On 7/16/07, Ronald Wiplinger [EMAIL PROTECTED] wrote:
1. Instead of using *1 (automon) I need to record each phone call at a
certain * box.
exten =
_1NXXNXX,1,MixMonitor(/var/spool/asterisk/monitor/${CALLERIDNUM}-${EPOCH}-${EXTEN}.wav)
exten = _1NXXNXX,2,Dial(Zap/R1/${EXTEN},90
Along the same lines -- Anyone know where I can get/extract the default
music on hold file from?
On 7/16/07, Cory Andrews [EMAIL PROTECTED] wrote:
Off topic, but involves an Asterisk deployment in a roundabout way.
Anyone here intimately familiar with Cisco Callmanager (Version 4-5), that
All you guys whining about delays for that past month: Ever thought the
problem is YOUR mail server? I have no problems at all. Messages arrive on
time.
On 7/16/07, Bill Maidment [EMAIL PROTECTED] wrote:
On Tue, 10 Jul 2007 13:15:20 -0500, The Asterisk Development Team wrote
The Asterisk
On 7/11/07, Jakub Głazik [EMAIL PROTECTED] wrote:
Asterisk [EMAIL PROTECTED]
Client hears pure silence when waiting for call answer. Music on hold
stops
when transferer pics a number and client doesn't even hear ringing.
Is this normal behaviour? How to change this?
Log says everything, MOH
I highly recommend the Sangoma cards. They have good support for Asterisk
also for other systems as well :) Asterisk does support Q.SIG that is not an
issue.
On 7/5/07, satish patel [EMAIL PROTECTED] wrote:
Dear all
I have asterisk 1.2 and now i want to install E1 card with
ODM is the same as WIP300. Probably the same phone as the D-Link Dual mode.
On 7/3/07, Ron Arts [EMAIL PROTECTED] wrote:
You might want to look at the Pirelli Dual Mode DP-L10.
I tested one, and sound quality and stability are much better
than the Nokia E61 or of any other wiFi phone I
The Proposed bill S704 reads It shall be unlawful for any person within the
United States, in connection with any telecommunications service or
IP-enabled voice service, to cause any caller identification service to
transmit misleading or inaccurate caller identification information,
Please tell
Replace with below. Actually Asterisk should only generate ringback when the
SIP phone is ringing.
On 6/25/07, satish patel [EMAIL PROTECTED] wrote:
exten = 222,1,Dial(SIP/222,r)
exten = 333,1,Dial(SIP/333,r)
exten = 555,1,Dial(SIP/555,r)
exten = 100,1,Dial(SIP/100,r)
exten =
Yes. I have so
On 6/25/07, Nick Seraphin [EMAIL PROTECTED] wrote:
Is this strictly a European phone? I can't find anyone who is selling
them in the US... at least not a company I've ever heard of or dealt with
before.
Tried Amazon.com, voipsupply.com, Tech Data, and 3 pages of Google
You might want to call:
302.338.9601
On 6/20/07, Dean Collins [EMAIL PROTECTED] wrote:
Hi Brian,
Trying to get in touch, please call or email
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).
On 5/31/07, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Thu, May 31, 2007 at 01:22:06PM -0700, Douglas Garstang wrote:
I'm trying to find a high port count ATA device. We have a lot (up to 110)
analog devices that we need to bridge to IP. Rather than go out and buy 110
ATA's, it
On 5/28/07, shadowym [EMAIL PROTECTED] wrote:
Thanks for all the replies. Seems there are at least 2 or 3 people giving
strong recommendations to iaxmodem/hylafax as a reliable (ie. business grade
production) solution. That is just the sort of feedback I was looking for.
My application is
On 5/24/07, shadowym [EMAIL PROTECTED] wrote:
So what is the bottom line? Does it work or not. I've heard stories it
works, it doesn't work, it kinda sorta works when it's not raining out side.
Everything under the rainbow.
What's the bottom line with recent updates on 1.2.x? Is it
spam running PRI the load on th e
machine shoots up to a crazy amount with Linux 2.6
I don't like to over-post but I just had to bring this one from the
archives to your attention:
On 4/11/07, Andrew Joakimsen [EMAIL PROTECTED] wrote:
I would suggest you avoid TDMoE its support is pretty
HP has a tool that is a free Open Source test tool / traffic generator
for the SIP protocol.
On 5/26/07, khawla khawla [EMAIL PROTECTED] wrote:
I am using Aserisk as a SIP server to interconnect differents PBX in
differents sites. I am now looking for a tool that can test the performance
of
On 5/9/07, Ritesh Agrawal [EMAIL PROTECTED] wrote:
Hi Folks,
Is there a way to find out the mobile/landline carrier name based on the
phone number?
For example, who is the mobile carrier for (415)2345678
I had heard about some query but just don't remember how/what?
Thanks in advance.
Ritesh
Malcom:
Great to know there are more loyal SuSE users like myself!
After you install your kernel source have you tried:
# cd /usr/src/linux
# make cloneconfig
# make prepare-all
The problem is SuSE does not provide the kernel headers, you need to
create them yourself.
Of course this assumes
I would suggest you at least look into DIALSTATUS.
Guys:
The dialplan is not a toy. You need to consider with care the results
of your actions. What Mike posted is an example of a bad dialplan in
the works.
On 5/11/07, Mike [EMAIL PROTECTED] wrote:
Hi,
I have a question of using 2 SIP
On 4/28/07, Noah Miller [EMAIL PROTECTED] wrote:
Nope. You can use a device like the Redfone fonebridge to convert the
PRI to TDMoE. Possible Downside: I've read some reports that say the
TDMoE module in asterisk is not so stable.
I highly suggest you don't use TDMoE. Especially not if you
On 4/27/07, Per Jessen [EMAIL PROTECTED] wrote:
Try your local mobile phone supplier. I used a headset that came with
one of my cell phones, and it worked great w/ my SPA-941.
Not a bad idea - which make was this for? None of my phones (Ericsson,
Nokia) have a 2.5mm socket, they're all
1) dont use MGCP -- SIP is better supported
2) Don't use Audiocodes, they blatantly ignore the GPL license.
On 4/19/07, J. David Bavousett [EMAIL PROTECTED] wrote:
Greetings;
We are trying to get Asterisk up and happy at our site-we tried VOIP
using Sphere about a year ago, spent a *boodle*
On 4/18/07, Gilles Ganault [EMAIL PROTECTED] wrote:
Hello
I'm about to order a GrandStream GXP-2000 and a Linksys SPA-921
I'd like to have some user feedback about how those phones perform, and
whether their LCD screen displays both the caller ID name and number (The
GrandStream BT-100 only
If that's what your phone is setup. Are you even using a SIP phone?
What does the PEER context contain?
Also, while Asterisk 1.2 and CALL WEAVER are basically the same
(besides that fact that CALL WEAVER is trying to fully support faxing
and Asterisk/Digium refuse to support correctly faxing)
I'd be glad to test the software, however I'm sure you'll find that
many people would be unwilling to provide their CDR (especially large
ones) because chances are it would contain alot of
personal/unidentifiable information.
On 4/17/07, Lenz [EMAIL PROTECTED] wrote:
Hello list,
I have been
Not that custom shouldn't work, but you just need to place them in
sounds/digits/custom not sounds/custom
On 4/15/07, Hermann Wecke [EMAIL PROTECTED] wrote:
Julian Lyndon-Smith wrote:
however, I get no errors, but still get the default Allison sounds
for the digits. Anyone got any clues on
On 4/15/07, Germán Rodríguez Vergara [EMAIL PROTECTED] wrote:
I'm looking for Optipoint 420 Standard SIP Firmware to make my first tests
with Asterisk and SIP, but I'm unable to find it. Could someone help me?
My understanding is Siemens sells an optiPoint 420 and an optiPoint
420S. The 420S
On 4/12/07, Mike [EMAIL PROTECTED] wrote:
Apr 12 23:06:16 WARNING[25593]: chan_sip.c:7036 check_via: '67.39.117.611'
is not a valid host
Using 1.2.13.
What does your sip show peer notarius-phone-1 print? Are you using
the xml configs on that Polycom?
I would suggest you avoid TDMoE its support is pretty much depreciated
and not supported by Digium. Does not work well with Kernel 2.6.x.
On 4/9/07, Michelle Dupuis [EMAIL PROTECTED] wrote:
Look at a channel bank with MGCP/TDMoE/USB interface (there are plenty
around). We too prefer to keep
Take a look at these
http://www.telephonydepot.com/product_p/105-056-4104.htm
http://www.telephonydepot.com/ProductDetails.asp?ProductCode=105%2D056%2D4108
I would suggest you avoid the AudioCodes units, AudioCodes blatantly
ignores the GPL and refuses to release even their kernel source.
On
How is the screen compared to the other Polycom products?
On 4/9/07, Jessee J Holmes [EMAIL PROTECTED] wrote:
Mike,
I don't have much information, except they are due for shipment soon (mid to
end of April to distribution from Polycom). We've demoed a couple and I
personally believe they'll be
On 4/11/07, Forrest Beck [EMAIL PROTECTED] wrote:
2) Have two servers with the same dialplan. One in each location.
Each server has it's own TDM cards installed. Phones on Site A will
register with the server on Site A, and phones on Site B will register
with the server on Site B. Then using
*zapata.conf
switchtype=qsig
On 4/10/07, George C. Attopany [EMAIL PROTECTED] wrote:
Hello,
Anyone to help with information on configuring Q.SIG in Asterisk ?
I run ASTERISK 1.4 with a Wildcard TE410P-Xilinx on Fedora Core 6
and Zaptel 1.4
I need to tie this ASTERISK system to a
On 4/5/07, Mike Lynchfield [EMAIL PROTECTED] wrote:
tried x+102 ?
NEVER do that. The call can fail for other reasons besides the
carrier. It can and will create conditions where your carrier properly
connects the call and then the call is re-dialed via another provider
or line.
On 4/6/07, Joe Acquisto [EMAIL PROTECTED] wrote:
While pondering several issues, poor quality PSTN POTS lines, potential cost
savings with multiple cell numbers, the FAX problems over TDM400p, etc, I
wondered about:
Poor quality POTS lines and fax problems do seem to be related. The
added
On 4/6/07, Joe Acquisto [EMAIL PROTECTED] wrote:
I find the source. IIRC, the problem was there was some dependency
(terminology ?), or package that was missing and I got complaints when trying
to compile. I can't recall what it was, but it is something that is included
in most distro's
On 4/5/07, Rob Schall [EMAIL PROTECTED] wrote:
Hey all,
I had a user complaining of calls which were dropping mid-conversation.
I looked into the time of one of the calls, and saw the following:
Apr 4 12:13:03 WARNING[6670] chan_zap.c: No D-channels available!
Using Primary channel 28 as
On 4/5/07, Rob Schall [EMAIL PROTECTED] wrote:
Yes, the connection seems solid and the cable is alright.
That doesn't mean the cable is the proper specification. Most people
use category 5 unshielded twisted pair which technically is not the
correct cable. While this is probably not the
Well I would wonder how Polycom even had any idea whom your vendor is.
On 4/2/07, Stephen Bosch [EMAIL PROTECTED] wrote:
Andrew Joakimsen wrote:
First-sale doctrine, unless your vendor did something illicit to
obtain Polycom phones there is nothing they can do about it.
What they can do
VGPS is a PHP/MySQL based provisioning system intended to generate
vendor-specific configuration files for Voice-over-IP (VoIP) devices
via a generic HTTP API.
Good luck...
http://sourceforge.net/projects/vgps
On 4/4/07, Forrest Beck [EMAIL PROTECTED] wrote:
I know this doesn't belong on this
On 4/4/07, John Schmerold [EMAIL PROTECTED] wrote:
What is that tone called where is stored and configured.
I'd like to replace the ring with an announcement that is played until
the call is picked up or put into voicemail.
The ring is called ring and is defined in indications.conf.
For build quality and footprint, the SPA-942s are hard to fault, but the web
interface isn't the most pleasant and remote provisioning is hit-and-miss (mainly
dependent on how cooperative your supplier is at getting you the provisioning
documentation/software).
As for Linksys, their
Bosch [EMAIL PROTECTED] wrote:
Andrew Joakimsen wrote:
Strange that I cant recall any other device that Asterisk was not
working with. So it seems that Polycom is not in compliance with
RFC2543. That and Polycom has been known to flat out refuse support if
you mention the word Asterisk
That's
And also you should post it on the voip-info wiki there is a page just
with bounties.
On 3/31/07, Edoardo Serra [EMAIL PROTECTED] wrote:
Salvatore Giudice ha scritto:
You could put a bounty on this. You may find someone who will be willing to
write this for money.
My Bounty for that feature
The logic of the macro is totally opposite of what it should be. I do
recall sending a corrected version of the script to someone a while
back, it might be on the mailing lists archive.
However, there is an option for the Dial() command to do exactly what you wish
p: This option enables
On 3/30/07, Gergo Csibra [EMAIL PROTECTED] wrote:
Friday, March 30, 2007, 5:02:08 AM, Matt wrote:
Wehh...
He activated the DND function of Linksys. It can be activate with *78
and deactivate with *79.
No because if that was a case his sip trace would show something along
the lines of 486
So does the P option in app_dial seems to me the easiest way to
implement is just hack app_dial so it wont prompt to record the name.
On 3/31/07, BJ Weschke [EMAIL PROTECTED] wrote:
On 3/31/07, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Peder @ NetworkOblivion wrote:
I also had a question
You need to setup a DNS SRV record.
On 3/28/07, Paolo Supino [EMAIL PROTECTED] wrote:
Hi
We've setup an Asterisk PBX recently and I encountered the following
problem: When [mac address]-registration.cfg file includes the FQDN of
the Asterisk PBX for the Polycom SoundPoint 501 phones it will
So does Grandstream (GXP-2000) and I am am not mistaken Aastra (480i) as well.
On 3/28/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Maxim Veksler wrote:
Thank you Rob for the detailed reply.
It solves one side of the problem (In a very cool and unexpected way I
must admit) but not
No, what happens is the SIP client has silence suppression enabled.
MOH works by symmetrical RTP. One frame of voice is received one frame
of MOH is sent. So if the client supresses silence frames there is no
MOH played during that time.
ALso if you see:
started music on hold
stopped music on
Strange that I cant recall any other device that Asterisk was not
working with. So it seems that Polycom is not in compliance with
RFC2543. That and Polycom has been known to flat out refuse support if
you mention the word Asterisk
On 3/28/07, Mike Hammett [EMAIL PROTECTED] wrote:
I was
In that respect I'd rather recommend the Linksys WIP300. After initial
frustration it does work great. Roaming actually works pretty well
with perhaps 250ms of silence and or distortion if you have really
good overlapping coverage. People like to blame WiFi for poor
scalability. No doubt it's a
;--
; Definitions of locally connected SIP phones
;
; type = user a device that authenticates to us by from field to place calls
; type = peer a device we place calls to or that calls us and we match by host
; type =
It would make more sense if you posted the musiconhold.conf file and
stated if you did or didn't install the asterisk_addons package with
mp3 support.
On 3/27/07, Klaverstyn, David C [EMAIL PROTECTED] wrote:
Hi All,
I have installed Asterisk 1.4.2 and have loaded ztdummy as I have no
Actually I think the additional wanpipe drivers are a major plus. For
PRI troubleshooting I even think there are tools that are not present
in Zaptel, or are much harder to setup (vs wanpipemon -g).
Commitment to multiple protocols, applications and platforms is also
another plus. Sangoma
This is the simplest solution I can think of:
http://www.smarthome.com/5070cw.html
On 3/26/07, Jay Milk [EMAIL PROTECTED] wrote:
Steve Totaro wrote:
Just get a Grandstream ATA and a handset with no buttons. So simple.
That doesn't really meet my needs -- I want to be able to dial-out, and
On 3/26/07, Olivier [EMAIL PROTECTED] wrote:
Hello,
1. Is it possible to install several SIP softphones on the same PC,
Yes. You can even install for example two softphones for Windows, two
for Linux and two for MacOSX (two is an imaginary number you can have
six on Windows, 27 on Linux and
Linksys will be glad to not provide you with any provisioning at all.
There is a program required to encode the config files, cant prove you
are vonage, too bad no file for you.
On 3/22/07, Chris Bagnall [EMAIL PROTECTED] wrote:
Greetings list,
Does anyone have any experiences they'd like to
Next generation bot nets -- forget about spam, telemarketing is the
next viral (literally) marketing concept!
On 3/20/07, Jon Pounder [EMAIL PROTECTED] wrote:
Quoting C F [EMAIL PROTECTED]:
Awesome, the first PABX virus is just around the corner now that M$ has some
bait for it to infect.
On 3/19/07, Brad Sumrall [EMAIL PROTECTED] wrote:
...music on hold...
Brad
Music on hold support is present, you can also add MP3 support with
asterisk-addons package, are you using MP3 without the correct format
installed? www.asterisk.org and download the add-ons package, read the
docs
I really have lost loads of faith for IAX. No authority found and
Rejected connect attempt messages for no apparent reason. Sometimes
computability issues between asterisk versions. Fax/T.38 support?? But
I have no complaints about when it actually does work.
Not that Asterisk has the best SIP
Like someone else said, go for either really big (4U) or also really
small (fanless ITX). 4U rackmount with a fanless PSU, 120mm fans can
flow a decent amount of air and be nearly silent, you just need to
shop around a bit. A fanless ITX with a compactflash card is great for
a small
There are two ways that you can implement reliable faxing:
1) Implement end-to-end QoS and insure that latency is very low and
there is no jitter. Should be 95% reliable, more or less depending on
the quality of the link. Most people don't have the ability. It's not
possible with ADSL, not
SIP or IAX? What are the relevant configs?
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I think the best we can hope for is a stable release of OpenPBX. T38
gateway there is not a single report of it working, this is the holy
grail of T38 faxing. I don't know if perhaps it is only designed for
T38 to Zap only? Txfax and rxfax won't even compile with the newer
asterisk 1.2 and newest
Yes, you can setup * key to do that, its a standard feature see the
docs of the voicemail application for details on how to do it.
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Or that its not even a new service?
On 3/5/07, Bruce Reeves [EMAIL PROTECTED] wrote:
Or the fact that www.virtualphoneline.com is part of DIDXchange and of
course you love it since you work for supertec.com, didxchange.com,
and virtualphoneline.com
On 3/5/07, Singer Wang [EMAIL PROTECTED]
I'll consider the offer if it includes your code being included with Asterisk.
On 3/3/07, Steve Underwood [EMAIL PROTECTED] wrote:
I'll do it for 30% less than they quote. :-)
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It is my understanding that pickup does not work across channel
technologies and Sipura does not support IAX. You can use the pickup()
application however.
On 3/1/07, Joseph [EMAIL PROTECTED] wrote:
How to configure Digium S101i adapter to work with pickupexten *8 ?
1. I want my users to dial certain number.
2. Record a voicemail with destination number.
No problems so far
3. Convert this Voicemail to Text.
Save the VM in database, but you need to build the speech-to-text
system somehow. The database will make the external application easier
to access
There is also an Ethernet/SIP overhead speaker. Voipsupply sells it.
On 3/2/07, Stefano Totaro [EMAIL PROTECTED] wrote:
Quite surprising, yes! :-)
I am from north east Italy, now I live in Verona (Romeo and Juliet's city
:).
I cannot do it connecting amp to the PBX. I have quite a long
No. Asterisk does not have fax support and there are no plans to add it.
You can however send the faxes as voice calls, however there is no
assurance as to its reliability. Maybe most will work but some will
without a doubt be predestine to fail.
On 3/2/07, --[ UxBoD ]-- [EMAIL PROTECTED]
No there is no fax support in Asterisk.
On 3/2/07, Khaled [EMAIL PROTECTED] wrote:
Dears
Any one know how to let t38 works on asterisk 1.2 or an distribution like
trixbox have asterisk 1.4
Regards
Khaled Chehab
System Integration Engineer
Xplorium Offshore.
Sakiet Al Janzir
If the remote end isnt up that could explain why the correct settings
dont work. If they told you to use those settings thats how it should
be provisioned, either:
1) You think your configuration is correct but you are wrong and it
indeed is incorrect
2) Your telco didn't correctly provision or
That's like saying a pinto is fast when you upgrade the engine. Well
Asterisk also supports T.38 for free... if you backport OpenPBX.org
fixes.
But realisticly ASTERISK DOES NOT HAVE FAX SUPPORT STOP CLUTTERING MY
INBOX WITH DISCUSSION OF FEATURE THAT DOES NOT EXSIST.
On 3/1/07, Zoa [EMAIL
Maybe.. if you dont expect to recieve calls to any device, then I just
wouldnt bother to register.
On 2/28/07, Ricardo Carvalho [EMAIL PROTECTED] wrote:
Can't I register multiple phones with the same user/password? That's
what I pretend to do, not ring groups...
Thanks,
Ricardo.
Audiocodes blatently violates the GPL... dont use their gear.
On 11 Feb 2007 19:11:51 -, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
I have 2 identical AudioCodes MP-112s. They have the same config except for
the SIP usernames/passwords and the device IP. The configs in extension.conf
and
What is the difference between using my proprietary asterisk-add on
than to using my proprietary email client (Microsoft Outlook) with my
GPL IMAP servers? You guys need to drop your BS elitist point of view,
It isn't your software, its talking to your software like any other
software does, the
Correct, if you want do do *8 + exten you need to use the dialplan and
the Pickup() applicaiton.
On 2/9/07, John Breen [EMAIL PROTECTED] wrote:
Ken Williams wrote:
i have two problems with my Grandstream GXP2000 :
1- When i wan pickup a call, that's don't work's (*8EXTEN)
and when
Asterisk 1.2 has no support of t.38 whatsoever, the call will drop
before t.38 is ever utilised, not even pass-thru.
1.4 Adds support for T.38 pass through only and no other sort of
faxing, the endpoint must support T.38 and you must send your call to
a T.38 gateway and you must not use NAT
AudioCodes is known to violate the GPL and not care at all about it.
On 12/13/06, Mike Clark [EMAIL PROTECTED] wrote:
Anyone have any experience with the Audiocodes MediaPack MP-118? We are
looking at options for a location that wishes to maintain 6 - 8 existing
analog phones in add
Perhaps you can write the functionality? I'm sure you can do a quick
hack of you modify app_voicechangedial.
On 2/2/07, François Delawarde [EMAIL PROTECTED] wrote:
Don't you think it could be an interesting feature in Asterisk? It
already does transcoding, why not gain when voice flow passes
, please let me know.
Andrew Joakimsen wrote:
Ok, I actually GOT a GXP-2000. It does not have a dialplan. You
cannnot dial without the handset off-hook. I do not seem to find a way
to use early dial for international calls in a practical way, not
being able to dial international calls
take a look at http://spc.pifiu.com
On 1/2/07, Erick Perez [EMAIL PROTECTED] wrote:
Hi, Anyone knows where to get the admin (not the end user) manual for
the linksys spa2102. This model is the 2 analog port+router.
There are a lot of advanced options that I would like to see what they do.
I know of the call pickup issues but what asterisk issue and what BLF issue?
On 1/25/07, Alberto Pastore [EMAIL PROTECTED] wrote:
Andrew Joakimsen ha scritto:
Actually I noticed just three days ago there is a new release, and the
releae notes seem to address
Disable TrMail and Pickup keys
-- Forwarded message --
From: Jeremy McNamara [EMAIL PROTECTED]
Date: Jan 22, 2007 5:22 PM
Subject: Hater
To: [EMAIL PROTECTED]
Continue to Hate on NuFone - Every time you post something people
already know that you don't have a clue.
Dial(ZAP/g1/18005551212,90,m) m will play music on hold while the call rings.
On 1/22/07, Darren Nay [EMAIL PROTECTED] wrote:
Hey All,
Is there an app available, or another method, to stream an audio file to a
caller while performing additional actions in the background? Regardless of
! after each time the call is dialed, yet it
still connects! Where is Thomson, they seemed to try to keep up for a
while...
On 12/21/06, Alberto Pastore [EMAIL PROTECTED] wrote:
Andrew Joakimsen ha scritto:
I too am wondering if someone has a contact at Thomson, some of the
softkeys need to either
Actually I noticed just three days ago there is a new release, and the
releae notes seem to address
Disable TrMail and Pickup keys
Disable call progress indication
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OpenPBX.org has better support, due to license issues and politial
bullshit I don't see Asterisk getting T.38 support that isnt a joke
(codec pass-thru?? LOL) for a long time. OpenPBX should have a stable
release within the month, if I am not mistaken they have a Release
Candiate #2 right now
What G723 codec do you have on Asterisk? What is your SIP.CONF? What
ATA/Phone is being used and what are the exact settings, especially
for the codec?
On 1/19/07, Phil French [EMAIL PROTECTED] wrote:
I am setting up Asterisk for use in a low bandwidth environment. As
bandwidth is precious and
Assuming your PRI supports timing from the remote end (CO) which I
highly suspect is the case, then you should set the asterisk machine
to be a slave to the CO timing and then set any other interfaces you
have to NOT be masters, so that the CO timing is always used. Assuming
you do this and
Perhaps someone could help you... if they actually had any knowledge
as to what your configuration is, which I doubt they do.
On 1/19/07, Eric Bishop [EMAIL PROTECTED] wrote:
On inbound calls from my SIP provider I get multiple warnings as follows:
WARNING[5351]: chan_sip.c:7086 check_via:
http://www.google.com/search?q=423+%22Interval+Too+Brief%22start=0ie=utf-8oe=utf-8client=firefox-arls=org.mozilla:en-US:official
Hint: Who develops Asterisk?
On 1/20/07, Thomas Madler [EMAIL PROTECTED] wrote:
Hi,
I'm trying to get my * server connected to a softswitch through an SBC. I
get
That's a great site! Perhaps it should be auto-sent to every poster
for the first 30 days of their membership:
Despite this, hackers have a reputation for meeting simple questions
with what looks like hostility or arrogance. It sometimes looks like
we're reflexively rude to newbies and the
I've actually found in many cases a lower bandwidth codec doesn't
improve at all and however it oftentimes makes the issue worse.
On 1/19/07, Martin Joseph [EMAIL PROTECTED] wrote:
On 2007-01-17 10:29:43 -0800, Yelson Vivas [EMAIL PROTECTED] said:
Hi Guys
I'm conecting 2 astersk servers
number is in there because on
mobile terminals, both the redirect and the ani are identified.
Anyone out there had any luck with the RDNIS before?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Andrew Joakimsen
Sent: Friday, January 19, 2007 9:20 AM
Ok, so why was the GPL license violated? Or did Mr. Torvalds offer you
an alternate license arrangement?
On 1/17/07, Evan Kirstel [EMAIL PROTECTED] wrote:
Sorry for the confusion..the MP202 is running Linux!
-Original Message-
From: Andrew Joakimsen [mailto:[EMAIL PROTECTED]
Sent
,Backgroud(busy-call-back-ltr) ; If you
had this sound file.
.
.
.
.
exten = s-BUSY,n,Voicemail(b123)
exten = s-BUSY,n,Hangup
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of Andrew Joakimsen
Sent: Thursday, 18 January 2007 3:25 PM
To: Asterisk Users
Is it not coming in as CallerID(RDIS)? The specifications for the
service don't seem too different from any other PRI.
On 1/16/07, Gary Mensenares [EMAIL PROTECTED] wrote:
Hi everyone!
I'm wondering if anyone on the list had the opportunity to work with an NTT
INS1500 ISDN PRI service before.
Most of the Cisco phones sold cheap are UNLICENSED (global spare)
thus you would not be able to purchase (or at least aren't supposed
to) the smartnet contracts, you need to buy the license ($100+) and
the contract ($10 or so)
On 1/10/07, Leo Ann Boon [EMAIL PROTECTED] wrote:
David Thomas
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