Could you share your AGI script?
CK
On Wed, Aug 18, 2010 at 5:43 AM, Johann Hoehn johann.ho...@ecommerce.comwrote:
On 08/17/2010 09:00 AM, Tino wrote:
Hello,
I would like to send sms to some external phone numbers from my
asterisk server. Is it possible to send sms via softphones like
Johann Hoehn wrote:
On 08/17/2010 09:00 AM, Tino wrote:
Hello,
I would like to send sms to some external phone numbers from my
asterisk server. Is it possible to send sms via softphones like
X-Lite ? . Any tips regarding this will be helpful
thanks
This is easy to do by using email to
Hi,
Use requirecalltoken=no in your peer configuration
Regards
On Wed, Aug 11, 2010 at 4:28 AM, bruce bruce bruceb...@gmail.com wrote:
Hello Everyone,
I am trying to diagnose issue with my IAX2 extension not working.
When I have iax2 set debug on all I see is this:
*Rx-Frame Retry[ No]
Avoid to use MySQL dialplan application, instead write an AGI script for
this purpose
On Tue, Aug 17, 2010 at 4:59 PM, Geraint Lee gera...@gmail.com wrote:
Right, I'm baffled.
I have:
exten = s,1,MYSQL(Connect DB1 127.0.0.1 geraint xxx amis2)
exten = s,n,MYSQL(Query NORESULT ${DB1} INSERT\
Hi,
I'm quite pleased with the asterisk/res_snmp integration (at least a
right one :) not some hackish scripted thingy)
but i felt it's missing quite a few datas.
What i would need is:
* Per channels, number of inbound call received since asterisk
startup (like a network interface)
*
Unless I've got some massive misunderstanding, yes we do
as described here
http://www.voip-info.org/wiki/view/Asterisk+RealTime
Why do you ask?
On 17/08/10 17:27, Zeeshan Zakaria wrote:
Ishfaq, do you use the asterisk real-time architecture?
Zeeshan A Zakaria
--
www.ilovetovoip.com
So why you need a reload when you are using the real-time architecture? Are
you adding contexts in extensions.conf? If yes then where are you using the
real-time?
I asked because to me it seems you didn't understand what Dan is asking, and
also seems you don't know how real-time works in regards
Hi
I'll give you an example process flow
Our customer logs onto our VoIP Portal and orders a SIP Geographic number
Geo number is successfully ordered which inserts an entry into our
extensions table in MySQL DB in the default context which has a goto
command to a newly created specific
man, see monast http://monast.sourceforge.net/
--
Renato dos Santos
shazaum.wordpress.com
2010/8/17 Matt Riddell li...@venturevoip.com
On 17/08/10 6:34 PM, Hans Witvliet wrote:
On Mon, 2010-08-16 at 13:35 -0400, Jamie A. Stapleton wrote:
Might be worth your time to check out:
Thanks for the details. I Agreed you know what you are doing. As for doing
it more elegantly, it is not possible to do a fruitful discussion without
knowing all the details of how your dialplan works, neither is this the goal
here. But I offer similar hosted PBX service, just use one context for
Look into astassisstant. I don't remember the website, but google will take
you their. It doesn't need any installations on the server, just a manager
user in manager.conf.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-08-18 8:39 AM, Shazaum shaz...@gmail.com wrote:
man, see monast
Hello Johann,
Thanks for your advice in this matter. But i am not sure how to pass the
numbers to be sent sms in the dialplan.
On Wed, Aug 18, 2010 at 3:13 AM, Johann Hoehn johann.ho...@ecommerce.comwrote:
On 08/17/2010 09:00 AM, Tino wrote:
Hello,
I would like to send sms to some
On Tue, 2010-08-17 at 09:42 -1000, Ben Schorr wrote:
Sorry, I should clarify - we have had a similar setup (IPSEC VPN, Polycom
331) working at a different location with a different handset for this same
firm. We've never gotten the phone/VPN to work at this particular site. I
was just
... using it as a tool and understanding what it does...
So one part of it's toolset identifys valid SIP accounts - and I was under
the impression that alwaysauthreject=yes was supposed to stop this...
However, it sends a request for a highly probably non-existent account,
then sends requests
Sending this to asterisk-users, in case anyone has AsteriskNOW
experience they can share.
Joe
-- Forwarded message --
From: Joe Wood sch...@gmail.com
Date: Wed, Aug 18, 2010 at 9:22 AM
Subject: AsteriskNow REGISTER'ing s@ extension for all inbound trunks
To:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joe Wood
Subject: [asterisk-users] Fwd: AsteriskNow REGISTER'ing s@ extension for
allinbound trunks
snip
Since you can see the CLI log, please post your asterisk version (core show
Le 18/08/2010 16:03, Tino a écrit :
Hello Johann,
Thanks for your advice in this matter. But i am not sure how to pass
the numbers to be sent sms in the dialplan.
agi(script,param1,param2,...,paramX) from your dialplan where script
lies in /var/lib/asterisk/agi-bin
On Wed, Aug 18, 2010 at
Hi,
My WaitExten() is not working as I expect it to. This is the relevant part
of my context. It is meant to receive incoming calls.
[incoming]
exten = xxx,1,Background(hello-world)
exten = xxx,2,WaitExten(7)
exten = _X,1,AGI(myAGI.php)
When I send the call from a .call, it works perfect, but
That is set and here is what I get:
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
REGREQ
Timestamp: 3ms SCall: 01217 DCall: 0 [44.55.66.77:4569]
USERNAME: 9988
REFRESH : 60
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX
Un-top-posting...
On 08/17/2010 09:00 AM, Tino wrote:
I would like to send sms to some external phone numbers from my asterisk
server. Is it possible to send sms via softphones like X-Lite ? . Any
tips regarding this will be helpful
On Wed, Aug 18, 2010 at 3:13 AM, Johann Hoehn
Hi,
My WaitExten() is not working as I expect it to. This is the relevant part
of my context. It is meant to receive incoming calls.
[incoming]
exten = xxx,1,Background(hello-world)
exten = xxx,2,WaitExten(7)
exten = _X,1,AGI(myAGI.php)
When I send the call from a .call, it works perfect, but
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kathryn Jones
Subject: [asterisk-users] WaitExten() always times out
Hi,
My WaitExten() is not working as I expect it to. This is the relevant part
of my context. It is meant to receive
sg01*CLI core show version
Asterisk 1.6.2.11 built by root @ localhost.localdomain on a i686
running Linux on 2010-08-16 15:17:26 UTC
On Wed, Aug 18, 2010 at 10:19 AM, Danny Nicholas da...@debsinc.com wrote:
From: asterisk-users-boun...@lists.digium.com
Thanks for you reply :).
I thought of that and tried replacing _X with a numbers it should match (9),
and it didn't work. It still times out as if no number was entered.
On Wed, Aug 18, 2010 at 2:11 PM, Danny Nicholas da...@debsinc.com wrote:
*From:*
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kathryn Jones
Subject: Re: [asterisk-users] WaitExten() always times out
Thanks for you reply :).
I thought of that and tried replacing _X with a numbers it should match
(9), and it
Hello list!
I am trying to get hold of ${CDR(duration)} and ${CDR(billsec)} variables in
h
It seems that these variables always return 0. I am using Asterisk version
1.6.2.11. Can't I get these values other than using CDR reccords ??
--
My .call file goes out to a pstn number.
That work around would be perfect :D, but I need the number given by the
caller.
On Wed, Aug 18, 2010 at 2:49 PM, Danny Nicholas da...@debsinc.com wrote:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
Hi,
Are you sure asterisk is receiving and processing DMTF correctly? Are
you using rfc2833, SIP INFO or inband DMTF? What is your asterisk
version? I use WaitExten(5) all the time, no matter if they are
single-digit or multiple-digit extensions.
Regards,
--
Ing. Miguel Molina
Grupo de
This is what I ended up doing, working fine now.
Cheers
On 18 August 2010 08:52, Nasir Iqbal na...@ictinnovations.com wrote:
Avoid to use MySQL dialplan application, instead write an AGI script for
this purpose
On Tue, Aug 17, 2010 at 4:59 PM, Geraint Lee gera...@gmail.com wrote:
Right,
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kathryn Jones
Subject: Re: [asterisk-users] WaitExten() always times out
My .call file goes out to a pstn number.
That work around would be perfect :D, but I need the number given by the
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tiago Geada
Subject: [asterisk-users] CDR variables
Hello list!
I am trying to get hold of ${CDR(duration)} and ${CDR(billsec)} variables
in h
It seems that these variables always
I must not be receiving them properly, since I can't make it work. I just
can't see why :P.
My asterisk version is 1.6.2.6. Like I said before, for outgoing .call files
WaitExten works fine, it's on incoming calls that I cannot receive the
number I need.
I had not checked my dtmf mode, this is
We don't use a context for that.
We set up dialplan code in a non asterisk part of MySQL called routing
types.
When a customer selects a DDI number they can choose a routing type to
use with it.
These routing types allow for variable substitution - i.e. if someone
adds the routing type
Hi Matt,
That's somewhat closer to what I do in my dialplan as well. But Dan
apparantly wants to add new contexts in his `extensions` table.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-08-18 6:02 PM, Matt Riddell li...@venturevoip.com wrote:
We don't use a context for that.
We set up
My thanks for previous help on fixing IXJ issues in 1.2.40; I now
have problems with a just-built 1.4.35 on the same host:
[Aug 18 17:26:48] WARNING[27209]: app_dial.c:1298 dial_exec_full: Unable
to create channel of type 'Phone' (cause 0 - Unknown)
== Everyone is busy/congested at this time
On Wed, 18 Aug 2010, Infra wrote:
My thanks for previous help on fixing IXJ issues in 1.2.40; I now
have problems with a just-built 1.4.35 on the same host:
[Aug 18 17:26:48] WARNING[27209]: app_dial.c:1298 dial_exec_full: Unable
to create channel of type 'Phone' (cause 0 - Unknown)
==
On Wed, 18 Aug 2010, Infra wrote:
On Wed, 18 Aug 2010, Infra wrote:
My thanks for previous help on fixing IXJ issues in 1.2.40; I now
have problems with a just-built 1.4.35 on the same host:
snip
I applied the patch for dialtone as per my issue on 1.2.40.
snip
the fxs line is now
Hi
to convert wav file use following
sox 'orgFile' -w -r 8000 -c 1 -s 'fixedFile'
while replace orgFile and fixedFile with actual filenames
If still now luck try with mp3
Regards
--
_
-- Bandwidth and Colocation Provided
I would rather use .call files. So easy to produce a text file...
On 18 August 2010 21:02, Steve Edwards asterisk@sedwards.com wrote:
Un-top-posting...
On 08/17/2010 09:00 AM, Tino wrote:
I would like to send sms to some external phone numbers from my asterisk
server. Is it possible
On Wednesday, August 11, 2010 11:08:37 am Tino wrote:
#!/bin/bash -x
T=$agi_uniqueid
I want to save value of 'agi_uniqueid' channel variable into a variable
called 'T' in my script
When executing and AGI from the dialplan, it will dump out it's variables
immediately, so you need to tell
On 08/19/2010 08:21 AM, Tiago Geada wrote:
I would rather use .call files. So easy to produce a text file...
On 18 August 2010 21:02, Steve Edwards asterisk.org
http://asterisk.org@sedwards.com http://sedwards.com wrote:
Un-top-posting...
On 08/17/2010 09:00 AM, Tino wrote:
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