Hello,
To help clarify, Jabber is receiving the incoming packets, but
Asterisk does not seem to be associating it with the gtalk
configuration and the call is not routed into any context. The remote
caller only hears continous ringing. However, outgoing, gtalk and
jabber work fine.
What could
On 13/06/11 19:44, Mike Diehl wrote:
Hi all,
I'm trying to provision my PAP2T's to use a SVR lookup to find the Asterisk
server. I'm using a provisioning file that contains an element like:
Proxy_1_ _sip._udp.example.com/Proxy_1_
However, the PAP doesn't seem to be able to find my server
Hi All;
My ISDN was working fine, and suddenly I start getting the below:
sig_pri.c:985 pri_find_dchan: Span 1: No D-chanannel anyway!
There is a Yellow Alarm, so what it could be the problem?
My configuration as following:
system.conf
span=1,1,0,ccs,hdb3,yellow
bchan=1-15,17-31
dchan=16
bilal ghayyad wrote:
There is a Yellow Alarm, so what it could be the problem?
Experience says you need to call your provider.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
--
ringinuse=yes will send your call to Agent only if her phone state is in 'In
use' or 'Not in use' BUT not when it is in
'ringing'.
So probably you can not achieve what you want with the current Queue()
implementation in Asterisk.
[SATISH]
On Mon, Jun 13, 2011 at 4:14 PM, Deka, Rajib IN MAA SL
Hy,
It doesn't seems to change anything.
Fortunately, the delay is small enough and the callers are not disturbed
by it so I give up improving this aspect.
regards,
Le 13/06/2011 23:09, Florent THOMAS a écrit :
A *great* thanks to you,
I will try it ASAP.
I'll let you know!
In my
Le 12/06/2011 20:41, Florent THOMAS a écrit :
Le 11/06/2011 17:54, Gordon Henderson a écrit :
On Sat, 11 Jun 2011, Florent THOMAS wrote:
Hy all of you,
Is anybody has a tutorial for integrate a siemens gigaset as180 and
connect it to Asterisk.
I've searched a lot and didn't found
Dear Doug;
But I am afraid it is a bug because I read something this in the below link:
https://issues.asterisk.org/view.php?id=17270
But maybe this was for old driver .. again, I am afraid if it is a bug.
DAHDI Version: 2.4.1
libpri-1.4.11.5
Any advise if the below message is a bug?
[Jun
bilal ghayyad wrote:
But I am afraid it is a bug because I read something this in the below
This bug is referring to Zaptel, not dahdi.
If things were working fine, and you haven't made any recent changes, in
my experience it's always been provider (99%) or cable (1%) issues when
I've lost
Hi,
I'm using a two-years old installation script for the first time on a
Squeeze (linux 2.6.32) platform.
For an unknown reason (might be an obvious one), Dahdi can't be loaded
anymore.
1. First of all, it seems /dev/dahdi content was previously (ie in Lenny)
owned by asterisk:asterisk
On Tue, Jun 14, 2011 at 9:44 AM, Olivier oza_4...@yahoo.fr wrote:
Hi,
I'm using a two-years old installation script for the first time on a
Squeeze (linux 2.6.32) platform.
For an unknown reason (might be an obvious one), Dahdi can't be loaded
anymore.
1. First of all, it seems /dev/dahdi
On Tue, Jun 14, 2011 at 03:44:32PM +0200, Olivier wrote:
Hi,
I'm using a two-years old installation script for the first time on a
Squeeze (linux 2.6.32) platform.
For an unknown reason (might be an obvious one), Dahdi can't be loaded
anymore.
1. First of all, it seems /dev/dahdi content
Hello all,
I'm having a problem with my intercom function that I use for under-chin
paging. I'm running 1.6.2.13 on this server, and we use Linksys SPA-942's
for our general phones. I have a global defined which has all the SIP
channels concatenated together - this is ${ALL-PAGE-EXTS}.
The
After a reboot, I can't reproduce the problem anymore which is quite
frustating.
2011/6/14 Tzafrir Cohen tzafrir.co...@xorcom.com
On Tue, Jun 14, 2011 at 03:44:32PM +0200, Olivier wrote:
Hi,
I'm using a two-years old installation script for the first time on a
Squeeze (linux 2.6.32)
The provider came to the site and they have a gateway, they connected the
gateway to the E1 cable, and we called the number and it answered !!!
So it is no more provider issue.
I start beleive, it is a bug maybe in the dahdi and libpri. I tried all the
possibilities, but no luck.
WHAT COULD
Struggling with an IP650 and 7 IP335s this morning. I have the following
hints defined (courtesy of FreePBX 2.9):
extensions_additional.conf:exten = 300,hint,SIP/300
extensions_additional.conf:exten = 301,hint,SIP/301
extensions_additional.conf:exten = 302,hint,SIP/302
Elliot,
You need to execute sendDTMF(1)
Articles are available with detailed setup description.
-Vladimir
On 6/14/2011 1:26 AM, Elliot Murdock wrote:
Hello,
To help clarify, Jabber is receiving the incoming packets, but
Asterisk does not seem to be associating it with the gtalk
You know I'm a flight engineer but non of the airlines wanted to hire
me because other than the self proclaimed title I have no clue how to
operate or maintain an aircraft.
The dictionary is probably wrong you should patch libpri
On Tue, Jun 14, 2011 at 11:43 AM, bilal ghayyad bilmar...@yahoo.com
Hi,
I'm having problems with audio coming to my asterisk PBX from eyebeam
softphone running on Windows XP machine, it has a lot of Jitter. I've
taken a lot of tcpdumps from Asterisk machine and jitter comes only from
softphones, audio comming and going to gateways, IP phones, etc has no
Anyone have recommendations for a gateway / ATA for business that can do
GroundStart? Preferably with an rj-21 - but okay if not..
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join
Dears;
To patch libpri: I just place the patch file in the libpri source directory and
then I run make and make install?
Or I need to compile the dahdi and asterisk also?
If the problem stayed, do I have to go for previous libpri version? Or for
previous dahdi version and asterisk version?
http://www.linuxtutorialblog.com/post/introduction-using-diff-and-patch-tutorial
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
bilal ghayyad
Sent: Tuesday, June 14, 2011 2:47 PM
To:
Robert Huddleston wrote:
Anyone have recommendations for a gateway / ATA for business that can
do GroundStart? Preferably with an rj-21 -- but okay if not..
I don't know of any ATA that will do GS
An RJ-21 is the designation for a 66 block with 25 pair connector on the
side
GS is
Hi,
1. Is there any manual entry about modprobe's options relative to a given
Dahdi driver (wctdm24xxp, for instance) ?
2. When loading a wctdm24xxp driver, is there any parameter to pass to
modprobe to configure a span in NT/point to point mode ?
3. After running dahdi_cfg -v , I can read
2011/6/9 Richard Mudgett rmudg...@digium.com
We have two pri line and I want to see how asterisk distribute
outgoing call per channels
I meant it use first last channel 47 or it will use first channel?
Or it will allocate dynamically ?
Extracted from chan_dahdi.c:
From: John Novack
Sent: Tue 6/14/2011 3:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Ground Start ATA / VOIP Gateway
Robert Huddleston wrote:
Anyone have recommendations for a gateway / ATA for business that can do GroundStart? Preferably
On 06/14/2011 03:11 PM, Olivier wrote:
Hi,
1. Is there any manual entry about modprobe's options relative to a
given Dahdi driver (wctdm24xxp, for instance) ?
2. When loading a wctdm24xxp driver, is there any parameter to pass to
modprobe to configure a span in NT/point to point mode ?
3.
Extracted from chan_dahdi.c:
Dial(DAHDI/pseudo[/extension[/options]])
Dial(DAHDI/channel#[c|rcadance#|d][/extension[/options]])
Dial(DAHDI/subdir!channel#[c|rcadance#|d][/extension[/options]])
Dial(DAHDI/ispan[/extension[/options]])
15906 root 39 19 94988 832 520 R 100.9 0.0 11:22.21 sysmon15913
root 39 19 94988 832 520 R 100.9 0.0 11:21.95 sysmon15905 root
39 19 94988 832 520 R 98.9 0.0 11:24.76 sysmon15908 root 39 19
94988 832 520 R 98.9 0.0 11:20.76 sysmon15909 root 39 19
I only need 4 fxs. I looked at the IAD2431 but it uses T1/E1 as WAN. If I
could assign Fast Ethernet to WAN that would be great. Budget is not that
great
From: asterisk-biz-boun...@lists.digium.com
[mailto:asterisk-biz-boun...@lists.digium.com] On Behalf Of Sum Ding Wong
Sent: Tuesday, June
Ya - customer is on a nice NEC SV8100.. The card is a ground start card..
they are currently being fed by a Cisco IAD2431 w/ RJ-21 punchdown
cross-connect.
But that IAD2431 uses T1/E1 as WAN.. They are doing away with the T1 and
want to use Ethernet for wan.
So IAD2431 would be great - but
Running asterisk 1.8.4.2, and occasionally we'll have a call drop and
the SIP retransmit error show up on the console. I actually think the
retransmit error is just a symptom of something else, possibly centered
around a timing issue.
I tried res_timing_dahdi and that worked for about a week,
The SV8100 can do either ground or loop
Assuming you can access the system it can easily be changed.
Programming manual here:
http://www.telecomcepts.com/downloads/SV8100/SV8100 Programming Manual_1.pdf
the original installer may have locked it down, but it CAN be changed.
John Novack
2011/6/14 Kevin P. Fleming kpflem...@digium.com
On 06/14/2011 03:11 PM, Olivier wrote:
Hi,
1. Is there any manual entry about modprobe's options relative to a
given Dahdi driver (wctdm24xxp, for instance) ?
2. When loading a wctdm24xxp driver, is there any parameter to pass to
modprobe
On 6/14/11 9:26 AM, Cassius Smith wrote:
Hello all,
I'm having a problem with my intercom function that I use for under-chin
paging. I'm running 1.6.2.13 on this server, and we use Linksys SPA-942's
for our general phones. I have a global defined which has all the SIP
channels concatenated
Hi,
My genconf_parameter is :
# grep -v ^# genconf_parameters
lc_countryfr
context_linesremote
group_lines1
bri_sig_stylebri
echo_canoslec
pri_termtype
SPAN/*NT
(I also tried various identations and syntax for pri_termtype line such as
Ok. Something isn't right. With a user that is local to my SIP user
database calls the queue phone number everything works without issue. It is
when a remote user (like someone from the PSTN) calls the queue phone number
that the caller gets put into the queue and the agent/member doesn't
On 6/14/11 4:25 PM, Russ Meyerriecks wrote:
On 6/14/11 9:26 AM, Cassius Smith wrote:
Hello all,
I'm having a problem with my intercom function that I use for under-chin
paging. I'm running 1.6.2.13 on this server, and we use Linksys SPA-942's
for our general phones. I have a global defined
Hey all
I am having instances where voicemail boxes will have a 1 message and
no 0 message this causes the user to be told that they have a message
that they can't get at. If I renumber the messages manually to start with
the 0 numbering then the user can get their messages. What
If you have experience with these phones...
We are trying to figure out how to transfer an established call on the
SPA504G while a second call is incoming. At present, the receptionist
has to answer every single incoming call before the XFER softkey is seen
again. This is completely
I'll have to look at that then - as I thought the card actually said Ground
Start on it.. I may have missed or it was scratched off the word loop start
From: John Novack [mailto:jnov...@stromberg-carlson.org]
Sent: Tuesday, June 14, 2011 5:20 PM
To: Robert Huddleston
Cc: 'Asterisk Users
that system can also handle IP trunks, though the equipment might not be
available to you or outside your budget window
How does this relate to Asterisk, or does it?
John Novack
Robert Huddleston wrote:
I'll have to look at that then -- as I thought the card actually said
Ground Start on
considering providing the sip trunking nyself via asterisk.
the sip trunking looks expensive - card and licenses from nec.
Sent from my iPhone
On Jun 14, 2011, at 6:06 PM, John Novack jnov...@stromberg-carlson.org wrote:
that system can also handle IP trunks, though the equipment might not be
One more piece to add. I had mentioned before that I could get a call from
a PSTN user to work the first time. So here is all the output of a Good
call from a PSTN user after I have performed a RELOAD on asterisks CLI
http://pastebin.com/9RSvQsmN
And when the caller or agent hangs this call up
Agreed NEC isn't cheap. Their products are generally pretty good and
robust though. I have an earlier one still working for 18 years and counting
Of course, when one considers the asterisk machine, configuration time,
firewall and the rise in sip hacking sip trunking can easily turn into
a
Hello,
Seems that it's been spotted and tracked at
https://issues.asterisk.org/jira/browse/ASTERISK-17993
--Elliot
On Tue, Jun 14, 2011 at 7:03 PM, Vladimir Mikhelson v...@mikhelson.com wrote:
Elliot,
You need to execute sendDTMF(1)
Articles are available with detailed setup description.
exactly my other concern - can just drop sip card in and put on the net - would
also have to get an sbc - which would be more than an ATA.
considering just using a cisco router (low end XM) and throwing a high density
voice card in it
Sent from my iPhone
On Jun 14, 2011, at 6:48 PM, John
On 6/14/11 4:37 PM, Russ Meyerriecks rmeyerrie...@digium.com wrote:
On 6/14/11 4:25 PM, Russ Meyerriecks wrote:
On 6/14/11 9:26 AM, Cassius Smith wrote:
Hello all,
I'm having a problem with my intercom function that I use for
under-chin
paging. I'm running 1.6.2.13 on this server, and we use
Dear;
Thanks a lot for guiding me.
Is it possible that the installation libpri-1.4.11.5 newer than the
libpri-1.4.11.5-patch?
Well, when I typed (note: I am trying to apply the libpri-1.4.11.5-patch for
the libpri-1.4.11.5):
libpri-1.4.11.5# patch -p0 -i libpri-1.4.11.5-patch
It gave me
Its probably not a bug so don't apply this patch. No D-Channel means
it cant sync up. It could be related to anything but the least likely
is that its a bug in libpri or dahdi.
Just go thru your configs, check and double check the cabling.
On Tue, Jun 14, 2011 at 7:27 PM, bilal ghayyad
Is it possible that the installation libpri-1.4.11.5 newer than the
libpri-1.4.11.5-patch?
Well, when I typed (note: I am trying to apply the
libpri-1.4.11.5-patch for the libpri-1.4.11.5):
libpri-1.4.11.5# patch -p0 -i libpri-1.4.11.5-patch
It gave me that patched detected as shown
Hi
Asterisk support dialout conference?.My requirement is when type a
CLI command with argument as a number ,asterisk should able to make a
call to that number and when connected ,that channel should entered in
to the conference room,like this I should able to add multiple users
into the
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