I do use just one context for voicemail. This is like the T-Mobile voicemail center (IE: 305-776-MAIL).voicemailservice.gsm is You've reached the voicemail system. If you have a voicemailbox on this system, press star. To leave a message for another customer enter their ten digit number now
something like this in da dialplanexten = 0078,1,Answer()exten = 0078,2,Wait(2)exten = 0078,3,MeetMe(0078,idpMs)exten = 0078,4,Hangup()notes:- change 0078 to your incoming no,
- so when you want to do the conference, just dial the defined extension number- or you can do a blind tranfer to the room
Hello !
I 'm trying T.38 faxig with spandsp using rxfax/txfax as fax terminal.
As another endpoint I 'm using Grandstream HT 486 ATA and a fax machine.
Has anybody success with the HT486 as T.38 terminal ?
ATA as originator: I managed only onetimes a successfull T.38 fax
session. The other
I have used AudioCodes MP 102, 104 and
108, both FXS and FXO. I have also used AudioCodes Mediant 2000. I can tell you
that these are good devices. There are also many other media gateways that have
a lot of facilities, but many of these implement those facilities in software.
AudioCodes
My problem is simple and Ive issued it about 3 weeks
ago. I want the UAs to authenticate with a number to the SIP server. Is this
possible?
For example, I configured an AT-RG613TX (Allied Telesyn
Residential Gateway). In its configuration it is not possible for me to skip
specifying a
HIi am trying to register mgcp gateways(Polycom 501, 601) to asterisk as a call agent, mgcp gateways are not registering to the call agent.Please help me on this if any one knows how to congigure the mgcp.conf on asterisk as well as an MGs.The following are the details of mgcp.conf on
Hi all,
Does Asterisk now support Intel's HMP platforms ? Does it support in
1.4 version ?
There's a special driver for Intel-based HMP hardware+software for ABE.
On the other hand, Asterisk has always been doing HMP :). In fact, I
would say Asterisk's success in HMP is one of the
Hi.
I'm trying to run a Junghanns quadBRI card with mISDN drivers.
I'm able to compile kernel mode user mode mISDN components
as well as chan_misdn.
The misdn-init config properly detects the card and starts
the hfcmulti driver; lsmod shows all required drivers are loaded.
However, the
I am getting this warning:-
Oct 23 15:47:22 WARNING[2124]: db.c:171 ast_db_put: Unable to put value '
192.168.1.12:5060:300:15553695861:sip:[EMAIL PROTECTED]:5060' for key '23'
in family 'SIP/Registry
I checked the file permissions. They are proper. There doesnot seem to be a
visible error. No
Hi,
Our Asterisk server has started core dumping more often, which seems to
be related to when it releases the clone lock, and it stops masquerading
the channel.
So, I wondered if anyone had an idea what causes these masquerading
attempts.
Our system is running: Asterisk
Jordan Novak wrote:
Has anyone created a GUI for this.
I am not sure what youre looking for but we developed a Voicemail Manager:
= http://sip-syndication.com
best regards,
Arnd
___
--Bandwidth and Colocation provided by Easynews.com --
Maybe I found the cause...
My Junghanns quadBRI PCI subsystem ID is 0xB552 (that is,
quadBRI version 2.0), while mISDN expects 0xB550 (quadBRI
version 1.0)
I'm wondering what differences lie in the two boards from a
driver's perspective... I'll try to recompile mISDN by
adding also
Arnd Vehling wrote:
Jordan Novak wrote:
Has anyone created a GUI for this.
I am not sure what youre looking for but we developed a Voicemail
Manager:
= http://sip-syndication.com
best regards,
Arnd
Hello Vehling,
This product of yours, does it manipulate, files on the Asterisk
You can, but it will demand a lot of work. We now work above
introduction of such decision on thin clients under control of
thinstation. As софтофона it is used mozphone (front-end), from the
thin client network_client (back-end).
___
--Bandwidth and
Sorry
You can, but it will demand a lot of work. We now work above
introduction of such decision on thin clients under control of
thinstation. As softphone it is used mozphone (front-end), from the
thin client network_client (back-end).
___
--Bandwidth
Hi Marian,
Thanks for the info, something I'll check into... we have recently
swapped the router over..
We receive a NEW message:
THEM US(new)
So the port forward (inbound) works ok..
We send them a reply:
THEM US(AUTHREQ)
And then they send us the INVAL...
THEM US
Hi Adrian,
yes this problem has happen only sometime, but i dont know exactly
when i has discover this - plase read my comments:
A)Calling directly via public IP's (port 4569 is forwarded on ADSL modem
to asterisk1) - not working
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX
Matt Florell wrote:
On 10/24/06, Steve Edwards [EMAIL PROTECTED] wrote:
On Sat, 21 Oct 2006, Jeremy McNamara wrote:
Steve Edwards wrote:
I have a farm of 7 1u's) with te410p's. When a call comes in, I
call an AGI
that creates a channel variable named GLOBALID. The GLOBALID is 2
digits
Do any *UK* users have an SPA3102 (the newer version of the
SPA3000/Sipura SPA3000) correctly detecting when an incoming PSTN call
has hung up?
I've read everything I can find, including an SPA3000 UK setup PDF that
lists UK ring etc tone settings, port impedances, disconnect tone
settings
Leo Ann Boon wrote:
Gregory Duchatelet wrote:
Hi all,
Does Asterisk now support Intel’s HMP platforms ? Does it support in
1.4 version ?
There's a special driver for Intel-based HMP hardware+software for
ABE. On the other hand, Asterisk has always been doing HMP :). In
fact, I would
Hi guys,
I've an asterisk 1.2.5 runing as production system. Now it becomes
very important to my customer an exact analysis of CDRs for their QoS
to their customers.
I've been analysing the CDRs, and i notice many entries like this:
Calldate |Channel|Source | Clid
|
On Tue, 2006-10-24 at 12:05 +0100, Faris Raouf wrote:
Do any *UK* users have an SPA3102 (the newer version of the
SPA3000/Sipura SPA3000) correctly detecting when an incoming PSTN call
has hung up?
I've read everything I can find, including an SPA3000 UK setup PDF that
lists UK ring etc
On Mon, Oct 23, 2006 at 08:23:12PM -0700, Lee Howard wrote:
If you don't mind saying, what is missing for full t.38 support?
Steve giving Digium a royalty-free license to his GPL software or a
pure-GPL branch of the Asterisk codebase, take your pick.
Why royalty-free? AFAICS there's
On Tue, Oct 24, 2006 at 10:22:33AM +0300, Paul Ianas wrote:
My problem is simple and I've issued it about 3 weeks ago. I want the
UAs to authenticate with a number to the SIP server. Is this possible?
For example, I configured an AT-RG613TX (Allied Telesyn Residential
Gateway).
Hello,
I followed the directions for setting up a user on
Asterisk IRC.
I type the following:
/msg #asterisk username register password
/msg #asterisk set alternative username
And I get /msg Nick Serv help register. I messaged
the moderator a couple of times to no avail.
Hi, all, we have some deployed Asterisk PABX, and we provide our
customers some customized queue report, they report a problem when
agent transfer call, the call duration includes the call time between
the transferer and transferee. They use cisco 7940 phone and use the
phone attended tranfer
Conrad Wood wrote:
On Tue, 2006-10-24 at 12:05 +0100, Faris Raouf wrote:
Do any *UK* users have an SPA3102 (the newer version of the
SPA3000/Sipura SPA3000) correctly detecting when an incoming PSTN call
has hung up?
I've read everything I can find, including an SPA3000 UK setup PDF that
It is brand new so I assume the firmware is the latest?: Software
Version: 3.2.6(GWa) Hardware Version: 1.1.5.
It just doesn't detect real hangups at all. If the person calling hangs
up, either before and after the call is answered, the SPA will
eventually timeout after about 30 seconds
Vitaly,
could you please be more spesific about all you did in order to get tis done, ill do anithing to aconplish this.
Thank You!
On 10/24/06, Vitaly Oborsky [EMAIL PROTECTED] wrote:
SorryYou can, but it will demand a lot of work. We now work aboveintroduction of such decision
Brian Candler wrote:
On Mon, Oct 23, 2006 at 08:23:12PM -0700, Lee Howard wrote:
If you don't mind saying, what is missing for full t.38 support?
Steve giving Digium a royalty-free license to his GPL software or a
pure-GPL branch of the Asterisk codebase, take your pick.
Why
Hi,
Does anyone know a what to use a different codec for calls which are
handset to handset (eg, G711) then when we have calls to the out side
world (via an asterisk server) to use a different codec(eg, G729)?
The idea is to reduce the bandwidth to the server for the majority of
calls,
When you listen to old messages, it would be better if Asterisk
reversed the order so that it starts at the most recent message and
then forwarding goes to the next oldest message, etc... The last
message would be the oldest. This makes more sense for old messages.
Also, is there a way
I think I understood what you want:
1- You want when someone dials an extension, do a Lookup in a database
using FWDCIDNAME
2- Then Dial the number that corresponds to this FWDCIDNAME in database
is that?
If it is so, i would recomend you to use AstDB - Asterisk Berkeley DB
(version1) -
You cant PM anyone if you arent registerd. When you
message nickserv copy exaclty how it is written in the MOTD (except the password
part).
- Original Message -
From:
Eddie
Johnson Jr
To: asterisk-users@lists.digium.com
Sent: Tuesday, October 24, 2006 2:13
PM
Conrad Wood wrote:
It is brand new so I assume the firmware is the latest?: Software
Version: 3.2.6(GWa) Hardware Version: 1.1.5.
It just doesn't detect real hangups at all. If the person calling hangs
up, either before and after the call is answered, the SPA will
eventually timeout after
You have polarity reversal detection and I do not (I did try with it on,
but it didn't help even though there I have measured a polarity reversal
on disconnect)
FWIW: I once had a nasty DSL filter that broke polarity reversal
detection.
You have 3ms On hook speed, I have less than 5ms.
Just a thought ... try reversing the Tip and Ring
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
You have polarity reversal detection and I do not (I did try with it on,
but it didn't help even though there I have measured a polarity reversal
on disconnect)
FWIW: I
On Tue, Oct 24, 2006 at 08:13:04AM -0400, Eddie Johnson Jr wrote:
I followed the directions for setting up a user on Asterisk IRC.
I type the following:
/msg #asterisk username register password
/msg #asterisk set alternative username
This is a strange way to attempt to write to the
ah. Do you have callerid from BT (bt line?). I signed up for something
called BT Privacy or so which is free and gives you callerid.
If you turn on logging (debug) on the sipura it'll log the received
callerid via syslog. Also helpful to check under info Last seen number
or so.
There is CLI
I am getting the following on my server when the problem happens:
Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 207 not
within window 209-209
Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 207 not
within window 209-210
Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received
Henry.L.Coleman wrote:
Just a thought ... try reversing the Tip and Ring
Henry L.Coleman CEO
Henry,
Apologies for answering the wrong message in my last post. I thought I
was answering the one from Conrad. Sorry!
By reversing the Tip and Ring you mean physically in the wiring or
somewhere
Just as a follow up.. on the OTHER server that is connected I'm seeing:
chan_iax2.c: Received VNAK: resending outstanding frames
On 10/24/06, Matt [EMAIL PROTECTED] wrote:
I am getting the following on my server when the problem happens:
Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received
Hi everybody!
Is it possible to order Asterisk to distribute calls to ZAP channels belonging
to one channel group (also called dial group) in any other way than in
sequential order (1,2,3 etc.)?
I would like to distribute calls equally between all available PRI spans.
Thanks in advance for
Hi Marian,
I think we worked it out... (time will tell now)..
Our gateway people were able to put IAX2 debug on, and then filter the
trace (manually!) so that we could compare call-flow.
Heres what they saw:
lon-pbx-backup-1*CLI
Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX
Hello All,I have several soundfiles that are recorded ub 44100Hz, 16-bit Mono. What is the best way and right tools to use to downsample these to 8000Hz so that they can be used with Asterisk. I've tried using sox with the -r switch and Audacity on the mac and Goldwave on Windows and they all
I'm not seeing any caller id in the syslog nor the last seen number
thing. (which helpfully just says , :-)
I'd be pretty sure that the device doesn't detect the cli. My one does
list the number under the 'last seen number thing'.
What sort of line is it? Straight BT? telewest? Some
Hi,
is there anybody who knows how to set music on hold for an ISDN channel?
In zapata.conf there is musiconhold parameter. Is there something
similar for misdn.conf?
TIA
Giorgio Incantalupo
___
--Bandwidth and Colocation provided by Easynews.com
Define diferent trunks for every PRI span and use RANDOM on your
dialplan before dialing!
On 10/24/06, Asterisk [EMAIL PROTECTED] wrote:
Hi everybody!
Is it possible to order Asterisk to distribute calls to ZAP channels belonging
to one channel group (also called dial group) in any other way
Can someone tell me if this indicates a problem? What does it mean when a macro
exits != 0 ?
Spawn extension (macro-syst_FindAppServer, s, 5) exited non-zero on
'SIP/xxx.yyy.142.186-b7515f98' in macro 'syst_FindAppServer'
Thanks,
Doug.
___
Hi,
I'm having no trouble using:
sox yourinputfile.wav -r 8000 -c 1 youroutfile.al resample -ql
Regards,
Tristan
Nate Criss a crit:
Hello All,
I have several soundfiles that are recorded ub 44100Hz, 16-bit Mono.
What is the best way and right tools to use to downsample these to
8000Hz
We had/have this problem, too - we eventually got it working (just by
constantly rebooting it), but it seems that something's not working
properly somewhere..
Can you look in your phone's boot log and see if you are getting any
errors? We were seeing errors relating to the phone not
I will sign in with good experiences with MP124 and Mediant 1000. I have an
MP202 under test.
--On Tuesday, October 24, 2006 10:10 AM +0300 Paul Ianas
[EMAIL PROTECTED] wrote:
I have used AudioCodes MP 102, 104 and 108, both FXS and FXO. I have also
used AudioCodes Mediant 2000. I can
Conrad Wood wrote:
I'm not seeing any caller id in the syslog nor the last seen number
thing. (which helpfully just says , :-)
I'd be pretty sure that the device doesn't detect the cli. My one does
list the number under the 'last seen number thing'.
What sort of line is it? Straight BT?
Yep, just swop the two wires. Sometimes the Tip and Ring get reversed
and most loop start interfaces don't really care (they work either way).
It's worth a try since if the disconnect is a reverse polarity flash then
the card may see not see this condition as it is already reversed.
I have a
Ignacio Ortega A. a écrit :
*Vitaly,*
could you please be more spesific about all you did in order to get tis
done, ill do anithing to aconplish this.
Have a look at the mailing list archive of MozPhone (moziax.mozdev.org):
back in August, Machula Viach made modifications in order to run
It means that it exited at priority 5 of the s extension in that context. (.. s, 5)It does not inherently mean anything bad, depending on if that is an accurate exit point in your Macro.Anthony
On 10/24/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Can someone tell me if this indicates a problem?
Steve and everyone...
I am using spandsp snapshot from oct 12, 2006. I am using asterisk 1.2.13.
When I am sending faxes I am only getting partial pages.
Internally using an IAXY connected to the fax machine I get 1 page of 3.
Extenally to a fax service using TDM2401E card I get the same
Hi,
I have a Polycom 501 that is currently unusable because I started a
firmware and sip upgrade that I can't complete.
The Ubuntu box address is set static at: 192.168.1.101.
The phone address is set static at 192.168.1.51.
The phone settings for the server menu are:
Server Type: Trivial FTP
Hi all, the lists seems to be littered with disconnect problems using
various equipment (TDM 400,Linksys etc etc.)
My question is very simple and could make for good solution to Asterisk
users.
Since * can detect various tones according to different country standards
would it be possible to
Title: Voicemail help
I would like to setup Asterisk for voicemail with CallManager 3.3(5). I would like to know what would be the best Distro of Linux to use and version, what version of Asterisk works best to interact with CallManager, and what H323 ChannelType works. As you probably read
Hello Dovid,
My firsts time doing this what is MOTD? I
also tried what you suggested /msg #asterisk username register and it did not
work. I must not be doing something correct because I had a couple of other people
try and not successful. Any suggetions?
Ed
From: [EMAIL
I use Fedora Core and it works fine. I'm not connected to call manager though. which version of Asterisk are you using?
-- Original message -- From: "Ward, Bill" [EMAIL PROTECTED]
I would like to setup Asterisk for voicemail with CallManager 3.3(5). I would like to know
I have tried both FC5 and 6. Asterisk works fine in both instances, for
example when i connect with an IAX2 softclient like Idefisk. I only encounter
the problem when I try to go through CCM.
-Original Message-
From: [EMAIL PROTECTED] on behalf of [EMAIL PROTECTED]
Sent: Tue
Carla Schroder wrote:
On Monday 23 October 2006 17:38, Edwin Lam wrote:
Re: [asterisk-users] Polycom SP4000 ftp problem
From: Edwin Lam [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Carla Schroder wrote:
Sooo...stick with tftp? :) Seriously, that's what it's for. tftp isn't
Edwin Lam wrote:
Carla Schroder wrote:
On Monday 23 October 2006 17:38, Edwin Lam wrote:
Re: [asterisk-users] Polycom SP4000 ftp problem
From: Edwin Lam [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Carla Schroder wrote:
Sooo...stick with tftp? :) Seriously, that's what it's for.
Hi Eddie,
Connect to irc.freenode.net, and then type this:
/msg nickserv register password
nickserv will tell you that your nick is now registered.
Then type this:
/j #asterisk
Say hi to CunningPike when you get there.
CP
On 24-Oct-06, at 1:12 PM, Eddie Johnson Jr wrote:
Hello Dovid,
We have a problem where callerid works 50% of the time on both lines. What
we are seeing in the logs is:
Oct 23 02:44:00 WARNING[28207] chan_zap.c: CID timed out waiting for ring.
Exiting simple switch
Oct 23 05:09:25 NOTICE[28840] chan_zap.c: Got event 17 (Polarity
Reversal)...
Oct 23 05:09:27
Unless of course the nick your using is used already in which case you will
have to change it with /nick newnick
-Original Message-
From: [EMAIL PROTECTED] on behalf of Anthony Rodgers
Sent: Tue 10/24/2006 1:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
This seems like a piece members of this list would find interesting...
===
There is growing concern over the interaction of VoIP systems
with the legacy PSTN, and the transmission of caller identity
data--most notably, Caller ID on the PSTN. It is not always
Anthony,
Thanks :)
Ed
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Rodgers
Sent: Tuesday, October 24, 2006 2:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Becoming a User on IRC
Hi Eddie,
On Tue, Oct 24, 2006 at 02:57:38PM -0400, Jay R. Ashworth wrote:
This seems like a piece members of this list would find interesting...
Further down, he notes:
The PSTN cannot turn on a dime and restrict ANI/CLID from many
clients using whitelist filters. Caller ID manipulation
On 2006-10-24 10:32:09 -0700, Henry.L.Coleman
[EMAIL PROTECTED] said:
Hi all, the lists seems to be littered with disconnect problems using
various equipment (TDM 400,Linksys etc etc.)
My question is very simple and could make for good solution to Asterisk
users.
Since * can detect various
This is probably the last time for a while is it
possible to develop a quick and simple solution for this problem Audio
works well, routing between SIP and h323... fine, but video still not providing
any signalling.
Thanks
___
(sorry for the second post)
We have a problem where callerid works 50% of the time on both lines. What
we are seeing in the logs is:
Oct 23 02:44:00 WARNING[28207] chan_zap.c: CID timed out waiting for ring.
Exiting simple switch
Oct 23 05:09:25 NOTICE[28840] chan_zap.c: Got event 17 (Polarity
On Tue, 2006-10-24 at 15:12 -0400, Jay R. Ashworth wrote:
On Tue, Oct 24, 2006 at 02:57:38PM -0400, Jay R. Ashworth wrote:
This seems like a piece members of this list would find interesting...
Further down, he notes:
The PSTN cannot turn on a dime and restrict ANI/CLID from many
Hi,
I having a problem with my asterisk, it is overwriting the Proxy Via
header with its own ip address and answering to the Proxy with the
modified header, so the Proxy is having problems to route the response.
I've tried with different versions of asterisk and nothing is changing,
and if I try
Hi,
I have a customer who experiences, once in a while, one-way audio...
That is... they can hear the person they called, but the person can
not hear them.
The customer is connected via IAX2 to our softswitch.
On the customer's end I have the following config in iax.conf:
[general]
bindport =
I have an asterisk box at a remote location (which I will call remote),
which registers to my local asterisk box (I'll call that one local), and
uses that to route calls to the outside world. The problem I am having
is that the remote location needs to lie about it's callerid sometimes,
Hi i got lots of this from the asterisk console what does this mean?format_wav.c:247 update_header: Unable to find our positionasterisk console:Oct 24 16:39:19 WARNING[4432]: format_wav.c:247 update_header: Unable to find our position
Oct 24 16:39:19 WARNING[2812]: format_wav.c:247 update_header:
Eric ManxPower Wieling wrote:
rename bootrom.ld to something else like bootrom.ld-disabled.
did that. it hung on sip.ld, rename sip.ld, it hung on
phone1.cfg. seems like if the file is bigger than say 1k.
it'll hang.
--
Edwin Lam [EMAIL PROTECTED]
Systems Engineer, Office General, Inc.
Ph:
Hello list,
I am encountering a bit of a problem in working with incoming calls with a
TDM2400 and * 1.2.4; when a call comes in, * will correctly detect the
ringing, but will sometimes report multiple Attempting native bridge.
What I do is basically that when a call comes in, I dial a
Hi all
I have installed 1.2.12.1 in FC5 with libpri.1.2.4
when i start
iam getting the following error and it quits
== Registered channel type 'Local' (Local Proxy Channel Driver)[chan_zap.so]Oct 23 16:16:07 WARNING[11084]: loader.c:325 __load_resource: libpri.so.1.0: cannot open shared
On Mon, Oct 23, 2006 at 04:17:45PM +0530, ram wrote:
Hi all
I have installed 1.2.12.1 in FC5 with libpri.1.2.4
when i start
iam getting the following error and it quits
== Registered channel type 'Local' (Local Proxy Channel Driver)
[chan_zap.so]Oct 23 16:16:07 WARNING[11084]:
J
I have seen your paper on the caller ID issue I can't agree with you more.
On 10/24/06, J. Oquendo [EMAIL PROTECTED] wrote:
On Tue, 2006-10-24 at 15:12 -0400, Jay R. Ashworth wrote:
On Tue, Oct 24, 2006 at 02:57:38PM -0400, Jay R. Ashworth wrote:
This seems like a piece members of this
Did you compile and install these in the correct order:
zaptel
libpri
asterisk
CP
On 23-Oct-06, at 5:47 AM, ram wrote:
Hi all
I have installed 1.2.12.1 in FC5 with libpri.1.2.4
when i start
iam getting the following error and it quits
== Registered channel type 'Local' (Local Proxy
When I call meetme:
exten = 1000,1,Answer
exten = 1000,n,Meetme(|||d)
Asterisk is complaing with:
-- Executing Answer(IAX2/xxx.yyy.142.204:4569-2, ) in new stack
-- Executing MeetMe(IAX2/xxx.yyy.142.204:4569-2, |||d) in new stack
-- Playing 'conf-getconfno' (language 'en')
But if we have asterisk and add on Steve's code wouldn't it (suppor to recieve a t.38 fax call and have spandsp decode it) work? What does Steve granting a license to Digium have to do with it? I don't care if Asterisk and the fax support don't come from the same place.
On 10/23/06, Lee Howard
On Wed, 2006-10-25 at 04:44 +0800, Mark Quitoriano wrote:
Hi i got lots of this from the asterisk console what does this mean?
format_wav.c:247 update_header: Unable to find our position
asterisk console:
Oct 24 16:39:19 WARNING[4432]: format_wav.c:247 update_header: Unable
to find
When you listen to old messages, it would be better if Asterisk reversed
the order so that it starts at the most recent message and then
forwarding goes to the next oldest message, etc... The last message
would be the oldest. This makes more sense for old messages.
Some people like
Have you tried using just ftp ?
- Original Message -
From: Marlin Unruh [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, October 24, 2006 7:28 PM
Subject: [asterisk-users] need help using tftp for polycom 501
Andrew Joakimsen wrote:
But if we have asterisk and add on Steve's code wouldn't it (suppor to
recieve a t.38 fax call and have spandsp decode it) work? What does
Steve granting a license to Digium have to do with it? I don't care if
Asterisk and the fax support don't come from the same
Hi there, I'm tring a basic asterisk settings.
I have a asterisk 1.2.7.1 running on a
I have a net with two computers and a router.
The router IP in the local net is 192.168.1.1,
The first pc has IP: 192.168.1.3 name datile3 . SO GNU Linux.
the second pc has IP: 192.168.1.4 name fissun . SO GNU
Cisco are worse. With the example files we were able to deploy and configure the Polycom phones with the newest firmware.With the sample files AND Cisco tech support we weren't even able to get them up to the latest version.
On 10/23/06, Dean Collins [EMAIL PROTECTED] wrote:
Lol, glad to hear it
Douglas Garstang wrote:
When I call meetme:
exten = 1000,1,Answer
exten = 1000,n,Meetme(|||d)
Asterisk is complaing with:
-- Executing Answer(IAX2/xxx.yyy.142.204:4569-2, ) in new stack
-- Executing MeetMe(IAX2/xxx.yyy.142.204:4569-2, |||d) in new stack
-- Playing
Doug wrote:
When I call meetme:
exten = 1000,1,Answer
exten = 1000,n,Meetme(|||d)
Asterisk is complaing with:
-- Executing Answer(IAX2/xxx.yyy.142.204:4569-2, ) in new stack
-- Executing MeetMe(IAX2/xxx.yyy.142.204:4569-2, |||d) in new
stack
-- Playing 'conf-getconfno'
Hi Tzafrir,
Thanks for your quick reply, I will look some downloads and install it as per your suggestion. I am using CentOS 4.3, kernel-2.6.9-34.01.EL
Thanks again.
Angel
- Original Message From: Tzafrir Cohen [EMAIL PROTECTED]To: asterisk-users@lists.digium.comSent: Monday, October
Hi there, I'm tring a basic asterisk settings.
I have a asterisk 1.2.7.1 running on a
I have a net with two computers and a router.
The router IP in the local net is 192.168.1.1,
The first pc has IP: 192.168.1.3 name datile3 . SO GNU Linux.
the second pc has IP: 192.168.1.4 name fissun . SO GNU
Hi All
I'm trying to understand how I would send my fax ?
If I use Word or what ever word processor or even an email client to
create what I want faxed.
I have *asterisk setup with and FXO Gateway that will make the call to
the fax number I dial
SIP extension 320 is the FXO gateway.
Andrew Joakimsen wrote:
But if we have asterisk and add on Steve's code wouldn't it (suppor to
recieve a t.38 fax call and have spandsp decode it) work? What does
Steve granting a license to Digium have to do with it? I don't care if
Asterisk and the fax support don't come from the same
You can use the fax server Hylafax ( http://www.voip-info.org/wiki/index.php?page=Asterisk+IAXmodem ) with IAXmodem (
http://iaxmodem.sourceforge.net/howto.php )You really don't want to be sending faxes over the internet via VoIP providers, not yet because there is no t.38 support for that. As
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