Hello!
I use similar setup.
Probably you need Answer() in receiving end. And wait(3) before
receiving fax.
T.38 works fine with 1.6.2.
Ilmars.
On 2010.05.05. 0:17, sean darcy wrote:
On 5/4/2010 7:32 AM, Miguel Amez wrote:
App_fax? I didn't hear about that. What's that?
Could you please
Hi Juan,
Thanks for your inputs, I tried with changes you suggested and find my
observation.
After adding context and extension able to make an outgoing call
[Digium-fxs to X-lite2000].
But not able to make incoming call [X-lite2000 to Digium-fxs]. Call
failed with,
(1) “*Call
i get may debug messages like this:
DEBUG[30684] channel.c: Internal timing is disabled
(option_internal_timing=0 chan-timingfd=-1)
Is because dahdi is not installed?
Can this be a possible cause of this behaviour?
On Tue, May 4, 2010 at 9:54 PM, nik600 nik...@gmail.com wrote:
Dear all
on
Tilghman Lesher wrote:
Okay, second idea is that you should very carefully examine your CDR table
layout and ensure that the columns that you have match EXACTLY what the
module expects you to have. If Asterisk expects you to have a column that you
don't (or the column type is wrong), that is
Dear List,
My Dial command:
exten = _X.,n,Dial(SIP/PBX2/1234,60,G(connect-jack^${EXTEN}^1))
exten = h,1,
[connect-jack]
exten = _X.,1,NoOp(${CHANNEL}) ; Leg A
exten = _X.,2,NoOp(${CHANNEL}) ; Leg B
The problem is: after answering, [connect-jack] both priorities are
executed, and right
Hi everyone,
I am trying to figure out the behavior of trustrpid
Basically its not behaving the way I expected it to or maybe I am
missing a configuration option or something else.
When a call from a phone is sent to the * box it has the following sip
headers:
From: From Phone
Ok..So what ip phone model do NAT?
Sebastian
On Wed, May 5, 2010 at 12:26 PM, Luki lugos...@gmail.com wrote:
However, when I connect a PC to that port, SPA922 works as bridge.
Exactly. The SPA9x2 has a 2-port switch; no NAT, no routing (unlike
the SPA2102, etc).
I think the 5.1 series
Hi!
[connect-jack]
exten = _X.,1,NoOp(${CHANNEL}) ; Leg A
exten = _X.,2,NoOp(${CHANNEL}) ; Leg B
The problem is: after answering, [connect-jack] both priorities are
executed, and right after executing them call drops.
The call legs drop because you do not do anything with them, since your
I am trying to change a 1.6 realtime statement into a 1.2 realtime
statement and I know much has changed. I wish I could just upgrade, but
alas not right now.
exten =x,n,Set(NULL1=${REALTIME(schedules,id,${SCHEDULE})})
comes back with
pbx.c:1371 ast_func_read: Function REALTIME not
Hi,
Great! I thought I won't see leg B channel while using M(), but I do!
:) M() did my day.
Thanks.
On Thu, May 6, 2010 at 4:29 PM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
Hi!
[connect-jack]
exten = _X.,1,NoOp(${CHANNEL}) ; Leg A
exten = _X.,2,NoOp(${CHANNEL})
On 6 May 2010, at 14:16, Sebastian Milioto wrote:
Ok..So what ip phone model do NAT?
I think you'd struggle to find one. If it's a requirement you're probably doing
something wrong...
S
--
_
-- Bandwidth and Colocation
Hi Garge -
exten =
,1,Asterisk_Application(Action) ;Dial(Zap/1/${Phone_Number_you want})
Two things:
1. There is no such thing as Zap anymore. Zap has been renamed to
Dahdi because of a trademark issue. So your extension should look
like:
exten = ,Dial(Dahdi/1/)
2. Do you
On 5/5/10, Adrian Marsh adrian.ma...@ubiquisys.com wrote:
Anyone have any experience with a Japanese local VoIP termination
supplier?
I've emailed a few companies looking to setup some PSTN to SIP and SIP
to PSTN termination, but no luck so far.
Thanks,
Adrian
--
Sent from my
Ok..So what ip phone model do NAT?
I think you'd struggle to find one. If it's a requirement you're probably
doing something wrong...
Definitely get a router. Plug the IP phone into the router, and then
you can plug the computer into the phone or the router.
- Noah
--
Dear list,
i have re-compiled again the source code of amr patch for 1.6
(https://asteriskvideo.svn.sourceforge.net/svnroot/asteriskvideo/amr/asterisk-1.6-AMR.patch)
The patch does not compile with the static function into frame.c
called :
static int amr_samples(unsigned char *data, int datalen)
On 5/5/10, Adrian Marsh adrian.ma...@ubiquisys.com wrote:
Anyone have any experience with a Japanese local VoIP termination
supplier?
I've emailed a few companies looking to setup some PSTN to SIP and SIP
to PSTN termination, but no luck so far.
Thanks,
Adrian
--
Sent from my
It is a building, with 24 separated rooms, each room will have a PC and a IP
Phone. Every room connected to a switch Cisco 2950.
I want keeping all PCs isolated behind a NAT (no access to neighbour's PC),
and still keep communication in same LAN between all IP Phones.
Should I take another
Hi all,
I have been testing several asterisk versions and I found out that all the
previus version of asterisk worked fine.
After 1.4.22 it cease to work.
In the change log referring to iax from 1.4.22 to 1.4.23 I found this:
On Thu, 6 May 2010, Sebastian Milioto wrote:
It is a building, with 24 separated rooms, each room will have a PC and a IP
Phone. Every room connected to a switch Cisco 2950.
I want keeping all PCs isolated behind a NAT (no access to neighbour's PC),
and still keep communication in same LAN
Philipp von Klitzing wrote:
Hi!
apps like playback do an implicit answer and this fires up the billsec
counter.
OK, here is my dialplan:
exten = _011X.,n,Playback(this-call-will-end-in)
exten =
_011X.,n,Dial(SIP/${ext...@${ldtrunk1},60,L(${ms}:3))
Is there any way
On 05/04/2010 07:41 PM, Leif Madsen wrote:
OK, I got sufficiently curious to make sure Skype for Asterisk still loaded
on
1.6.2.7. It does for me, but I had to run make install in my Skype source
directory. One of the modules loaded, but the 'skype' CLI command was not
available until
It is a building, with 24 separated rooms, each room will have a PC and a IP
Phone. Every room connected to a switch Cisco 2950.
I want keeping all PCs isolated behind a NAT (no access to neighbour's PC),
and still keep communication in same LAN between all IP Phones.
Should I take another
I see the following in SPA922 System tab (new firmware)
VLAN Settings Enable VLAN:yesnoEnable CDP:yesno VLAN ID:PC Port VLAN Highest
Priority:01234567No Limit Enable PC Port VLAN Tagging:yesnoPC Port VLAN ID:
VLAN ID:1 for all Phones, and VLAN 2, 3, 4, 5..,24 for each PC. This
should work,
-Original Message-
From: asterisk-users-boun...@lists.digium.com on behalf of Noah Miller
Sent: Thu 5/6/2010 10:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] OT: NAT in SPA922
It is a building, with 24 separated rooms, each room will
On Thu, 6 May 2010, Sebastian Milioto wrote:
I see the following in SPA922 System tab (new firmware)
VLAN Settings Enable VLAN:yesnoEnable CDP:yesno VLAN ID:PC Port VLAN Highest
Priority:01234567No Limit Enable PC Port VLAN Tagging:yesnoPC Port VLAN ID:
VLAN ID:1 for all Phones, and VLAN
In case anybody was following this thread,
wanted to let people know that the fix made it into SVN,
and is packaged into
1.6.2.8-rc1
Huge thanks to Kevin and Tilghman
On Wed, Apr 21, 2010 at 3:40 PM, David Backeberg dbackeb...@gmail.com wrote:
issue opened.
Hi!
Should I take another approach on that?
Put each PC in its own VLAN. Keep all the phones in one VLAN.
Note: VLANs are an organisational tool, and do not really add security.
If you want to go with VLANs in thise case then rather consider port
based VLAN (configured in the switch
Yes, I purchased licenses for Fax for Asterisk and yes I called tech support
and had the WORST experience I have ever had with any technical support
call.
I am running Asterisk 1.6.2.6 and:
FAX For Asterisk Components:
Applications: 1.6.2.0_1.2.0
voipgw01Digium FAX Driver: 1.6.2.0_1.2.0
Am I missing something here? I see
if (needvideo) { /* only if video response is appropriate */
add_line(resp, m_video-str);
add_line(resp, a_video-str);
add_line(resp, hold); /* Repeat hold for the video stream */
} else if
I can confirm that the following fixes my problem:
--- chan_sip.c (revision 261450)
+++ chan_sip.c (working copy)
@@ -10357,12 +10357,22 @@
strlen(connection) + strlen(session_time);
if (needaudio)
len += m_audio-used + a_audio-used + strlen(hold);
+
On Wednesday 05 May 2010 18:29:26 Neeraj Chand wrote:
---
Message: 10
Date: Wed, 5 May 2010 10:26:34 -0500
From: Tilghman Lesher tles...@digium.com
Subject: Re: [asterisk-users] CDR to MS-SQL via ODBC issue
To:
Does anybody have T.38 faxing working with Flowroute? I am running
Asterisk 1.6.2.7-rc3 with spandsp 0.0.6-pre17. I can successfully
receive faxes over ulaw. I enabled T.38 with t38pt_udptl=yes in
sip.conf. When I receive a fax it tries to negotiate T.38 and
Flowroute sends back a Bad Request
On 05/06/2010 05:46 PM, Ryan Wagoner wrote:
Does anybody have T.38 faxing working with Flowroute? I am running
Asterisk 1.6.2.7-rc3 with spandsp 0.0.6-pre17. I can successfully
receive faxes over ulaw. I enabled T.38 with t38pt_udptl=yes in
sip.conf. When I receive a fax it tries to negotiate
On Thu, May 6, 2010 at 5:54 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 05/06/2010 05:46 PM, Ryan Wagoner wrote:
Does anybody have T.38 faxing working with Flowroute? I am running
Asterisk 1.6.2.7-rc3 with spandsp 0.0.6-pre17. I can successfully
receive faxes over ulaw. I enabled
I wasn't sure how the lines were counted. Here is the debug output
from Asterisk where it is building the invite packet. I looked at the
a=T38 lines and nothing is standing out to me.
Ryan
[May 6 13:29:05] DEBUG[32389] chan_sip.c: Header 0 [ 47]: INVITE
sip:+num...@x.x.x.x:5060 SIP/2.0
[May
I'm migrating an application running on a fairly old 1.4 (or 1.2?)
version of Asterisk to some boxes running 1.6.0.27
The application takes an inbound INVITE like:
mumble-fratz-sip%3afoo%40bar@asteriskbox.abc.com:5062
The older version of asterisk replies with a 200 OK and a Contact:
header
Hi Luki,
Thank you so much.. The soft xx worked perfectly. The rtptimeout is
excellent also.
Regards,
S.
On 5 May 2010 23:59, Luki lugos...@gmail.com wrote:
Are there any CLI commands to free this up or any other ways without
having
to restart asterisk.
Did you try soft hangup channel?
On Thu, May 6, 2010 at 7:11 PM, Warren Selby wcse...@selbytech.com wrote:
On Thu, May 6, 2010 at 5:54 PM, Kevin P. Fleming kpflem...@digium.com
wrote:
On 05/06/2010 05:46 PM, Ryan Wagoner wrote:
Does anybody have T.38 faxing working with Flowroute? I am running
Asterisk 1.6.2.7-rc3 with
I am trying to figure out the behavior of trustrpid
Basically its not behaving the way I expected it to or maybe I am
missing a configuration option or something else.
When a call from a phone is sent to the * box it has the following sip
headers:
From: From Phone sip:1001 at
On Fri, May 07, 2010 at 02:20:24AM +, crjw wrote:
| I am trying to figure out the behavior of trustrpid
|
| Basically its not behaving the way I expected it to or maybe I am
| missing a configuration option or something else.
|
| I had a similar situation in which playing with trustrpid
I think this is a motel kind of situation and a PVLAN serves the
situation right. Put all the ipphones in the voice vlan as suggested,
make a seperate isolated vlan for the PCs, this will restrict traffic
between the clients.
Rgds,
Vineet Bhojnagarwala RCDD, NTS, OSP
Spear Networks Pvt
Alternatively, if using normal vlans, this can also be achieved by
enabling access list on the switch and restrict traffic flows.
Generally this is done on a layer 3 switch, don't think it will
support on your switch model.
Rgds,
Vineet Bhojnagarwala RCDD, NTS, OSP
Spear Networks Pvt Ltd
Dear all,
We have been using asterisk for 4 years. Now we have got problems which
occurs during the attended transfer.
During attended transfer, sometimes we cannot hear the sound of 'pbx-transfer'.
I cannot understand why this is happening?
log is :
-- Started music on hold, class 'default',
Is there anything special that has to be done to make video calls work?
It doesn't seem to work for me (no video).
What CODECS are supported?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
44 matches
Mail list logo
