk 1.2beta1..
Anybody done this?
When I noticed that this particular release doesn't have the tx and rx fax
apps on the tree as older ones.
Anybody knows what happened?
___
--Bandwidth and
True, same thing shows, only the ftp host is down, www is up.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|trixter http://www.0xdecafbad.com
|Sent: Miércoles, 21 de Septiembre de 2005 08:23 p.m.
|To: Asterisk Users Mailing List - Non-Commercial
On 9/21/05, Matt Roth <[EMAIL PROTECTED]> wrote:
- What format are you recording to?- What codec are the SIP calls being placed over?
We are recording to the PCM format and using the G711 uLaw codec. Highvoice quality is essential to our application (we are a call center) sowe partnered with MCI t
Bartosz Jozwiak wrote:
Does anybody use SoundPoint IP Attendant Console for Polycom IP 601 with
asterisk ?
Is it going to work with hints in dial plan ?
Since it is not even shipping yet (it was just announced two days ago),
the answer is no.
However, we have had a test unit for some time (a
Bartosz I don't know that anyone has laid hands on it yet, they just
unveiled it at VON earlier this week, they should be shipping next week.
Cory Andrews
Partner / Purchasing
VOIPSupply.com
++
454 Sonwil Drive
Buffalo, NY 14225
++
v - 800.398.VOIP E
Hi guys.
We have currently Asterisk CVS-v1-0-08/15/05-15:53:48
connected in SIP with a Cisco AS5300 (IOS 12.3). One
PRI is connected to the Cisco gateway.
The problem we have is that on incoming PSTN calls to
the AS5300, relayed in SIP to Asterisk, the callerID
name is not being transmitted. We
On Wed, 2005-09-21 at 20:13 -0500, Anton Krall wrote:
> Guys, is Steve's ftp site down? DNS says ftp.soft-switch.org doesn't exist.
>
> Anybody else seen this?
Its happening here. I checked a few things, domain is not expired,
joker DNS is serving this domain and its up. www works, so it is a h
Does anybody use SoundPoint IP Attendant Console for Polycom IP 601 with
asterisk ?
Is it going to work with hints in dial plan ?
http://www.polycom.com/products_services/0,1443,pw-34-182-12104,00.html
Thanks for any help.
Bartosz
-
This mail sen
Guys, is Steve's ftp site down? DNS says ftp.soft-switch.org doesn't exist.
Anybody else seen this?
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/
Hi,
I'm setting up Asterisk as a voicemail with SER. My problem is,
when a caller that is not registered with asterisk (no username and
password in sip.conf) it prompts "403, Forbidden" . I need all calls
from outside of my network to reach asterisk for my users' voicemails,
because anon
Have customer go out to WalMart and buy the ceapest phone they have.
Plug it into the CO lines. See if theyget the Ping!! If so tell
customer to call BellSouth and open a trouble ticket, it sounds like you
you have a short in the pair. Very posible with the storms we have been
having in the past
On Wed, Sep 21, 2005 at 08:58:41PM +0630, Vamsi Pottangi wrote:
> Is it possible to go beyond 250 concurrent ZAP channels with some tweaking
> or workaround ? Meetme uses zap channels, so we could have a max of 250
> conference ports. Is it possible to higher this ?
>
> "An Asterisk system can onl
Mark Phillips wrote:
I was at VON in Boston today and saw on the Digium stand a Cisco 7960
with a picture of Tux and the Asterisk log on its display. I WANT IT!
Anyone know where I can download this file please?
http://www.loligo.com/asterisk/cisco/79xx/2003-04-27.examples/asterisk-tux.
Take a look in the WIKI here, scroll down to Logo Displayed on 79XX Screen
http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx
Cory Andrews
Partner / Purchasing
VOIPSupply.com
++
454 Sonwil Drive
Buffalo, NY 14225
++
v - 800.398.VOIP Ext 22
f -
Guys.
Anybody running asterisk cvs-head and the latest unicall from steve using
r2mfc (in Mexico by any chance)?
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium
Guys. Im using cvs-head from around may and when tring to use automon
(hitting #3) the files are left as .WAV but when trying to open thru winamp
or media player, they complain of bad codecs as if the files werent wavs...
Anybody had issues like this?
__
Hi,
I already made working.
Thanks for the help,
Ryan
Rudolf Ladyzhenskii wrote:
Hi,
You need a single extension to call voicemail. I am using 100.
extensions.conf
exten =>100,1,VoiceMailMain(${CALLERIDNUM})
exten =>100,2,Hangup()
Now, if you simply call VoiceMailMain() without paramet
I was at VON in Boston today and saw on the Digium stand a Cisco 7960
with a picture of Tux and the Asterisk log on its display. I WANT IT!
Anyone know where I can download this file please?
--
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
__
Hello,
Does anyone have experience with Soyo G668 phones crashing? The
crashes appear to occur mostly when dialing out or hanging up. We are
using v 1.42 of their SIP firmware.
Regards,
Ilan
___
--Bandwidth and Colocation sponsored by Easynews.com --
> Hi,
>
> Has anybody seen a non commercial, or freeware, or GPL, or even
> CHEAP... POP/IMAP to Text-to-speech?
>
> I have a working version for POP3 using festival. It DOES
> work... it even cleans the email contents to get the actual content. It
> works great with Outlook
Hi there,
Yes, if I call a cellphone number from a fixed phone (land line) i must dial 044. that's why I'm using it.
My asterisk box has only 1 pots connection to the pstn (from TELMEX). Though, all my outgoing calls go through it, fixed phone lines (other TELMEX users)
and supposedly to cellphon
Remember that 044 is dialed from Mexico land-line
phones. Not used when dialing from outside.
- Original Message -
From:
Alchaemist
To: asterisk-users@lists.digium.com
Sent: Wednesday, September 21, 2005 5:01
PM
Subject: [Asterisk-Users] Re: How can i
call to
Hi (Hola!):
Thanks for your replies.
Rene, I've already posted my dialplan, but i think it didn't reach the list.Here are my configuration files:
"Claudio Canseco" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]...
Hi, thanks for your replay Alex:
Right now a have an Aste
Hi again...
How are you
dialling this?
90446612345678 ? or 0446612345678 ?
Also
another possibility is that the card is sending the DTMF when it haven't yet get
the tone from your PSTN? just thinking... in that case you can use the 'w' in
the dialstring to get a wait dela
Hi Claudio (Hola)
The reason is
surely that you have a conflict with the prefix commonly used in mexico for cell
phones (044)
You will
have to review all your extensions.conf and related files, to make sure the
calls are routed correctly.
Regards!
Alchaemist
"Cl
I call
to Mexico (fixed and cellular phones) via an IAX2 link. And it gives me no
problems at all.
What
kind of trunk are you using?
Maybe
you should post your dial plan... probably there is a mistake
there...
Rene
Kluwen
Chimit
-Original Message-From:
[EMAIL PROTECTED]
I installed astersik 1.2beta and from that point the led that indicate a new call flash.
The ATA installed is an AZATEL.
Any idea what can I check?
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lis
Ok, I tried xlite (SIP softphone) and I could get into the
voicemail now. However, I got busy signal when I called
any Idefisk softphone from xlite. From Idefisk calling xlite
seems fine.
Min
> -Original Message-
> From: Min Qiu
> Sent: Wednesday, September 21, 2005 4:37 PM
> To: [E
Very rarely we come across real success stories using Asterisk as a part
of a great solution, and when I see one, I want to share it with you.
Though it is not mentioned in the news item, it is a fact that Iareanet
uses Asterisk as the core for their messaging part of the solution and
today they ar
[EMAIL PROTECTED] wrote:
I am trying to create an IVR system that uses both POTS and IP phones
and I have a few problems that I encountered with the IP SIP phones
(Grandstream Budge Tone 102).
1. When a user hits the hook fast enough, the user can create multiple
IVR connections that gives the
> Hi,
>
> I have a little situation here :( Perhaps somebody can give me a
> hand with it.
>
> I have an Asterisk working, and in another office, a Sipura
> SPA-3000.
> I configured the SPA and I have the extension working, the incomming
> trunk working, but the outgoi
Hi, thanks for your replay Alex:
Right now a have an Asterisk server on a Dell Optiplex GX110 (PIII 666MHz, 320 RAM) with no soundcard.
With an X100P clone card (an ambient modem).
Everything looks good, I've been able to make local calls trough PSTN, IAX, SIP.
I only have 1 POTS line, and 4
Hi,
Has anybody seen a non commercial, or freeware, or GPL, or even
CHEAP... POP/IMAP to Text-to-speech?
I have a working version for POP3 using festival. It DOES
work... it even cleans the email contents to get the actual content. It
works great with Outlook emails and
Yes, the configbits are exist and correct (see below). I was
wondering if the softphone could make difference. That is the
digits were send too fast/too slow for the *'s voicemail. Any
one here able to use Idefisk check voicemail? Can you share
your configuration?
Thank you,
Min
[default]
On Wed, Sep 21, 2005 at 08:58:41PM +0630, Vamsi Pottangi wrote:
> Is it possible to go beyond 250 concurrent ZAP channels with some tweaking
> or workaround ? Meetme uses zap channels, so we could have a max of 250
> conference ports. Is it possible to higher this ?
>
> "An Asterisk system can onl
Might be this bug:
http://bugs.digium.com/view.php?id=4528
try adding 't' or 'T' to the Dial of the Zap if it's outbound.
If that's not the problem, use the manager API to send a call from the
meetme room to an extension that does Monitor for a specified period of
time. That is how we do it in th
Hi list! I´m trying to put to work the "callprogress=yes" for outgoing
calls using the FXO port of my TDM400 digium board.
The main reason to try that is that I DON´T WANT the asterisk to start
billing my calls before the PSTN called party answers the line. With the
callprogress disabled, aste
Claudio:
In order to receive help from this list,
you need to include more information.
How are you connecting to the carrier?
What are you using as terminals? Softphone?
Which one? SIP or IAX2? Hardphone? Brand and model.
Contents of your extensions.conf, zapata.conf,
and zaptel
Hi,
I tried to use Monitor(wav,filename) function in dialplan to record Meetme
conference. When I monitored on IAX2 or SIP channels in that conference It
recorded all audio (in and out) but when I monitored on ZAP channels I could
hear only IN audio and piece of OUT audio (announcement get pin and
On Wed, September 21, 2005 20:23, Min Qiu said:
> While on the subject, how the password works?
>
> I failed to access the voicemail by using the demo config.
> Password 4242 does not seem to work. I'm using softphone
> Idefisk v1.24.
The extension has to exist in voicemail.conf, and the second
Hi,
I have a little situation here :( Perhaps somebody can give me a
hand with it.
I have an Asterisk working, and in another office, a Sipura
SPA-3000.
I configured the SPA and I have the extension working, the incomming
trunk working, but the outgoing trunk (peer) doe
Hi,
I tried to use Monitor(wav,filename) function in dialplan to
record Meetme conference. When I monitored on IAX2 or SIP channels in that
conference It recorded all audio (in and out) but when I monitored on ZAP
channels I could hear only IN audio and piece of OUT audio (announcement
On 21/09/05, Benjamin Lawetz <[EMAIL PROTECTED]> wrote:
> I've got a friend who's spending 6 months on the other side of the world. So
> before he left I configured him a softphone on his laptop to connect to my
> asterisk so he can call home free of charge.
>
> Unfortunately, he just found out he
I think its the best you can do.
Maybe there should be some option to be set for the monitor command to
buffer, with a warning that it will eat memory.
Its also not needed to buffer the complete call at once, just buffering
and writing to disk every 10 seconds would already be a big improvement
This might be of help as well?
http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+Disconnect+Supervision&diff=3
"Liu Peter" <[EMAIL PROTECTED]> wrote in message
news:[EMAIL PROTECTED]
1) how to config callprogress=yes ? in extensions.conf?
could you give me an example?
2) you means rec
Hi Peter
1) how to config callprogress=yes ? in extensions.conf? could you give me an
example?
Not in extensions.conf, but in zapata.
http://www.voip-info.org/wiki-Asterisk+config+zapata.conf
In my case, busydetect=yes and busycount=5 were the key to getting it right.
2) you means record the c
I see there are a
few phones out now that support IAX2. I was wondering what opinions people
have about the viability of using IAX2 phones or adapters vs SIP phones and
adapters and where they think IAX2 is heading. I am well aware of the
advantages of using IAX2 through NAT and for inter
I'm having lots of problems with sipura spa1001's and spa2002's. Asterisk
claims they are busy when they aren't. Other times, it claims to be ringing
them, but they aren't really ringing. I have done the following to try to
resolve the problem:
1) I upgraded all my spa1001's and
Hi,
I tried to use Monitor(wav,filename) function in dialplan to
record Meetme conference. When I monitored on IAX2 or SIP channels in that
conference It recorded all audio (in and out) but when I monitored on ZAP
channels I could hear only IN audio and piece of OUT audio (announcement
It's true that the average Asterisk implementation doesn't have enough
RAM, but we are replacing a legacy NorTel switch in a call center. If
you look at the cost of traditional PBXs, the cost of additional memory
starts to look a little better. = )
Now for some quick math:
1 minute of PCM a
While on the subject, how the password works?
I failed to access the voicemail by using the demo config.
Password 4242 does not seem to work. I'm using softphone
Idefisk v1.24.
Thanks,
Min
> -Original Message-
> From: Rich Adamson [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, September
Here is my extensions.conf file for debugging
__
Yahoo! Mail - PC Magazine Editors' Choice 2005
http://mail.yahoo.com
extensions.conf
Description: 3949034846-extensions.conf
___
--Bandwidth and Colocat
I've got a friend who's spending 6 months on the other side of the world. So
before he left I configured him a softphone on his laptop to connect to my
asterisk so he can call home free of charge.
Unfortunately, he just found out he has horrible internet connection.
Bandwith and latency is ok, the
The problem is that then it won't work on systems with little memory. 50
streams would eat memory like crazy.
Zoa
Matt Hess wrote:
In light of the I/O bottleneck problem I'd have to ask why asterisk
can't just buffer incoming audio and then flush a complete audio file
to disk.. I'm assuming t
> On Monday 19 September 2005 12:38, Rich Adamson wrote:
> > The g(6) adds a 6 db gain for zap calls that end up recording a Voicemail
> > message.
> ...
>
> > * 'g(#)' the specified amount of gain will be requested during message
> > recording (units are whole-number decibels (dB))
>
>
I would think memory would be the limiting factor. A 3-4 minute wav
file is what, 30Meg or so? And there is one for each end of the call,
so that's 60Meg. Now let's say it's a 15 minute call and then are 10 of
them at once. That's 30Meg x 5 (5 times the length of my estimate) x 2
(each leg)
Hello,
I have installed oh323 channel driver. Outgoing calls to
H.323 world do not include RFC2833 in the outgoing TerminalCapabilitiesSet
despite that userInputMode=RFC2833 has already been set.
Does anyone know how to make RFC 2833 DTMF relay work over
oh323 channel?
Kind regards,
On Wednesday 21 September 2005 13:52, Adam Robins wrote:
> Should I plug in the actual IP addresses instead of host=dynamic? Also,
> I do not currently have "register" statements.
> In iax.conf for these.
register => each to the other.
-A.
___
--Bandwi
> On Mon, 2005-09-19 at 10:38 -0600, Rich Adamson wrote:
> > For those that have experienced low VM recording volumes when using
> > a Digium TDM04b (or similar analog pstn card), a work around has been
> > committed to cvs-head.
>
> Does this mean that tracking down the cause of the low volume is
Its probably an IRQ sharing problem.
- Original Message -
From: "Alan Bunch" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, September 19, 2005 10:09 AM
Subject: [Asterisk-Users] Pinging ...
> Ok, if I missed something in the wiki please
On 02:57, Thu 22 Sep 05, Matt Riddell wrote:
> Tomasz Chmielewski wrote:
> > How can I manipulate the incoming callerID number (and add 0 before it)?
>
> exten => s,1,Answer()
> exten => s,2,SetCIDNum(0${CIDNUM})
> exten => s,3,...
And when using CVS head this will become:
exten => s,1,Set(CALLER
I have two Asterisk boxes that I thought were trunked, but based on not
seeing the (T) in iax2 show peers, now I'm not sure.
Server 192.168.xxx.1 extensions.conf has:
Exten => _2XXX,1,Dial(IAX2/interoffice:[EMAIL PROTECTED]/${EXTEN})
Server 192.168.xxx.1 iax.conf has:
[general]
trunk=yes
[interof
Hi,
I've been trying to dial out to a cellphone, but all my calls get redirected to 066 (the emergency number at my city, like 911)
does anyone know how to fix this, any ideas,?
does anyone from mexico has done this?
Any comment will be highly appreciated,
Regards,
Claudio
_
I am trying to create an IVR system that uses both POTS and IP phones
and I have a few problems that I encountered with the IP SIP phones
(Grandstream Budge Tone 102).
1. When a user hits the hook fast enough, the user can create multiple
IVR connections that gives the appearance of an echo that i
In light of the I/O bottleneck problem I'd have to ask why asterisk
can't just buffer incoming audio and then flush a complete audio file to
disk.. I'm assuming that recordings vary in length.. the problem with
this idea is what happens if 50 recordings all complete at the same
time.. a dump li
On Monday 19 September 2005 12:38, Rich Adamson wrote:
> The g(6) adds a 6 db gain for zap calls that end up recording a Voicemail
> message.
...
> * 'g(#)' the specified amount of gain will be requested during message
> recording (units are whole-number decibels (dB))
How in the hell do
Hi
I am trying to get a SNOM 360 to monitor other extensions i.e. when someone
makes a call to/from another extension, one of the LED's on the SNOM 360 will
change state. I am using 1.0.9/bristuff-8l.
I have 2 extensions - 2001 is a SNOM 190, 2002 is a 360 - both are running
the latest
It depends on the ATA, and our router, etc... Typically in the range between
1 and 2
->-Original Message-
->From: [EMAIL PROTECTED]
->[mailto:[EMAIL PROTECTED] On Behalf Of
->Adrien Laurent
->Sent: Monday, September 19, 2005 12:23 PM
->To: asterisk-users@lists.digium.com
->Subje
Ok, if I missed something in the wiki please point me there with the
correct search terms.
Asterisk 1.0.7 (AAH really)
4 co lines from Bellsouth into a Diguim T400P.
Polycom 501 x 4 on the desktops.
My problem is on calls to or from the CO I hear a "pinging" (thing sonar
ping in a submarine
> I've been trying to diagnose why my server has a constant idle time of 90%
> even when nothing is running.
>
> After finally discovering what "hi" means in 'top' (it means hardware
> interrupts) I find that this percentage always averages around 7-10%.
>
> How can I find out what is causing th
Matthew Boehm wrote:
I've been trying to diagnose why my server has a constant idle time of 90%
even when nothing is running.
After finally discovering what "hi" means in 'top' (it means hardware
interrupts) I find that this percentage always averages around 7-10%.
How can I find out what is ca
On Mon, 2005-09-19 at 10:38 -0600, Rich Adamson wrote:
> For those that have experienced low VM recording volumes when using
> a Digium TDM04b (or similar analog pstn card), a work around has been
> committed to cvs-head.
Does this mean that tracking down the cause of the low volume issue was
not
Try in www.asterisk-es.org
-Mensaje original-
De: Sebastian Milioto [mailto:[EMAIL PROTECTED]
Enviado el: lunes, 19 de septiembre de 2005 15:08
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: [Asterisk-Users] Asterisk in Spanish
Hi all,
I've been installing [EMA
> > On a related note, I wanted our phones to display "city, st" for the
> > caller-ID name in the event that none was provided.
>
> Interesting code. What sort of memory does * take up when you load up
> all those CLID values?
>
> Nathan
>
I am a little late to this thread, but the answer is
Olle E. Johansson wrote:
Vahan Yerkanian wrote:
What is the proper way of adding hints to multiple extensions?
In my case extensions are the same as the sip usernames, while as per
http://www.voip-info.org/tiki-index.php?page=Asterisk%20presence
exten => 1234,hint,SIP/1234 works,
exten =
On Wed, 21 Sep 2005, Thorsten Lockert wrote:
> On Sep 21, 2005, at 8:27 , Armin Schindler wrote:
> > Is there any solution for using ChanIsAvail() in a macro?
>
> Yes. Fix app_chanisavail.c such that it says "if (ast_goto_if_exists(..."
> instead of "if (!ast_goto_if_exists(...". Somone bungled
All,
This message has generated a lot of responses, so I'm going to address
each of them here in an attempt to consolidate the thread.
Matt,
- I'm very interested in the specifics of your setup.
- How much space is on the RAM disk?
features.conf is devoid of #
the queue doesn't have h in it.
only have tT
--- Kevin Bockman <[EMAIL PROTECTED]> wrote:
> Crystal Stream, Incorporated wrote:
> > I am getting this on the console once people call
> in
> >
> >-- outgoing agentcall, to agent '1001', on
> > 'Local/[EMAIL PROTE
I think I got it but just to be sure, where do I find that
setting on sipura 841?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
WilliamsSent: Miércoles, 21 de Septiembre de 2005 10:22
a.m.To: Asterisk Users Mailing List - Non-Commercial
DiscussionSubj
Crystal Stream, Incorporated wrote:
Here is the full "transaction"
-- outgoing agentcall, to agent '1001', on
'Local/[EMAIL PROTECTED],1'
-- Called Agent/1001
-- Executing Macro("Local/[EMAIL PROTECTED],2",
"sipline|3044") in new stack
-- Executing Dial("Local/[EMAIL PROTECTED],2
Hi All.
After some input, I created a V1.1 version of my ODBC VM retrieval from
the ODBC_Storage
It now uses either Mysql or unixODBC drivers to connect to the database
I didn't have php compiled with unixODBC so i had to recompile it in
"./configure --with-unixODBC --with-mysql --with-apxs2=./
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