] On Behalf Of Tilghman
Lesher
Sent: Wednesday, September 16, 2009 4:03 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] res-crypto dependencies
On Wednesday 16 September 2009 14:34:05 Michelle Dupuis wrote:
I'm trying to enable res_crypto on a 1.4 installation, but menuconfig
says ssl is needed
, September 16, 2009 4:43 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] res-crypto dependencies
On Wednesday 16 September 2009 15:10:57 Michelle Dupuis wrote:
On Wednesday, September 16, 2009 4:03 PM, Tilghman Lesher wrote:
On Wednesday 16 September 2009 14:34:05 Michelle Dupuis wrote:
I'm
I have internal mailboxes that I don't want visible to callers going through
the directory. Is it possible (in * 1.4) to hide mailboxes fom the
directory, without creating a new context?
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Are you using the # symbol in the extension name or to access a feature
(eg: outside line)??
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez
Sent: Monday, September 07, 2009 1:12 PM
To: Asterisk
Start with simple mail testing (forget asterisk)
Does mx1.datagrama.net accept messages for testu...@mydomain.com ? Try a
telnet session first...
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joan Antoni
Terre
Sent:
Do a quick search for SMTP commands - to simulate a complete session via
telnet.
Most MTA's will check sender and recipient for validity, relaying, etc. Be
sure both are reasonable and acceptable to host using telnet first.
If you are new to sendmail.cf, read the instructions at the top of
Check your hostname settings, hosts file, and order of name resolution...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bernd
Petrovitsch
Sent: Monday, August 24, 2009 5:26 PM
To: Asterisk Users List
I'm working on a script that needs to determine the extension (eg: 123) of
the phone that initiated the call, or CALLERID number if an externall
caller.
Is there a simple way to do this?
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Does anyone have an example of how to create a custom filename for the
(combined in/out) audio file captured through automon?
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Yes - we typically install behind NAT. The issue will usually be your
firewall setup ...assuming you have setup your peers for NAT.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ricardo
Martins
Sent:
Just out of curiosity, how are you planning to use it? (Reading email,
etc?)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of equis software
Sent: Monday, June 08, 2009 7:58 AM
To: Asterisk Users List
Subject: [asterisk-users]
, 2009 2:15 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Suddenly the voice became garbage
(likerobot)using Asterisk 1.4.19.2
Michelle Dupuis escribió:
You're not alone...we never found the cause of this (rare) occurance...
-Original Message-
From: asterisk-users-boun
You're not alone...we never found the cause of this (rare) occurance...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Sunday, May 31, 2009 8:58 PM
To: Asterisk Users List
Subject:
Just check the version of the card (5v vs 3v) - I don't think PCI X is
compatible with the older 5v cards.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Wednesday, May 27, 2009 9:20 AM
I created a mysql table and lookup script for this. One one server were we
could not use mysql, we created an array of exchanges and compared to those.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean
Pick a release and stick with it as long as you can. Only when you have to
jump, pick a new release, test the hell out of it, and then leave it alone.
Too many people try to keep on the latest release...
_
From: asterisk-users-boun...@lists.digium.com
I'd like to setup a single extension for which all INBOUND and OUTBOUND
calls are recorded to a wav file. I took a look at the wiki:
http://www.voip-info.org/wiki/view/Asterisk+record+calls
but it's not too helpful. Can someone show some sample code in out
recording?
Thanks,
MD
I want to record calls in wav format. Can someone tell me how many MB of
storage per minute each recording requires (assuming SIP / uLaw codec / full
duplex recording)
Thanks,
MD
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I have a simple asterisk install (1.4.18), and want to use call parking. I
can successfully park a call (I see on the CLI that the call is parked to
701). Everything is pretty default.
However, I can't pickup a call from another phone. When I dial 701 from a
phone, asterisk can't find that
, February 25, 2008 8:18 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Parked calls - can't pickup
On Mon, 2008-02-25 at 20:03 -0500, Michelle Dupuis wrote:
However, I can't pickup a call from another phone. When I dial 701
from a phone, asterisk can't find that extensions
Will the built-in T.38 support eliminate the need for spandsp? I'm curious
how this will affect iaxmodem.
MD
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob Hillis
Sent: Saturday, February 23, 2008 7:12 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] FXO
Users List
Subject: Re: [asterisk-users] FXO Cards - T38
Michelle Dupuis wrote:
Will the built-in T.38 support eliminate the need for spandsp? I'm
curious how this will affect iaxmodem.
Why on earth would you want to eliminiate spandsp? (which
app_fax from asterisk addons appears
I have setup hylafax today, along with iaxmodem. I'm just starting the
debugging process and see the following message every 60 seconds:
[Feb 19 17:44:16] NOTICE[1996]: chan_iax2.c:6021 update_registry:
Restricting registration for peer 'iaxmodem0' to 60 seconds (requested 300)
Can someone
registration for
peer 'iaxmodem0' to 60 seconds
Michelle Dupuis escribió:
I have setup hylafax today, along with iaxmodem. I'm just starting
the debugging process and see the following message every
60 seconds:
[Feb 19 17:44:16] NOTICE[1996]: chan_iax2.c:6021 update_registry
Wow...where did you get that answer?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Paul Hales
Sent: Tuesday, February 19, 2008 8:42 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] ISDN2 facility code...
I have just been given the
Just something I noticed: your third line from extensions.conf begins with
s, while the other two begin with _X.
Michelle Dupuis
Technical Support Specialist
Generation Software - Linux and Asterisk solutions and support. Visit us at
www.generationd.com http://www.generationd.com
The Aastra's also have a range of interested firmware bugs that
support/development just can't seem to fix. Do a search for aastra
hang/lockup and you will find what I mean. They look very nice though! For
a simple home deployment, Aastra's are probably great.
-Original Message-
Well, we can already integrate to major platforms via SMTP. The real value
is in deep integration to the most popular email platform in business:
Exchange.
I would love to see smart Exchange integration, where deleting the VM
attached email will delete the corresponding message from asterisk.
There is a bug in the 480 firmware where if the callerid of the incoming
call is malformed (or basically the Aastra doesn't like, for example have a
# sign in the number), the phone won't ring.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Can someone explain what the facilityenable setting does in zapata.conf
I've read the wiki archive, but it's not even clear what an ISDN
facility is.
Thanks,
MD
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asterisk-users
channel bank?
Exists?
To: asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; DelSp=Yes;
format=flowed
Quoting Michelle Dupuis [EMAIL PROTECTED]:
We have a client with a Nortel PBX with digital phone
We have a client with a Nortel PBX with digital phone sets. Due to T1
problems (old firmware), we are interested in trying a FXO channel bank.
Is there a channel bank (or equivalent) which emulates Meridian digital
phone sets? In order words, an FXO channel bank that's Meridian digital?
Can someone advise on how to go about finding someone QUALIFIED to make
changes to libpri?
We have a pilot stuck on hold, due to old buggy PRI software on a meridian
PBX. Upgrading the meridian software is not an option, sowe would like
to have libpri changed to compensate for the bug.
Is
read from the cfg file
and set, but it wasn't the case.
Same problem happened to your setup?
On Nov 7, 2007 6:33 PM, Michelle Dupuis [EMAIL PROTECTED] wrote:
Use the web interface of the phone to retrieve the config file that
you uploaded. Is it only partially there?
-Original
Have a look at the smartCID script on www.generationt.com
It allows you to have a database of numbers and override the name (and
number), flag numbers for screening, etc.
MD
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jon Weisman
Sent: Thursday, November 08,
Are all of your settings getting lost, or just some? We've encountered some
interesting bugs in the Aastra's...(tech support said wait for the next
firmware release, for 8 months - and yes there have been firmware releases
in-between).
If your tftp is in fact working, strip you .cfg down to the
Use the web interface of the phone to retrieve the config file that you
uploaded. Is it only partially there?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Roi Stork
Sent: Wednesday, November 07, 2007 9:27 PM
To: Asterisk Users Mailing List
We are connecting an asterisk box to a Nortel Option 61 via a T1 with PRI.
We have hit a problem we cannot overcome; specifically, the Nortel is asking
for the ROSE information element (IE) over the PRI connection. This causes
libpri to drop the connection, with cause 100. The Nortel cannot turn
: [asterisk-users] PRI over T1 calls dropping, cause 100
Michelle Dupuis wrote:
I have a T1 link from asterisk 1.2.23 (also tried with 1.4.13) to a
Meridian Option 61C. Calls either way drop with error
Channel 0/23, span 1 got
hangup, cause 100. Can anyone offer insight into the cause
The T1 was setup as tie line, not a trunk. The Bell guy tried setting up
the line 2 ways:
1. As a trunk. This did not work because:
a) When he typed in the access code for the trunk on a phone set (and
then any numbers), the call never appeared on the Asterisk side.
b) The Bell guy said
Well, I'll bite and get the war going.
I'm putting in my first Sangoma card at the moment...so I have some current
experience.
The card installs great. Hardware compatibility is good (tried in a few
machines). Documentation on website is weak.
Tech support...mixed. I spent a lot of time
I have a T1 link from asterisk 1.2.23 (also tried with 1.4.13) to a Meridian
Option 61C. Calls either way drop with error Channel 0/23, span 1 got
hangup, cause 100. Can anyone offer insight into the cause and
solution/workaround? (I tried upgrading to Ast 1.4.13, and upgrading
matching zaptel
:[EMAIL PROTECTED] On Behalf Of
Michelle Dupuis
Sent: Tuesday, October 30, 2007 9:47 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Digium Vs sangoma Hradware
Well, I'll bite and get the war going.
I'm putting in my first Sangoma card
We are planning a very large Asterisk deployment, using Wifi SIP phones.
We've done installs using Spectralink and the SVP to manage congestion at
the access points, but we have a client that doesn't want Spectralinks.
Anyone have experience with an alternative congestion management (AP
: Re: [asterisk-users] Treating T1 as trunk in/out, not individual
lines
Michelle Dupuis wrote:
I'm tying a Nortel option 61 to asterisk via T1. I don't want to split each
of the t1 channels out into individual lines (tied to a specific extension)
- so a trunk in and out.
Assuming PRI over T1
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Treating T1 as trunk in/out,
not individual lines
Michelle Dupuis wrote:
Ok..so how would the CALLED and CALLERID ID be presented to
Asterisk
when using PRI signaling.
When you are using a PRI, you'll see
I've got a few Cisco 7921 wifi phones to use with an Asterisk pilot. For
the purpose of the pilot (i.e. low investment) I want to configure the
phones from the keypad.
Each phone shows settings locked! whenever I try to edit the network
profiles. I can't seem to unlock them! Hopefully there
That did the trick! It appears that all of the config is retrieved from a
.cnf.xml file, so there wasn't much more I could do at the phone level other
than set the networking parameters.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Yehavi
I have a new Sangoma A102 and I'm trying to get it running in asterisk. A
look through the dmesg log shows the card is detected and the various
channels created. However, when I start asterisk I get the error below.
Any ideas?
My zapata.conf is below.
Thanks,
MD
== Registered custom
I'm connecting a T1 PCI card to a Nortel Option 61 switch T1 card. My
Sangoma A102D shipped with 2 T1 cables - which I assume are straight
through. Do I need to make crossover cables for this scenario?
Thanks
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I have a new asterisk system with a T1 card. It appears that running ztcfg
-vv is required in order for asterisk to start properly.
Is this correct? Are people adding this command to the asterisk startup
script?
Thanks
___
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I'm tying a Nortel option 61 to asterisk via T1. I don't want to split each
of the t1 channels out into individual lines (tied to a specific extension)
- so a trunk in and out.
Assuming PRI over T1 signaling, how would I pass the CALLED and CALLER info
across the channels so each side knows
List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get sangoma A102D setup
on asterisk
On Fri, Oct 26, 2007 at 04:00:30PM -0400, Michelle Dupuis wrote:
I have a new Sangoma A102 and I'm trying to get it running in
asterisk. A look through the dmesg log shows the card
Huff [mailto:[EMAIL PROTECTED]
Sent: Saturday, September 08, 2007 11:34 AM
To: Michelle Dupuis
Subject: Re: [mythtv-users] Real Time Clock Alarm Broken with 2.6.22+ kernel
On 9/7/07, Michelle Dupuis [EMAIL PROTECTED] wrote:
Craig,
I wrote an nvram-wakeup replacement call acpi-wakeup
I have to connect a Meridian S1 to Asterisk for a slow migration to VoIP.
What is the best way to connect them?
1. Is a T1 the best solution?
2. Can I pass Caller and Callee information across the link?
If a T1 is best, I recall a standalone T1-SIP device a year ago on this
list. Does anyone
Over a year ago I saw a discussion about a standalone device which converted
a T1 in/out to SIP in/out (over 10/100 LAN). Anyone recall what this device
is?
(I'm looking for a standalone device - not a PCI card).
Thanks
___
Sign up now for AstriCon
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Meridian S1 to Asterisk via T1
Michelle,
On Fri, 7 Sep 2007, Michelle Dupuis wrote:
I have to connect a Meridian S1 to Asterisk for a slow
migration to VoIP.
What is the best way to connect them
Anyone know if Asterisk offers SVP support (Spectralink protocol)
-MD-
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- Non-Commercial Discussion
Subject: Re: [asterisk-users] Best wifi IP phone for asterisk
Michelle Dupuis wrote:
We're looking at a large wifi phone deployment, and we're looking for
wifi phones that:
1. Are SIP compliant (Asterisk friendly) 2. Provision capable (ideally
TFTP of own MAC address
Same here. We have commercial call center clients on Unlimitel. They've
had a few outages during business hours, but Unlimitel is responsive.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dr. Michael J.
Chudobiak
Sent: Monday, July 02, 2007 5:29
: [asterisk-users] Best wifi IP phone for asterisk
On Tue, 26 Jun 2007, Hendrik Visage wrote:
On 6/25/07, Michelle Dupuis [EMAIL PROTECTED] wrote:
We're looking at a large wifi phone deployment, and we're looking
for wifi phones that:
HAve a look at the Linksys WIP 300 (or something) Can
Does anyone know if Asterisk can natively support the SVP protocol from
SpectraLink?
Thanks,
MD
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I can't find reference to TFTP for provisioning - does this phone support
it?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marcus Franke
Sent: Monday, June 25, 2007 10:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
We're contemplating Linksys WIP 330 phones, but we're concerned about
configuration effort. Does anyone have the file format for an XML file to
configure this phone? We got auto-provisioning to D/L a file, but the XML
file format seems to be a secret...
Thanks,
Michelle
We're looking at a large wifi phone deployment, and we're looking for wifi
phones that:
1. Are SIP compliant (Asterisk friendly)
2. Provision capable (ideally TFTP of own MAC address)
3. Industrial quality (no cheap plastic stuff).
4. Well documented (and none of the only telco's get
Have a look at poptop (PPTP server) - pretty straight forward. Great if you
have Windows clients.
If you have Linux clients (or want a permanent tunnel), there are other
options.
Michelle Dupuis
Technical Support Specialist
Generation Software - Linux and Asterisk solutions and support. Visit
The IAXMODEM might get you half way there...but if you want to connected it
to a windows box (which I assume is why you use the RAS acronym), you'll
have to look for remote serial port software.
-MD-
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
How about forking the process when the AGI launches, and pass the PID back
to Asterisk in a variable. When the call ends (caught at the h), call
another AGI script to kill/stop that pid.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
I've reposted with a more meaningful subject - hopefully someone will
replyWe have an Asterisk v1.2.16 box registering with an ITSP using SIP.
The registration succeeds, and is confirmed with SIP SHOW REGISTER.
However, we frequently (every few minutes) see this on our console:
REGISTER
Ast 1.4 will pass through T.38, but not terminate/originate T38. Be sure
you understand the implications for your fax termination
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cesar Benjamin
Garcia Martinez
Sent: Saturday, May 05, 2007 3:26 PM
We have an Asterisk v1.2.16 box registering with an ITSP using SIP. The
registration succeeds, and is confirmed with SIP SHOW REGISTER. However,
we frequently (every few minutes) see this on our console:
REGISTER attempt 1 to [EMAIL PROTECTED]
REGISTER attempt 2 to [EMAIL PROTECTED]
Any
installation with configuration. Without ascii
config files (or a tool from the mfg to create binary config files from a
script), each soft device must manually configured.
Michelle Dupuis
Technical Support Specialist
Generation Software - Linux and Asterisk solutions and support. Visit us
Look at a channel bank with MGCP/TDMoE/USB interface (there are plenty
around). We too prefer to keep fxs/fxo hardware outside of the * box.
Michelle Dupuis
Technical Support Specialist
Generation Software - Linux and Asterisk solutions and support. Visit us at
www.generationd.com
You don't need the cfg files (or a tftp) to boot the phones or register.
There are some sample configs lying around, but Aastra's are very poorly
documented (and their firmware still has big bugs - so don't modify from
default too much). We've setup a number of 480i's and got very frustrated
with
Start with a codec check (sounds like the CNG tone frequencies are out of
spec)...
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Wulter
Sent: Wednesday, March 28, 2007 4:58 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
Asterisk isn't a simple apt-get and run type program...have a look at the
asterisk wiki for help getting started. There's a lot to configure
MD
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josu Lazkano
Lete
Sent: Tuesday, March 27, 2007 11:20 AM
To:
If something changed on its own 6 hours ago (i.e. you didn't touch
Asterisk or its config), then look beyond Asterisk. Did you do a hard
reboot yet? Memory check? Reseat PCI cards and memory? Check power
supply? New drivers/software loaded? (Did YUM run in the background?)
If everything
We regularly install * on Fedora (clients with lots of leading edge hardware
like Fedora). No problems
I expect you will only encounter * 1.4.x errors like everyone else.
Michelle Dupuis
Technical Support Specialist
Generation Software - Linux and Asterisk solutions and support. Visit
We installed a quad xeon 3ghz which transcoded ~100 active channels (as a
gateway). Take a look at the codec demands (in asterisk show codecs I
believe) and scale from there. This box was 60% loaded - which is all we're
comfortable with before latency goes too high.
Michelle Dupuis
Technical
Can one do an in-place update to 1.4.1 from 1.4.0 ? (Just compile and
install)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
Development Team
Sent: Friday, March 02, 2007 7:04 PM
To: undisclosed-recipients:
Subject: [asterisk-users] Asterisk
For an all electronic solution, use fax2mail and mail2fax (from
www.generationd.com). For a fancier all VOIP solution consider hylafax.
For analog only you can plug your fax machine in as you suggest. For a step
up, buy an ATA with T.38 capability and plug your fax machine into that.
MD
You will likely have latency issues - causing choppiness. Start with a
traceroute to validate latency.
Michelle Dupuis
Technical Support Specialist
Generation Software - Linux and Asterisk solutions and support
Visit us at www.generationd.com
-Original Message-
From: [EMAIL
frequencies and what's left? If
there's a lot of noise, then the other party is doing a bad job encoding the
DTMF. Otherwise we can start to chase your machine causes
Michelle Dupuis
Technical Support Specialist
Generation Software - Linux and Asterisk solutions and support. Visit us
T.38 won't work over the H.323 leg of your call (even with Open H.323),
chan_h323 won't support it.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Octavarium
Sent: Tuesday, February 27, 2007 12:51 PM
To: asterisk-users@lists.digium.com
Subject:
Have you tried starting Linux with irqpoll / noapic? Sounds like a BIOS
bug..
MD
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marco
Parisotto
Sent: Tuesday, February 27, 2007 3:27 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] TE212P on FC6 -
List - Non-Commercial Discussion
Subject: Re: [asterisk-users] H323-to-SIP proxy
What about the SIP leg?
- Mensaje Original -
De: Michelle Dupuis [EMAIL PROTECTED]
Para: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Enviados: martes 27 de febrero de
Isn't there a zap dummy (or something that uses the RTC) included in
Asterisk 1.40 that creates the timing source? We don't install any external
timing sources and we don't have choppyness problems on pure sip
connections...
Jason - is this on a standard PC motherboard (or a mini device like
I have a high volume asterisk 1.40 installation and I ran a SIP SHOW
CHANNELS. (see partial output below). My questions are:
1. wc-l of the output shows 4000 lines. Does this mean 2000 active calls?
(2 channels per call)
2. The latter part of the output shows unkn for Form column. Why does
My asterisk install is showing the following every 1/2 second:
chan_sip.c:2739 auto_congest: Auto-congesting SIP/my.domain.net-0e118a90
There are lots of calls going through.
1. What can I do about this?
2. Is there a way to limit the number of calls (responding to invites with
no
Have you tried SMSSEND? It's open source, available as RPM.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michiel van
Baak
Sent: Sunday, February 25, 2007 6:03 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Sending SMSa
On
We have a * box with sip in, and h.323 out. When the H.323 call setup is
underway, will Asterisk translate the progress/status/result codes to SIP
automatically?
Ordo we have create our own result codes in SIP headers?
Thanks,
MD
___
I'm working on calls coming in to an asterisk box as H.323, and going out as
SIP to a remote device (a VoiceMaster). The remote device is refusing the
calls with SIP error 406 (Not Acceptable).
I have attached the SIP debug output below. It looks like codecs overlaps -
can anyone see why the
I don't think Asterisk plays a role in this (unless I'm missing your point).
A simply script to ping your server room will do. Upon failure, the script
could initiate a PPP connection outbound.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
I'm looking at setting up an asterisk box dedicated to SIP-H323 conversion
(a 3rd party is currently converting the protocols for us).
1. Is it worthwhile to split this functionality onto a second server? Or
should we let the ast pbx handle the conversion? (we have a couple hundred
active
@lists.digium.com
Subject: Re: [asterisk-users] Mini-ITX board + FXO PCI card?
At 10:09 11/02/2007 -0500, Michelle Dupuis Henderson wrote:
We use a lot of mini-itx pc's, including the pCI slot. I don't think
any of the systems have shared an irq with the PCI slot
Thanks for the tip. In that case
We use a lot of mini-itx pc's, including the pCI slot. I don't think any of
the systems have shared an irq with the PCI slot
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vincent
Delporte
Sent: Tuesday, February 13, 2007 5:29 PM
To:
If the PSTN side is only complaining about conversations with a single phone
on the SIP side, look at the SIP phone.
Check the settings for that SIP phone/PC (VAD disabled, NIC settings,
runaway processes). Do PSTN callers here choppiness from the SIP phone
caller?
-Original Message-
on SIP
Well, the PSTN side is complaining about a random phone on the SIP side.
Yes, they do hear choppiness.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michelle
Dupuis
Sent: Monday, February 12, 2007 2:49 PM
To: 'Asterisk Users Mailing List - Non
you how
it's set up there, but the network technicians said it is enabled.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michelle
Dupuis
Sent: Monday, February 12, 2007 3:19 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE
I would suggest you grab the menu from the .conf file and paste it into the
new setup. (After even a little asterisk experience, they should be able to
get away from the gui).
The sound files could be copied as well. I'm guessing from your question
that you/your client may not having Linux
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