Hi
All,
Does anybody on
this list know where I can get Digium FXO/Intel 735, Digium FXS
boards in India ?
Similarly I am
also trying to lay my hands on the Grandstream IP phones but have been unable to
find a source.
Thanks
--
Salil
_
but approximately 40 minutes or
longer of
total recording in MSGSM format) It will no longer respond to my DTMF
escape
digits.
In my agi-test.agi file I simply something similar to the following.
$result = $AGI->record_file($wavfile, WAV, 12345 , 7, 1);
As expected it will wait for up to
Hi Dave,
On Sun, 17 Oct 2004, David Croft wrote:
>
> I have tried your config and variations on it but have the same problems.
Sorry to hear that you're still having problems. If you email me your
sip.conf and extensions.conf then I'd be happy to take a look.
> Placing a call out using intervi
Title: (Another) Queue log analyser
Ben,
I would definitely have use for this application, fantastic start. When will you be making the source available?
In my reports I use the CLID to look at calls for different agents i.e. call volume by agent.
Warm Regards
Shad Mortazavi
--
Hi
everyone,
Is anyone using
Xten eyeBeam Video softphone with Asterisk? It supports few types of H.263
codecs for video. I have tried to use it with Asterisk with enabled video
support in sip.conf and allowed h263, but in the moment I click to start
sending video I get this error in Ast
I don't have any soft phones setup, the SNOM receives the calls no problems when the
SNOM tries to dial out it says "Not Found:" on the phone display, on the asterisk
console with "asterisk -vgc" when I try to dialout I only get
chan_sip.c:7561 handle_request: Unknown SIP command 'PUBLISH'
Hello James.
You have "context = sip" in your sip.conf, but you do not have any dialplan
for outgoing calls in that context.
Add to the [sip] context:
include => outgoing
Regards,
Joris
- Original Message -
From: "James Bean" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, Octobe
Or better (to avoid loops) leave the [sip] context intact (like you had
originally) and change in sip.conf your context to 'from-sip', and add in
your extensions.conf:
[from-sip]
include => internal
Regards,
Joris
- Original Message -
From: "Joris Trooster / Interstroom" <[EMAIL PROTECT
Henry Devito wrote:
I am in the process of writing an app to do this with Cisco phones7940/60.
The feature on most PBX's is Page Groups, This allows paging through the
speaker phones.
This sounds interesting.
Can I help in testing?
Are you writing it in C or is it an agi script?
--
respectfully, Jo
Hi,
I do apologise I only have a basic understanding of VoIP and H323, here
is my situation, any help would be very much appreciated.
I am trying to coax my asterisk 1.0.1 box using oh323 0.6.3b with
openh323 13.5 & pwlib v1.6.6 (I purchased 1 G.729 license from digium
and installed it correctly
ind a source.
>
>Thanks
>--
>Salil
>
>
>-- next part --
>An HTML attachment was scrubbed...
>URL:
>http://lists.digium.com/pipermail/asterisk-users/attachments/20041018/8b86b31d/attachment-0001.html
>
>
As explained on the wiki page
http://www.voip-info.org/wiki-Cisco+7940-7960+auto-answer+config you don't
just do a dial(phone1&phone2) you put all the phones through to a
conference, the one drawback of this is that you have to set one of the
cisco's lines to autoanswer, which you probably won't wa
Look at the debug commands. I think this it what you are talking about?
P.S. Please when you start a new thread send a new message and don't reply
to an old one and just change the subject. I sort my messages by header as
I'm am sure others on this list do.
-Original Message-
From: [EM
I would be very interested in the script that allow me to use
saft_asterisk and non-root user on suse 9.1.
Regards Claus
>>> [EMAIL PROTECTED] 10-09-2004 06:22:23 >>>
On Thu, 9 Sep 2004, Martin Mielke wrote:
> Hi all,
>
> due to the rather big email traffic regarding this issue, I decided
to
>
hi
how can I, using mysql sipfriends, allow one user to use g.729 while
disallowing this codec on another user?
thanks
roy
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Am Montag, 18. Oktober 2004 14:21 schrieb Roy Sigurd Karlsbakk:
> hi
>
> how can I, using mysql sipfriends, allow one user to use g.729 while
> disallowing this codec on another user?
>
> thanks
>
> roy
>
There is currently no support for allow clauses unless you decide to add it.
Jens
__
On Mon, 2004-10-18 at 06:51 -0500, Henry Devito wrote:
> Look at the debug commands. I think this it what you are talking about?
>
> P.S. Please when you start a new thread send a new message and don't reply
> to an old one and just change the subject. I sort my messages by header as
> I'm am s
Title: Polycom IP-XXX with shared registration
Hi All,
I am trying to get the Polycom IP-XXX range working with the hint system and subscribe/notify for simple busy or available status.
Has anybody got this working, I found the Wiki which explains the SNOM system but this does not seem to
On Mon, 2004-10-18 at 13:35 +0200, [EMAIL PROTECTED] wrote:
> Hi.
>
> Is possible to caprure calls with asterisk?
>
> I have a calling from onde device to another. While it´s ringing I´d
> wish to capture the calling from another device which has permissions
> to make it. is it possible?
Check
On Mon, 2004-10-18 at 07:35 -0500, Steven Critchfield wrote:
> On Mon, 2004-10-18 at 06:51 -0500, Henry Devito wrote:
> > Look at the debug commands. I think this it what you are talking about?
> >
> > P.S. Please when you start a new thread send a new message and don't reply
> > to an old one a
Hi list!
I found a wiki about asterisk and voice only door phones but I would like
a video door phone.
Are there any video doorphones available that will work with * or that can
be connected to * via an interface?
Thanks for any suggestions!
Remco
___
What about my problem with the RecordFile not recognizing escape digits in
long recorded calls - after about 20 mins?
Anyone?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco Barende
Sent: Monday, 18 October 2004 10:44 PM
To: Asterisk Users List
Sub
I am writing this in C, well trying to write this in C. I will let you know
when it is ready for testing. I found the solution in the WIKI to be clunky
for the install I am proposing to a company that will have 250 phones and
want to page through the phones with no overhead paging.
-Original
On Mon, 18 Oct 2004 13:35:39 +0200 (CEST), [EMAIL PROTECTED]
<[EMAIL PROTECTED]> wrote:
> Is possible to caprure calls with asterisk?
If the calls pass through Asterisk, then yes it's possible. The
keyword for this is "monitor".
You'll find info on this on the Wiki:
http://www.voip-info.org/tik
On Debian SID with gcc 3.3.4. Got app_conference from CVS (today).
I edited the Makefile. Buit no matter what I do,
always the same error appears:
In file included from /usr/include/bits/types.h:31,
from /usr/include/sys/types.h:31,
from /opt/asterisk/include/as
Linus Surguy wrote:
examples of things which I have actually been asked about. There are
a number of protocols based in 2600Hz tones (most US) and 2280Hz
tones (mostly Europe), which are probably still spread quite widely
in low density point-to-point connections. If there is anything you
need,
Dan,
Can I import all the settings from a previous version into 0.9.9b to save
re-inputting all the info?
Dave
=
Hi all,
Thanks to the great work of Steve Kann on the iaxclient library,
now DIAX is able to support the foll
HI Mike,
You wouldn't be trying to connect to Comindico in Australia by any
chance?
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Mike O'Connor
> Sent: Monday, 18 October 2004 02:05
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] chan_h3
If you are for bulk deployment of the phones in large numbers, without
losing your skin along with your shirt, I would recommend buying ATCOM
Phones. You can get them at $55.00 a pop in Bulk and $65 to $70 in
retail. These phones have all the basic features.
Try the link below for an OEM version a
Hi David,
- Original Message -
From: "David J Carter" <[EMAIL PROTECTED]>
Can I import all the settings from a previous version into 0.9.9b to save
re-inputting all the info?
For the moment you can only do that by direct editing of the diax.cfg file
You just need to copy the section [REGI
Hi!
I frequently get errors like "Call failed to go through, reason X" in
/var/log/asterisk/messages
Are the reasons explained anywhere? I did not find any info.
Thanks,
Carlos
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Is there a way to get the Call ID off of a call that runs through *
without loading any kind of billing CDR platform?
If not, I think it would be a great addition to * if the Call ID was
passed as variable (in AGI).
Thanks,
- Darren
___
Asterisk-User
Summary: how to force the alaw codec upon a call between Firefly &
Grandstream BT100?
Not sure whether this is a problem with FireFly, with Asterisk, with
both or just with me ;-)
I have:
disallow=all
allow=alaw
in the general section of my sip.conf.
Using Ethereal on the PC running FireF
Hi Darren,
It is today, check the variables CALLERID, CALLERIDNUM & CALLERIDNAME
/Danny
At 15:58 2004-10-18, you wrote:
Is there a way to get the Call ID off of a call that runs through *
without loading any kind of billing CDR platform?
If not, I think it would be a great addition to * if the Ca
On Mon, 2004-10-18 at 09:58 -0400, Darren Sessions wrote:
> Is there a way to get the Call ID off of a call that runs through *
> without loading any kind of billing CDR platform?
>
> If not, I think it would be a great addition to * if the Call ID was
> passed as variable (in AGI).
It would be
Call-ID as in SIP Call-ID *not* Caller ID.
:)
Thanks though Danny.
On Oct 18, 2004, at 10:02 AM, Danny Froberg wrote:
Hi Darren,
It is today, check the variables CALLERID, CALLERIDNUM & CALLERIDNAME
/Danny
At 15:58 2004-10-18, you wrote:
Is there a way to get the Call ID off of a call that runs t
Steve - it'd be really cool if you knew what you were talking about.
There is a distinct difference between a Call ID and Caller ID.
Guessing by your need to immediately label everyone a 'newbie' and the
fact you don't know what a SIP Call-ID is, I can only speculate as to
your technical expertis
Hello all,
I'm trying setting up an IVR on a Asterisk Soho PBX.
My problem is when I dial the IVR extensión from an Asterisk internal extension
all goes well, but when I dial the external number of the IVR, e.g. 119235656,
the PSTN number of my asterisk, I get the same IVR menu but when I pres
Hi,
I’ve got some Problem sound problems when i try
to connect a second channel on my ISDN line. I use chan_capi with AVM Fritz
card USB.
Did someone have I idea???
Second question:
What is the best solution for ISDN outgoing line
thanks
Realtime in cvs-head allows this.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Jens Kübler
> Sent: Monday, October 18, 2004 7:33 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users]
> I like the fact that the HandyTone has 2 ports, does this mean I can
> configure two different phone numbers, one for each port or is it for tow
> outgoing lines only? Also, I can offer someone a second phone line without
> any additional equipment etc.
It actually only has one port. The second
I've been messing around with my config files and now iax->iax calls do
not work anymore.
I think I did not change anything that would break this but yet it does.
If I do 'iax2 show peers' the remote server show up OK on the console yet
I still get an error.
The server that is placing the call
If anyone has had success in using a Swissvoice IP10S and the SIP firmware
and would like to help me, please contact me offlist. I have successfully
upgraded the phone to SIP and it registers to asterisk, but that's about it.
No dialtone, won't answer a ringing call, etc..
THanks,
Matthew
___
I think the Iaxy gets better every day, however our company for now uses
Sipura Devices. The 2 VOIP ports you're looking for can be found in the
Sipura SPA-2000. The 1 VOIP, 1 PSTN port (like the HT 486) is in the
SPA-3000 and for the cheapest, you can get a 1 VOIP port model in the
SPA-1000. We've
On Mon, 2004-10-18 at 10:18 -0400, Darren Sessions wrote:
> Steve - it'd be really cool if you knew what you were talking about.
>
> There is a distinct difference between a Call ID and Caller ID.
>
> Guessing by your need to immediately label everyone a 'newbie' and the
> fact you don't know wh
Josh Krueger said:
>> I like the fact that the HandyTone has 2 ports...
>
> It actually only has one port. The second port you speak of is a backup
> line. You plug a pots line into the "Line" port, and its a failover line
> in case the internet connectivity goes down. Kind of a good thing to have
Thanks Joe. The polycom phones look pretty good. The only odd thing is
that they make you use a reseller in order to get updates ;(
Sincerely,
Jonathan Miller
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Joe Greco
> Sent: Sunday, October
So Asterisk gurus out there, is there a nice clean way in the dialplan
to determine if the caller is coming from a transferred call, and on
the unavailable context in the dial, instead of going to e-mail go
back to the transferee?
If anyone has this sort of logic or could spit out an extensions
Does the new Realtime config in the CVS head support setting and clearing
MWI for sip clients?
Thanks,
Karl Putz
Forte Communications
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Is there any way to send a caller to a certain context when the caller is
forwarded from another pstn number. We are using * as a voicemail server for
our cusotmers and we are currently providing each vm customer a did to send
the caller to when their line is busy. I would like setup * to take t
> +++ Gunnar Schaller [14/10/04 22:40 +0200]:
>> I have also ast_data and extensions in mysql, works fine for me. Your
>> crash might be something other, I dont't think it's ast_data. Do you
>> have more infos to the crash? Logfiles?
>>
>> Gunnar
>>
> what kind of log files can i provide here
What do you mean? If you mean storing the fact that a phone has a message
waiting in the database, I don't believe that happens. The RealTime is still
in development so it may be added.
Matthew
- Original Message -
From: "Karl H. Putz" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List
Hi Folks,
I shelled out for some licences for the X-Ten Pro phone so that we could
use it whilst away from the office. Only problem seems to be that I
can;t seem to work out how to make it tell me if I have VM without
dialing the VM system.
Any ideas?
Thanks
--
Mark Phillips, G7LTT/KC2ENI
Ran
Hmmm,
I have been working on similar. I thought that while some of the SIP
information is in the database there are still parameters Asterisk is
looking for in the sip.conf.
Looking at the code just now chan_sip.c first looks to the sip.conf to
obtain settings. Once Asterisk sip channel has open
Currently setting mailbox= in sip.conf along with appropriate additional
info is required to set and clear MWI for sip clients.
MySQL peer table does not include the mailbox variable and, while ast_data
does include the mailbox variable, the polling architecture of chan_sip does
not currently work
On Fri, 15 Oct 2004, Matthew Boehm wrote:
What 'should' happen is that if someone dials any extension starting
with 3, Dial attempts to dial it. If there is no such channel, set
DIALSTATUS and goto priority n + 101. Then check result of DIALSTATUS.
If DIALSTATUS is equal to CHANUNAVAIL then got
There is a setting in your sip.conf called mailbox. If you add
this setting to your config it will send a message waiting signal
to your soft phone.
.o---o.
Brian Fertig
Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
813.864.3161x107
You must have some old code. That is not RealTime enabled code that you
posted. RealTime works differently.
Matthew
- Original Message -
From: "Race Vanderdecken" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<[EMAIL PROTECTED]>
Sent: Monday, October
I can't get Asterisk to load this morning after an update and I am not
sure why.
The error I am getting is:
== Parsing '/etc/asterisk/zapata.conf': Found
Oct 18 07:59:46 ERROR[16384]: chan_zap.c:6181 mkintf: Unable to get
parameters
Oct 18 07:59:46 ERROR[16384]: chan_zap.c:9109 setup_zap: Unabl
Thanks Dan.
It seems to crash less too. Is there any way to enter DTMF tones?
Thanks
Peter
-Original Message-
From: Dan [mailto:[EMAIL PROTECTED]
Sent: 18 October 2004 14:48
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] DIAX
Ah, yes. That function is supported in RealTime. In RealTime, the database
table contains 1 column for each possible sip.conf configuration option.
So yes, [EMAIL PROTECTED] will be supported in RealTime SIP.
Matthew
- Original Message -
From: "Karl H. Putz" <[EMAIL PROTECTED]>
To: "As
Hello all,
I'm trying setting up an IVR on a Asterisk Soho PBX.
My problem is when I dial the IVR extensión from an Asterisk internal
extension all goes well, but when I dial the external number of the IVR,
e.g. 119235656, the PSTN number of my asterisk, I get the same IVR menu
but when I press
You should set the ALERT_INFO variable before sending a Dial command to
the phone as per the example below:
exten => 101,1,SetVar(ALERT_INFO=)
exten => 101,2,Dial(SIP/1234)
(Valid values are:
Bellcore-dr1 - standard phone ring cycle
Bellcore-dr2 - "Long-Long"
Bellcore-dr3 - "Short-Short-Long"
Be
> -Original Message-
> From: Matthew Boehm [mailto:[EMAIL PROTECTED]
> Sent: Monday, October 18, 2004 12:11 PM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
> Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] New Realtime config and MWI
>
>
> Ah, yes. That function is
> In other words, I'm
> looking for sample zaptel.conf, zapata.conf, and extensions.conf files
> that are INCREDIBLY simple, allowing outbound (and maybe even inbound
> ;-) calls, so I can get something working, and build from there. Any
Here are a few good starting points:
http://www.fnords.org
Hi all,
I'm trying to put together a list of gear w/prices to implement an
asterisk system. Does anyone know a good place to buy polycom phones?
Their website isn't much help. Specifically looking for IP500 and IP600
phones. Thanks again!
Sincerely,
Jonathan Miller
ACCS.net
Hi,
Wilson Pickett wrote:
I can see why this is not the case. However, if some kind soul would
make a Windows command line "iaxyprovision.exe" I'd be happy.
You can compile it with Cygwin. Or download from:
ftp://ftp.planetarium.com.br/pub/util/voip/iaxyprov/
Download cygwin1.dll if you don't have
Hi,
I've also tried the chan_sccp on my 12 sp last week and found it
extremely unstable.
I have not been able to get caller id on the 12sp to work with
chan_skinny. Have you been able to send it a button template?
Thanks,
-Ryan
On Sun, 2004-10-17 at 21:27, Jason Price wrote:
> maybe gmail is acti
>>
I can't believe how excited I am about a friggin piece of telecom hardware
but this is getting to be adictive. What a geek ;)
>>
I am with you here, it's been a long time since I stayed up days on end
without sleep just to mess with geeky stuff. It's like discovering computers
for the first t
Are you using chan_sip v1.538?
Matthew
- Original Message -
From: "Robert Jackson" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Monday, October 18, 2004 11:25 AM
Subject: RE: [Asterisk-Users] New Realtime config and MWI
> -
Jonathan Miller said:
> Does anyone know a good place to buy polycom phones?
Got an IP500 for $165 with shipping a couple weeks back from
www.pagecomputers.com. Looking now, they're back ordered. I'll likely be
buying 6 or 8 more from them in a week.
Only hitch so far is that I had to get the f
Thanks it I got it working using 1.0.1 i was using the latest cvs version
before which was not patching right.I think now i even got that fixed.
+++ Gunnar Schaller [18/10/04 17:29 +0200]:
>
>
> > +++ Gunnar Schaller [14/10/04 22:40 +0200]:
> >> I have also ast_data and extensions in mysql, wo
Hi Peter,
From: "Whisker, Peter" <[EMAIL PROTECTED]>
Thanks Dan.
It seems to crash less too. Is there any way to enter DTMF tones?
What do you mean by 'enter DTMF tones'?
When you are in a call, you can enter DTMF tones using the
numeric keypad.
Please give me more details.
Best regards,
Dan
I have a couple of Polycom phones, bootrom 2.5.0, SIP 1.3.1.0056. Works
great with Asterisk when I power on the phone. However, after some time,
say an hour, I cannot receive calls on this phone. On Asterisk, when I
do "database show", it does show the phone in there, but it cannot reach
the phone.
no it looks for one but sends it a default one. chan_sccp crashes
asterisk if you hit one of the speedial keys , plus it will not make
any calls after the first one then it unregisters and sits.. i sent a
email off the to chan_sccp dev people waiting to hear what they have
to say.. im willing to do
Henry Devito wrote:
I am writing this in C, well trying to write this in C. I will let you know
when it is ready for testing. I found the solution in the WIKI to be clunky
for the install I am proposing to a company that will have 250 phones and
want to page through the phones with no overhead pa
I have a couple of Polycom phones, bootrom 2.5.0, SIP 1.3.1.0056. Works
great with Asterisk when I power on the phone. However, after some time,
say an hour, I cannot receive calls on this phone. On Asterisk, when I
do "database show", it does show the phone in there, but it cannot reach
the phon
Ah, I see your point and apologize for the misunderstanding. Large
deployments could be a problem, thats for sure.
If you need the MAC address I would suggest a cheap Dell wireless router (as
many others do I am sure) as it logs the MAC addy along with the DHCP addy.
- Original Message
The way the asterisk C code works is to look through a list of
user/extensions/peers to find the search object. O = (N/2)
After looking through the "Linked-List" of "users" and not finding it
Asterisk then looks to the database using the mysql_user function call
to find what it is looking for. It
Thanks,Steve Totaro[EMAIL PROTECTED]www.totarotechnologies.com
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They were for me.. But back up now..
brian
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Monday, October 18, 2004
1:43 PM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users]
Voicepulse down for anyone else?
Thanks,
Steve Tota
Does VoicePulse use Level3 ? If so there is a
reported problem in the Washington DC area that seems to have been
corrected.
- Original Message -
From:
Steve Totaro
To: [EMAIL PROTECTED]
Sent: Monday, October 18, 2004 11:43
AM
Subject: [Asterisk-Users] Voicepulse
On Mon, 2004-10-18 at 13:34 -0400, Steve Totaro wrote:
> Ah, I see your point and apologize for the misunderstanding. Large
> deployments could be a problem, thats for sure.
>
> If you need the MAC address I would suggest a cheap Dell wireless router (as
> many others do I am sure) as it logs the
Could be this:
From: Jon Lewis <[EMAIL PROTECTED]>
Cc: [EMAIL PROTECTED]
Subject: Re: Level 3 US east coast "issues"
On Mon, 18 Oct 2004, Grant A. Kirkwood wrote:
Level 3 experiencing widespread "unspecified routing issues" on the US east
coast. Master ticket 1086844. Anyone have more specific info
Google.com
Thank you,
Steve Maroney
On Mon, 18 Oct 2004, Jonathan Miller wrote:
>
> Hi all,
>
> I'm trying to put together a list of gear w/prices to implement an
> asterisk system. Does anyone know a good place to buy polycom phones?
> Their website isn't much help. Specifically lookin
I am wondering how I can inquire on the current call status and the
codec used for that call.
show channel provides information but am I missing something here, I
really want to know which codec has been used for the channel.
--
Scott Henderson
==
Finite
Well I am a few minutes outside of DC so thats probably it. Quite a bit of
hops through Level3 for me.
110 ms <10 ms10 ms 192.168.2.1
2 <10 ms10 ms10 ms 10.91.40.1
310 ms10 ms10 ms fe-3-6-ar01.howardcounty.md.md02.comcast.net
[68
.87.59.69]
410 ms
Yep, my DID's have been out all night and all day so far. Can't get anyone
on the phone or through their ticket system.
Their site was down for part of the night too.
I think it has something to do with the general issues across the net.
Thanks,
Todd Routhier
Lightwave Technologies, LLC.
Over half of the places I've called off of Google/Froogle that list the
Polycom IP phones do not actually have them in stock, so don't just assume
if you see it listed there you can get it.
MATT---
-Original Message-
From: Steve Maroney [mailto:[EMAIL PROTECTED]
Sent: Monday, October 18,
> -Original Message-
> From: Matthew Boehm [mailto:[EMAIL PROTECTED]
> Sent: Monday, October 18, 2004 12:52 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] New Realtime config and MWI
>
>
> Are you using chan_sip v1.538?
>
Only on my t
Is it like a game?? Let's see who can be an ass first!!!
"Someone asked a question that I can refer them to google for!! I better
mash that reply button so I can be the cool guy!!!"
It's ridiculous to point this user to Google for this question unless you
are pointing him towards a previous thre
Have you tried voipsupply.com? Or even EBay? A lot of ppl sell them New or
Virtually New on EBay, with Buy It Now for instant purchasing. I find that
voipsupply.com is cheaper than the average EBay "Buy it Now" price. I also
find that voipsupply.com does business on EBay as b2tech or something like
Can you give me more info on "general issues across the net" ?
Yeah, VoicePulse seems to be having issues, it's usual though. I wish they
weren't the only place I knew to get flat rate incoming DID's Nationally in
the U.S from.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL P
Try sip show channels or iax2 show channels instead of just "show channels"
Tells you what Codec is in use for the active channels.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott
Henderson
Sent: Monday, October 18, 2004 1:58 PM
To: Asterisk Users M
Hi!
Does anybody know how to convert .gsm file format
to .g729 in order to use it for an IVR system?
Thanks in advance.
Vïctor
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On Mon, 2004-10-18 at 12:54, Paul Dugas wrote:
> Jonathan Miller said:
> > Does anyone know a good place to buy polycom phones?
>
> Got an IP500 for $165 with shipping a couple weeks back from
> www.pagecomputers.com. Looking now, they're back ordered. I'll likely be
> buying 6 or 8 more from th
> Is it like a game?? Let's see who can be an ass first!!!
>
> "Someone asked a question that I can refer them to google
> for!! I better mash that reply button so I can be the cool guy!!!"
Very perceptive -- but don't be fooled by the few loud ones. There are
a good number of smart people on
What kind of user interface is most common when managing an
Asterisk PBX? If this question sounds odd, here is what I am after.
As a computer user who happens to be blind, I see Linux and
FreeBSD as a real God-send as I do lea sure activities as well as my
job which is the feeding
From: Fabian Garcia
[mailto:[EMAIL PROTECTED]
Sent: Friday, October 15, 2004
7:25 PM
To: [EMAIL PROTECTED]
Subject: How to make asterisk send
email notification of voicemessages?
Hi,
I’ve been trying to have Asterisk to email user each
time a voice message is left.
Hi everybody.
With te module oh323 SIP to SIP calls can be autorized in the same * server
in an h323 gatekeeper. I need to do it this way because all the original
system has been built under h323, so, it's easy to integrate the Asterisk
with the rest of the system this way.
My question is: is the
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