On Mon, Sep 13, 2010 at 4:33 PM, Stanislav Korsei kor...@rinogo.com wrote:
Hello!
I've created clean installation of Asterisk 1.6.2.11 with spandsp 0.0.5.
When i try to receive fax I get:
[Sep 13 00:45:59] WARNING[3283]: app_fax.c:432 transmit_audio: channel
'SIP/crocus-ua-0004' refused
On 09/14/2010 04:33 AM, Stanislav Korsei wrote:
Hello!
I've created clean installation of Asterisk 1.6.2.11 with spandsp 0.0.5.
When i try to receive fax I get:
Why install 0.0.5? Its ancient. the world has moved on.
[Sep 13 00:45:59] WARNING[3283]: app_fax.c:432 transmit_audio: channel
Hello!
I'm trying to make fax work on Asterisk 1.6. I've installed DAHDI,
marked spandsp as app_fax, but faxes are not going trough,
although application itself installs successfully. I've been using rx_fax
tx_fax on 1.4 and everything worked fine.
Can you recommend any specific solution to this
On Wed, Sep 8, 2010 at 4:18 PM, Stanislav Korsei kor...@rinogo.com wrote:
Can you recommend any specific solution to this problem or way to install
app_fax?
Not without specific debugging about what problems you're seeing. You
get a lot of information when faxes succeed or fail. Try a fax and
Thought a different succinct subject line must drum up an answer or two...
Also, this has been tested from two different carriers: We're getting
an average of 2/10 call success rate.
-- Forwarded message --
From: Joe Wood sch...@gmail.com
Date: Thu, Aug 26, 2010 at 6:58 PM
First off, let me first say that this is not a one-way audio problem.
Sometimes I can get 'her' to speak to me, other times I can't for a
long time.
I'm just using a very simple system to dump people into MeetMe().
Nothing fancy.
I do have the following in my modules.conf:
preload =
On 08/05/2010 06:25 PM, Roderick A. Anderson wrote:
Kevin P. Fleming wrote:
On 08/05/2010 03:52 PM, Roderick A. Anderson wrote:
I have a Linux-Vserver guest running CentOS 5.5 with Asterisk 1.6
installed from the asterisk.org and digium.com repositories.
I have Asterisk starting (service
I have a Linux-Vserver guest running CentOS 5.5 with Asterisk 1.6
installed from the asterisk.org and digium.com repositories.
I have Asterisk starting (service asterisk start) but see errors about
dahdi in /var/log/asterisk/messages.
... ERROR[25658] codec_dahdi.c: Failed to open
On 08/05/2010 03:52 PM, Roderick A. Anderson wrote:
I have a Linux-Vserver guest running CentOS 5.5 with Asterisk 1.6
installed from the asterisk.org and digium.com repositories.
I have Asterisk starting (service asterisk start) but see errors about
dahdi in /var/log/asterisk/messages.
Kevin P. Fleming wrote:
On 08/05/2010 03:52 PM, Roderick A. Anderson wrote:
I have a Linux-Vserver guest running CentOS 5.5 with Asterisk 1.6
installed from the asterisk.org and digium.com repositories.
I have Asterisk starting (service asterisk start) but see errors about
dahdi in
Hi all,
My latest Asterisk system is based on Debian squeeze with Asterisk
1.6.2.6-1 and SIP only. One of my favorite features that I had working
with Asterisk 1.4 is the PrivacyManager. However, this was not
straightforward, because anonymous SIP calls arrive with
${CALLERID(num)} =
Try removing the quotes in your n(true) priority.
Thanks,
--Warren Selby
On Aug 2, 2010, at 7:40 PM, Jaap Winius jwin...@umrk.nl wrote:
Hi all,
My latest Asterisk system is based on Debian squeeze with Asterisk
1.6.2.6-1 and SIP only. One of my favorite features that I had working
with
Quoting Warren Selby wcse...@selbytech.com:
Try removing the quotes in your n(true) priority.
From FAILED? That makes no difference: with or without the quotes,
the result is always 0, which leads in the Dial() rule being executed.
Actually, though, that's not even relevant, because before
El 29/06/10 15:28, Mark Deneen escribió:
We are experiencing intermittent DTMF problems here, with the
following setup:
ITSP - PIX - Asterisk (g729, RFC2833 for DTMF).
I am running Ubuntu server 10.04, but Asterisk is compiled by us and
not installed from the software repository.
On Thu, Jul 1, 2010 at 7:09 PM, Miguel Molina mmol...@millenium.com.cowrote:
I've experienced a similar DTMF issue with recent asterisk 1.4 versions
(1.4.32, 1.4.33.1), I'm not sure about 1.6.2.X. What happens here is
that the DMTF activated features, like disconnect (default *) or blind
, 2010 16:52
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.6 and multiple parking
Hello there
You should have a look at features.conf
Regards Aksel
Fra: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun
Of Aksel Celasun
Sent: Monday, June 28, 2010 16:52
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.6 and multiple parking
Hello there
You should have a look at features.conf
Regards Aksel
Fra: asterisk-users-boun
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Wednesday, June 30, 2010 11:27
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk 1.6 and multiple parking
Actually, I should simply have tried. I did need
-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk 1.6 and multiple parking
Here is my only question left about parkinglots in 1.6. How does the
parkinghints=yes parameter work?
I've tried using core show hints , but there are never any hints. Even
when a call is actually parked
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, June 30, 2010 13:38
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk 1.6 and multiple parking
In 1.4 you set up the lots you want to monitor
We are experiencing intermittent DTMF problems here, with the following
setup:
ITSP - PIX - Asterisk (g729, RFC2833 for DTMF).
I am running Ubuntu server 10.04, but Asterisk is compiled by us and not
installed from the software repository. Essentially, DTMF works for some
time, but at some
Hi,
One of the big features of 1.6 was described as multi-tenant parking.
Basically, parking people in different lots so the sales dept. could only
pick up their calls, and tech support theirs and no mix up was possible.
I can only find the original announcement and others asking the same
Hello there
You should have a look at features.conf
Regards Aksel
Fra: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] På vegne av Mike
Sendt: 28. juni 2010 21:39
Til: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Emne: [asterisk-users
Michael wrote:
I am attempting to setup Asterisk to work with Gtalk.
I am using the following versions:
Slackware Linux 12.0
Asterisk 1.6.2.9
GNU TLS 2.8.6
Iksemel (svn v25)
OpenSSL 0.9.8o
It all compiles however about 10 seconds after starting Asterisk it crashes.
If there is any
Hello,
I am attempting to setup Asterisk to work with Gtalk.
I am using the following versions:
Slackware Linux 12.0
Asterisk 1.6.2.9
GNU TLS 2.8.6
Iksemel (svn v25)
OpenSSL 0.9.8o
It all compiles however about 10 seconds after starting Asterisk it crashes.
To mitigate this issue I have moved
Hi,
I have read the docs, and now I want to attempt to setup Asterisk 1.6. I
am not going to complicate it with load balancing, etc. The setup is just 1 SIP
line - no other in-house connections. All inbound traffic. I intend to keep
this simple. Imagine that I sell pies in my
Hello Platt,
Thank you for help.
I have tested and it works fine.
--
Please discover scientific miracles of CORAN
http://www.55a.net/
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
Hello,
Thank you for your reply.
The first proposed solution has resolved the problem for a test in the local
network. Another test is planned today later with a client in the same NAT
and another in the public internet with a public static ip address.
Do you have any advice for that case?
Thank you for your reply.
The first proposed solution has resolved the problem for a test in the local
network. Another test is planned today later with a client in the same NAT
and another in the public internet with a public static ip address.
Do you have any advice for that case?
Hello All,
I have installed Asterisk 1.6 with openVPN in the same machine. I have set
up a VPN connection between 2 SIP clients and Asterisk using x-lite.
The 2 clients connects to Asterisk. SIP signaling goes ok over the vpn
tunnel.
When attempting to make a call between the clients, the
mosbah.abdelkader wrote:
Hello All,
I have installed Asterisk 1.6 with openVPN in the same machine. I have
set up a VPN connection between 2 SIP clients and Asterisk using x-lite.
Just a guess, set canreinvite=no in the sip.conf for each of the end points
Doug
--
Ben Franklin quote:
Hello All,
I have installed Asterisk 1.6 with openVPN in the same machine. I have set
up a VPN connection between 2 SIP clients and Asterisk using x-lite.
The 2 clients connects to Asterisk. SIP signaling goes ok over the vpn
tunnel.
When attempting to make a call between the clients,
Hi all!
I have installed a quite old Asterisk 1.6.2.0-rc2 with latest DAHDI on
Ubuntu 9.10 from repository. It is working now but mysql logging is very
strange. All calls have logged in mysql cdr table, which is fine, but
disposition is 'NO ANSWER' even if I had talked on phone. Duration is
How to enable cdr_mysql.conf in Asterisk 1.6?
I have installed asterisk-addons which compiled mysql support,
module show is showing cdr_addon_mysql.so
but cdr_mysql.conf was not created in /asterisk directory
Is there any configuration file to enable mysql support?
Comping cdr_mysql.conf from
http://hostseries.com/asterisk-cdr-logging-in-mysql/
http://www.asterisk.net.au/tutorial/10/
http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql
http://www.spiration.co.uk/post/1327/asterisk-addons%20setting%20up%20mysql%20cdr%20for%20Asterisk
On Fri, Oct 30, 2009 at 11:35 AM, Joseph
On Friday 30 October 2009 01:05:26 Joseph wrote:
How to enable cdr_mysql.conf in Asterisk 1.6?
I have installed asterisk-addons which compiled mysql support,
module show is showing cdr_addon_mysql.so
but cdr_mysql.conf was not created in /asterisk directory
Is there any configuration file
Thanks Prince (good links) and Tilghman.
I'm using Gentoo installation of Asterisk-1.6.1.8-r1 that just showed up on
portage.
I've emerged(installed) asterisk-addons and this file usually creates necessary
drivers and copy cdr_mysql.conf file into /etc/asterisk (it worked in past
verions 1.2
I just jumped to asterisk-1.6.1.8 and I calls will not go through to my
asterisk.
Same setup with asterisk-1.4 and calls get accepted.
sip show registry (asterisk-1.6):
Host dnsmgr Username Refresh State
sip.actio.pl:5060 N 4589835105 Registered
sip show
On 10/30/09 12:05, Joseph wrote:
I just jumped to asterisk-1.6.1.8 and I calls will not go through to my
asterisk.
Same setup with asterisk-1.4 and calls get accepted.
sip show registry (asterisk-1.6):
Host dnsmgr Username Refresh State
sip.actio.pl:5060 N 4589835
Hi,
I set up my asterisk so it does dial out on DAHDI/1 which is on an FXS modul.
But the phone that is attached to the line does nothing at all.
asterisk-CLI shows a lot of
-- DAHDI/1-1 is ringing
-- DAHDI/1-1 is ringing
-- DAHDI/1-1 is ringing
Even when I lift up the handset during (and no
- Eckhard Jokisch e.joki...@orange-moon.de wrote:
Hi,
I set up my asterisk so it does dial out on DAHDI/1 which is on an FXS
modul.
But the phone that is attached to the line does nothing at all.
asterisk-CLI shows a lot of
-- DAHDI/1-1 is ringing
-- DAHDI/1-1 is ringing
--
On Mon, Oct 12, 2009 at 05:59:16PM +0200, Eckhard Jokisch wrote:
Hi,
I set up my asterisk so it does dial out on DAHDI/1 which is on an FXS modul.
But the phone that is attached to the line does nothing at all.
asterisk-CLI shows a lot of
-- DAHDI/1-1 is ringing
-- DAHDI/1-1 is ringing
Hi ,
I;ve Asterisk 1.6.0 with static agents (sip softphones with extns 100 101
) in a queue..When a caller arrives in queue , it lands on first 100 , 100
then does a blind transfer to 101 .. so that the caller can converse with
101 .. strangely enough the queue_log shows :
Hi ,
I;ve Asterisk 1.6.0 with static agents (sip softphones with extns 100 101
) in a queue..When a caller arrives in queue , it lands on first 100 , 100
then does a blind transfer to 101 .. so that the caller can converse with
101 .. strangely enough the queue_log shows :
Sriram escribió:
Hi ,
I;ve Asterisk 1.6.0 with static agents (sip softphones with extns 100
101 ) in a queue..When a caller arrives in queue , it lands on
first 100 , 100 then does a blind transfer to 101 .. so that the
caller can converse with 101 .. strangely enough the queue_log
I've downloaded and installed Trixbox 2.8 (asterisk 1.6) ..I encounter 2
problems for dynamic agents login and logout -
1. When agent from sip phone dials *11 , he is prmpted to enter extension
number first - but if he feeds the extension number, asterisk doenst allow
him to
Kevin P. Fleming
Envoyé : vendredi 21 août 2009 17:11
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Asterisk 1.6 t 38 passthrough
BERGANZ François wrote:
I have that problem:
[Aug 21 15:57:35] WARNING[5198]: chan_sip.c:6737 get_ip_and_port_from_sdp
Hello,
How can I do t-38 passthrough with asterisk 1.6 ?
I know how to do with 1.4 but not with 1.6
Thank you
Cordialement,
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
___
-- Bandwidth and Colocation
BERGANZ François wrote:
How can I do t-38 passthrough with asterisk 1.6 ?
I know how to do with 1.4 but not with 1.6…
There is no difference, the identical configuration should work. I would
recommend using the 1.6.0.14 or 1.6.1.5 release candidates (or any later
releases) as they contain a
-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Kevin P. Fleming
Envoyé : vendredi 21 août 2009 15:05
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Asterisk 1.6 t 38 passthrough
BERGANZ François wrote:
How
BERGANZ François wrote:
When I receive a fax it is in g711
After pickup, the fax invite again with T38 in the SDP.
Have I something to insert in the dialplan or other to let the T38
passthrough ?
No.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive
-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Kevin P. Fleming
Envoyé : vendredi 21 août 2009 15:31
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Asterisk 1.6 t 38 passthrough
BERGANZ François wrote:
When I receive
BERGANZ François wrote:
I have that problem:
[Aug 21 15:57:35] WARNING[5198]: chan_sip.c:6737 get_ip_and_port_from_sdp:
Failed to read an alternate host or port in SDP. Expect audio problems
[Aug 21 15:57:35] WARNING[5198]: chan_sip.c:17425 handle_request_invite:
Failed to set an
Hello D Tucny,
Your solution works indeed well, thanks for it:)
pepesz
Monday, August 3, 2009, 6:20:39 AM, you wrote:
2009/7/31 pepesz76 pepes...@o2.pl
Dear All,
I'n trying to make a simple call forwarding, however I have small
problem when evaluating an expresion.
Here is my
2009/7/31 pepesz76 pepes...@o2.pl
Dear All,
I'n trying to make a simple call forwarding, however I have small
problem when evaluating an expresion.
Here is my extensions.conf
...
; Unconditional Call Forward
exten = _#21*X.,1,Set(DB(CFIM/${CALLERID(num)})=${EXTEN:4})
exten =
Dear All,
I'n trying to make a simple call forwarding, however I have small
problem when evaluating an expresion.
Here is my extensions.conf
...
; Unconditional Call Forward
exten = _#21*X.,1,Set(DB(CFIM/${CALLERID(num)})=${EXTEN:4})
exten = _#21*X.,2,Hangup()
exten =
Does Asterisk 1.6 fully support RFC4235?
Or is it the same implementation as 1.4?
Thanks.
-- James
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now:
On Tue, Jun 30, 2009 at 9:21 AM, Deric Pagederic.p...@nisc.coop wrote:
I've set up an outbound .call system for customer callbacks and the like.
Calls are going out over analog lines and I'm trying to use the
WaitForSilence routine to make sure the phone has stopped ringing before
starting
I've set up an outbound .call system for customer callbacks and the
like. Calls are going out over analog lines and I'm trying to use the
WaitForSilence routine to make sure the phone has stopped ringing before
starting message playback. The problem is that if I set the first
argument of
Hi on the list,
does anyone of you have experience with asterisk 1.6 and mISDN, pri
primarily?
Thanks in advance Regards,
Christophorus
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asterisk-users mailing list
To
Hello List.
We are having some problems using t.38 together with a Cisco voice router at
one of our providers end.
We are using the new digium asterisk fax module to generate the fax, and when
we use together with our internal Audiocodes Mediant 2000 gateways, we have no
issues what so
On Wed, May 13, 2009 at 3:30 AM, Jon Schøpzinsky j...@firstcom.dk wrote:
We are having some problems using t.38 together with a Cisco voice router at
one of our providers end.
We are using the new digium asterisk fax module to generate the fax, and
when we use together with our internal
: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [asterisk-users] Asterisk 1.6 T.38 generation towards a Ciscovoice
router
On Wed, May 13, 2009 at 3:30 AM, Jon Schøpzinsky j...@firstcom.dk wrote:
We are having some problems using t.38 together with a Cisco voice router at
one of our
On Wed, May 13, 2009 at 9:21 AM, Jon Schøpzinsky j...@firstcom.dk wrote:
I used wireshark to debug the problem, and I can see that the cisco equipment
is correctly sending t.38 packets to asterisk, and the whole re-invite
process is successful.
The problem is, that Asterisk discards the t.38
Thank you .. appreciated.
Best Regards,
--
SplatNIX IT Services :: Innovation through collaboration
- Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote:
On Tuesday 28 April 2009 15:35:14 --[ UxBoD ]-- wrote:
HI,
I am trying to setup CDR with ODBC and MySQL but get the
HI,
I am trying to setup CDR with ODBC and MySQL but get the following error :-
[Apr 28 21:30:01] ERROR[14567]: cdr_odbc.c:133 odbc_log: Unable to retrieve
database handle. CDR failed.
I can successfully connect with iSQL so ODBCINST and ODBC ini files must be
okay. I have modified
On Tuesday 28 April 2009 15:35:14 --[ UxBoD ]-- wrote:
HI,
I am trying to setup CDR with ODBC and MySQL but get the following error :-
[Apr 28 21:30:01] ERROR[14567]: cdr_odbc.c:133 odbc_log: Unable to retrieve
database handle. CDR failed.
I can successfully connect with iSQL so ODBCINST
MaxGao wrote:
hi,all
i have just set up asterisk 1.6.0.7 rc1 with spandsp 0.0.5 pre4 to
ReceiveFAX, link to a E1 (DE410P) using dahdi
this can receive the fax from E1 successfully, but i see many error
message in the log like this:
[Mar 16 09:24:38] ERROR[23540] channel.c: ast_read() called
hi,all
i have just set up asterisk 1.6.0.7 rc1 with spandsp 0.0.5 pre4 to ReceiveFAX,
link to a E1 (DE410P) using dahdi
this can receive the fax from E1 successfully, but i see many error message in
the log like this:
[Mar 16 09:24:38] ERROR[23540] channel.c: ast_read() called with no recorded
asterisk 1.4.23.2 and spandsp 0.0.4 get the same error nowbut less times
than other version ...
[Mar 16 10:12:50] DEBUG[23749]: chan_dahdi.c:7115 do_monitor: Monitor doohicky
got event Alarm on channel 1
[Mar 16 10:12:50] DEBUG[23752]: chan_dahdi.c:4731 __dahdi_exception: Exception
on 11,
On Sun, Mar 15, 2009 at 9:57 PM, MaxGao ss...@126.com wrote:
and many times when reciving tax , the E1 card will down , all the channel
get red alarm...
[Mar 16 09:49:19] DEBUG[20928] chan_dahdi.c: Monitor doohicky got event
Alarm on channel 2
[Mar 16 09:49:19] WARNING[20928] chan_dahdi.c:
Hi,
My setup is:
IPPhone1 --- Asterisk1 with B410P Patton 4638 --- Asterisk2 ---
IPPhone2
I want to evaluate Asterisk1 in TE/PtmP mode.
So, Patton box is configured in NT/PtmP (with 3 BRI links between both
systems).
Anyway, asterisk -rx pri show spans keeps replying :
PRI span 1/0:
Olivier wrote:
Hi,
My setup is:
IPPhone1 --- Asterisk1 with B410P Patton 4638 --- Asterisk2 ---
IPPhone2
I want to evaluate Asterisk1 in TE/PtmP mode.
So, Patton box is configured in NT/PtmP (with 3 BRI links between both
systems).
Anyway, asterisk -rx pri show spans keeps
Hi all,
I saw that there was an auto-provisioning feature on asterisk 1.6.x for
the Polycoms. But no real documentation.
I would like to know how, exactly, does the network has to be
configured to allow that. I used to provision my Polycom phones with the
help of tftp or ftp. But if there's a
Mike wrote:
I've read some sources claiming that Asterisk does not need DAHDI for
timing in 1.6.1. Is this true? Searching the web, all I can find is
pages celebrating the fact but no actual documentation on which version
it was introduced in and how one would go about configuring an
We made a very simple application to insert the cost of a call into the
CDR table that Asterisk uses. We recently upgraded to Asterisk 1.6 and
I noticed that my application stopped working.
The reason is that my application depends on a field called route to
be NULL so that it
Folks,
I've read some sources claiming that Asterisk does not need DAHDI for
timing in 1.6.1. Is this true? Searching the web, all I can find is
pages celebrating the fact but no actual documentation on which version
it was introduced in and how one would go about configuring an external
time
Wilton Helm wrote:
I still am not quite on the same page with you, though. There are a lot of
commands that aren't function calls that go into various config files. The
most basic and obvious one is
exten
There must be a hundred of these and I don't know where they are listed with
Thanks all very much for the help pointers.
I've found all of the documentation on asterisk (especially 1.2-1.4) to
be more than adequate, and the voip-info wiki to be almost complete for many
things I've had to do in the past.
I also back in 2004 was able to bring up several high end large
New to Aserisk 1.6 and find the 'installation tutorials' seem low to non
existent.
You go to the main Asterisk page (digium.org) and really just old install
instructions for 1.2 are in the examples.
Download links only give you asterisk itself and not dahdi or libpri
which also are needed to run
I meant digium.com.
Yay for messups!
It's been one of those weeks.
Really.
New to Aserisk 1.6 and find the 'installation tutorials' seem low to non
existent.
You go to the main Asterisk page (digium.org) and really just old install
instructions for 1.2 are in the examples.
Download
you can use any 1.4 how to but just use dahdi (both modules and tools)
David
2009/1/27 Steve Gladden aster...@michiganbroadband.com
I meant digium.com.
Yay for messups!
It's been one of those weeks.
Really.
New to Aserisk 1.6 and find the 'installation tutorials' seem low to non
Hi Steve -
New to Aserisk 1.6 and find the 'installation tutorials' seem low to non
existent.
Welcome to Open Source!
Seriously, look at the README files accompanying asterisk, dahdi, and
libpri. They will give you compilation/installation instructions.
You can also search this list with
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Steve Gladden wrote:
Is 1.6 so cutting edge that I should not expect to find complete
documentation (yet)like I seem to be expecting very easily?
Most of what is applicable to 1.4 is applicable to 1.6. I'm running 1.6
without any hiccups -- YMMV.
New to Aserisk 1.6 and find the 'installation tutorials' seem low to non
existent.
I first looked at * about four months ago and rapidly came to the same
conclusion. Even with the O-Reilly book, which I purchased in paper, although
it is freely downloadable, I feel there is a huge dearth of
On Tue, Jan 27, 2009 at 11:24:38AM -0700, Wilton Helm wrote:
New to Aserisk 1.6 and find the 'installation tutorials' seem low to non
existent.
I first looked at * about four months ago and rapidly came to the same
conclusion. Even with the O-Reilly book, which I purchased in paper,
Wilton Helm wrote:
[snip]
My conclusion after installing a worthless * demo (that actually does
allow two SIPs to talk to each other) is that Asterisk is not of any
value to anyone other than a person who makes a full time career out
of running Asterisk systems. I've installed and
Wilton Helm wrote:
[snip]
My conclusion after installing a worthless * demo (that actually does
allow two SIPs to talk to each other) is that Asterisk is not of any
value to anyone other than a person who makes a full time career out
of running Asterisk systems. I've installed and
Thanks for the reply. I have looked at the links you provided and I think they
will be useful. I may have some issues with drivers for the HFC, but I guess I
won't know until I try it.
Wilton
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YMMV. Mine certainly did. For the better.
My comments were more negative than I intended. My installation is worthless
at this point because it is only a cookbook example and I haven't tried to
modify it to meet my needs. I didn't intend to imply that Asterisk is
worthless, just that I've
On Jan 27, 2009, at 10:50 PM, Wilton Helm wrote:
YMMV. Mine certainly did. For the better.
My comments were more negative than I intended. My installation is
worthless at this point because it is only a cookbook example and
I haven't tried to modify it to meet my needs. I didn't intend
I'm impressed that you picked up 6502 assembly out of an even larger
vaccum considering there was no 'net back then to help at all. Did
you install a PBX on an Atari?
No, I interfaced a Rockwell AIM to a 300 station Philips electromechanical PABX
(designed and built about 100 interface cards,
On Tue, Jan 27, 2009 at 12:50:42PM -0700, Wilton Helm wrote:
I just got a very nice posting from Tzafir showing me a web domain
I didn't even know existed.
It only includes documentation generated by 'make docs' . And is
actually linked from the README itself.
I'm not abandoning it by any
It actually does contain references of all applicaitons, CLI commands, and
such.
Where? I saw some examples, but I've never found an organized list of
commands. I'd love it.
Wilton
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On Tuesday 27 January 2009 15:05:57 Wilton Helm wrote:
It actually does contain references of all applicaitons, CLI commands, and
such.
Where? I saw some examples, but I've never found an organized list of
commands. I'd love it.
For applications, Appendix B, and for dialplan functions,
Thanks for engaging with me on this. I picked up the book and I see what you
mean about Appendix B. I had under-appreciated it probably because of a
paradigm shift I need to make. I think you meant Appendix E rather than F for
dialplan.
I still am not quite on the same page with you,
Wilton Helm wrote:
Thanks for engaging with me on this. I picked up the book and I see
what you mean about Appendix B. I had under-appreciated it probably
because of a paradigm shift I need to make. I think you meant
Appendix E rather than F for dialplan.
I still am not quite on the
2009/1/17 Steve Gladden aster...@michiganbroadband.com
The scenario we have is fax send/recieve software that ONLY talks T38
and an asterisk box.
We have ITSP providers that do NOT talk T38 but G711 only.
As you're using an IP connection, chances are you'll get issues with faxing
if you
On Sat, Jan 17, 2009 at 11:51 AM, Steve Gladden
aster...@michiganbroadband.com wrote:
The scenario we have is fax send/recieve software that ONLY talks T38
and an asterisk box.
We have ITSP providers that do NOT talk T38 but G711 only.
Does asterisk have the capability to take the T38 call
What you need is a so called T38Gateway application.
there is a patch o the tracker which you might want to try:
http://bugs.digium.com/view.php?id=13405
klaus
Steve Gladden schrieb:
The scenario we have is fax send/recieve software that ONLY talks T38
and an asterisk box.
We have ITSP
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