We are looking for people to help test our termination services before
going into production. The more problems, the better, we want to debug
everything before we start selling. In exchange for your help, you will
get a discount on services once we are fully operational.
If you are
Figures... So... Everybody went to FWD :) ?
It mostly works, does IAX, so, yeah.
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Hello,
Anybody can recommend a IAX2-native IP phone?
Preferably buyable online..
TIA.
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Hello,
in January, there was announced a graphical asterisk load tester
on this list.
Can anyone tell me, where I can find this tool?
Regards,
Henry
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hello my friend is trying to get his spa-1001
(sipura) 1001 connected to my asterisk box.
he reset his spa-1001 to factory defaults I emailed
him the voip-info page I found on google and yes I did look on google anyways he
isn't able to get the thing to connect to it eg getting a dial tone,
Rather than caching the data, which would remember the data past it's
useby date (which is never a fixed value, the useby date is when the
data in the DB is modified, which could be at anytime!), we should
simply read the entire extension in a single select, and cache that
answer for the life
Dear sir,
I'm interested in your project. Can you tell more about it??
Regards.
Daniel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, May 22, 2005 4:23 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users]
chawki hammoud wrote:
--- Christian Wengel [EMAIL PROTECTED] wrote:
Hi!
Could you post your musiconhold.conf and
modules.conf, please?
MfG Christian
This is the modules.conf file:
[modules]
autoload=yes
noload = pbx_gtkconsole.so
noload = pbx_kdeconsole.so
load =
snacktime wrote:
Personally I would rather see realtime load everything into memory and
not go to the database unless something has changed or you reload.
How would this be different than just storing the configuration
statically in the database, and reading it at startup and at subsequent
Why disregard from MX? :)
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of JetSpeak
|Sent: Lunes, 23 de Mayo de 2005 01:01 a.m.
|To: asterisk-users@lists.digium.com
|Subject: [Asterisk-Users] Looking for people to test calls
|
|We are looking for
In article [EMAIL PROTECTED],
Rod Bacon [EMAIL PROTECTED] wrote:
I'm new to H.323 and I have noticed that there are two separate channel
drivers for * available - the inbuilt one, and oh-323. I had trouble
compiling oh-323 with the current cvs stable, so I tried the inbiult one
(with
I have that same problem just now. I`m trying to find some
solution with serveral tests, using IOS v.12.3(8r)T7 on the C2821 box with two
PRI ports.
When i find something, it`ll be posted here, and i`m
awaitingto do it also from your side.
-b
- Original Message -
From:
Anton Krall wrote:
Why disregard from MX? :)
You might want to check the archives, or Google for Vonage staff arrested in
Mexico, or something along those lines..
flynn
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I would like to hear stories from people using TAPI, CTI or CallerID
software with Asterisk.
What are you guys using, setup examples, etc.
Has anybody sucessfully integrated SugarCRM with Asterisk and how did you do
it.
Are you running callerid software? Did you stumble into problems like using
Personally I would rather see realtime load everything into memory and
not go to the database unless something has changed or you reload.
Then maybe add a new manager command like the following:
Then don't use realtime extensions, use the static database functions in
realtime to load your
I think there is a misconception here but then again I might be wrong.
As far as I know, COFETEL, the Mexican regulatory entity for
telecommunications prohibits companies from selling voip LD services in
Mexico without the appropiate licenses. But, nothing states that people in
Mexico can buy
Tony Mountifield wrote:
In article [EMAIL PROTECTED], I wrote:
In article [EMAIL PROTECTED],
Michael Manousos [EMAIL PROTECTED] wrote:
Can you get an ethereal trace on a call with that problem?
Run an RTP analysis on the captured stream (Tools Menu) and save
the contents of the RTP packets
Title: ZAP/DTMF
Hi community!
If somebody is interesting in issues I found then
configuration I have described above has strange behaviour ONLY when Asterisk 1
has Sangoma 4 PORTs card. Pairs Digitum/Digitum and Sangoma 2ports/Digitum are
OK - DTMF are passed thru E1.
I am going to
Vamsi Pottangi wrote:
Hi All,
There is a parameter simultaneousMax=10 in oh323.conf.
Had anybody tried out what is the maximum value that can be achieved ?
What is the maximum number of simultaneous h323 calls can the oh323
driver can handle.
I tried to get it only till 30 to 40 simultaneous
Hi, how I can connect Astrisk to my iptel account???
I have try to this configuration, but it doesn't work:
In sip.conf:
register = my_account_name:[EMAIL PROTECTED]
[iptel.org]
type=friend
host=iptel.org
fromuser=my_account_name
secret=
nat=yes
in extensions.conf:
[fromiptel]
exten =
Hi,
I've a busy server where one E1/PRI span has been behaving oddly.
Here's the span:
Primary D-channel: 16
Status: Provisioned, Up, Active
Switchtype: EuroISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
T200 Timer: 1000
T203 Timer: 1
T305
+++ Mark Johnson [20/05/05 11:08 -0400]:
Ok, guys... Please be gentle with me. I have what is going to be the
strangest question you will have ever heard, but I have no idea what to
tell this person.
I set up Asterisk 3 or 4 weeks ago, everything is running smooth. My
receptionist has
Hello Everybody,
I wished to know that the .c files
in Asterisk in the /usr/src/asterisk directory, can they be modified to change
the behavior of Asterisk? If yes, could you please tell me as how does one go
about modifying the code and also which are the files that are modified? It
Hello Everybody,
This is Bharat here. Are the
protocols RTP and SIP modified to suit according to the features one desires?
If one can, please kindly do let me know as to how one goes about modifying the
same. And also which are the file that are modified to make changes in the rtp
and
Hi there,
I need your help. Please le me know if it is possible to have following
implementation in place:
Asterisk server #1 (ast1) has server SIP clients with extensions 17XX
Asterisk server #2 (ast2) has server SIP clients with extensions 16XX
All I need that extensions from ast1 be able to
I've got an Asterisk box at a client; last week, it just stopped answering
the phone. Outbound calls still went, but inbound -- no dice. Asterisk
didn't even acknowledge that the line was ringing. A reboot fixed it --
though, clearly, I can't have them rebooting all the time.
So:
-
I have recently upgraded my firmware from v6.3 to v7.4. Now when the phone is booted or rebooted, the initial screen "Initializing Vlan" takes forever to initialize before it initializes IP. Any ideas/Thoughts? (Trying not to Revert back to v6.3).
-C
Hi all
What headset do people use with the GXP-2000? Any recommondations for
or against particular models?
Thanks
Peter
--
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
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Are you connecting the phone to a trunk port with a voice vlan defined?
Peter
-Original Message-
From: Christopher Kenna [mailto:[EMAIL PROTECTED]
Sent: 23 May 2005 12:38
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Cisco 7960 v7.4
I have recently upgraded my
On Fri, May 20, 2005 at 02:17:57PM -0600, Rich Adamson wrote:
Guess that makes about 9 out of 10 happy customers... anyone want
to make that 90 out of 100?
We've only dealt with them a couple time. However, our first order consisted
of a Vegastream ATA which was just flakey and a Snom 190 that
Hi
Im trying to put up an sip pbx system for my company
but im getting some problems when Im trying to call from server (
branch A ) to server ( branch B )
This is my extentions.conf :
exten = 3003,1,Dial,SIP/[EMAIL PROTECTED]
Nope, nothing is defined except for the default vlan.
-C
[EMAIL PROTECTED] 5/23/2005 7:53 AM
Are you connecting the phone to a trunk port with a voice vlan defined?Peter -Original Message- From: Christopher Kenna [mailto:[EMAIL PROTECTED] Sent: 23 May 2005 12:38 To:
Try creating a trunk to the switch and creating a voice vlan, this can be the
same as the data vlan. My phones (on 7.4) pause for about 5 seconds at the VLAN
stage
Config from one of my switches
interface GigabitEthernet1/0/6
description Carolines Study
switchport trunk encapsulation dot1q
I'm using astapi for outgoing tapi driven calls from asterisk by Omni,
works great. Couple of memory default problems but that could be ram
(who knows).
I'd like to have incoming name popup from outlook but haven't tried
anything yet.
Cheers,
Dean
-Original Message-
From: [EMAIL
If your looking to link 2 asterisk boxes
might I suggest IAX. Much more efficient in the way bandwidth
is utilized between the locations. Also
if you want to use your sip solution, have you setup the other
end point in your SIP.CONF? I have never
got IP dialing to work in asterisk but it
Thanks Steve
I realised the other day that I don't want the Cisco to register with
credentials. There is in fact a hidden credentials command in 12.3(8)T.
What I did was take away all registration commands from my sip-ua block in
the Cisco.
I am using [EMAIL PROTECTED], so I have created a
Mark,
Try writing the sip.conf stanza as:
[192.168.44.23]
context=from-pstn
host=192.168.44.23
type=friend
insecure=very
The 'insecure=very' allows any calls from this IP address to match.
Alistair Cunningham,
Integrics Ltd,
+44 (0)7870 699 479
http://integrics.com/
Mark Dutton wrote:
Hi B
Do you mean I must do this in my sip.conf
file on eatch server
Branch A
register= 3001:[EMAIL PROTECTED]
/3001
Branch B
register= 5001:[EMAIL PROTECTED]
/5001
thx
Q
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian C. Fertig
Sent: 23 May
Terrible call quality all week last week. Using a different provider
for the time being.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Sunday, May 22, 2005 10:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users]
My voipjet calls *were* doing that, two, three weeks ago, so I signed up
for livevoip. Now livevoip is so bad that I fell back to voipjet. We
really get what we pay for, don't we? 2c/min providers I have access to
have had consistently good call quality.
-Original Message-
From:
Slash-dot is pointing to this article on Asterisk and Pingtel.
http://www.theregister.co.uk/2005/05/22/pingtel_voip/
Paul
Paul Mahler
www.signate.com
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Joel Duffield wrote:
Hey steve
I remember a tip somewhere where they used a conference room and added all
the users into that conference muted, then kicked them out at the end of the
call. Sorry I can't remember at all where this was but it looked like it
could work. How did you get the
Adnan Ahmed wrote:
Hello ,
I want some tips guidance i am sure this topic discuss alot in list,i
try my best to solve it by myself try googling looking wiki everywhere
but no luck question is iax-iax trunking not working setting,trying
each n every option
server2 iax.conf:
[general]
Peter Braidwood wrote:
Try creating a trunk to the switch and creating a voice vlan, this
can be the same as the data vlan. My phones (on 7.4) pause for about
5 seconds at the VLAN stage
Holy crap! 5 seconds?!? Mine take at least a minute, if not 2. Can the voice
vlan also be the default vlan?
i'll give that a try, thanx
-c
[EMAIL PROTECTED] 5/23/2005 8:46 AM
Try creating a trunk to the switch and creating a voice vlan, this can be the same as the data vlan. My phones (on 7.4) pause for about 5 seconds at the VLAN stageConfig from one of my switchesinterface
2c/min providers I have access to have had consistently good
call quality.
I'll gladly pay 2c/min for quality calls. We pay nearly $1/min to the US
from here (Yes, you read it right, 1 dollar per minute), so 1c/min extra
won't be noticed. At 2x/min our calls are practically free.
There's a
Title: Re: [Asterisk-Users] spa-1001 with asterisk?
When configuring the SPA-1001 you need to put the port you are using after the IP address in the configuration menu. For example... If your asterisk server is at 192.168.4.34 and you are using port 5060 for SIP, then you need to type in
Hi All,
i'm going crazy trying to make asterisk work with the following
hardware:
02:05.0 Network controller: Cologne Chip Designs GmbH ISDN network
controller [HFC-PCI] (rev 02)
Subsystem: Cologne Chip Designs GmbH ISDN Board
Flags: bus master, medium devsel, latency 16, IRQ 17
Can the H.323 channel be configured as a gateway from another system?
Or can it be configured as an endpoint on another system?
Or can it only be connected to actual endpoints like phones? If either
of the first two are yes, does anybody have a sample h323.config file?
The samples that come
Yep I totally agree, I think a lot of carriers are trying to compete on
price because that is the easiest but I think there is already a
backlash, I know that I was reconsidering using packet 8 for my
Australian calls at one stage because of quality (they seemed to have
picked up the quality a
The voice and data vlan can be the same, however that makes it trivial for
others to record phone conversations if this is a concern. Having a separate
vlan for voice makes it a *bit* harder!
I have only ever used Cisco switches, however .1q is an open standard so it
should work with other
I have been trying to figure a way to SendDTMF into a MeetMe room using
the Manager API.
I can't redirect everyone into another context and then bring them back
because that would mess up my logic.
I am trying to use local channels and the originate Action to accomplish
this.
Exten: 3441115
Title: Asterisk-Users Digest, Vol 10, Issue 174
Hello Adam,
I am busy all day until after 6pm EST. If you want
you can call me then at 1-914-591-2211.
Regards,
Greg
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
CollardSent: Monday, May 23, 2005 12:54 AMTo:
Version 0.118 - 23. may 2005
* Drag and Drop Call and Channels buttons
* Show Channels as buttons on the Panel
* Show Queue information in very large letters on new panel
* Schedule Wakeup Calls (good for hotels)
Download: http://ipswitchboard.thorben.dk
Does anybody know of a WINDOWS application (preferably freeware) that will
simply playback asterisk GSM sound files, I don't want to record them, just
playback the ones that are currently there.
Any help would be greatly appreciated
cheers
___
From: Quintin
[mailto:[EMAIL PROTECTED]
Sent: 23 May 2005 02:08 PM
To:
'asterisk-users@lists.digium.com'
Subject: sip to sip
Hi
Im trying to put up an sip pbx system for my company
but im getting some problems when Im trying to call from server (
branch A ) to server
I tried that, but it is not working for me with [EMAIL PROTECTED] v1.0 :-(
-b
Mark,
Try writing the sip.conf stanza as:
[192.168.44.23]
context=from-pstn
host=192.168.44.23
type=friend
insecure=very
The 'insecure=very' allows any calls from this IP address to match.
Alistair Cunningham,
Well yes and no. If they have static
IPs then you only need to setup a context as such:
You would assign the following information
on your Branch B server with BranchAs information.
[branchA]
type=friend
defaultip=xxx.xxx.xxx.xxx
context=default
insecure=yes
host=xxx.xxx.xxx.xxx
Hi everybody,
I have an Asterisk working only with IVR functions,
with no AGENTS/QUEUE configurations.
Today, looking at the logs generated by * I find
the next line (in event_log file):
Mar 22 09:59:02 asterisk[15081]: Queued call to
Zap/51/ABCDEFGHI expired without completion after 2
Stankiewicz Michael wrote:
i followed this how-to:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20zaphfc%20install26
having in response no sign of life.
If the module doesn't even get installed, or the kernel does not report
any card as recognized, you could tweak the initialization
Steve Clark wrote:
Joel Duffield wrote:
Hey steve
I remember a tip somewhere where they used a conference room and added all
the users into that conference muted, then kicked them out at the end of the
call. Sorry I can't remember at all where this was but it looked like it
could work. How
Use Apple QuickTime
Best Regards
Erick W.
- Original Message -
From: Brett, Gary [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, May 23, 2005 9:34 AM
Subject: [Asterisk-Users] play gsm files in windows
Try QuickTime for Windows. With this program you can play files with
extension gsm.
I have tried this and it works perfectley!!!
Anatoliy Kounitksiy
Does anybody know of a WINDOWS application (preferably freeware) that will
simply playback asterisk GSM sound files, I don't want to record them,
Brett, Gary wrote:
Does anybody know of a WINDOWS application (preferably freeware) that will
simply playback asterisk GSM sound files, I don't want to record them, just
playback the ones that are currently there.
Any help would be greatly appreciated
You could try Audacity
Eric,
Do you know of one that can convert or record?
.o---o.
Brian Fertig
NOC/Network Engineer
Systems Engineer
Planet Telecom, Inc.
Tampa, FL Office
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
http://www.nch.com.au/wavepad/
Brett, Gary wrote:
Does anybody know of a WINDOWS application (preferably freeware) that will
simply playback asterisk GSM sound files, I don't want to record them, just
playback the ones that are currently there.
Any help would be greatly appreciated
cheers
On Mon, May 23, 2005 at 03:34:44PM +0100, Brett, Gary said:
Does anybody know of a WINDOWS application (preferably freeware) that will
simply playback asterisk GSM sound files, I don't want to record them, just
playback the ones that are currently there.
See the Wiki:
What third party programs are available for parsing the queue_log file
and CDR file? I know about XC-AST, but management would prefer a php
based solution.
What have other admins done to retrieve detailed call information about
the queue system? Anyone develop their own that they don't mind
Try setting
defaultip=192.168.44.23
Too
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of barney
Sent: Monday, May 23, 2005 4:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Getting a Cisco gateway to
thanks a lot,
i've googled around hunting for an answer to my biggest doubt: the
cross-cable.
i understand that it looks like an cat-5 cross-cable and how it has to
be done, but ... why 8 wires ?
i found this image: http://www.gcom.com/home/support/t1crossover.html
and that one:
I am in the process of doing mine now. It works ok here and there not
100% as of yet. But its written in PHP
.o---o.
Brian Fertig
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Johann
Sent:
On Mon, 2005-05-23 at 10:29 -0500, Johann wrote:
What third party programs are available for parsing the queue_log file
and CDR file? I know about XC-AST, but management would prefer a php
based solution.
What have other admins done to retrieve detailed call information about
the queue
On 03:05, Mon 23 May 05, Anton Krall wrote:
I would like to hear stories from people using TAPI, CTI or CallerID
software with Asterisk.
What are you guys using, setup examples, etc.
Has anybody sucessfully integrated SugarCRM with Asterisk and how did you do
it.
Are you running callerid
Johann wrote:
What third party programs are available for parsing the queue_log file
and CDR file? I know about XC-AST, but management would prefer a php
based solution.
What have other admins done to retrieve detailed call information about
the queue system? Anyone develop their own that
Mike,
The cable needs to be a cross-over cable when connecting directly between 2
T1s, bypassing PSTN. One side of isdn has to be configured as TE and the
other as NT. Only 4 wires are needed (not full 8 wires) to build a T1
cross-over. If you are connecting the systems thru pstn, you need
Stankiewicz Michael wrote:
thanks a lot,
i've googled around hunting for an answer to my biggest doubt: the
cross-cable.
i understand that it looks like an cat-5 cross-cable and how it has to
be done, but ... why 8 wires ?
The plug is a standard RJ45 one, but only the 4 inner wires are
See the Wiki:
http://www.voip-info.org/wiki-Asterisk+sound+files
you can doit with SOX on yor server or Linux box
This is how you convert wav files to gsm files used by Asterisk
$ sox inputfile.wav -r 8000 -c 1 outputfile.gsm resample -ql
Best Regads
Erick W
- Original Message -
Hi,
I've connected a TE110P from digium with a 2 E1 to a Siemens PBX using
asterisk from ubuntu linux.
Everything is working as expected. This box is being used as a H323
gateway to the pstn. There are few complains but it is working pretty
well overall.
There is one thing that is bothering
Title: Message
Is there a softphone
for windows that supports IAX?
I can't seem to find
anything out there...maybe im looking in the wrong places...
Jeromy
Grimmett
VoipEmpire.com
[EMAIL PROTECTED]
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Calling all SIP gurus--
I'm trying to register my asterisk to an ISP's SIP gateway.
I'm getting authentification errors.
Here's the results of SIP DEBUG against it's IP.
[I've tweaked all confidential fields so as to protect the innocent
(namely, me).]
--- (9 headers 0 lines)---
Responding
Hi All,
Today I got a couple of ZyXEL Prestige 2000W WiFi phones. I'm having a
problem making SIP calls although I can receive calls just fine. When I
try to make a call the phone makes some sound (like bup bup bup bup bup
bup beep beep) and then I just hear hissing background noise (not too
In article [EMAIL PROTECTED],
Jeromy Grimmett [EMAIL PROTECTED] wrote:
Is there a softphone for windows that supports IAX?
I can't seem to find anything out there...maybe im looking in the wrong
places...
Firefly. See http://www.virbiage.com/firefly/techfaq.php
Cheers
Tony
--
Tony
Torsten Krueger wrote:
On Sat, 21 May 2005, Marcin wrote:
Is there a simply way to allow dialout from ISDN modem to
outside number through Asterisk?
We've done this several times with Junghanns Cards - nearly no problem,
just the normal dialplan entries.
Thanks a lot. I'm amazed it's so easy.
Hey Johann!
Just thought I would mention our upcoming 2.6 release of PhoneCALL. We
already have routines in there that check system logs, and Asterisk logs.
You can check it out at:
http://www.vecsector.com/phonecall
Click on the DEMO on the right-hand side.
User/Pass: demo/demo
Look on
I have them all as IP's. I tried registering with myself and that stayed up
longer then the ones going out on the net, but when I did a reload then that
one died too. I had this problem when I was running 1.0.5 and still have it
with 1.0.7. does this function have any dependancies that I don't
Send me your current app_meetme.c and I will patch it for you so that
you can do it, and then you can provide that patch back to the
community.
On 5/23/05, Michael Blood [EMAIL PROTECTED] wrote:
I have been trying to figure a way to SendDTMF into a MeetMe room using
the Manager API.
I can't
I get this error message in my syslog.
I have searched the list but I can't seem to find a answer that solves the
problem.
chan_sip.c: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed
for '192.168.100.2'
The asterisk server is running dev head. The server is the 100.1 ip. I have
3
R2 use MF for signalling.
Jorge mendoza
Panitaxx wrote:
I have a setup for a 30 incoming channels with telcel. The incoming is
R2, they told me the outgoing is MF not R2. If the other channels are
fxo, you should change your zaptel.conf so you can use zapata.conf
and comment out those
You might try IAXComm. It's a bit immature but works fairly well on Windows, and is cross-platform as well. However, I've found SIP clients to be better generally and better supported. Regards, Bryce Chidester Rhino Equipment Corp. [EMAIL PROTECTED] SIP: [EMAIL PROTECTED] +1 (480) 940-1826
Does anybody know of a WINDOWS application (preferably freeware) that will
simply playback asterisk GSM sound files,
WinAmp with a gsm plugin (google it).
--Luki
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Does anybody know of a WINDOWS application (preferably freeware) that will
simply playback asterisk GSM sound files, I don't want to record them, just
playback the ones that are currently there.
Most people already have it : Quicktime
hth
___
Howdy,
Am Montag, den 23.05.2005, 18:41 +0200 schrieb Marcin:
Torsten Krueger wrote:
On Sat, 21 May 2005, Marcin wrote:
Is there a simply way to allow dialout from ISDN modem to
outside number through Asterisk?
We've done this several times with Junghanns Cards - nearly no problem,
just
The standard Windows recorder will play GSM files. You must make sure
you set the correct values. Codec, playback rate, etc.
Walt Reed wrote:
On Mon, May 23, 2005 at 03:34:44PM +0100, Brett, Gary said:
Does anybody know of a WINDOWS application (preferably freeware) that will
Hi,
How do you transfer (using say blind transfer) a call to an extension
that is currently busy on another call? You don't want the call to be
transferred to voicemail, it must stay in 'hold' until the extension
becomes available, and then immediately ring that phone.
Thanks,
Thomas
Is it possible to use openh323's t38modem with asterisk and spanddsp?
or would hylafax have to be thrown into the mix? If it is possilble
how would I go about getting astersik to see it?
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If I run sox (on Linux), just specifying the input and output files by
with the right extensions, it will convert a raw gsm file to a wav
format file while retaining the gsm compression:
sox vm-youhave.gsm vm-youhave.wav
This is without any additional options. The output file is playable on
Am Montag, den 23.05.2005, 12:07 -0400 schrieb Jorge Verastegui:
Hi,
I've connected a TE110P from digium with a 2 E1 to a Siemens PBX using
asterisk from ubuntu linux.
Everything is working as expected. This box is being used as a H323
gateway to the pstn. There are few complains but it
Michiel... Can you explain a bit more how the files on your website work? I
see they are php agis.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Michiel van Baak
|Sent: Lunes, 23 de Mayo de 2005 10:58 a.m.
|To: asterisk-users@lists.digium.com
That sounds like a call queue...
http://www.voip-info.org/wiki-Asterisk+call+queues
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thomas
Andrews
Sent: Monday, May 23, 2005 10:12 AM
To: Asterisk
Subject: [Asterisk-Users] How do you transfer a call to
make sure you have 'canreinvite=no' in sip.conf
Cheers
Giles
- Original Message -
From: Derek Conniffe [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Monday, May 23, 2005 5:39 PM
Subject: [Asterisk-Users] ZyXEL
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