On Jun 14, 2006, at 10:15 PM, Crazy Boy wrote:
Hi,
Thank you for your response. I have a doubt. May I know what is meant
by simultaneous calls? Looking forward for your response.
Simultaneous- as in at the same time.
There are dictionaries online also, you might try them.
On Wed, Jun 14, 2006 at 04:49:28PM -0700, Mike Fedyk wrote:
Tzafrir Cohen wrote:
On Wed, Jun 14, 2006 at 01:15:03PM -0700, shadowym wrote:
FreePBX or AAH(aka trixbox) requires 256MB of RAM minimum to run properly.
Just sitting there doing nothing on my test system it is using 170MB.
Anyone has any experience with these cards? Looks suspicious like the X101P.
http://www.cuphone.com/products/ppg/index.htm
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On Wed, Jun 14, 2006 at 10:48:35PM -0700, Mike Fedyk wrote:
Tzafrir Cohen wrote:
On Wed, Jun 14, 2006 at 11:51:06AM -0400, Mike wrote:
Hi,
I'm stuck writing a Web GUI because nothing out there is exactly what I
need. I'm not writing something as extensive as what _is_ out there, but
On Thu, Jun 15, 2006 at 02:30:46PM +0800, Leo Ann Boon wrote:
Anyone has any experience with these cards? Looks suspicious like the X101P.
http://www.cuphone.com/products/ppg/index.htm
Considering that the X100P/X101P is basically a simple winmodem card
(a card that has the basic chips for
I am using aah 2.6
Please any one knows how to add a directory ? to be dialed
from *411
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without
On Wed, Jun 14, 2006 at 06:28:38PM +0200, Mimmus wrote:
Hi,
using Sangoma drivers:
- doing 'lsmod', I see:
zaptel ... wanpipe,wctdm24xxp,wctdm,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor2
I'd like to avoid loading all these modules. What have I to do?
Because you try to modprobe them all instead of
I bought one over a year ago along with the USB phone. Was never able
to get the card to work properly with anything (even the software it
came with). For less money I got am X100P clone on ebay and that works
great.
Leo Ann Boon wrote:
Anyone has any experience with these cards? Looks
I am using AddQueueMember to add a local channel to a queue. My
(simplified) dial plan is
[AddMember]
exten = 789,1,AddQueueMember(SomeQ|Local/[EMAIL PROTECTED])
[Queue]
exten = 123,1,Queue(SomeQ|nt|||120)
exten = 123,2,Hangup()
exten = h,1,NoOp(InQ)
[Agent]
exten = 456,1,Dial(SIP/456)
exten
No I dont believe so. The address book is a new feature as it is very
basic in my opinion and even editing it on the phone is difficult.
I would expect a web based editing feature to be implemented at some
point and once that is done it should be possible to do a mass update of
the phones.
On
On 15 Jun 2006, at 02:59, Daniel Salama wrote:
Does anyone know how many simultaneous calls can a WRTG54GS handle?
Assuming SIP phones are connected locally using G711.u codec and
the WRTG54GS connects to a remote Asterisk server using IAX2
trunking using GSM codec.
Very few (2 perhaps)
Hi,
I had the same problem using Realtime, seams not to be possible to use both...
Whatever, I use another way than gotoIf to achieve this function:
[to_ext]
switch=Realtime/extd
_X.,1, set(numdial=${EXTEN})
_X.,2,Goto(${CALLERID(num)},1)
_X.,3,Goto(to_ext_def,${EXTEN},1)
I am using [EMAIL PROTECTED] version 2.6 how can I update the
existing to be 2.8.
Any ideas
regards
*
No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without
Calls going on at the same moment in time.
Crazy Boy wrote:
Hi,
Thank you for your response. I have a doubt. May I know what is meant
by simultaneous calls? Looking forward for your response.
ThanksRegards,
Chandra.
*/Zoa [EMAIL PROTECTED]/* wrote:
Crazy Boy wrote:
Hi Friends,
Unless I'm misunderstanding what you're looking to do, Aaron has hit the nail
on the head here. You need to set it up so that the secondary, tertiary, etc.
boxes are weighted differently. That way, you need not know or care about the
weights directly within the dialplan.
Regards,
- Brad
Do take a look at the OpenSlug on the nslu2 - The nice thing
about the 'Slug' is that you can add a USB harddrive for swap and
voicemail, so it is more 'expandable' than the WRTG54
I should warn you I have never tried trunking IAX on my slug,
I will do at some point
Tim, are you running
On Thu, 2006-06-15 at 11:33 +0300, Khaled Chehab wrote:
I am using [EMAIL PROTECTED] version 2.6 how can I update the existing to
be 2.8.
Any ideas
Ask on the AAH lists?
--
Dave Cotton [EMAIL PROTECTED]
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Hi I make a Digital Receptionist ,but how can I
attach it to an extension
Help
regards
*
No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without
Khaled Chehab wrote:
Hi I make a Digital Receptionist ,but how can I attach it to an extension
[EMAIL PROTECTED] is now called TrixBox. You'll get a lot more support at
http://www.trixbox.org
cYa,
Avi
--
National Manager - Special Projects
Melbourne / Sydney / Canberra / Hobart / London
On 6/14/06, Cory Andrews [EMAIL PROTECTED] wrote:
Sangoma is NOT releasing a single T1 with echo cancellation.
But AFAIK they ARE releasing (have released?) a dual T1/E1 card with
hardware EC, which may be sufficient compromise.
Cheers,
Steve
___
Is there anyway to force an autopickup on a cisco 7940 / 60 from the
dialplan ?
My problem is that I am originating a call from the AMI, with the
internal user being called first, and then connecting to external user.
However, sometimes the internal user doesn't pick up the phone, so the
Hello
everybody,
I have some problems
with the quality of the sounds file when I record a lot of
call.
I see of the
internet that there is maybe something to do with "Bus Masterisng" with the SCSI
card (53c1030) and the RNIS card (sangoma E1).
Do you have a idea
for me ?
Thanks,
That is the first thing I did.. rxgain seems to help to make it louder
when the two PSTN lines are bridged but also affects
the audio quality adversely. It's very touchy.. it also causes
horrible feedback (loud screeching when lines bridge).
I've upgraded zapata and libpri too.
JD
Carlos
Hi,
I am familiar with asterisk, though never actually tinkered with one
myself ... so i don't know the full extent of its capabilities.
I am facing a request to bridge a sip network and an h323 network.
I would like to operate the sip with ser as the proxy and some
gatekeeper on the h323 side
Hello,
There are 3 PRIs connected to the card each from
different operators. Especially echo occured on span 3 is really annoying.
Configuration files are as follows. Is there something wrong in conf ?
Zapata.conf --
[channels]
context=default
Hello
Eicon Networks produce a Single T1 card with hardware echo cancellation.
http://www.eicon.com/worldwide/products/MediaGateways/diva-server-vt1-pr
i.htm
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Davies
Sent: 15 June 2006 11:43
To:
Hi Gareth,
Gareth Blades wrote:
No I dont believe so. The address book is a new feature as it is very
basic in my opinion and even editing it on the phone is difficult.
I would expect a web based editing feature to be implemented at some
point and once that is done it should be possible to
Steve Davies wrote:
On 6/14/06, Cory Andrews [EMAIL PROTECTED] wrote:
Sangoma is NOT releasing a single T1 with echo cancellation.
But AFAIK they ARE releasing (have released?) a dual T1/E1 card with
hardware EC, which may be sufficient compromise.
I find it much cheaper to pick up a
Off course for the price, you could by a four port sangoma with echo cancel
David Waugh wrote:
Hello
Eicon Networks produce a Single T1 card with hardware echo cancellation.
http://www.eicon.com/worldwide/products/MediaGateways/diva-server-vt1-pr
i.htm
-Original Message-
From:
Title: Re: [Asterisk-Users] Echo Problem with T411P
Try setting echocancelwhenbridged=yes. Also, in your zaptel, you only need to define one span as the clock master, so should be like
zaptel.conf--
span = 1,1,0,ccs,hdb3,crc4
bchan = 1-15,17-31
dchan = 16
span =
Hi
Carlos,
- send us your interrupts configuration: cat
/proc/interrupts
- tryadjusting line impedancy; see fxotune on zaptel source
directory.
cheers,
Fabay
-Mensaje original-De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]En nombre de
news.asterisk.usersEnviado el: Jueves,
Hi List,
Just a little question about the MYSQL(Clear ) function,
Do I need to clear the resultid after an update or an Insert query ?
Because I don't want to overload mysql ...
I'm doing this actually and I have troubles I don't understand
resultid is set in the following macro and
On Thu, 15 Jun 2006, Sean Cook wrote:
Off course for the price, you could by a four port sangoma with echo cancel
If you look at the price, the echo-cancel and the ports only, then you might
be right. But people with professional intentions would do the comparison
more accurate and look into
Hello Strom,
thanks for your reply, but I guess your answer is missing (???).
Bye,
Stefan
Am Donnerstag, 15. Juni 2006 02:18 schrieb Strom Carlson:
-Original Message-
From: Stefan-Michael. Guenther (in-put GbR) [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: 6/10/06
This may be slightly
off topic.
I am using FreePBX,
and using it's backup feature.
Here is the question
part:
I would like to copy
my backup off the asterisk server.
From your
experiences, which approach seems more resilient to failure:
Push the backup from
asterisk to another server
Hi, guys.
May be someone could give me advise?
I am trying to find good VOIP provider ONLY for OUTGOING calls with low
per channel cost and cheap rates on Eastern Europe, Turky and xUSSR.
Should support g729 or g723 codecs, SIP or IAX connectivity.
--
Hi
Please I want to make a schedule to make list of extensions in a conference,
automatically the system call them and put them in a conference mode
Can any one help me
Regards
*
No employee or agent is authorized to conclude any binding
On Thu, Jun 15, 2006 at 08:20:29AM -0400, BerkHolz, Steven wrote:
This may be slightly off topic.
I am using FreePBX, and using it's backup feature.
Here is the question part:
I would like to copy my backup off the asterisk server.
Please define what is it exactly that you want to
Did someone use a 26xx, 36xx or 53xx as a T38 Gateway?
I need to know if we can register an ata like Sipura 2100 to the cisco
equipment, or we need to register the cisco on asterisk in order to complete
the circuit.
The documentation on the Cisco is always referred to the call manager. All
that
On 6/15/06, Idris AVCI [EMAIL PROTECTED] wrote:
Hello,
There are 3 PRI's connected to the card each from different operators.
Especially echo occured on span 3 is really annoying. Configuration files
are as follows. Is there something wrong in conf ?
Have you verified that the provider on
We're using a Cisco 3660 as a T.38 gateway for fax reception, but as
Asterisk does not support T.38 in pass through mode yet what we're doing
is sending a SIP REFER message (via the Transfer application) to our SIP
provider (when we detect fax tones) to redirect the call to the Cisco
gateway.
13Kbs
Google is a wonderful tool. Learn to use it!
bp
On 6/14/06, Pablo Allietti [EMAIL PROTECTED] wrote:
On Fri, Jun 09, 2006 at 04:45:51PM -0400, William Piper wrote:GSMand what is the size in KB that gsm spent?
bpOn 6/9/06, Pablo Allietti [1]pablo@lacnic.net wrote:hi all, i saw in digium that
Sounds like there maybe a codec issue. If you are using g729, make sure you have licenses.
bp
On 6/14/06, Mimmus [EMAIL PROTECTED] wrote:
Hi,calling a partner on the other side of a SIP trunk, call gets disconnectedimmediately after answer. This is the content of log file:
Jun 14 16:25:14
Asterisk guy wrote:
are there any open source sip softphone (Window OS version )?
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Hello everyone,
I am not an expert in Asterisk programming
yet. So, can someone help me put my first steps on how to use AGI to access
MySQL tables and do queries. Any reference or help is appreciated. My target is
to get Festival to read TTS from data stored in MySQL table based on the ID
Are you still interested on tests?
Regards,
Carlos Alperin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, April 25, 2006 2:46 PM
To: asterisk-users@lists.digium.com; users@openser.org; [EMAIL PROTECTED]
Subject:
Voxee.com supports both SIP and IAX2, as well as GSM, iLBC, uLAW, aLAW
and G729 codecs. Their rates to international locations are based on
6/30 billing, and vary a lot from location to location. You can view
their rates here: http://www.voxee.com/rates.xls
You may also want to look at
Walid,
Check the ASTCC agi script ; it just does exactly
that:
http://www.voip-info.org/wiki-ASTCC
Cheers,
Fred
- Original Message -
From:
Walid Azab
To: asterisk-users@lists.digium.com
Sent: Thursday, June 15, 2006 11:14
Subject: [Asterisk-Users] AGI to read
Is any way to get an configuration example of that?
I know that we cannot send the T.38 through the Asterisk, that is I'm trying
to avoid that, but there is no way to register the Sipura 2100 on the Cisco.
It's not a Gatekeeper or a VoIP Server.
However there is 1.2.7.1 patch for the T.38
Hi everyone,
I noticed that the waiting message indicator
does not lit when I have a message in my voice mail. Any suggestion why this is
happening?
Thanks
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To
None of those are open source that I recall.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Derek Whitten
Sent: Thursday, June 15, 2006 6:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Thanks a lot Fred. I will give it a try. I
am using [EMAIL PROTECTED] V2.8 by the
way.
Regards
Walid
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Frederic
JeanSent: Thursday, June 15, 2006 3:24 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
I don't know if you keep your eye on the -biz list or not. But you
should if you don't. Plainvoip.com just anounced last weekend they are
offering blended US48/Canada Termination @ .007 w/ free Toll Free
termination. They support IAX/SIP w/ all major codecs including g723
and g729.
On 15/06/06, Walid Azab [EMAIL PROTECTED] wrote:
Hi everyone,
I noticed that the waiting message indicator does not lit when I have a
message in my voice mail. Any suggestion why this is happening?
You probably need to change the bulb.
--
Peter Bowyer
Email: [EMAIL PROTECTED]
On Thu, Jun 15, 2006 at 06:06:40AM -0700, Derek Whitten wrote:
Asterisk guy wrote:
are there any open source sip softphone (Window OS version )?
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To
For me it does not work, my extensions looks like this:
exten = 555,1,MeetMeCount(500|count)
exten = 555,2,Gotoif,$[${count} = 1]?6
exten = 555,3,Meetme,500|xApMs|1234
exten = 555,4,Playback,goodbye
exten = 555,5,Hangup
exten = 555,6,Goto(from-internal-custom,556,1)
exten = 555,7,hangup
exten =
I did try fxotune.. it did wonders on echo and audio quality when
dialing in.
Before the zapata upgrade it did nothing (the -d -b options weren't
there).
I found a page on voip-info.org that talked about using the new version
and
followed what was there:
amportal stop
./fxotune -d -b 2
minisip (http://www.minisip.org) - it is LGPL, GPL ... windows support
is in testing status
On 6/15/06, Kerry Garrison [EMAIL PROTECTED] wrote:
None of those are open source that I recall.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
It sounds nice, but, how many calls can you get on the NSLU2? Say the
SIP phones are talking either G711.u or GSM only and the IAX trunk is
GSM only.
Thanks,
Daniel
On Jun 15, 2006, at 4:15 AM, Tim Panton wrote:
On 15 Jun 2006, at 02:59, Daniel Salama wrote:
Does anyone know how many
I have to check about the configuration as I'm not the one who did it.
Bear in mind that what we did was for fax reception, so the SIP REFER is
being sent to our fax provider, not to the Cisco gateway. The gateway
just receives the call after it's been redirected.
Carlos Alperin wrote:
Is
I noticed that the waiting message indicator does not lit
when I have a message in my voice mail. Any suggestion
why this is happening?
Double check:
- mailbox=ext@context for your extension in sip.conf
- if your 'home directory' is under /var/spool/asterisk/voicemail/context
context is usually
If you had approx $300 per phone as a budget and needed to buy 30 phones
for a new system, which would you get?
Things needed:
At least 2 lines
mini-switch to go to the PC as there is only 1 network connection per
physical station
Screen to be able to show callerid info and a few lines afterward
What about define timeout or interrupt
exten = t,1,Hangup
exten = i,1,Hangup
at the end of your extension definition
It works for me always
Carlos Alperin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bartosz
Wegrzyn - asterisk
Sent: Thursday, June
We developed a commercial SIP phone for windows platform. What do you
want to know ?
-Original Message-
From: Kerry Garrison [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 15, 2006 4:31 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] open
Someone has written a script that uses telnet to log into the phone,
and issue the test commands to answer the call. Search the lists it
should be somewhere.
On 6/15/06, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
Is there anyway to force an autopickup on a cisco 7940 / 60 from the
dialplan ?
On 6/15/06, Khaled Chehab [EMAIL PROTECTED] wrote:
If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or
disclose its content
Hi, I have a asterisk/voip newbie and I am sorry if my quetion is banal.
I used in my private LAN, Express Talk on Windows XP and Asterisk latest
version on Fedora Core 4 , with this configuration in Express Talk
Lines menu:
Setting for Line: Default Line Settings
Full 'friendly' Display Name:
I have always been
an advocate for Broadvoice. Their service although is a little shoty at
times has been an extremely cheap service that works with Asterisk. 19.99
a month for unlimited (Some say now really unlimited but I average 3500 minutes
a month so pretty fine for me) calling and to
Openwengo
Laurent
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: jeudi, 15. juin 2006 15:40
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] open source sip softphone (Window OS version )
On Thu, Jun 15, 2006 at
I'm bridging a Cisco Call Manager 3.2 system (h323 only) to an asterisk SIP setup. It works. There are issues, but that has more to do with Unity voicemail than the h323 implementations.
On 6/15/06, Cesc [EMAIL PROTECTED] wrote:
Hi,I am familiar with asterisk, though never actually tinkered with
I was just told that for my forthcoming system I will be getting a data
T-1 instead of a voice T-1. Given that all of the handsets will be voip
phones, no analog at all, do I need echo cancellation? I looked at the
voip-info wiki and it seems to me that the answer should be no but I
would like
Is there anyway to force an autopickup on a cisco 7940 / 60 from the
dialplan ?
Unfortunately, no.
However, the 7940 can use two SIP accounts, the 7960 six accounts. This is
how I implemented intercoms:
Set the first (top) line to register to an account that will be used to
ring, set the
Cisco CP-7960G
Cisco-CP-7940G
Polycom IP601
Or
(2) Linksys SPA-942 - 1 on each side of my desk
Cory Andrews
Executive Vice President
++
VoIPSupply.com
PBXSelect.com
++
454 Sonwil Drive
Buffalo, NY 14225
voice - 800.398.VoIP X3402
fax - 716.630.1548
e - [EMAIL
Polycom 601, hands down.
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Warren
Sent: Thursday, June 15, 2006 10:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Best $300 VoIP phone for asterisk?
If
Steve Davies wrote:
On 6/15/06, Idris AVCI [EMAIL PROTECTED] wrote:
Hello,
There are 3 PRI's connected to the card each from different operators.
Especially echo occured on span 3 is really annoying. Configuration
files
are as follows. Is there something wrong in conf ?
Have you verified
This information may help. Not sure if this is exactly what you need, but can't hurt to try to help...
exten = _XXX,1,Goto(Context_Ext,s,1)
-
[Context_Ext]
exten = s,1,Wait(0.5)exten = s,2,DigitTimeout,1exten = s,3,ResponseTimeout,1exten = s,4,Background(...YourMessageHere...)
.
SIP or SCCP?
On 6/15/06, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
Is there anyway to force an autopickup on a cisco 7940 / 60 from thedialplan ?My problem is that I am originating a call from the AMI, with the
internal user being called first, and then connecting to external user.However,
Tzafrir Cohen wrote:
On Thu, Jun 15, 2006 at 08:20:29AM -0400, BerkHolz, Steven wrote:
This may be slightly off topic.
I am using FreePBX, and using it's backup feature.
Here is the question part:
I would like to copy my backup off the asterisk server.
Please define what is it
Hi all,
I've got a fairly simple setup, a 4 port zaptel T1 card with 1 PRI and 2
flat em T1's coming into it, and its working perfectly except for one
small thing. When people dial any of the numbers on either the PRI or
the T1 from outside of the pbx, they dont get the ringing sound. It
We user mySQL in certain instances of AGI and it works great. This information has helped us quite a bit. If you are using php - make sure you are using the CLI version.
http://phpagi.sourceforge.net/
Here is a code snip of how we are doing it.
#!/usr/local/bin/php -q?php/*** @package
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Greets all sorry for the posting but this one has me confused. I
compiled Zaptel and am
experiencing something strange. I followed the instructions on
README.udevs to the tee,
set permissions, etc. and I'm confused as to what gives. Zaptel
compiled
Daniel Salama wrote:
It sounds nice, but, how many calls can you get on the NSLU2? Say the
SIP phones are talking either G711.u or GSM only and the IAX trunk is
GSM only.
Thanks,
Daniel
Unless someone has ported zaptel (and ztdummy) to run on the mipsel you
won't be doing any
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 14, 2006 7:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] DUNDi Not Able to Handle Complex
FailoverSituations
On Wed, 14 Jun 2006, Douglas
-Original Message-
From: Watkins, Bradley
[mailto:[EMAIL PROTECTED] Behalf Of Watkins,
Bradley
Sent: Thursday, June 15, 2006 2:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] DUNDi Not Able to Handle Complex
FailoverSituations
Hi,In your script, use
signalling=fxo_ks, you have TDM04B?I have TDM400 with fxo modules and use signalling=fxs_ksRegardsOn 6/15/06, news.asterisk.users
[EMAIL PROTECTED] wrote:
I did try fxotune.. it did wonders on echo and audio quality when
dialing in.
Before the zapata upgrade it
Kevin
Fleming has said on numerous ocassions that this is known not to work, and is
not supported.
-Original Message-From: Benjamin Stocker
[mailto:[EMAIL PROTECTED]Sent: Tuesday, June 06, 2006 4:31
AMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users] Asterisk
Hi,
I set a preference order of the codecs to my
sip.conf
[general]port = 5060 ; Port to bind
tobindaddr = 0.0.0.0 ; Address to bind tocontext = default ; Default for
incoming calls of not registered phonesdisallow = allallow =
g729allow = g723allow = alawallow = ulaw
Connected a 'Sipura
Warren wrote:
I was just told that for my forthcoming system I will be getting a data
T-1 instead of a voice T-1. Given that all of the handsets will be voip
phones, no analog at all, do I need echo cancellation? I looked at the
voip-info wiki and it seems to me that the answer should be no
Steve Totaro wrote:
http://www.voip-forum.com/news.php?p=187
I am battling this same problem and cannot for the life of me figure
out how to work around it. The above link looks promising but leads
to a server error page.
Basically here is what I need to do. I have three asterisk boxes.
Hello all!
It's come time where I need to add another programmer to our team.
You should have at least 3 years of work experience with PHP/MySQL.
Please send me your resume and a few code samples if you can.
If you can only work part-time or full-time, please include that in your
response.
Dunno if anyone else has seen this yet:
http://www.scmagazine.com/us/news/article/563800/vulnerabilities+put+asterisk+telephone+systems+risk/
--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
___
What are the advantages of the 601 over the 501? I do not have PoE here
(at least not yet) and do not need the additional lines (except maybe on
the attendant's console).
Is the 601 better than the 501 or does it just have the ability to
handle more lines?
W
Watkins, Bradley wrote:
Polycom
Hi All,
Does anyone have any experience getting a 7936 to work with Asterisk?
Do you need to use SCCP or is there a SIP image for the phone? I have a few 7960G's and they are working with SIP, just curious if the config of the conference phone is the same and if anybody has any good setup links.
Hello to everybody.I need of urgent help. I need to create a group in that this have access barely the connection between extensions. In hypothesis some those persons can do connection by pstn or voip. Someone has some hint of as I should proceed?
Andre Luiz Martins
On Thursday 15 June 2006 10:22, Warren wrote:
I was just told that for my forthcoming system I will be getting a data
T-1 instead of a voice T-1. Given that all of the handsets will be voip
phones, no analog at all, do I need echo cancellation? I looked at the
voip-info wiki and it seems to
Matthias Fechner wrote:
Hi Gareth,
Gareth Blades wrote:
No I dont believe so. The address book is a new feature as it is very
basic in my opinion and even editing it on the phone is difficult.
I would expect a web based editing feature to be implemented at some
point and once that is done
in which extension,
the thing is that when every (voip) user disconnects ,
the zap channel is still connected to the conference,
What about define timeout or interrupt
exten = t,1,Hangup
exten = i,1,Hangup
at the end of your extension definition
It works for me always
Carlos Alperin
It
seems that I am having a heck of a time explaining my attempts at distributing
ACD Queues to the list. Here's one little problem, that's only a piece of the
puzzle.
dundi.conf:
180q
=
global_dundi_q_pbx1,100,IAX,dundi1:[EMAIL PROTECTED]/${NUMBER},nopartial180q
=
If there is an Answer(), Playback(), Background(), or anything else that
answers the line before Dial(), you must have set up
/etc/asterisk/indications.conf or Asterisk will not know how to do
inband ringback.
Derek wrote:
Hi all,
I've got a fairly simple setup, a 4 port zaptel T1 card with
Hello Strom,
thanks for your reply, but I guess your answer is missing (???).
Bye,
Stefan
Am Donnerstag, 15. Juni 2006 02:18 schrieb Strom Carlson:
-Original Message-
From: Stefan-Michael. Guenther (in-put GbR) [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: 6/10/06
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