Just for fun I did something similar at one point.
0-9 A-D and * and # make a character set of 16 characters, perfect for encoding
as hex.
Take your string, get the ASCII value of each character, convert it to hex, and
add it to the "encoded" string.
Just before dialing, replace all "e" with #
I experienced a similar problem, asked here, and go NO response. I put CEL on
the shelf until an Asterisk 1.8 release has working CEL.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cegadsl
Sent: Friday, Feb
Polycom phones can be set to include their MAC in the User Agent string.
Useragent: PolycomSoundPointIP-SPIP_550-UA/3.3.4.0085_0004f2233929
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini
S
Yes, but it doesn't seem to indicate if the timeout is in seconds of
milliseconds.
pbx*CLI> agi show commands topic get data
-= Info about agi 'get data' =-
[Syntax]
get data [] []
[Description]
Stream the given , and recei
I get this on 1.8.x as well. I assume it is a harmless bug.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Nowrot
Sent: Wednesday, February 22, 2012 3:32 PM
To: Asterisk Users Mailing List - Non-Commer
Are you reading STDIN to initialize your AGI? If not Asterisk may ignore your
AGI commands. I recommend using one of the many AGI libs out there.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall
: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dahdi Answer a Call On ringing State.
Hi Eric,
but in this case dialing is not completed ring is still going on, so it should
not answered
thanks
Dhaval
On Thu, Feb 16, 2012 at 7:52 PM, Eric Wieling wrote
First. move to Europe. Asterisk's SMS support is for SMS over PSTN, which is
supported by carriers in Europe, but not in the USA. You would use DAHDI.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
FXO ports are considered Answered as soon as dialing completes. This is the
way analog FXO ports work. Use PRI or SIP if you need correct Answer
supervision.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
The Missed Calls list is broken on 4.0.x (see the release notes). The notes
say the issue only happens for a couple of Polycom models, but that is not my
experience.
We use 3.3.x
On 3.3.x I needed to add the following to make Intercom work with FreePBX.
-Original Message---
This is my first 1.8 install. I'm trying to set up CEL and am getting the
following errors. Does anyone have any ideas on where to look? res_odbc and
cdr_adaptive_odbc appear to be working, the CDRs are working.
[Feb 11 23:44:36] WARNING[23125]: cel_odbc.c:699 odbc_log: Column type -9
(fie
Only the higher end Polycoms support Siren7 and Siren14. I believe only the
VVX and SoundStation IP phone support those codes.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Klaverstyn
Sent: Thursda
In the Asterisk CLI "core show application ivr" The IVR app is poorly
documented, but I believe it may do what you need to do.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yaroslav Panych
Sent: Friday, Fe
We had similar problems, updating to the latest 1.8.x seems to have solved the
issue for at least one number we were having issues with.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz
Sent: Monday,
Can they answer the call by pressing the line key when simply picking up the
handset does not answer the call? If so, then the users are not properly
seating the handset in the cradle.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@list
See http://www.voip-info.org/wiki/view/Asterisk+func+CONNECTEDLINE pay special
attention to the sendrpid note.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gunnar Schaller
Sent: Monday, January 16, 2012 3:2
This symptom usually means you are doing an attended transfer instead of a
blind transfer.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roland
Sent: Monday, January 16, 2012 10:57 AM
To: Asterisk Users Mail
"w" is only allowed as part of the dialed TN on FXO and FXS ports.
Dial the TN normally, use the D() option to Dial to send post answer digits.
i.e. Dial(DAHDI/g0/12345,240,D(w))
See "core show application dial"
-Original Message-
From: asterisk-users-boun...@lists.digium.co
...@lists.digium.com] On Behalf Of Joseph
Sent: Saturday, January 07, 2012 1:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [SOLVED] Asterisk 10.0 & 1.4 - iax codec are not
compatible
On 01/07/12 13:27, Eric Wieling wrote:
>This means you are allowin
This means you are allowing guest calls. A VERY bad thing.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Saturday, January 07, 2012 1:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discu
Chances are the incoming call is not matching anything in iax.conf. turn on
iax debug, try a call, post the results. Maybe someone familiar with IAX can
help you.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behal
Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk 1.8.8 - caller ID not working.
On 01/06/12 18:15, Eric Wieling wrote:
>Putting in a Wait(n) is only (sometimes) needed to wait for the CallerID NAME
>on PRI or BRI.
Putting wait(5) in my dial plan doesn't
Putting in a Wait(n) is only (sometimes) needed to wait for the CallerID NAME
on PRI or BRI.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Friday, January 06, 2012 6:06 PM
To: Asterisk U
iday, January 06, 2012 10:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk 1.8.8 - caller ID not working.
On Fri, Jan 6, 2012 at 8:16 AM, Eric Wieling wrote:
All "screwing up with Asterisk" is supposed to be documented
All "screwing up with Asterisk" is supposed to be documented in the relevant
UPGRADE*.txt files. Have you checked them?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Thursday, January 05, 2012
both audio and video its
sent to the other client as video call .I tried settings the
SIP_CODED_INBOUND and outbound also ... but no luck
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Providing which version of Asterisk you are using might be helpful.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:09 PM
To: Asterisk Users Mailing List - Non-
You should confirm with "ps -aux | grep asterisk"
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Tuesday, January 03, 2012 1:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
S
The UPGRADE*.txt files included with the Asterisk tarballs give a nice summery
of the major changes between each Asterisk verison.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker
Sent: Wednesday,
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, December 28, 2011 3:19 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk 1.8.
The issue is not fixed in 1.8.8.0 either.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B
Sent: Wednesday, December 28, 2011 3:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
I suspect nobody responded because this topic has been discussed over and over
again. Search the mailing list archives.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator
TOOTAI
Sent: Tuesday, Dec
-users] Codec warnings after upgrade to 1.8
On Fri, Dec 23, 2011 at 10:40 AM, Eric Wieling wrote:
I'm getting various codec related warnings after upgrading to 1.8. Did
I miss something in the UPGRADE file? Does Asterisk no longer transcode 8-)?
WARNING[
I'm getting various codec related warnings after upgrading to 1.8. Did I miss
something in the UPGRADE file? Does Asterisk no longer transcode 8-)?
WARNING[11123]: channel.c:4909 ast_write: Codec mismatch on channel
DAHDI/i1/12124221200-74 setting write format to g722 from ulaw native formats
" -- Got SIP response 480 "Temporarily Unavailable" back from 10.10.11.203"
this is why you are getting congestion instead of NOANSWER. Fix that and add
a timeout to your dial and it should work.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users
Adtran PoE switches.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Wednesday, December 21, 2011 2:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] OT -
en't moved from 3.2.x firmware yet.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Friday, December 16, 2011 4:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Unless you live in the Netherlands, your CallerID does not start with 31.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kaushal Shriyan
Sent: Friday, December 16, 2011 7:38 PM
To: Asterisk Users Mailing Lis
Did you run your old configurations thru the Polycom script to convert them to
work with 3.3+?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marco Mooijekind
Sent: Friday, December 16, 2011 4:41 PM
To: Aster
Confirm your web server user is running as the same user as asterisk.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Germann
Sent: Friday, December 16, 2011 3:06 PM
To: asterisk-users@lists.digium.com
Sub
>From cdr.conf.sample:
; Normally, CDR's are not closed out until after all extensions are finished
; executing. By enabling this option, the CDR will be ended before executing
; the "h" extension so that CDR values such as "end" and "billsec" may be
; retrieved inside of of this extension.
;endb
No.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James
Courtier-Dutton
Sent: Friday, December 16, 2011 5:30 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] SIP Trunk
Hi,
I have a situatio
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Latham
Sent: Thursday, December 15, 2011 1:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Best PBX for Call
Asterisk uses libcap to do "root-like" things when running as non-root.
Setting the DSCP/QoS value of packets requires root access, but Asterisk seems
to manage just fine using libcap (not libpcap, that is different).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[ma
Wav49 is GSM wrapped in a MS header.
You should be able reverse the order of the two items without harm.
If you remove formats, then Asterisk won't find the existing messages or
greetings in the format you removed.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailt
If you want to use your own DSP transcoder, try asking on asterisk-dev. If you
simply want to use a hardware based transcoder Digium and Sangoma have cards.
Sangoma: http://sangoma.com/products/hardware_products/transcoding.html
Digium: http://www.digium.com/en/products/hardware/voice
-Orig
The SIP server has no way to tell the device is no longer available until the
next time the device registers (or the server tries to send a call to the
device).
ASTERISK has the qualify feature, which uses a SIP OPTIONS packet to probe the
peer every min or so.
-Original Message-
Fro
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: Friday, November 11, 2011 8:45 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] 10.0.0-rc1: dahdi doesn't see card
Thanks.
Show us /etc/asterisk/chan_dahdi.conf (and any #include'd files)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: Friday, November 11, 2011 5:51 PM
To: asterisk-users@lists.digium.com
Subject: [
users] DID Provider Issues
I've seen this service a number of times, however they all lose DNIS when they
do this. Do you provide RDNIS?
On Fri, Nov 11, 2011 at 10:47 AM, Eric Wieling wrote:
I work for a CLEC which as VoIP services.
We allow customers to spec
I work for a CLEC which as VoIP services.
We allow customers to specify a telephone number to send calls to if, for some
reason, the call cannot be sent to the customer. Usually this is for when the
customer's circuit is down.
-Original Message-
From: asterisk-users-boun...@lists.dig
The Asterisk source tree has a document with instructions on getting a
backtrace from the segfaults so you can report it on the issue tracker.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Thursd
Asterisk does not know if the user is dialing "2" or dialing "2666", so it must
wait for a timeout. Rewrite your dialplan so there are no ambiguous
extensions in the context and it will work as expected.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk
being said, I was
under the impression that only the local Telcos have control over the phone
numbers.I take it that this is not correct?
Cheers,
Berry.
On Fri, Nov 4, 2011 at 10:35 AM, Eric Wieling wrote:
> Why not go direct to Verizon Business (they provide nationwide wholesale SIP
> se
Why not go direct to Verizon Business (they provide nationwide wholesale SIP
services) or Level3 for your SIP interconnect? Leave the local telco out of it.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nic
In your example the CallerID number will always be "start". Not what he is
looking for.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, November 03, 2011 9:38 AM
To: 'Asteris
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: Th
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 Behalf
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 wrote
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: asterisk-users-boun...@lis
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: asterisk-users-
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 Message-
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
[mailto:asterisk-
hdi-channels.conf': == Found
== Parsing '/etc/asterisk/chan_dahdi_additional.conf': == Found
Thanks,
Michael.k
On Thu, Oct 6, 2011 at 9:44 PM, Eric Wieling wrote:
What happens when you do the module load chan_dahdi.so command?
-Original Message-
6, 2011 at 9:17 PM, Eric Wieling 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-bo
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
[mailto:asterisk-users-boun...@lists.digium.co
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...@lists.digium.com] On Behalf Of michael k
Sent: Thursday, October 06, 2011 11:40 AM
To: Ast
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 Mail
clues?
Thank you for answers,
Best regards.
On Sun, Sep 18, 2011 at 8:37 PM, Eric Wieling wrote:
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
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
[mailto:asterisk-users-boun.
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 dial
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 AM
"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 wi
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 a
>-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 ide
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,
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 [h@pbxmax-dia
> -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 Ca
> -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
> -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: [aste
> -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 cod
> -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] In
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: asterisk-user
> -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
>
>
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: asterisk-users@lists.
> -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] Captur
> -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
> Subjec
> -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
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 creat
te that as soon as the call lands on asterisk, we pass the control
> over to adhearsion. Does that affect how events are handled in asterisk?
>
> --
> Thanks,
> Ishwar.
>
>
>
> On Thu, Jul 28, 2011 at 6:37 PM, Eric Wieling wrote:
>
>
>
>
>
> -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 o
> -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] file2
> -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. Digiu
use the f/F option T.38 or T.30 on recevie fax.
This option was added as part of a patch in 1.8 and is in the 1.10/2.0 branch.
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
________
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