I understand the desire to try, but you are trying too hard. Getting a soft
modem to work with Asterisk is. like trying to push a string up a 10 foot
pipe.
At the least, buy an inexpensive FXO device from someone like Grandstream
and use it via Ethernet to work with Asterisk. If you have
Hey folks,
I can park a call with #70 after enabling that feature in
features.conf. However, once I retrieve the call from the parking
lot, #70 cannot be used to park it again. Worse yet, none of the
keys defined in the featuremap work anymore, include blindxfer or
automon.
Any ideas what may be
Hi folks,
I was using the following featuremap:
blindxfer = *1
disconnect = *9
atxfer = *2
parkcall = *7
automixmon = *0
and everything worked.
Then the need arouse to use some features like automixmon during
a conference, but MeetMet has the * key bound to the
(admin) menu. Thus, in
On Mon, 15 Jun 2009, Shiva Kumar wrote:
Hello Asterisk-users,
I am new to Asterisk. I got SIP Calls to work between two computers using a
soft phone and asterisk in the middle. Since then, I have been trying to get
my soft phone to make a PSTN call with terrible failure for about two days
now.
martin f krafft wrote:
Hey folks,
I can park a call with #70 after enabling that feature in
features.conf. However, once I retrieve the call from the parking
lot, #70 cannot be used to park it again. Worse yet, none of the
You fail to mention the version of Asterisk that you're working
also sprach Doug Lytle supp...@drdos.info [2009.06.16.1142 +0200]:
I can park a call with #70 after enabling that feature in
features.conf. However, once I retrieve the call from the parking
lot, #70 cannot be used to park it again. Worse yet, none of the
You fail to mention the version of
On Sun, Jun 14, 2009 at 03:10:03PM +1000, Alex Samad wrote:
On Sun, Jun 14, 2009 at 06:28:09AM +0300, Tzafrir Cohen wrote:
[snip]
The scripts for downloading the post-build firmware were moved to the
separate dahdi-firmware package (sadly it has not made it into the
archive yet). As
Can you confirm that currently there is no way to update the caller
ID via the manager interface once the B leg is ringing or connected?
Looks like this would be feasible with the functions introduced in
https://issues.asterisk.org/view.php?id=8824 ([patch] Remote (called)
Party Identification -
On Sun, Jun 14, 2009 at 06:28:09AM +0300, Tzafrir Cohen wrote:
[snip]
Although I think I did see it download the firmware
The scripts for downloading the post-build firmware were moved to the
separate dahdi-firmware package (sadly it has not made it into the
archive yet). As the firmware
martin f krafft wrote:
Yes, please. It's good to know that this is a known bug.
Please remember, the patch is for 1.4
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
Index:
Hi all,
I am new to Asterisk and have a basic question. I am using Asterisk as PSTN
gateway and want to connect Asterisk to a PSTN switch. We have ordered a Digium
T1 card.
I have installed the basic asterisk software and tried a call with dummy
extension. Can someone please tell
Hi
it seems like chan_dahdi.so is missing in debian asterisk 1.4.21
so I have upgraded to 1.6 and no I can load chan_dahdi.so
Command 'module load chan_dahdi.so' failed.
[Jun 16 21:22:30] WARNING[4360]: loader.c:417 load_dynamic_module: Error
loading module 'chan_dahdi.so':
On Tue, Jun 16, 2009 at 09:04:37PM +1000, Alex Samad wrote:
On Sun, Jun 14, 2009 at 06:28:09AM +0300, Tzafrir Cohen wrote:
[snip]
Although I think I did see it download the firmware
The scripts for downloading the post-build firmware were moved to the
separate dahdi-firmware
On Monday 15 June 2009 11:44:27 pm Antoine Patte wrote:
Hello,
I try to set up a gateway gtalk to sip.
I test asterisk 1.6.1 and 1.4.21 from debian repository and the result
is identical : no sound during the call.
I had the same issue, only from gtalk to asterisk with some connections and
Alex Samad wrote:
it seems like chan_dahdi.so is missing in debian asterisk 1.4.21
so I have upgraded to 1.6 and no I can load chan_dahdi.so
Command 'module load chan_dahdi.so' failed.
[Jun 16 21:22:30] WARNING[4360]: loader.c:417 load_dynamic_module: Error
loading module
also sprach Doug Lytle supp...@drdos.info [2009.06.16.1314 +0200]:
Please remember, the patch is for 1.4
Right, and I found the corresponding lines in 1.6. But there are
more questions now:
- snprintf(returnexten, sizeof(returnexten), %s||t, peername);
+ snprintf(returnexten,
On Tue, Jun 16, 2009 at 07:03:06AM -0500, Kevin P. Fleming wrote:
Alex Samad wrote:
it seems like chan_dahdi.so is missing in debian asterisk 1.4.21
so I have upgraded to 1.6 and no I can load chan_dahdi.so
Command 'module load chan_dahdi.so' failed.
[Jun 16 21:22:30]
On Tue, Jun 16, 2009 at 02:35:08PM +0300, Tzafrir Cohen wrote:
On Tue, Jun 16, 2009 at 09:04:37PM +1000, Alex Samad wrote:
[snip]
dahdi_genconf generates configuration. It is a tool intended to help you
and not a required step.
It defaults to using mg2[1]. You can tell it to use a
On Tue, Jun 16, 2009 at 02:35:08PM +0300, Tzafrir Cohen wrote:
On Tue, Jun 16, 2009 at 09:04:37PM +1000, Alex Samad wrote:
On Sun, Jun 14, 2009 at 06:28:09AM +0300, Tzafrir Cohen wrote:
[snip]
Although I think I did see it download the firmware
The scripts for downloading
Alex Samad wrote:
some question I have now is when i do a dahdi show channel 1 i get these
interesting results
Echo Cancellation:
128 taps
currently OFF
I have a hardware echo can and I have asked for it to be turned on !
This is nothing new; the echo canceller on a channel
Hi all! Do anybody has a full working environment using externip on an asterisk
box behind a nat? I tried with two diferent boxes (Elastix-1.4.24 e
Trixbox-1.4.22-3)and the asterisk do not replace neither contact, neither sdp
headers info with the externip informed on sip.conf general
Yes - we typically install behind NAT. The issue will usually be your
firewall setup ...assuming you have setup your peers for NAT.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ricardo
Martins
Sent:
On Tue, 16 Jun 2009, Ricardo Martins wrote:
Hi all! Do anybody has a full working environment using externip on an
asterisk box behind a nat? I tried with two diferent boxes
(Elastix-1.4.24 e Trixbox-1.4.22-3)and the asterisk do not replace
neither contact, neither sdp headers info with
Hi Rob,
sip show users presents the following.
Username Secret Accountcode Def.Context ACL NAT
2100 default
No RFC3581
And that was what I needed! Thanks for the hint Rob, I just had to
move the context
sean darcy escribió:
For our internal fax machines, I'm checking if the faxes are going to
branch offices. If they are, I want to capture and email them to the
branches. I've set up extension 8447 to test this.
A fax machines is connected via an SPA 2102 on 173. Any calls from 173
are
Tzafrir Cohen wrote:
Duh. Ignore this. You asked about the hardware EC. The hardware EC can
be activated regadrdless of the software EC you use.
(Not sure exactly how. Anybody?)
It's automatic; nothing needs to be specified in /etc/dahdi/system.conf
at all. If chan_dahdi is configured to
On Tue, Jun 16, 2009 at 9:56 AM, Danny Nicholasda...@debsinc.com wrote:
The problem is the Asterisk Read function. It is set to accept as many 0-9
and * as you want to throw at it, then stop on # or timeout. Unless you
disable the # stops, you can't use # in features. I would strongly
The problem is the Asterisk Read function. It is set to accept as many 0-9
and * as you want to throw at it, then stop on # or timeout. Unless you
disable the # stops, you can't use # in features. I would strongly caution
against that because you would almost certainly break something else.
LESC - proceed at your own peril...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan Moore
Sent: Tuesday, June 16, 2009 10:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Anyone have any idea on how to force marker bits on in RTP ?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian
Marsh
Sent: 10 June 2009 14:50
To: Asterisk Users Mailing List - Non-Commercial
You can't change the actual callerid, but why not load a variable and update
that?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp
Kempgen
Sent: Tuesday, June 16, 2009 6:02 AM
To: Asterisk Users
Yes Gordon. I'm using nat=yes and I don't have an ALG enabled
router/firewall. I used the sip debug output on the asterisk(s) and
could see the sdp headers as they were gererated by asterisk, with the
wrong (internal) address on it.
Asterisk is sending the audio to the correct way, the public
I'm not sure if your problem is addressed by this:
https://issues.asterisk.org/view.php?id=14546 . If that's the case it was
solved in version 1.4.25
Best regards,
Santi
2009/6/16 Ricardo Martins rpopp...@gmail.com
Yes Gordon. I'm using nat=yes and I don't have an ALG enabled
Philipp Kempgen wrote:
Can you confirm that currently there is no way to update the caller
ID via the manager interface once the B leg is ringing or connected?
Looks like this would be feasible with the functions introduced in
https://issues.asterisk.org/view.php?id=8824 ([patch] Remote
In the B leg, check for the variable value instead of Callerid(num).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp
Kempgen
Sent: Tuesday, June 16, 2009 10:27 AM
To: Asterisk Users Mailing List -
On Tue, 16 Jun 2009, Ricardo Martins wrote:
Yes Gordon. I'm using nat=yes and I don't have an ALG enabled
router/firewall. I used the sip debug output on the asterisk(s) and
could see the sdp headers as they were gererated by asterisk, with the
wrong (internal) address on it.
Asterisk
On Tue, Jun 16, 2009 at 2:23 AM, martin f krafftmadd...@madduck.net wrote:
Hey folks,
I can park a call with #70 after enabling that feature in
features.conf. However, once I retrieve the call from the parking
lot, #70 cannot be used to park it again. Worse yet, none of the
keys defined in
On Tue, 2009-06-16 at 13:02 +0200, Philipp Kempgen wrote:
Can you confirm that currently there is no way to update the caller
ID via the manager interface once the B leg is ringing or connected?
Correct. Well, at least not with 1.6.0 or 1.6.1 or 1.6.2 branches.
Looks like this would be
Hi all, tks for your time, I could solve the problem on the box that is
behind the iptables firewall. I rewrote the rules and the externip is
coming with the messages, working with either public and nated uas.
I will try further with the asterisk box behind a linksys firewall.
The rules I
FYI,
Got the Asterisk to Cisco CallManager working over h323. After many
days of trying it was a pretty simple fix.
This is what I had:
[globals]
CISCOTRUNK=H323/callman02
[cisco]
exten = _8XXX,1,Dial(${CISCOTRUNK}/${EXTEN:1...@172.16.200.10:1720)
So if I just write it out without the
Hi to all
excuse me but i don't understand what is the correct configuration
needed to setup a transfer with REFER in SIP.
I've tried the transfer() application, but i've experienced some
problem, i can't reproduce the error in a clear debug environment but
randomly the call crash before to be
Anyone contacted by a Mike Low of Toronto who does business under a list
of names including Zigit, ZonFon, Call Cheap, Kallback King, Cell-0.com,
etc. and etc.
would be well advised to contact me offline before proceeding to provide
services or consulting assistance.
--
George Pajari (dCAP),
On Tue, Jun 16, 2009 at 09:42:34AM -0500, Kevin P. Fleming wrote:
Tzafrir Cohen wrote:
Duh. Ignore this. You asked about the hardware EC. The hardware EC can
be activated regadrdless of the software EC you use.
(Not sure exactly how. Anybody?)
It's automatic; nothing needs to be
On Tue, Jun 16, 2009 at 08:06:57AM -0500, Kevin P. Fleming wrote:
Alex Samad wrote:
some question I have now is when i do a dahdi show channel 1 i get these
interesting results
Echo Cancellation:
128 taps
currently OFF
I have a hardware echo can and I have asked for it to
On Wed, Jun 17, 2009 at 07:16:53AM +1000, Alex Samad wrote:
On Tue, Jun 16, 2009 at 08:06:57AM -0500, Kevin P. Fleming wrote:
Alex Samad wrote:
[snip]
Default law: ulaw
I have a alaw:1-4 in the conf file, but it doesn't seem to take
That is not valid syntax for
On Wed, Jun 17, 2009 at 07:08:10AM +1000, Alex Samad wrote:
On Tue, Jun 16, 2009 at 09:42:34AM -0500, Kevin P. Fleming wrote:
Tzafrir Cohen wrote:
Duh. Ignore this. You asked about the hardware EC. The hardware EC can
be activated regadrdless of the software EC you use.
(Not
On Wed, Jun 17, 2009 at 01:23:19AM +0300, Tzafrir Cohen wrote:
On Wed, Jun 17, 2009 at 07:08:10AM +1000, Alex Samad wrote:
On Tue, Jun 16, 2009 at 09:42:34AM -0500, Kevin P. Fleming wrote:
Tzafrir Cohen wrote:
Duh. Ignore this. You asked about the hardware EC. The hardware EC can
On Tue, Jun 16, 2009 at 10:13 AM, Miguel Molinammol...@millenium.com.co wrote:
sean darcy escribió:
For our internal fax machines, I'm checking if the faxes are going to
branch offices. If they are, I want to capture and email them to the
branches. I've set up extension 8447 to test this.
A
Yep, heard that a few times before, not just on this list.
George Pajari wrote:
Anyone contacted by a Mike Low of Toronto who does business under a list
of names including Zigit, ZonFon, Call Cheap, Kallback King, Cell-0.com,
etc. and etc.
would be well advised to contact me offline before
Hello, all. The little bit of reading I've done on lua makes me eager
to give it a try. However, when I try to install it (Asterisk 1.6.1.1
on CentOS 5.3), it is not available in menuselect. I have installed lua
and lua-devel. I've seen very little about it in my Internet searches.
What else
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