Hi,
I am trying to setup a GSM-Card for Asterisk. I currently use a vgsm I
from voismart (http://www.voismart.it/) but the driver is very bad
(compile-problems and no echo cancellation).
Is there anybody out there who can recommend me another piece of
hardware (pci card)? I need 1 or better
Hi
After recently upgrading to 1.8.3 I have noticed that the nat setting
for my peer in my sip table is not making it into the realtime cache.
For example
* Name : 501
Realtime peer: Yes, cached
Secret : Set
MD5Secret: Not set
Remote Secret: Not set
Context :
I totally agreed with Leif Madsen that viable options are available and time
and effort spent on winmodem should be carefully considered.
My system also works with an ATA as PSTN gateway and VOIP SIP provider for
DID and inbound/outbound service. It will save time much more time and
effort while
Hi
Scratch that
The value name has changed from Nat to Force Rport
Back to the drawing board
On Wed, 2011-03-02 at 08:57 +, Ishfaq Malik wrote:
Hi
After recently upgrading to 1.8.3 I have noticed that the nat setting
for my peer in my sip table is not making it into the realtime
Hi
With an FXO module + Zaptel, I'd like to know if there are ways to
know when the remote party has answered the phone, whether calling
through a callfile or by sending DTMF's.
I read about {CHANNEL(state), ChanIsAvail(), and ${DIALSTATUS}: Are
those reliable ways to know when the
Hi,
We are new to IP phone firmware upgradation (Sorry if it is a re-post of
previous question(s)).
Recently we have bought a cisco 7942G IP phone.
It currently has SIP 42.9-0-2SR1S firmware loaded on it.
We do not see any option to configure a SIP Proxy where we can provide SIP
Server
Your error is in front of you.
format_wav.c:148 check_header: Not in mono 2
[Feb 28 22:27:07] WARNING[2736]: file.c:386 fn_wrapper: Unable to open
format wav
Your wav file is not in proper format.
Must be mono, and at 8khz, 16bit
You can resample by using this command:
sox
On Wed, Mar 02, 2011 at 10:54:14AM +0100, Gilles wrote:
Hi
With an FXO module + Zaptel, I'd like to know if there are ways to
know when the remote party has answered the phone,
Any chance thi information is available through polarity reversal?
In thise case:
answeronpolarityswitch
Try using openvox gsm cards.
http://www.openvox.cn/store/g400p-p-63.html?cPath=25zenid=1fb3262c83d14d02b40fb6f577c7ebb7
its
cheaper as well...
On Wed, Mar 2, 2011 at 2:13 PM, Thorsten Göllner t...@ovm-group.com wrote:
Hi,
I am trying to setup a GSM-Card for Asterisk. I currently use a vgsm I
Hello folks,
for a customer of us we are trying to make asterisk and windows RTC
library work together, but without success.
We *need* to use gsm codec, so in the peer section we have
disallow=all
allow=gsm
the sip signaling is ok, and when sniffing we got this session description:
INVITE from
Hi,
We are planning to set up a prototype IVR system in Urdu language using
Asterisk. For speech recognition, we will be using our own engine built
using Sphinx, and for text to speech synthesis (for run time generation of
responses based on user queries), we have a system for Urdu built in C++
It seems like it is a v1.8 only function at present (unless a backport is
released).
From http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
-
Asterisk 1.8 will allow to read SIP response codes in the dialplan via
${HASH(SIP_CAUSE,channel-name)}
Asterisk 1.8 also comes with
It's all I use now.
I was luckily enough to be involved with quite a bit of the beta testing
in the UK - and, although there are a couple of 'nice-to-haves' missing,
they are excellent handsets. Polycom sound quality at Grandstream
prices ;)
I particularly like the 'use your own screen logo'
Hello
I haven't found an example on how to compare the value of a string
variable with spaces in it, and the While loop below never exits:
== extensions.conf
exten = start,n,Set(MYVAR=Dummy value)
exten = start,n,NoOp(${MYVAR})
;BAD TOO
;exten = start,n,While(!$[${MYVAR} : Some
Changing
exten = start,n,While($[${MYVAR} != Some string])
to
exten = start,n,While($[${MYVAR} != Some string])
does the trick for me.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
On Wed, Mar 02, 2011 at 10:05:35AM +1000, Stuart Longland wrote:
There's also regulatory requirements: here in Australia since I'm
plugging into the PSTN, it needs to carry the ACMA's regulatory
compliance mark. So buying something from overseas isn't an option.
It's less of an option for me
On Wed, 2 Mar 2011 13:37:57 -, Andrew Thomas
a...@datavox.co.uk wrote:
exten = start,n,While($[${MYVAR} != Some string])
Thanks Andrew.
--
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New to
Hi,
On Wed, Mar 02, 2011 at 05:20:07PM +0800, asterisk asterisk wrote:
I totally agreed with Leif Madsen that viable options are available and time
and effort spent on winmodem should be carefully considered.
Indeed, but I never suggested anywhere using a winmodem. The modem I
mentioned is a
On Wed, 2 Mar 2011 14:06:12 +0200, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
Any chance thi information is available through polarity reversal?
In thise case:
answeronpolarityswitch = yes
hanguponpolarityswitch = yes
Thanks for the tip. Google returned a bunch of discussions about
I have a FreePBX system with PRI trunks that's doing a number of things very
nicely, but frustrating me in one area.
I am using a Grandstream GXW-4008 in an off-premises location to provide
POTS service on four ports (this device worked fine in an early
application using a hardware VPN to the
I have the following situation
I'm using Action Originate to originate a call for a costumer. Originate goes
to a context that call the dial application. Before the application (Dial using
the G option) to be invoked i'm setting the variable cellphone like this:
[firstcontext]
exten =
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luiz Gustavo
Chiaretto
Sent: Wednesday, March 02, 2011 8:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Doubt about cdr on asterisk
I have
I'm running asterisk on a Freebsd with 2 Nic's.
Inside NIC is 192.168.5.x where the phones are.
Outside NIC used to be a public IP with the ISP's device set to
bridging, but the new WiMAX router only offers me the public ip
94.18.x.x on the outside,
and forwarding everything to 192.168.1.50
On Wed, 02 Mar 2011 15:03:46 +0100, Gilles codecompl...@free.fr
wrote:
Does someone know if it's OK to use all those for a TDM with an FXO
module, or should only some be used together?
In addition, based on people's experience, is CHANNEL() reliable to
detect call progress?
;Down, Rsrvd,
On 3/2/2011 9:46 AM, Leif Neland wrote:
Some of the phones are being disconnected with Asterisk saying no reply
to critical packet
What kind of phones are they? I might have nothing to do with your
network configuration; try adding to sip.conf [general]:
session-timers=refuse
--
Jeremy
Thanks for your answer Danny,
I thought there was another solution using some cdr options.
Best Regards.
Luiz Gustavo Chiaretto
- Original Message -
From: Danny Nicholas da...@debsinc.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
ok thanks for your response
i have created an agent in sip
sip.conf
[222]
type=friend
context=agents
host=dynamic
dtmfmode=auto
disallow=all
allow=alaw
allow=ulaw
qualify=yes
context=test
i have add in extensions.conf the fil below but when i check in
*var/spool/asterisk/monitor
there is no
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Wednesday, March 02, 2011 11:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] records inbound and outbound
thank you i have one question waht is 3009 is the called
Regards
2011/3/2 Danny Nicholas da...@debsinc.com
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
elharit
*Sent:*
I made a sub-context 3009 in default to let me call from my phone sipphone
to my phone 144 and record the conversation.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Wednesday, March 02, 2011
Un-top-posting...
2011/3/2 Danny Nicholas da...@debsinc.com
How I did it
exten = 3009,1,Answer()
exten = 3009,2,MixMonitor(test.wav|av(0)V(0))
exten = 3009,3,Dial(SIP/144)
exten = 3009,4,Hangup()
From: asterisk-users-boun...@lists.digium.com
On Wednesday 02 March 2011 07:06:31 Andrew Thomas wrote:
It seems like it is a v1.8 only function at present (unless a backport
is released).
From http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
-
Asterisk 1.8 will allow to read SIP response codes in the dialplan via
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Wednesday, March 02, 2011 12:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] records inbound
When I switched to 1.8 from 1.4 I am getting this error
pbx.c:4055 pbx_extension_helper: No application 'Wait,1' for extension
(default, s, 1)
http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands
This page says its in 1.0 and I dont think has been removed.
Did
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Wednesday, March 02, 2011 3:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Question on Asterisk 1.8
On Wed, 2011-03-02 at 16:33 -0500, Jerry Geis wrote:
When I switched to 1.8 from 1.4 I am getting this error
pbx.c:4055 pbx_extension_helper: No application 'Wait,1' for extension
(default, s, 1)
http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands
This
Best guess is that syntax changed from 1.4 to 1.8. Change line to
Exten = s,1,Wait(1)
Danny
Your correct. it was a syntax change. the above works.
jerry
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-- Bandwidth and Colocation Provided by
Hello,
I am looking at implementing Asterisk for a project I'm working on.
I need to authenticate a user against a database, and implement CC
processing; shouldn't be a problem with PHP-AGI.
In addition, I need to do several other things.
After a user has been authenticated, they will be
Jerry Geis wrote:
Your correct. it was a syntax change. the above works.
I've always used Wait(#) in my 1.4.x dial plans.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kyle Baczynski
Sent: Wednesday, March 02, 2011 3:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Functionality
Hi Danny,
Thank you. I will look through the archives, then.
If anybody can provide any specific threads or key phrases I might
search for (sometimes there are buzzwords for these things, you know), I
would much appreciate it. I'm new to Asterisk.
Thank you
---
Kyle Baczynski
On Wed, 2
I'm using 1.8.3, and have 2 sip providers. Both are set with
qualify=yes. Each of them sometimes have qualify times 10+ times the
other. For instance, one will be at 10-15ms, the other at 200ms.
Is there a way I can route an outgoing call to the provider with the
lower qualify time?
sean
On 3/03/11 11:29 AM, sean darcy wrote:
I'm using 1.8.3, and have 2 sip providers. Both are set with
qualify=yes. Each of them sometimes have qualify times 10+ times the
other. For instance, one will be at 10-15ms, the other at 200ms.
Is there a way I can route an outgoing call to the provider
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: Wednesday, March 02, 2011 4:29 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] how to use qualify times to route calls
I'm using
On 3/03/11 11:34 AM, Danny Nicholas wrote:
getprov.agi does sip show peers and gets the qualify time from status.
The low value is returned in the variable BESTPROV.
If you're going to do that, you could probably knock something up with
the SIPPEER function - SIPPEER(status).
--
Cheers,
On 03/02/2011 05:34 PM, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: Wednesday, March 02, 2011 4:29 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
On Wed, 2 Mar 2011, sean darcy wrote:
That would be a great idea, but would stretch my limits.
Isn't that what makes it fun?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867
What value do you get from the hangup cause, are they different?
I think can you use a gotoif checking the hangup cause.
On Wed, Mar 2, 2011 at 12:43 PM, Tilghman Lesher tilgh...@meg.abyt.eswrote:
On Wednesday 02 March 2011 07:06:31 Andrew Thomas wrote:
It seems like it is a v1.8 only
Call them.
On Wed, Mar 2, 2011 at 8:17 PM, Robert Augustyn
robert.augus...@linqone.com wrote:
Hi,
Is there a way of finding out what block of phone numbers were issued to
Roger’s business customers in my end of the woods?
Thanks,
Sincerely,
Robert Augustyn
--
Hm,
I did not think about that I just assumed that they would not give it as
this are the contact number of their clients ...
I believe that I have seen it somewhere on the web cannot find it though.
Sincerely,
Robert Augustyn
p:519.997.3106 ext:802
m:519.817.2503
Hi,
I am running a service where I play full songs but MP3 files kept on
crashing my server. I resorted to wav but the quality is really poor
after converting..or even sometimes not audible at all! Do you guys
know of a better way I can convert mp3 to wav and restore quality?
Below is the script
Is there any way to make a Cisco 7936 conference phone work in version 1.8?
--
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New to Asterisk? Join us for a live introductory webinar every
On Wednesday 02 March 2011 19:17:03 Robert Augustyn wrote:
Is there a way of finding out what block of phone numbers were issued to
Roger’s business customers in my end of the woods?
You can find out from NANPA, the registry which assigns blocks of phone
numbers. Note that due to phone
Hi!
My customer want's to allow calls to landlines in EU and US and disallow
calls to cells in EU. Rest of countries are blocked.
Country blocking is easy... Is there a service that allows checking
phone number? Maybe some specific Enum? I ask for number and server
responds with info, for
I don't remember exact name but there are two authorities which provide
real-time portability information online but you need subscription.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher
www.numberingplans.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Piotr Górski
Sent: Thursday, March 03, 2011 12:02 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Testing from where
Try to convert into gsm instead wav.
sox test.wav -r 8000 -c1 test.gsm
Am 03.03.2011 06:20, schrieb Timothy Smith:
Hi,
I am running a service where I play full songs but MP3 files kept on
crashing my server. I resorted to wav but the quality is really poor
after converting..or even sometimes
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