Well that answers that question. I see that t38modem provides an H232
modem - is this unsuitable for HylaFAX's purpose? (ignoring the fact
that it requires a kernel recompile on most newer distros.)
Steve Underwood wrote:
Rob Hillis wrote:
Last time I heard IAXModem didn't support T.38
Hi Rob,
Rob Hillis wrote:
Well that answers that question. I see that t38modem provides an H232
modem - is this unsuitable for HylaFAX's purpose? (ignoring the fact
that it requires a kernel recompile on most newer distros.)
Steve Underwood wrote:
Rob Hillis wrote:
Last time I heard
Unfortunately there is only one port, clearly labelled handset
On 31/12/2007, at 11:34 PM, dave cantera wrote:
glenn,
check your handset cord... it might be plugged into the wrong port
in the back of the phone. perhaps the headset jack...
daveC
Glenn Gillen wrote:
Hey all,
I've
Hello
Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0
and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd
like to play PCM WAV files instead of eg. GSM. Per...
www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk
... I recorded a
On Jan 1, 2008 3:36 PM, Vincent [EMAIL PROTECTED] wrote:
Hello
Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0
and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd
like to play PCM WAV files instead of eg. GSM. Per...
Glenn Gillen wrote:
Unfortunately there is only one port, clearly labelled handset
On 31/12/2007, at 11:34 PM, dave cantera wrote:
glenn,
check your handset cord... it might be plugged into the wrong port
in the back of the phone. perhaps the headset jack...
daveC
Push the cord all
On Tue, 1 Jan 2008 17:23:29 +0530, Godson Gera [EMAIL PROTECTED]
wrote:
s,2,Playback(/usr/local/lib/asterisk/test_wav_out)
And asterisk will automatically pickup the file that it can play with any
asterisk supported format from the specified path.
OK. Is there a way to tell Asterisk which codec
Vincent wrote:
On Tue, 1 Jan 2008 17:23:29 +0530, Godson Gera [EMAIL PROTECTED]
wrote:
s,2,Playback(/usr/local/lib/asterisk/test_wav_out)
And asterisk will automatically pickup the file that it can play with any
asterisk supported format from the specified path.
OK. Is there a way to tell
On Jan 1, 2008 3:36 PM, Vincent [EMAIL PROTECTED] wrote:
Hello
Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0
and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd
like to play PCM WAV files instead of eg. GSM. Per...
Dear all
I have cisco phone 7974 i have useing SIP protocol to register phone
on Asterisk and it is working fine but i have one problem when how do i use
confranceing between 2 party i am not talking about meetme confrance i am
taking about phone confranceing like press flash key
I'm not looking at T.38 , at this time its terminating a SIP trunk with
multiple DID's for fax.
I'm using this configuration with linksys PAP ATA and satisfied with
results.
I'm looking at removing these ATA 's and using Asterisk ( or giving it a try
) for terminating fax.
Last time I heard
vincent,
here is a script that I used to convert a single wav file or the entire
directory... no file specified on launch, converts all files in the
current directory...
creates a logfile, although trivial...
daveC
#!/bin/sh
#
# convert-all.sh
#
# convert all *.wav files to .gsm .au formats
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Thanks nice script.
But why au files in addition to gsm?
- Original Message -
From: dave cantera
To: Asterisk Users Mailing List - Non-Commercial Discussion ; [EMAIL
PROTECTED]
Sent: Tuesday, January 01, 2008 11:27 AM
Subject: Re: [asterisk-users] [1.4 + FreeBSD 6.2]
I asked this question last week and never got an answer. I also didn't find
the answer in the wiki.
I think it would be nice if asterisk would check the database again if the user
re-registers, but it doesn't seem to do that. A periodic update would be ok
too, but it doesn't seem to do that
On Tue, Jan 01, 2008 at 11:27:54AM -0500, dave cantera wrote:
vincent,
here is a script that I used to convert a single wav file or the entire
directory... no file specified on launch, converts all files in the current
directory...
creates a logfile, although trivial...
daveC
#!/bin/sh
#
Adam Moffett wrote:
I asked this question last week and never got an answer. I also
didn't find the answer in the wiki.
I think it would be nice if asterisk would check the database again if
the user re-registers, but it doesn't seem to do that. A periodic
update would be ok too, but
Hi,
Before I report a bug on http://bugs.digium.com, I
would like to know if someone is seeing the same error
message.
Personally I am not using wctdm24xxp but other modules
such as wcte12xp and wctdm. The latter modules load
fine and are compiled with pci_register_driver as
expected.
The only
On Tue, Jan 01, 2008 at 10:24:24AM -0800, Vieri wrote:
Hi,
Before I report a bug on http://bugs.digium.com, I
would like to know if someone is seeing the same error
message.
Personally I am not using wctdm24xxp but other modules
such as wcte12xp and wctdm. The latter modules load
fine
Check your extensions.conf
On Jan 1, 2008 11:33 AM, lists65 [EMAIL PROTECTED] wrote:
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--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
Is it a custom kernel that has no PCI support?
It's a custom 2.6.22 with
# grep -i pci /usr/src/linux/.config
# Bus options (PCI, PCMCIA, EISA, MCA, ISA)
CONFIG_PCI=y
# CONFIG_PCI_GOBIOS is not set
# CONFIG_PCI_GOMMCONFIG is not set
#
ATT or Verizon. I think those are the only ILECs left, right?
On Dec 31, 2007 9:26 AM, Steve Finkelstein [EMAIL PROTECTED] wrote:
Senad,
Mind if I ask who that provider is?
Thanks.
Sent from my iPhone
On Dec 31, 2007, at 8:10 AM, Senad Jordanovic [EMAIL PROTECTED] wrote:
Justin Case
Then I suggest you prepare yourself for a lot of pain. Fax over the
'net without T.38 is almost guaranteed to not work.
Al lists wrote:
I'm not looking at T.38 , at this time its terminating a SIP trunk
with multiple DID's for fax.
I'm using this configuration with linksys PAP ATA and
Andrew Joakimsen wrote:
ATT or Verizon. I think those are the only ILECs left, right?
Don't forget the company formerly known as US Worst Now Quest
John Novack
On Dec 31, 2007 9:26 AM, Steve Finkelstein [EMAIL PROTECTED] wrote:
Senad,
Mind if I ask who that provider is?
What are you using for a PSTN gateway?
From: Brian Alexander [mailto:[EMAIL PROTECTED]
Sent: Monday, December 31, 2007 3:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] One Way Delay in Audio Over Analog
I have been trying to track down the
Andrew Joakimsen wrote:
Check your extensions.conf
Hahahahaha!
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
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REALY?? Humm I have been doing this for over a year and we receive over 400
faxes a month! 8 iaxmodems with DID's from a real SIP provider. And this
connection is used for ALL office traffic, mail, VPN, webmail, and DNS. NO echo
and no voice quality issues. Now we do have a 12mb down 768k
Yes you can use res_conf_pgsql.so is present in asterisk 1.4
On Oct 7, 2006 1:22 AM, John Miloo [EMAIL PROTECTED] wrote:
Hello Comunity,
How can I get Asterisk realtime working with Postgres? (without ODBC)?
Thanks
John
/doc/realtime.txt in Version 1.4 Beta2
Currently there are
Jonn R Taylor wrote:
REALY?? Humm I have been doing this for over a year and we receive over
400 faxes a month! 8 iaxmodems with DID's from a real SIP provider. And this
connection is used for ALL office traffic, mail, VPN, webmail, and DNS. NO
echo and no voice quality issues. Now we
I have it setup to email me any failed fax connections. Most of the faxes come
from remote offices, distributors and customers.
Jonn
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Tuesday, January 01, 2008 2:35 PM
To: Asterisk Users
Jonn R Taylor wrote:
I have it setup to email me any failed fax connections. Most of the faxes
come from remote offices, distributors and customers.
Same here, but HylaFAX won't send you any logs of attempts that haven't
at least negotiated a fax transmission. Call comes in, tries to
On Tue, 2008-01-01 at 13:48 -0600, Jonn R Taylor wrote:
REALY?? Humm I have been doing this for over a year and we receive
over 400 faxes a month! 8 iaxmodems with DID's from a real SIP
provider. And this connection is used for ALL office traffic, mail,
VPN, webmail, and DNS. NO echo and
[Footers trimmed to protect my precious bandwidth. MY PRECIOUS!]
Yes, but he didn't qualify it. You can get a T1 with unlimited minutes in the
US -- as long as those minutes are local-only. Long distance is another
matter, although most providers sell their voice T1s with a block of long
If I had ANY failed faxes I would here about it. Iaxmodem creates a log of its
own, so when I get a connection that fails hylafax sends the failure to me. One
of the things that I found is you need to add nojitterbuffer to the iaxmodem
config file, only use g711, and you must have QOS enabled
Jonn R Taylor wrote:
If I had ANY failed faxes I would here about it. Iaxmodem creates a log of
its own, so when I get a connection that fails hylafax sends the failure to
me. One of the things that I found is you need to add nojitterbuffer to the
iaxmodem config file,
Really? I'll have
Jonn R Taylor wrote:
One of the things that I found is you need to add nojitterbuffer to the
iaxmodem config file
The reason that you need the nojitterbuffer in the iaxmodem config file
is because you're actually getting at least some jitter.
IAXmodem's jitterbuffer simply fills-in gaps due
I have always said that if some one said it can't be done, they did not try
hard enough.
FYI... I love this.
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
Jonn
-Original Message-
From: [EMAIL
That is correct. I found that out awhile ago with our internal fax. It would
not connect, but the external faxes coming in over SIP worked.
Jonn
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Howard
Sent: Tuesday, January 01, 2008 3:50 PM
To:
I'd say consider yourself very lucky. I know I did some testing here
some time ago with faxing over VoIP.
* One extension to another over G711a with both extensions on the
same LAN - worked 95% of the time
* One extension on my Asterisk server to an Extension on a friend's
NOT true and I have proven that for the last year.
Jonn
_
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Hillis
Sent: Tuesday, January 01, 2008 4:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.4 Fax
I'd
Jonn R Taylor wrote:
FYI... I love this.
Ben Franklin quote:
I truly believe it. But, it being a Franklin quote is in some dispute.
I like it all the same.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither
Jonn R Taylor wrote:
I have always said that if some one said it can't be done, they did not try
hard enough.
FYI... I love this.
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
As the person
Guys!
what i was looking here was a simple hint/recommendation for installing
IaxModem and Hylafax.
Let me try it myself and see how feasible this solutions is.
On Jan 1, 2008 5:02 PM, Steve Underwood [EMAIL PROTECTED] wrote:
Jonn R Taylor wrote:
I have always said that if some one said it
Steve,
One of the main reasons that this works is controlling the data to and from the
internet. I have spent the last 10 years building networks for ISP's. The key
is getting the data from point a to point b in tact and in order.
I did not get lucky as you put it. I am a network engineer and
Jesse,
We have multiple installations of this scale and a few with far more
concurrent call paths (250+). In our experience, Asterisk scales
nicely to these levels as long as you are realistic about what you
expect of the server. For instance, we rarely, if ever, convert
signal to TDM.
tzafrir,
thanks for the note... yep, it is useless...
daveC
Tzafrir Cohen wrote:
On Tue, Jan 01, 2008 at 11:27:54AM -0500, dave cantera wrote:
vincent,
here is a script that I used to convert a single wav file or the entire
directory... no file specified on launch, converts all files in
Hi, We currently testing a trixbox/asterisk installation and have used Freepbx
to set-up and configure the box and it is running tremendously well. We have an
generic IVR configured to which can transfer callers to a child IVR. This child
IVR has a number of options to send the caller off to
my final ivr is this, he works me very well
exten = 110,1,GotoIfTime(08:00-18:00|mon-fri|*|*?110,in)
exten =110,n,Dial(SIP/111,86,Tt)
exten =110,n,Dial(SIP/112,86,Tt)
exten =110,n,Hangup()
exten = 110,n(in),Set(TIMEOUT(digit)=2)
exten = 110,1,Answer()
exten = 110,2,Background(introm)
exten =
I put something together like this for a finance company - Asterisk
looked up the callerid in a MySQL database, and put the call into a
queue, with a higher priority if the call was from certain clients.
If the callerid was not found, it them allowed for a clientid and
pincode to be entered.
On Tuesday 01 January 2008 20:40:19 troxlinux wrote:
my final ivr is this, he works me very well
exten = 110,1,GotoIfTime(08:00-18:00|mon-fri|*|*?110,in)
exten =110,n,Dial(SIP/111,86,Tt)
exten =110,n,Dial(SIP/112,86,Tt)
exten =110,n,Hangup()
exten = 110,n(in),Set(TIMEOUT(digit)=2)
Uh,
On Tue, 01 Jan 2008 16:10:47 +0100, MatsK [EMAIL PROTECTED] wrote:
The codec is specified (for a sip device) in sip.conf, like this:
Good to know. Actually, I'll have Asterisk save voicemails as WAV and
move the files to the www's htdocs, and send an e-mail to users with
the link they'll just
On Tue, 01 Jan 2008 11:27:54 -0500, dave cantera
[EMAIL PROTECTED] wrote:
here is a script that I used to convert a single wav file or the entire
directory... no file specified on launch, converts all files in the
current directory...
Thanks for the script. I'll keep it handy.
On Tue, 1 Jan 2008 21:05:11 +0530, Godson Gera [EMAIL PROTECTED]
wrote:
Asterisk automatically takes care of saving CPU issue as it picks the file
that have less translation cost
Yes, but that's OK for files that I use in the IVR, but not for
voicemail messages. The CPU is too slow to handle
I think perhaps you are the exception rather than the rule.
Maybe you were able to engineer your network so that fax works without
any of the FoIP protocols - good luck to you if you have. For /most/
people, it's unlikely they would have sufficient control over their WAN
segment to ensure that
Mike Dent wrote:
Hi,
just wondered if it was the same firmware on both devices?
thanks
Mike
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Hi there,
is there any howto how do i configure a asterisk/trixbox for mail2fax?
The fax must be send over sipgate or other SIP peers. (i dont have
any normal telephones connected).
What i wanne do is somethink like this:
Subject: +49691234567
Attache: *.pdf
The attched pdf have to be send ;)
On Jan 2, 2008 12:23 AM, Daniel [EMAIL PROTECTED] wrote:
Hi there,
is there any howto how do i configure a asterisk/trixbox for mail2fax?
The fax must be send over sipgate or other SIP peers. (i dont have
any normal telephones connected).
Do people even read the mail list anymore, or do
2008/1/2, Daniel [EMAIL PROTECTED]:
Hi there,
is there any howto how do i configure a asterisk/trixbox for mail2fax?
The fax must be send over sipgate or other SIP peers. (i dont have
any normal telephones connected).
What i wanne do is somethink like this:
Subject: +49691234567
Hi List;
I heared that IAX is good for NATing issues, but I do
not know if it can help me in that senario:
I have two Asterisks machines in different sites and
both are behind NAT (both have private IP address), I
need to link these two asterisks with IAX trunk (if it
help really in such
If you have zaptel 1.2.22.1 and kernel 2.6.22 could
you please do the following and see if it does the
same for you?
# modprobe wctdm24xxp
FATAL: Error inserting wctdm24xxp
(/lib/modules/2.6.22-gentoo-r9/misc/wctdm24xxp.ko):
Unknown symbol in module, or unknown parameter (see
dmesg)
dmesg:
Apparently not. I'm sure as heck not going to get involved in this
argument again! :)
Bill Hackensack wrote:
Do people even read the mail list anymore, or do they just land on
this planet, subscribe to the list, and ask the same questions that's
been asked over and over and over and over and
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