Hello.
I would like to know if is possible to send mass sms with an php agi script
through asterisk?
For example: I have about 50 cellphone numbers I would like to text whenever
theres a meeting, I should load the numbers from a database and send a message
via web with php and have asterisk
Hello.
I would like to know if is possible to send mass sms with an php agi script
through asterisk?
For example: I have about 50 cellphone numbers I would like to text whenever
theres a meeting, I should load the numbers from a database and send a message
via web with php and have asterisk
, didn't think this wasnt an asterisk related question.
-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 11:42 AM
To: asterisk
Subject: [asterisk-users] Assistance
Hello.
I'm looking to buy a FXO card to do some testing with two phone lines I have at
home and was looking in ebay some and found some cheap ones but, the I've never
heard of the brand or manufacturer: chinaroby. They run for about $99 plus
shipping. Have any one used these? or please
Hello.
I'm trying to set TTS with Cepstral and Swift but can't get it to work. I get
this error when testing it:
-- SIP/101- Playing 'welcome.gsm' (language 'es')
-- Executing [...@local-calls:3] Swift(SIP/101-, Hello this is
ceptral) in new stack
[Jul 28 18:29:16]
Do you have cepstral installed and have the voice(s)
registered ?
try: swift --voices
asterisk:~# swift --voices
Swift command-line synthesis program
Version 5.1.0 of July 2008
Copyright (c) 2000-2006, Cepstral LLC.
Voice | Version | Lic? | Gender | Age | Language | Sample Rate
...@jeremykister.com
Subject: Re: [asterisk-users] app_swift.c:338 engine: Failed to set voice
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Wednesday, July 28, 2010, 9:08 PM
On 7/28/2010 8:33 PM, Landy Landy
wrote:
asterisk:/home/landysaccount# grep ^[a-z
Date: Thursday, June 17, 2010, 1:47 AM
On 6/17/10 12:49 AM, Steve Edwards
wrote:
On Wed, 16 Jun 2010, Landy Landy wrote:
I'm unable to place any calls through a2billing. I
followed instructions
here: http://trac.asterisk2billing.org/cgi-bin/trac.cgi/wiki/F.A.Q
to
DISABLE PIN number
Of Landy Landy
Sent: Tuesday, June 15, 2010 9:53 PM
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] a2billing for residential
voip usage
I copied the config to the a2billing.conf in /etc/asterisk
folder. I'm still not able to place any calls yet
, 2010, 1:05 AM
you see lot of documentation on wiki
Google them many success case you see
Ram
On Tue, Jun 15, 2010 at 7:01 AM, Landy Landy landysacco...@yahoo.com wrote:
Hello List.
I just installed a2billing with asterisk 1.6 and got it working. The only
problem is that I'm trying to setup
I copied the config to the a2billing.conf in /etc/asterisk folder. I'm still
not able to place any calls yet. Looks like I have to read more on how to
configure trunks and providers whick got me confused. I'll learn though.
--- On Tue, 6/15/10, Vardan Harutyunyan hvarda...@gmail.com wrote:
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of Landy Landy
Sent: Tuesday, June 15, 2010 9:53 PM
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] a2billing for residential
voip usage
I copied the config
Hello List.
I just installed a2billing with asterisk 1.6 and got it working. The only
problem is that I'm trying to setup something to manage who's using the most
minutes in the house. I noticed a2billing only works for callin cards setups,
or maybe I didn't configure it correctly for what I
Hello List.
I am having problems retreiving voicemails on my system. I noticed when someone
leaves a message through the pstn line I can't hear anything. I tested leaving
a message from one of the extensions and that can be heard. I don't know if is
the type of card I'm using for analog (
I have this:
[menu]
exten = _X.,1,answer()
exten = _X.,2,wait(1)
exten = _X.,n,GoTo(ivr,s,1)
[default]
include = record
include = incoming
include = menu
[local-dial]
exten = _1XX,1,Verbose(. In local-dial context, dialing exten: ${EXTEN}
.
exten =
Hello List.
I am puzzled and how asterisk listens to calls or connections from clients.
When I do a netstat -nat I don't see asterisk listening on port 5060. Now, I'm
testing a server with three network interfaces: two to the internet doing
load balancing and the other to our LAN. I would
See http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf
Search for bindaddr.
Or udpbindaddr for 1.6.2+...also,
tcpbindaddr, tlsbindaddr if you plan
on adding TCP/TLS SIP support to asterisk.
Thanks to everyone who replied for clarifying.
--
Hello All.
I would like to know what codec is being used during a call. For example if I
have 3 channels on 3 active calls how can I find what codec is beeing used by
each client?
Thanks in advanced.
--
_
--
Hello. Happy New Year to everyone.
I have a small WISP and would like to have customers to call our number to
check their balance. I am planning on writing an AGI with php so it can get the
customer info from the customer database. I don't know how to interact with the
caller while in the agi
--- On Sat, 1/2/10, Landy Landy landysacco...@yahoo.com wrote:
From: Landy Landy landysacco...@yahoo.com
Subject: [asterisk-users] Help getting info from caller
To: asterisk-users@lists.digium.com
Date: Saturday, January 2, 2010, 9:01 AM
Hello. Happy New Year to everyone.
I have a small
I was able to test the script, here is what I have:
[CODE]
#!/usr/bin/php -q
?php
//ini_set(include_path,
.:../:./includes:../include:/var/lib/asterisk/agi-bin/includes);
//include( ./includes/optimum_config.php );
$CONF['host'] = 'server';
$CONF['user'] = '';
--- On Wed, 12/16/09, Landy Landy landysacco...@yahoo.com wrote:
From: Landy Landy landysacco...@yahoo.com
Subject: Re: [asterisk-users] Best way ro run 2 or more asterisk servers?
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Wednesday
peering useful: http://astrecipes.net/index.php?n=204Thanksl.
I followed exactly what' on that tutorial and can't get it to work. Now,
I tried:
example
Server1
[server2]
type=peer
context=from_client
host=server2-ip
Server2
[server1]
type=peer
context=from_client
host=server1-ip
I'm trying to get two server communicate with each other and call from one to
the other but, I'm having a lot of problems.
I tried to create a iax trunk between the two:
At the server:
[client]
type=friend
username=asterisk2
authuser=asterisk2
fromuser=asterisk2
secret=sss
auth=md5
Date: Wednesday, December 16, 2009, 1:26 AM
trust both the side giving IP address
in the sip.conf
I did this in the iax.conf file
[client]
type=friend
username=asterisk2
authuser=asterisk2
fromuser=asterisk2
secret=sss
auth=md5
context=from_client
host=172.16.0.11
trunk=yes
qualify=yes
Hello List.
I have a question regarding connecting two asterisk servers. I'm trying to
learn how asterisk comunicates from server to server. I already have a server
running smoothly now, I'm installing another one to test it along side the
actual one.
I would like to run different scenarios:
, 2009, at 9:16 PM, Landy Landy landysacco...@yahoo.com
wrote:
Same thing:
== Using SIP RTP CoS mark 5
-- Executing [...@outbound:1]
Answer(SIP/102-096a48c8, ) in
new stack
-- Executing [...@outbound:2]
Verbose(SIP/102-096a48c8, In
timeofday ) in new stack
Hello List.
I would like to know how I can use two or more service providers with asterisk
to be used randomly for ei, if an user tries to make a call I would like to
randomly use a provider. It doesn't matter where the call is destined to.
Thanks.
Hi List.
Don't know if I already posted about this problem but, if I have I apologize
for the double post.
I am trying to test a time of day extension dialing 80, all I'm trying to test
is if is morning I would like asterisk to say Good Morning but, when I run
the test I get the following
Same thing:
== Using SIP RTP CoS mark 5
-- Executing [...@outbound:1] Answer(SIP/102-096a48c8, ) in new stack
-- Executing [...@outbound:2] Verbose(SIP/102-096a48c8, In timeofday
) in new stack
In timeofday
-- Executing [...@outbound:3] GotoIfTime(SIP/102-096a48c8,
Hello.
I am currently testing an asterisk server using the default codecs, I have
allow=all, and noticed everytime I test it in a wireless lan the latency
rockets off the roof to over 1000ms. I would like to test g729 since it uses
less bandwidth but, read somewhere I have to buy a license per
You only need to purchase 10 licenses, if all 10 clients
will be making calls at the same time.
Ok. Does this apply only for outbound calls using a voip provider and/or
applies to calls within the lan?
___
-- Bandwidth and Colocation
List.
How can I resolve this problem?
I've searched on the web but, can't really find a solution.
Please help.
--- On Wed, 11/25/09, Landy Landy landysacco...@yahoo.com wrote:
From: Landy Landy landysacco...@yahoo.com
Subject: [asterisk-users] Unable to open sound file error
To: Asterisk
list. People might use your account to
call satelite
lines for EUR 7,50 per minute. This kind of mistakes might
bankcrupt
you :-(
I hope this helps.
Erik
On 19 nov 2009, at 22:36, Landy Landy wrote:
Can someone please share with me a sip configuration
to connect
Hello.
I have a question regarind sound files in asterisk 1.6. I have a sound package
in ulaw format and I would like to know if I have a sip extension with
allow=alaw would asterisk convert that file to the codec the user is allowed to?
I am having a problem playing a file that exist in
How about adding:
insecure=invite,port
--- On Mon, 11/23/09, Tim Uckun timuc...@gmail.com wrote:
From: Tim Uckun timuc...@gmail.com
Subject: [asterisk-users] can't get pap2 to register from outside the LAN.
To: asterisk-users@lists.digium.com
Date: Monday, November 23, 2009, 8:25 PM
I
the past week and weren't
working. After restarting asterisk I'm able to use my provider via asterisk to
make calls.
I would like to thank those who helped me.
--- On Fri, 11/20/09, Landy Landy landysacco...@yahoo.com wrote:
From: Landy Landy landysacco...@yahoo.com
Subject: Re: [asterisk-users
.
--- On Thu, 11/19/09, Landy Landy landysacco...@yahoo.com wrote:
From: Landy Landy landysacco...@yahoo.com
Subject: Re: [asterisk-users] can't call through voip provider
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Thursday, November 19
Ok. I do NOT have ports 1-2 opened in. I guess I
should try that and see if it works.
I will open ports 5060 - 5070 and 1 - 100100 and do
some test tonight. I will keep you posted.
I ran this test and there was no difference.
I still can't get through.
---
Retransmitting
Can someone please share with me a sip configuration to connect an asterisk
server to a voip provider since my configuration isn't working for me.
thanks.
--- On Thu, 11/19/09, Landy Landy landysacco...@yahoo.com wrote:
From: Landy Landy landysacco...@yahoo.com
Subject: Re: [asterisk-users
: Thursday, November 19, 2009, 5:11 PM
On Thu, Nov 19,
2009 at 3:36 PM, Landy Landy landysacco...@yahoo.com
wrote:
Can someone please share with me a sip configuration to
connect an asterisk server to a voip provider since my
configuration isn't working for me.
thanks.
Who
I have the conf provided in last post.
exten = _9.,1,Dial(SIP/voipprovider/${EXTEN:1})
Yes, I have that in the dialplan.
Does sip show registry show that it's registered
successfully?
*CLI sip show registry
Host dnsmgr Username Refresh State
Hello.
Please help me with this, I can find any solution on this pls help. Your help
will be very appreciated. Thanks.
--- On Tue, 11/17/09, Landy Landy landysacco...@yahoo.com wrote:
From: Landy Landy landysacco...@yahoo.com
Subject: Re: [asterisk-users] can't call through voip provider
-users] can't call through voip provider
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Wednesday, November 18, 2009, 9:28 AM
On Wed, 2009-11-18 at 06:01 -0800,
Landy Landy wrote:
Please help me with this, I can find any solution on
this pls
:03 PM
What does your provider see when you
attempt to call them?
Thanks,
--Warren Selby
On Nov 18, 2009, at 3:38 PM, Landy Landy landysacco...@yahoo.com
wrote:
Thanks for replying.
But how come I'm able to use a softphone to place
calls from withing
the lan? I really
in (you can cut this to as small as 1-10004).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of Landy Landy
Sent: Wednesday, November 18, 2009 4:13 PM
To: Asterisk Users Mailing List - Non-Commercial
Discussion
asterisk-users@lists.digium.com
Date: Monday, November 16, 2009, 9:51 PM
On Mon, Nov 16,
2009 at 2:40 PM, Landy Landy landysacco...@yahoo.com
wrote:
snip
I don't know what else to try. When I try to call I get
this at the cli:
== Using SIP RTP CoS mark 5
-- Executing
Hello.
Sorry to repost this message but, I don't have the original message in my inbox
nor in my sent box.
Well, last week I posted a problem I am having trying to use an asterisk server
use a voip provider and a pstn. Pstn works fine but, I cant even connect to my
provider's server. I don't
According to my provider they´re not receiving any request from us but, now
everytime I try to place a call through them I´m getting:
*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
100(Unspecified)D 5060
) and try
again...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of Landy Landy
Sent: Saturday, November 14, 2009 10:15 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Can't connect to voip
Pre-judging people doesn't work on mailing lists given
the
inherent language barriers, etc.
I believe language barriers can cause many problems when trying to communicate.
I might say something in another language trying to translate a phrase or
something, that might not have the same
Have you tried nat=yes in the
definition in sip.conf?
Yes, I have that definition in sip.conf. Now, I'm getting the following error
-- SIP/voipprovider-094132d8 is making progress passing it to SIP/102-09423d58
-- Got SIP response 603 Declined back from 208.xx.xx.xx
--
Hello.
I'm trying to test an Asterisk server by using a VOIP provider for
international calls but, I'm having problems trying to get my server
communicate with theirs. I don't know if I'm having all these issues becuase
I'm behind NAT or what. I have the following in my server's sip.conf:
exted != exten
Ok. That was the actual error, I guess I needed some sleep.
Thanks.
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
I am testing an ivr but I'm having problems. The call keeps looping and it
doesn't hangup the call after passing three times through the menu. Here's my
conf:
exten = s,n,NoOp(Here's Count)
exten = s,n,NoOp(${COUNT})
;123,n,Set(COUNT=$[${COUNT} - 1])
exten = s,n,GotoIf($[${COUNT} =
Do you mean that incoming calls on your PSTN line works as
they should,
but not when they reach the voicemail? or that incomming
calls on PSTN
are always mute?
Incoming calls on PSTN line work as they should but, when someone leaves a
voicemail message the messege is mute. When I try to
:00PM
-0700, Landy Landy wrote:
Hello.
I have a server installed with asterisk 1.6. I have a
PSTN line that
comes in to one of those clone cards. Everything seem
to be working
fine. The only problem I have is that I can't get
voicemails coming
from the PSTN line. All other: SIP, IAX
Hello.
I have a server installed with asterisk 1.6. I have a PSTN line that comes in
to one of those clone cards. Everything seem to be working fine. The only
problem I have is that I can't get voicemails coming from the PSTN line. All
other: SIP, IAX work fine. I can hear those ok but, when
I have a similar problem with DAHDI. If my server gets rebooted, I can't make
any calls until the a call come in from outside. From there I can answer the
call and DAHDI works fine afterwards.
--- On Mon, 9/28/09, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
From: Tzafrir Cohen
In your case: is the problem reset by restarting asterisk?
'dahdi
resstart'?
The problem does not reset by restarting asterisk.
I've noticed that I can call other sip phones but, when trying to call out, I
get the same (Busy/Congested/Not-Available) congested messege.
I also found this weird, I thought my equipment was the problem. Good to know
about this issue so, Digium takes care of the problem.
I'm running:
asterisk-1.6.1.5
dahdi-linux-2.2.0.2
libpri-1.4.10.1
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