Robert Bielik skrev:
Ok, now pretty much everything is up 'n running, however when I try to send
an ANSWER (or any) command to *, it replies with
org.asteriskjava.manager.response.ManagerError Permission Denied. In
manager.conf for the *-java client, I have
read =
Hello,
I remember having a similar problem sometime back but I do not remember
the solutions. My Xorcom FXS bank is not showing up in /proc.
Here some helpful output:
r...@pbx1:~# zaptel_hardware
usb:002/006 xpp_usb- e4e4:1161 Astribank-modular
USB-firmware
pci::01:05.0
On Mon, Nov 02, 2009 at 12:19:41PM +0530, DHAVAL INDRODIYA wrote:
hello all,
i have 2 asterisk boxes on that 1 have public IP Address and another is only
have local IP address
now on public IP there are some 7 DID forwarded , now i want to forward 3
DID out of 7 DID to local machine we
On Mon, Nov 02, 2009 at 02:49:05AM -0500, ALEX BALASHOV wrote:
In a manner of speaking.
Top-posting, on top of your other sins.
Please spare us this capital punishment.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406
I have Asterisk and Digium AEX808B
What are please the commands that i can run on Asterisk to get the information
about the connected lines from PSTN to see the parameters of them
and as well the corresponding files in Asterisk that i can change into, to tune
these parameters to be matched
On Mon, Nov 02, 2009 at 09:33:21AM +0100, Loic Didelot wrote:
Hello,
I remember having a similar problem sometime back but I do not remember
the solutions. My Xorcom FXS bank is not showing up in /proc.
Here some helpful output:
r...@pbx1:~# zaptel_hardware
usb:002/006 xpp_usb-
Tzafrir Cohen wrote:
Top-posting, on top of your other sins.
Please spare us this capital punishment.
An entirely fair point.
Nevertheless, I eagerly await your similarly convicted petitions aimed
at curbing illiterate, obnoxious and indolent attempts to get others
to do extensive work on
Alex,
You forgot to clip the extra from the quote, shame on you!
On Mon, Nov 2, 2009 at 9:47 AM, Alex Balashov abalas...@evaristesys.com wrote:
Tzafrir Cohen wrote:
Top-posting, on top of your other sins.
Please spare us this capital punishment.
An entirely fair point.
Nevertheless, I
I have the same result with Asterisk 1.4.21 on a Debian Lenny server
--
-- --
Marc LEURENT
lf...@leurent.eu
Le mercredi, 28 octobre 2009 12.27:59, Marc Leurent a écrit :
Hello, when I remove a peer from my sip.conf and just do a reload, the peer
is still ping with SIP OPTIONS until I restart
On Sun, Nov 01, 2009 at 04:13:22PM -0500, cov...@ccs.covici.com wrote:
Hi. When I dial a Dahdi extension using asterisk 1.6.0, and there is no
answer, the extension hangs up, but the dial status is busy instead of
no answer. How do I get this to work -- do I need to update dahdi? The
card
Hi,
I am using this version zaptel-1.4.12.9.svn.r4653.
Best regards,
Loïc.
On Mon, 2009-11-02 at 10:38 +0200, Tzafrir Cohen wrote:
On Mon, Nov 02, 2009 at 09:33:21AM +0100, Loic Didelot wrote:
Hello,
I remember having a similar problem sometime back but I do not remember
the solutions.
hello friends
friend i had just finished my chapters of asterisk. ill be
configuring asterisk in for home for r/d purpose. i am having p4 machine
with 1 GB RAM, ill be configuring asterisk on centos 5.3, the only doubt
which i am having is which hardware ill have to buy to configure asterisk.
i
Hi,
can I reset and reload the firmware while the kernel modules are loaded?
Should the files appear in /proc once the firmware has been loaded
correctly or do I need to unload/reload the kernel modules?
Best regards,
Loic.
On Mon, 2009-11-02 at 10:38 +0200, Tzafrir Cohen wrote:
On Mon, Nov
aster...@opensourcesolution.in wrote:
hello friends
friend i had just finished my chapters of asterisk. ill be configuring
asterisk in for home for r/d purpose. i am having p4 machine with 1 GB
RAM, ill be configuring asterisk on centos 5.3, the only doubt which i
am having is which
When we can expect to have a res_fax and res_fax_degium module for asterisk
V 1.6.2
Regards
Khaled Chehab
NGN Eng.
Untitled
Operations Office - Lebanon
Office : +961 1 868686 ext 115
Mobile: +961 3 045212
E-mail: kche...@xplorium.com
On Mon, 2009-11-02 at 09:37 +, aster...@opensourcesolution.in wrote:
hello friends
friend i had just finished my chapters of asterisk. ill be
configuring asterisk in for home for r/d purpose. i am having p4
machine with 1 GB RAM, ill be configuring asterisk on centos 5.3, the
only doubt
We have the 870 working great in our test environment so far.
Garth van Sittert
BSC (Physics Comp Sci)
Technical Director
BitCo
08600 24826
www.bitco.co.za
--[ UxBoD ]-- wrote:
Anybody tried one with Asterisk yet ? Views ?
Best Regards,
- Garth van Sittert ga...@bitco.co.za wrote:
| We have the 870 working great in our test environment so far.
|
|
| Garth van Sittert
| BSC (Physics Comp Sci)
| Technical Director
| BitCo
| 08600 24826
| www.bitco.co.za
|
|
|
| --[ UxBoD ]-- wrote:
| Anybody tried one with Asterisk yet
Hi,
you can do print the dialstatus to the console e.g.:
exten = s,n,NoOp(${DIALSTATUS})
More info:
http://www.voip-info.org/wiki/view/Asterisk+cmd+NoOp
Bye,
Patrick
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Hello,
Does anyone know where I can get an up to date guide on installing
CDR_MSQL?
VOIP-Info has old information.
Many thanks
Dan
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asterisk-users mailing list
Hello,
Let me explain a scenario
There are different Asterisk Servers at different Remote locations.
Recording in different formats for FIVE seconds reveals that
Format : Size
wav : 84 KB
gsm : 8.3 KB
sln : 84 KB
It can be recorded in any format. This is size for five seconds only. We
need to
Hi,
at first: why do you use capitals for your name? Don't do that if you
don't have a very good reason.
You can convert wav to mp3 on the recording server and then send it to
the central system.
Bye,
Patrick
On Mon, Nov 2, 2009 at 1:11 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.com wrote:
hi,
i have new supermicro server (centos5, 2.6.30.9/2.6.27.34/2.6.18-distro
kernels, wanpipe 3.5.6)
card is:
1 . AFT-A101-SH : SLOT=4 : BUS=8 : IRQ=11 : CPU=A : PORT=1 : HWEC=32 : V=36
and i have this in log
irq 17: nobody cared (try booting with the irqpoll option)
Pid: 0, comm: swapper Not
Thanks Patrick.
First: I dont do that intentionally.
Thanks for suggestion. Let me investigate it.
On Mon, Nov 2, 2009 at 5:34 PM, Patrick Plattes patr...@erdbeere.netwrote:
Hi,
at first: why do you use capitals for your name? Don't do that if you
don't have a very good reason.
You can
After conversion from .wav to .mp3 the size remains almost the same.
On Mon, Nov 2, 2009 at 5:46 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.comwrote:
Thanks Patrick.
First: I dont do that intentionally.
Thanks for suggestion. Let me investigate it.
On Mon, Nov 2, 2009 at 5:34 PM, Patrick
On Mon, Nov 02, 2009 at 10:38:45AM +0100, Loic Didelot wrote:
Hi,
can I reset and reload the firmware while the kernel modules are loaded?
Yes, there shouldn't be a problem with that.
Should the files appear in /proc once the firmware has been loaded
correctly or do I need to unload/reload
On Mon, Nov 02, 2009 at 05:11:53PM +0500, ABBAS SHAKEEL wrote:
Hello,
Let me explain a scenario
There are different Asterisk Servers at different Remote locations.
Recording in different formats for FIVE seconds reveals that
.WAV (wav49, wav/gsm) should be playable by most systems.
--
Hello,
I now have this which looks better.
r...@pbx1:~# lsusb
Bus 002 Device 010: ID e4e4:1162
Bus 002 Device 001: ID :
Bus 001 Device 004: ID 0403:e6c8 Future Technology Devices
Bus 001 Device 003: ID 0403:6001 Future Technology Devices
Bus 001 Device 001: ID :
On Mon, Nov 02, 2009 at 02:28:36PM +0100, Loic Didelot wrote:
Hello,
I now have this which looks better.
r...@pbx1:~# lsusb
Bus 002 Device 010: ID e4e4:1162
Bus 002 Device 001: ID :
Bus 001 Device 004: ID 0403:e6c8 Future Technology Devices
Bus 001 Device 003: ID 0403:6001
svn 6466 from trunk.
Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Sun, Nov 01, 2009 at 04:13:22PM -0500, cov...@ccs.covici.com wrote:
Hi. When I dial a Dahdi extension using asterisk 1.6.0, and there is no
answer, the extension hangs up, but the dial status is busy instead of
no
On Sun, 2009-11-01 at 18:50 -0500, Thomas Perron wrote:
Where is everyone located? I am in Virginia, USA
There are literally thousands of people on this mailing list, so I doubt
it's worth having everyone tell you where they're from. That being
said, I'm also in Virginia (near Fredericksburg),
FWIW, I convert all of my files to WAV for Web reading using SOX. Sox will
let you put all your files into the compressed gsm format for storage (sox
file.wav file.gsm), then you can just reverse the process for presentation
(sox file.gsm /tmp/file.wav)
_
From:
As I understand this thread, you want two different contexts based on the
number you dial. If you dial 1703... the big10 context should be executed.
If 1567... then Cleveland is executed. Is this correct? If so
Then this is what the two lines in [default] should read:
exten =
Hi,
is there any way from outside change x,y an z after a call is bridged?
maybe with AMI interface?
best regards
Thomas
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To UNSUBSCRIBE or update
Hello,
Does anyone know how to set the remote party id?
Thanks
Dan Journo
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On Mon, 2 Nov 2009, Patrick Plattes wrote:
you can do print the dialstatus to the console e.g.:
exten = s,n,NoOp(${DIALSTATUS})
More info:
http://www.voip-info.org/wiki/view/Asterisk+cmd+NoOp
A better practice would be to use verbose() -- an application with
greater functionality written
because read=agi lets you read agi events, not send agi actions, agi in
write= must be set too if you want to send agi commands.
On Mon, Nov 2, 2009 at 3:25 AM, Robert Bielik robert.bie...@xponaut.sewrote:
Robert Bielik skrev:
Ok, now pretty much everything is up 'n running, however when I
Hi all,
I am wondering what people are doing for security when registering IP phone's
remotely if you do not have the equipment to do a VPN tunnel at the remote
site. The phone I would be working with mainly is the Polycom lineup.
Thanks,
Connor Spiess
dnsmgr.conf:
enable=yes
refreshinterval=300
regards.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH
Sent: Friday, October 30, 2009 3:28 AM
To: 'Asterisk Users Mailing List -
You can restrict on IP address, MAC address and port type and that's just
what I know. If someone want's through bad enough you're going to have a
problem, but you can at least slow down or stop casual hackers.
_
From: asterisk-users-boun...@lists.digium.com
Hi,
Try Berofix / beronet - tested with Tylersburg Supermicro mb's - works perfectly well.
Jacek
-Original Message-From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of marek cervenkaSent: Monday, November 02, 2009 1:38 PMTo:
Hello Connor,
You might be able to start with this link:
http://www.voip-info.org/wiki/view/Asterisk+sip+md5secret
You can also go even further if you know the IP address of where the
phones are coming from by using the permit/deny options:
On Fri, 30 Oct 2009, hbk wrote:
Hi,
I have played with the 820 for some weeks, mostly love it excellent speech
quality. Even got the mini browser running
showing my favorite webcam, this could be put to real use too:)
Issues so far:
Some embarrassing crashes while speaking, was able to
Probably after 1.6.2 has been officially released beyond the release
candidate stage.
Thanks,
--Warren Selby
On Mon, Nov 2, 2009 at 4:14 AM, Khaled W Chehab kche...@xplorium.comwrote:
When we can expect to have a res_fax and res_fax_degium module for
asterisk V 1.6.2
Regards
What version of asterisk are you installing?
Thanks,
--Warren Selby
On Mon, Nov 2, 2009 at 5:59 AM, Dan Journo d...@keshercommunications.comwrote:
Hello,
Does anyone know where I can get an up to date guide on installing
CDR_MSQL?
VOIP-Info has old information.
Many thanks
Dan
Just my .02; You shouldn't use outlying features like fax on rc releases -
these aren't usually but can be (b)leading edge stuff.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Monday, November 02, 2009
There is something called Opensky that claims to allow making and receiving
skype calls
on your SIP device (including Asterisk). However I haven´t tested it yet.
http://latestgeeknews.blogspot.com/2009/02/opensky-skype-interface-gateway-who.html
2009/11/1 hbk fo...@online.no
Hi,
I get
Remco Barendse wrote:
On Fri, 30 Oct 2009, hbk wrote:
Hi,
I have played with the 820 for some weeks, mostly love it excellent speech
quality. Even got the mini browser running
showing my favorite webcam, this could be put to real use too:)
Issues so far:
Some embarrassing crashes
The mortality rate on power supplies, diplays and the number or
broken
receiver hook swicthes on the lot of Snom 360's i bought 3 years
ago is
outright embarrassing.
That's odd. We've had Snom 190s, 320s, and 360s running day in day out
for years with not a single issue.
Would say
--- On Sat, 10/31/09, Martin asteriskl...@callthem.info wrote:
On Sat, Oct 31, 2009 at 5:27 AM,
Tzafrir Cohen tzafrir.co...@xorcom.com
wrote:
I'm not sure if handling of overlap hasn't changed
since.
But can you provide a trace of how Asterisk sees
things? e.g. 'pri
intense debug
When executing the following command :
[r...@nagios ~]# /usr/local/nagios/libexec/check_nrpe -H ip_address -c
check_asterisk_peers
I get the following output :
NRPE: Unable to read output
Somewhere Nagios does not have enough rights to question Asterisk about
the sip peers.
These are the
On 11/02/09 07:28, Steve Edwards wrote:
On Mon, 2 Nov 2009, Patrick Plattes wrote:
you can do print the dialstatus to the console e.g.:
exten = s,n,NoOp(${DIALSTATUS})
More info:
http://www.voip-info.org/wiki/view/Asterisk+cmd+NoOp
A better practice would be to use verbose() -- an
We were thinking about doing something similar as well. A lot of people are
asking for this. If there is anybody else interested, we could share the
load
I was thinking about creating a context like @agents, so that when you do
the log-on you basically add Local/1...@agents as a member of the
To avoid boring everybody else to death with the discussion, I created a
mailing list for that on Google Groups - see http://tinyurl.com/yjtf62s
Thanks
l.
2009/11/2 Lenz Emilitri lenz.lo...@gmail.com
We were thinking about doing something similar as well. A lot of people are
asking for this.
I have sent this before but here is how I do agent login and queue:
; Agent login logout
exten = *20,1,Verbose(2,Doing agent login/logout)
exten = *20,n,Answer()
exten = *20,n,wait(.0.5)
exten = *20,n,Read(AgentNumber,agent-user)
exten =
Hi,
Does anyone have an up to date guide for setting up fax 2 email with asterisk?
Thanks
Dan
IT Maintenance Contract Clients can now access our Instant Chat Service to
receive immediate remote IT support. Click the chat link below for
I can only tell you that it worked before...
Nov 2 17:22:28 VERBOSE[11329] logger.c: -- Extension '1004' in context
'from-pstn-deviate-custom' from '7034' does not
when you have overlapdial turned on it should have checked if there's
a potential matching extension
which you have it right
Un-top-posting...
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Connor
Spiess
I am wondering what people are doing for security when registering IP
phone's remotely if you do not have the equipment to do a VPN tunnel at
the remote site. The phone I would be working with
Actually, both. You can (AFAIK) specify 5060, 1 etc and UDP/TCP, etc.
Of course, I have been wrong at least once before :)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Monday,
On Mon, Nov 02, 2009 at 03:18:03PM -, Dan Journo wrote:
Does anyone know how to set the remote party id?
I guess someone does.
If you provide more details you'll have better chances of getting a good
answer.
Version of Asterisk?
Why do you want to set the remote party ID?
How have you
On 2 Nov 2009, at 17:22, Dan Journo wrote:
Does anyone have an up to date guide for setting up fax 2 email with
asterisk?
So you can fax them obnoxiously long signatures?
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Un-top-posting...
On Mon, 2 Nov 2009, Danny Nicholas wrote:
You can restrict on IP address, MAC address and port type and that's
just what I know. If someone want's through bad enough you're going to
have a problem, but you can at least slow down or stop casual hackers.
Aren't MAC
We want to disconnect our PSTN fax line and transfer the number over to
our asterisk server.
I need to get it up and running before we can put in the order to
transfer the fixed line number over to SIP.
Thanks
Dan
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Sorry Steve,
Forgot to remove it before sending the email.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Howes
Sent: 02 November 2009 17:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Monday 02 November 2009 11:46:49 am Steve Edwards wrote:
Un-top-posting...
On Mon, 2 Nov 2009, Danny Nicholas wrote:
You can restrict on IP address, MAC address and port type and that's
just what I know. If someone want's through bad enough you're going to
have a problem, but you can
Is the a way to execute the macro AFTER connecting to the channel:
M(x[^arg]) - Execute the Macro for the *called* channel before connecting
to the calling channel.
doesn't work for me as I need to listen to the macro progress as it is sending
DTMF tone and respond from the
Dan Journo wrote:
I need to get it up and running before we can put in the order to
transfer the fixed line number over to SIP.
Faxing over SIP is never a good idea.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve
--- On Mon, 11/2/09, Martin asteriskl...@callthem.info wrote:
I'd double check that you really have overlapdial=yes for
those
channels ... it should be declared
before channel = keyword in
zapata.conf/chan_dahdi.conf
I declared overlapdial in zapata.conf:
switchtype = euroisdn
signalling
Thank you ,
I did that
sip set debug ip 198.163.0.103
SIP Debugging Enabled for IP: 198.163.0.103
I checked the /var/log/asterisk files and there is no information
there.
Could you please inform where am I suppose to see the debug information
?
tks
Jair
Ott Rose wrote:
you
can
I've heard mixed reports.
Some say they've had no problems, some say that faxes fail most of the
time.
I want to try it out and see how it goes.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
I've heard of people who go to casinos and come home with a couple
thousand bucks winnings, too. But the truth is that invariably the vast
majority of people who gamble don't win.
http://hylafax.sourceforge.net/docs/fax-over-voip.pdf
Everyone wants to see if they're lucky. The smart ones,
2009/11/2 Doug Lytle supp...@drdos.info
Dan Journo wrote:
I need to get it up and running before we can put in the order to
transfer the fixed line number over to SIP.
Faxing over SIP is never a good idea.
And why would that be? I think that faxing over SIP using T.38 is a
fantastic
I have been getting the following message every time I make a call for
the past few months:
[Nov 2 13:08:18] WARNING[9859]: file.c:1273 waitstream_core: Unexpected
control subclass '-1'
Everything seems to be working so I do not know if this is important.
I am using Asterisk
Lee Howard wrote:
I've heard of people who go to casinos and come home with a couple
thousand bucks winnings, too. But the truth is that invariably the vast
majority of people who gamble don't win.
http://hylafax.sourceforge.net/docs/fax-over-voip.pdf
Everyone wants to see if they're
Kevin P. Fleming wrote:
Lee Howard wrote:
I've heard of people who go to casinos and come home with a couple
thousand bucks winnings, too. But the truth is that invariably the vast
majority of people who gamble don't win.
http://hylafax.sourceforge.net/docs/fax-over-voip.pdf
Lee Howard wrote:
FAX over SIP does not necessarily mean FAX over VOIP. FAX over SIP can
also describe T.38, which is not as much of a gamble as FAX over VOIP :-)
Does Asterisk 1.4 support T.38?
Only for passthrough between SIP channels; Asterisk 1.6.0 and later also
support T.38
Christian Victor wrote:
2009/11/2 Doug Lytle supp...@drdos.info mailto:supp...@drdos.info
Faxing over SIP is never a good idea.
And why would that be? I think that faxing over SIP using T.38 is a
fantastic idea.
As far as I know, T.38 isn't supported under 1.4
Doug
--
Ben
Kevin P. Fleming wrote:
Lee Howard wrote:
I've heard of people who go to casinos and come home with a couple
thousand bucks winnings, too. But the truth is that invariably the vast
majority of people who gamble don't win.
http://hylafax.sourceforge.net/docs/fax-over-voip.pdf
Everyone
How do these fax2email providers run their service?
Do they all use physical lines rather than use the internet?
Thanks
Dan
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas
Kenyon
Sent: 02 November 2009
On Mon, Nov 2, 2009 at 12:22 PM, Dan Journo
d...@keshercommunications.comwrote:
Hi,
Does anyone have an up to date guide for setting up fax 2 email with
asterisk?
You can buy this shrink-wrapped from Cisco if you're willing to pay what
they're asking. There are probably other vendors who
David Backeberg wrote:
On Mon, Nov 2, 2009 at 12:22 PM, Dan Journo
d...@keshercommunications.com mailto:d...@keshercommunications.com
wrote:asterisk app_fax (which depends on SpanDSP from Lee Howard).
SpanDSP was written by Steve Underwood.
Doug
--
Ben Franklin quote:
Those who would
On Mon, Nov 2, 2009 at 3:38 PM, Dan Journo d...@keshercommunications.com
wrote:
How do these fax2email providers run their service?
Do they all use physical lines rather than use the internet?
If you read far enough back in the archives, you'll find somebody who
claimed they used
asterisk-1.4
We're in the process of replacing an ancient Centigram-based, but
Fujitsu-labelled, voicemail system with an Asterisk solution. The system
will interface with a Fujitsu F9600 switch and use the SMDI module in
Asterisk 1.6.1.x to communicate the calling information needed to make the
interplay
Ciao,
I installed Xlite on Windows Vista, the IP connection (ping) is working,
shall I check something else ?
Thanks in advance.
2009/11/1 Farooq Hussain farooqhussain...@gmail.com
Dear Giancarlo,
On which OS your are installing XLITE. If you are trying to connect XLITE
using Winodws XP
Testing a new gateway and have a Rhino Channel Bank... Sending a test
fax and everything works fine (Receive the fax fine) But I notice this
in the log
Google search didn't return much of anything...
DAHDI hook failed returned -1 (trying 1): Device or resource busy
On Mon, Nov 2, 2009 at 4:26 PM, Robert Grignon rgrig...@fleetone.com wrote:
Testing a new gateway and have a Rhino Channel Bank... Sending a test fax
and everything works fine (Receive the fax fine) But I notice this in the
log
Google search didn't return much of anything...
DAHDI hook
Dan Journo wrote:
How do these fax2email providers run their service?
I've not the faintest Idea, the provider I use afaict outsource it.
Do they all use physical lines rather than use the internet?
Thanks
Dan
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Hello, short question: is there a possibility to use asterisk as an outbound
proxy? iam open for any suggestions, use asterisk trunk, dirty patches, ugly
workarounds, everything.
What is want to build is:
SIP Phone - via TLS/SRTP - Asterisk as outbound proxy - via UDP/RTP -
VoIP-Provider
So
Kristijan Vrban wrote:
Hello, short question: is there a possibility to use asterisk as an
outbound proxy? iam open for any suggestions, use asterisk trunk, dirty
patches, ugly workarounds, everything.
What is want to build is:
SIP Phone - via TLS/SRTP - Asterisk as outbound proxy - via
2009/11/2 Doug Lytle supp...@drdos.info
Christian Victor wrote:
2009/11/2 Doug Lytle supp...@drdos.info mailto:supp...@drdos.info
Faxing over SIP is never a good idea.
And why would that be? I think that faxing over SIP using T.38 is a
fantastic idea.
As far as I know, T.38
On 11/03/2009 04:25 AM, Thomas Kenyon wrote:
Kevin P. Fleming wrote:
Lee Howard wrote:
I've heard of people who go to casinos and come home with a couple
thousand bucks winnings, too. But the truth is that invariably the vast
majority of people who gamble don't win.
My mental plan orginilly was:
1.- Creating a macro that acceps ARGs like.
a.- agent
b.- queue/s
In the macro we could have the voice respose for the loging. I am using on
1.4 the following procedure.
* the agents call to 21Agentid to loging, and it is promt just for the
passwd
* the agents call
Is the a way to turn the ring tone OFF during dialing?
When I'm in a macro mode I have to listen to ring the tone for 20sec before
macro finish and I get connected.
--
Joseph
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On 11/02/09 19:56, Joseph wrote:
Is the a way to turn the ring tone OFF during dialing?
When I'm in a macro mode I have to listen to ring the tone for 20sec before
macro finish and I get connected.
I've found a better solution, setting musing on hold before calling party
answers:
m: Provide
Proxy is not the correct term to describe this scenario, but yes, it
is possible in principle.
Kristijan Vrban wrote:
Hello, short question: is there a possibility to use asterisk as an
outbound proxy? iam open for any suggestions, use asterisk trunk, dirty
patches, ugly workarounds,
Hi,
Yesterday I've got a core dump from Asterisk, other times I was able to
discover what this core dump was related with through gdb Ouput info,,
but this time.. I'm really lost. Could some one please help me
GDB output is at
http://pastebin.com/m603e6a74
Any help would be appreciated.
Hi Guys,
Does anyone know how to make the custom build of Asterisk 1.6.1 work with
Exchange 2007 UM? It always times out with system unavailable and won't go
through. I have TCP enabled and all the trunk and outgoing settings configured.
When I use TrixBox CE 2.8.0.2, it works. But custom
On Tue, Nov 03, 2009 at 01:18:14AM -0300, Fernando Berretta wrote:
Hi,
Yesterday I've got a core dump from Asterisk, other times I was able to
discover what this core dump was related with through gdb Ouput info,,
but this time.. I'm really lost. Could some one please help me
GDB output
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