Tzafrir,
Le dimanche 30 octobre 2011 à 10:30 +0200, Tzafrir Cohen a écrit :
On Sat, Oct 29, 2011 at 08:14:55PM +0200, Eric van der Vlist wrote:
Hi,
Xorcom astribanks get initialized straight on when using Ubuntu 11.10
packages but I am having a hard time to get the same result running
have
installed Boris Derzhavets' packages:
https://launchpad.net/~bderzhavets/+archive/seabios163 and updated my
host definition to emulate USB 2.0 but I still have the same issue.
Have I missed something?
Thanks,
Eric
You cannot echo cancel SIP. Removing echo must be done before PSTN is
converted into SIP. i.e. your PSTN/SIP gateway.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Diego Alejandro
Sanchez Quiroga
Sent:
CallerID is your country code + city/area code + telephone number. Do not set
the leading 0, that is not part of the Caller*ID.
Example London UK number, country code 44, area code 20, number 1234-5678:
Set(CALLERID(num)=442012345678
-Original Message-
From:
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DID and how the caller id will appear
Dear,
Callerid you need to add parameter in chan_dahdi.conf file. so what is you
chan_dahdi.conf file ?
Best Regards,
Mahesh
On Wed, Oct 19, 2011 at 6:46 PM, Eric Wieling ewiel
Upgrade to 1.8.7.1 There was a bug fixed recently (I think in 1.8.6, but might
have been 1.8.7) which caused Asterisk to sometimes not transcode when it
should.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
I am assuming you are using a provisioning server.
If the phone is running firmware 3.2 or earlier you can access the phone web
interface and confirm the dialplan active on the phone is the same as what you
set in the config file on the server.
-Original Message-
From:
When you set bindaddr=0.0.0.0 Asterisk will not bind to any specific IP and the
OS will choose the source IP of the packet.Let me repeat this: THE OS PICKS
THE SOURCE IP.
If your OS routing tables are correct, then the packets will be sourced from
the correct IP.
-Original
Permit deny in your example applies only to incoming calls to Asterisk from the
device which authenticates as context1. A very illogical name for a SIP
peer/user/friend, but I've seen stranger things.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
/chan_dahdi_additional.conf': == Found
Thanks,
Michael.k
On Thu, Oct 6, 2011 at 9:44 PM, Eric Wieling ewiel...@nyigc.com wrote:
What happens when you do the module load chan_dahdi.so command?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Looks like you do not have chan_dahdi.so loaded in Asterisk.If you don't
install DAHDI before you install Asterisk, then Asterisk will not be built with
support for DAHDI.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
:17 PM, Eric Wieling ewiel...@nyigc.com wrote:
In the Asterisk CLI run the commands module unload chan_dahdi.so and
module load chan_dahdi.so.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun
In the USA ordering BRI service is discouraged by the telcos and is very
uncommon. In Verizon NE CLECs are not even permitted to order BRIs.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tamer Higazi
Sent:
I always use the recalc option to show translations, it seems to provide much
more accurate numbers.
Example: core show translation recalc 20
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tony Mountifield
Try module load chan_zap.so in the CLI. You should see whatever errors are
generated.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of NaJIm
Sent: Thursday, September 29, 2011 5:52 PM
To: Asterisk Users
Currently if asterisk loses its connection to the postgresql it does not
attempt to reconnect. I have searched all over for a setting that would have
asterisk attempt to reconnect but I can not find anything. Is there something I
am missing?
Thanks!
-Eric
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Catalin S.
Sent: Sunday, September 18, 2011 4:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] single registration per user
Hello Eric,
Is about outgoing calls from multiple devices
Asterisk only allows one device per peer to register. If a 2nd device
registers, the first registration is overwritten.
You can use permit/deny to limit which IPs a device can register from.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
It does on PRI.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Friday, September 16, 2011 7:32 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Monitoring second leg being
dialplan,
which is working, as well as MoH dialplan to see if this could be the cause
of crashing.
How do I test this?
Is it a call file that can handle this without ringing my extension first,
like internal system calling?
Thanks,
--Eric
If I read Kevin's post correctly, his statement applies to ALL echo cancellers,
not just software EC.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gustavo Santos
Sent: Wednesday, September 14, 2011 10:52
sox -h will list the formats supported by your install of sox. If mp3 is not
listed, then your sox does not support mp3. This is not uncommon. Many Linux
distros do not ship support for patent encumbered formats. Either stop using
mp3 (this is what I suggest) or compile and install sox
Assuming SIP sip show channels will show you which codec is used for each
call leg. However it does not track transcoding.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RSCL Mumbai
Sent: Wednesday, August
It is possible to use Asterisk as a dialup PPP server, but only if you are
doing PRI between the telco and Asterisk (see core show application DAHDIRAS).
You could bring analog POTS lines into a dialup server (Portmaster maybe?) if
PRI is too expensive. Can outsource your dialup customers to
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet
Sent: Friday, August 26, 2011 6:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Looking for ideas
Take a look at the A(x) and m options to dial. In the Asterisk CLI core show
application dial for a the docs to Dial().
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Boykin
Sent: Thursday, August 18,
# termtype: unknown
bchan=1-23
dchan=24
echocanceller=mg2,1-23
/etc/asterisk/chan_dahdi
[channels]
language=en
context=Incoming-Pri
switchtype=dms100
signalling=pri_cpe
group=1
channel = 1-23
Thanks,
Eric Merkel
--
_
-- Bandwidth
On Tue, Aug 16, 2011 at 10:48 AM, Shaun Ruffell sruff...@digium.com wrote:
On Tue, Aug 16, 2011 at 10:31:54AM -0400, Eric Merkel wrote:
I am having a problem with a new PRI turn-up on dahdi 2.5.0 and
asterisk 1.8.5 that I have not seen before. The PRI is setup as B8ZS,
ESF and the span shows
I am having a problem with ${DIALSTATUS} and )=CDR(disposition) disagreeing.
Below is a dialplan snippet and the resulting CLI output. This is running in
an 'h' extension.
Noop(DIALSTATUS=${DIALSTATUS})
Noop(CDR(disposition)=${CDR(disposition)})
-- Executing
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Lee Howard
Sent: Tuesday, August 09, 2011 7:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FAX Issues
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Jim Boykin
Sent: Tuesday, August 09, 2011 12:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DAHDI
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of CDR
Sent: Monday, August 08, 2011 9:42 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Version 1.8 strange expression error
This
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Landy Landy
Sent: Friday, August 05, 2011 12:42 PM
To: asterisk
Subject: [asterisk-users] Assistance sending mass sms to cellphones
Hello.
I would
Add qualify=yes to the peer (aka trunk) This is not about SIP response codes.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Shaun Wingrin
Sent: Thursday, August 04, 2011 4:21 AM
To:
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Zeeshan Ali Shah
Sent: Thursday, August 04, 2011 9:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Paul Belanger
Sent: Thursday, August 04, 2011 10:47 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Customizing sip response codes
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, August 04, 2011 3:00 PM
To: j...@sunfone.com; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re:
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Lyle Giese
Sent: Wednesday, August 03, 2011 8:16 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Need a volunteer for a Patch
On
If it doesn't go green when you put a hard loopback on the port, then contact
Digium support.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Dave George
Sent: Tuesday, August 02, 2011 10:52 PM
To:
Could it be this bug? https://issues.asterisk.org/jira/browse/ASTERISK-17742
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Sunday, July 31, 2011 7:48 AM
To:
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Friday, July 29, 2011 9:06 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc: jim.smith...@debsinc.com
Subject: Re:
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Ishwar Sridharan
Sent: Friday, July 29, 2011 9:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Capturing
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Nikhil
Sent: Thursday, July 28, 2011 9:03 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a
...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Ishwar Sridharan
Sent: Thursday, July 28, 2011 2:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Capturing call Reject/Decline events on a PRI
line
Hi Eric,
There weren't any
In order to get the proper encoding for Asterisk, you must provide the
correct values for each of these characteristics. In your case, they
are as follows:
rate = 8000
data size = 8-bit (byte)
data encoding = gsm
channels = 1 (mono)
Therefore, the command you would use to
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Mike
Sent: Thursday, July 28, 2011 3:47 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Disabling Polycom
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Bryant Zimmerman
Sent: Tuesday, July 26, 2011 3:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] file2ban
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of CDR
Sent: Saturday, July 23, 2011 1:39 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Securing Asterisk
I beg to differ. Digium is
Asterisk does not support changing codecs on the fly.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matteo Campana
Sent: Friday, July 22, 2011 10:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
] On Behalf Of Matteo Campana
Sent: Friday, July 22, 2011 11:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question about codec re-negotiation in asterisk
1.4.X
On Fri, Jul 22, 2011 at 4:46 PM, Eric Wieling
ewiel...@nyigc.commailto:ewiel...@nyigc.com
-negotiation in asterisk
1.4.X
Eric
With 1.8.x I use.
exten = Process,1,Set(SIP_CODEC=ulaw)
And the system kicks the call over to ulaw. Now this is just prior to the
answer so I don't know if it meets your criteria. But it works great to enforce
inline T.30 audio faxes. I also use the f/F option T.38
Sent from my Computer
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Steve Edwards
Sent: Wednesday, July 20, 2011 4:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Sent from my Computer
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Lee Archer
Sent: Monday, July 18, 2011 7:04 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Seg Faults with
- Non-Commercial Discussion
Subject: Re: [asterisk-users] Seg Faults with 1.6.2.19
Hi Eric, are you using ODBC?
Regards
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Eric Wieling
Sent: 18 July 2011
Sent from a computer
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Mark Rosedale
Sent: Wednesday, July 13, 2011 10:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Sent from my Toshiba Satellite A106 computer
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Steve Edwards
Sent: Wednesday, July 13, 2011 5:07 PM
To: Asterisk Users Mailing List - Non-Commercial
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Tuesday, July 12, 2011 3:42 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Monitoring connection to VoIP provider?
On
Sent from a computer
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Mark Rosedale
Sent: Tuesday, July 12, 2011 4:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
(agents,h,3)
-- Executing [h@agents:3] Hangup(SIP/223-6ec45a88, )
in new stack
== Spawn extension (agents, h, 3) exited non-zero on
'SIP/223-6ec45a88'
== End MixMonitor Recording SIP/223-6ec45a88 srvradio*CLI
2011/7/8 Eric Wieling ewiel...@nyigc.com
Show us the CLI output
Good point, folks, and thank you. I don't know yet whether they'll using
something that is really a DS1-over-IP but that's what it sounds like.
Since this would be a new Asterisk install rather than a legacy PBX, I
probably prefer to go with somebody who can do plain SIP/IAX trunking (and
not
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
salaheddine elharit
Sent: Friday, July 08, 2011 6:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] timeout with
Discussion
Subject: Re: [asterisk-users] timeout with outbound calls
i have tested this solution and i have the same issue
in my case want to call a phone number 06 from my
snom phone (sip223)
the issue still the same
any help please
2011/7/8 Eric Wieling ewiel...@nyigc.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
ADAK, INDRANIL (ATTSI)
Sent: Thursday, July 07, 2011 8:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] simple
A carrier I like will be introducing PRI over IP, presumably going thru some
sort of gateway box (I'm guessing by Adtran but no data yet). Has anybody
set up successfully to work directly with such a feed without bothering to
take it down to T1 and use a T1/PRI card?
Thanks
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Tim Nelson
Sent: Thursday, July 07, 2011 4:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Anybody doing
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Cassius Smith
Sent: Wednesday, July 06, 2011 4:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] single keypress
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Dan Journo
Sent: Saturday, July 02, 2011 8:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] SIP Peer Name
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Kaushal Shriyan
Sent: Friday, July 01, 2011 8:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Starting
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
michael k
Sent: Thursday, June 30, 2011 11:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cannot figure
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Ernie Dunbar
Sent: Wednesday, June 29, 2011 6:34 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] No audio format found to offer.
That's not the password.
I switched it to that in the config file for realism.
I always give some honey out to those who have a sugar tooth.
Any ideas on the fix?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
Emailed wrong address
-Original Message-
From: ERIC HERRON [mailto:e...@lanline.com]
Sent: Monday, June 27, 2011 10:58 AM
To: 'asterisk-users@lists.digium.com'
Subject: FW: [asterisk-users] Asterisk 1.8.4 - Google iCal not working
Forgot link..
That's not the password.
I switched
Please note that the include is part of the chan_dahdi.conf config
file.
Sawan
On Fri, Jun 24, 2011 at 12:15 AM, Eric Wieling ewiel...@nyigc.com
wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Sawan Vithlani
Sent: Thursday, June 23, 2011 10:12 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Problem with detecting fax on
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Kevin P. Fleming
Sent: Thursday, June 23, 2011 11:41 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Problem with detecting fax on
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Inbound SMS
On Wed, Jun 22, 2011 at 10:23 AM, Administrator TOOTAI ad...@tootai.net
wrote:
Le 22/06/2011 01:10, ERIC HERRON a écrit :
I know Asterisk 1.8 can send out texts via SMS()
Can I send Asterisk a text
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
ERIC HERRON
Sent: Wednesday, June 22, 2011 10:27 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Inbound
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Eric Wieling
Sent: Wednesday, June 22, 2011 10:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Inbound SMS
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Jerry Geis
Sent: Tuesday, June 21, 2011 10:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Question on pause
: Re: [asterisk-users] Question on pause in dialing
Try setting echotraining=no in chan_dahdi.conf.
IIRC that pause is for the echotraining to train the echo canceler.
Eric,
Is there another setting ? This had no effect.
This is happening while I am entering the number to dial, I
nexusone/nexusone (Unspecified)D 0Unmonitored
How could I prevent these calls from dropping?
---
Eric Smith
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
, Jun 20, 2011 at 6:33 PM, ERIC HERRON e...@lanline.com wrote:
I have an asterisk 1.4.26 mte running.
Sometimes inbound caller ID displays asterisk
These calls do not show up on the CLI nor the CDR.
I read somewhere that these are asterisk hack attempts.
Is this true?
What
I know Asterisk 1.8 can send out texts via SMS()
Can I send Asterisk a text via a DID and it do something?
So far I am reading that it cannot but I do not know if there have been
updates.
Thanks,
-E
--
_
-- Bandwidth
It is not included. It was supposed to be included in 1.6.3, but that verison
of Asterisk was never released, it became 1.8.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Arjan Kroon|Mobillion
Sent:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Ryan Wagoner
Sent: Monday, June 20, 2011 7:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Connected Line
If you can't ping between the two end points, then you can't do direct RTP.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Terry Brummell
Sent: Monday, June 20, 2011 8:16 AM
To: Asterisk Users Mailing
@lists.digium.com
Envoyé le : Lun 20 juin 2011, 16h 39min 32s Objet : Re:
[asterisk-users] Re : Re : Direct RTP with Asterisk
On 20/06/11 13:18, Eric Wieling wrote:
If you can't ping between the two end points, then you
can't do direct RTP.
precisely. If 10.10.9.1 isn't reachable from
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Jonas Kellens
Sent: Monday, June 20, 2011 3:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Get second cipher
I have an asterisk 1.4.26 mte running.
Sometimes inbound caller ID displays asterisk
These calls do not show up on the CLI nor the CDR.
I read somewhere that these are asterisk hack attempts.
Is this true?
What is the best way to defend from this?
I know a secure password
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Carlos Chavez
Sent: Friday, June 17, 2011 2:53 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk voicemail distribution groups
List - Non-Commercial Discussion
Subject: Re: [asterisk-users] No audio after a reinvite changing codec
Inviato da iPhone
Il giorno 16/giu/2011, alle ore 16:37, Eric Wieling
ewiel...@nyigc.com ha scritto:
We experience the same thing. The solution we use is to
not change codecs
We experience the same thing. The solution we use is to not change codecs in
the middle of a call. I assumed it was an issue with our upstream.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Larry
This was a bug in 1.4, 1.6.x, and 1.8. It is fixed in the latest release of
each of the Asterisk versions. Check the Changelog for 1.8.4, you might see
the bugtracker ID with the patch.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
The latest 1.8.x solved the problem for us on multiple servers.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Wednesday, June 15, 2011 9:29 AM
To: 'Asterisk Users Mailing List -
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:
See sip.conf.sample included in Asterisk 1.8.x. On my system the relevant
section starts on line 541. STATUS NOTIFICATIONS (SUBSCRIPTIONS)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Jerry Geis
In my experience this is usually caused by REINVITES. Disable reinvites (aka
directmedia in recent Asterisks) and see if that helps.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Florent THOMAS
Sent:
Either use ExtensionState or watch for Hold/Unhold events.
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+ExtensionState
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Jerry Geis
Sent:
Either use ExtensionState or watch for Hold/Unhold events.
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action
+ExtensionState
Eric
Thanks I have 2 questions:
1) trying to use the command and not getting anything of value:
Response: Success
ActionID: 1
Message: Extension
All major changes are listed in the UPGRADE.txt files included in the 1.8
tarball.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
satish patel
Sent: Wednesday, June 08, 2011 9:57 AM
To: asterisk-users
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