Maybe it is something to do with AGI - Dial command.
IFAIK you can't control Dial via AGI script.
From http://www.voip-info.org/wiki/view/Asterisk+AGI :
Dialing out
If the AGI application dials outward by executing Dial, control over
the call returns to the dialplan and the script loses contact
Hello List.
We are having some problems using t.38 together with a Cisco voice router at
one of our providers end.
We are using the new digium asterisk fax module to generate the fax, and when
we use together with our internal Audiocodes Mediant 2000 gateways, we have no
issues what so
Hi,
I installed Digiums Free Fax for Asterisk and found out, that it
automatically retries failed faxes, is there a way to stop that?
Thanks
Markus
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asterisk-users mailing
And also
this is the macro for failover
[macro-trunkdial-failover-0.3]
exten = s,1,GotoIf($[${LEN(${FMCIDNUM})} 6]?1-fmsetcid,1)
exten = s,2,GotoIf($[${LEN(${GLOBAL_OUTBOUNDCIDNAME})} 1]?1-setgbobname,1)
exten = s,3,Set(CALLERID(num)=${IF($[${LEN(${CID_${CALLERID(num)}})}
Hi,
We've just finished adding support for writing AGI scripts in a variety
of popular scripting languages to Asterisk-Java.
The FastAGI server in Asterisk-Java allows you to move your AGI scripts
to a dedicated server and increases performance by eleminating the need
to start the language
Hi All,
Sorry to hijack this post but I am confused. What is the advantage of using
this Digium Fax For Asterisk product when you can use Asterisks' 1.6.x module
app_fax or Asterisks' 1.4.x agx-ast-addons with the app_txfax and app_rxfax
modules?
Regards
David.
-Original Message-
Hi All,
I was wondering if there's any way in Asterisk 1.4.21.2 to playback a wav
file to a channel using the AMI?
I've had a play and, as there wasn't a Playback command implemented directly
in the AMI, I thought about maybe calling an AGI script from the AMI to do
this but it seems there's no
Hi,
it was was my fault, there is no retry ... sorry to bother you.
@David:
I wasn`t very conviced about Spandsp, after trying several versions it
worked, but not well.
We are sending faxes via SIP. When sending faxes from our 1.6 Asterisk
to our 1.4 Asterisk 50%+ Faxes failed.
T.38 worked once
On Wed, May 13, 2009 at 3:43 AM, Markus Weiler
markus_wei...@mailworks.org wrote:
I installed Digiums Free Fax for Asterisk and found out, that it
automatically retries failed faxes, is there a way to stop that?
You already claimed that this isn't actually the case. I will tell you
that the
Hi folks
I am still thinking about the best way to fit this into the config
files, but in the meantime I would like to offer some additional info
in support of my argument for both signalling hold and sending MOH
media. This is quoted from the SIPConnect recommendation from The SIP
Forum, an
On Wed, May 13, 2009 at 7:39 AM, Markus Weiler
markus_wei...@mailworks.org wrote:
I wasn`t very conviced about Spandsp, after trying several versions it
worked, but not well.
spandsp has been revised to the point that it's now at 0.0.6pre11,
released this month. I've had quite the opposite
On Wed, May 13, 2009 at 3:30 AM, Jon Schøpzinsky j...@firstcom.dk wrote:
We are having some problems using t.38 together with a Cisco voice router at
one of our providers end.
We are using the new digium asterisk fax module to generate the fax, and
when we use together with our internal
I just inherited a client that is using a Switchvox system. I normally
install a CentOS based system with freePBX and some custom endpoint
management stuff for Polycom phones. This Switchvox is making me feel a
bit stifled. I am having nightmares of another recent encounter with
Trixbox
Thats not entirely true.
I am using astcc.agi which does exactly this (actually is DeadAGI): dials
the call, and when call is finished, control is given back to the agi script
(for updating cdr and billing the call).
What I am trying to do is just add a small portion of code in the 'trytrunk'
Hi All,
I'm trying to find a software package to do the following sip proxy
work:
I've an A*k server A that needs to be decommissioned, from the USA, and
replaced by server B, in the UK. Both servers are on public internet
IPs.
Whilst the client migration happens, I want to divert all the
I used wireshark to debug the problem, and I can see that the cisco equipment
is correctly sending t.38 packets to asterisk, and the whole re-invite process
is successful.
The problem is, that Asterisk discards the t.38 packets with the error message
I sent, and therefore the T.38 session never
You appear to be quite correct on your Google analysis. From my light
reading of Digium's description of Switchvox, it is pretty much Asterisk
for Dummies; the sole interface and maintenance of the system is a web
interface. The concept apparently is that Digium provides a running system
Hello everyone,
A month ago I took on an issue on the Asterisk issue tracker
(https://issues.asterisk.org/view.php?id=11797) dealing with multicast RTP
paging.
This is the ability to send audio to phones (the phone must support it) and
have it played out the speakerphone. Using multicast RTP
I have a Sangoma A400 card with two FXS ports. They work fine,
however as I have analog phones connected, I have no way of telling
the phone I am done dialing. Pressing # works fine, but then Asterisk
passes that # over to the POTS line, and about every 5th call, for
some reason that is
Redirect traffic with iptables like this:
Host ~# iptables -t nat -I PREROUTING -d OLD_PUBLIC_IP -j DNAT --to
NEW_PUBLIC_IP
I'm not sure if this will work for SIP. You may need the proxy to change info
in the sip messages between server and client.
--Dave
From:
Here is the AMI packet I use to do this:
Action: Originate
Channel: Local/do_playb...@cfmc_cdi_private
Exten: do_chanspy
Context: cfmc_cdi_private
Priority: 1
Variable: CfMC_ActionID=callE1330
Variable: CfMC_WhatToPlay=cfmc/song
Variable: CfMC_WhoHear=SIP/GXP280-16-0844e290
ActionID: callE1330
cb wrote:
I have a Sangoma A400 card with two FXS ports. They work fine,
however as I have analog phones connected, I have no way of telling
the phone I am done dialing. Pressing # works fine, but then Asterisk
That's what the digit and response timeouts are for. I have:
;
That's superb, thanks very much Jim.
J.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Dickenson
Sent: 13 May 2009 15:54
To: Asterisk User MailList
Subject: Re: [asterisk-users] Playback to channel using
I would try with a b2bua.
Here's a good (imho) example:
http://www.b2bua.org/
As a second step to take, I would do automatic tftp/http provisioning for
the devices you have (unless you are talking about softphones). This way you
can specify whichever sip server you want for your devices.
Hi David,
Thanks for the reply. That's pretty much what I've already tried, but
with no luck on the production machines. In testing it worked, but the
public IPs and single NICs were causing issues (we believe)
So I was looking for a proxy-type solution.
Adrian
I have an application where we receive calls on an inbound PRI. After
hours, our Asterisk box dials our answering service on an outbound PRI
and then bridges the caller to the answering service. The flow looks
like this:
(CALLER)INBOUND_PRI -- CONTEXT -- GOSUB(Incoming) --
AstriCon 2009 is still 5 months away, but the time will fly! We're
looking for speaker presentations for 2009's conference - do you have
something you want to talk about? Submit your talk proposal today -
the proposal window closes on June 1. AstriCon is the single largest
conference
On Wed, May 13, 2009 at 9:21 AM, Jon Schøpzinsky j...@firstcom.dk wrote:
I used wireshark to debug the problem, and I can see that the cisco equipment
is correctly sending t.38 packets to asterisk, and the whole re-invite
process is successful.
The problem is, that Asterisk discards the t.38
Hello!
I am trying to mount a remote directory for voicemail using sshfs. However,
whenever Asterisk attempts to write the file, it fails, because SSHFS cannot
lock the directory. Is there a solution to this problem or an alternative
method for using a remote directory for voicemail?
Thanks,
Probably a permissions problem. Check out this article
http://ubuntu.wordpress.com/2005/10/28/how-to-mount-a-remote-ssh-filesystem-
using-sshfs/
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Elliot Murdock
Sent:
Hi,
I'm connecting Asterisk v. 1.4.10 to Zanzibar Open IVR that acts as a SIP
trunk. Since recognition didn't work correctly, I've troubleshot with
Wireshark and saw that RTP stream is first send to one port on SIP trunk and
then when first RTP packet arrives in opposite direction (from TTS
Tunnel samba or nfs through ssh, rather than using sshfs, then mount using once
of those more ubiquitous standards.
-Dave
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Elliot Murdock
Sent: Wednesday, May 13, 2009 1:09 PM
To:
Hellos,
I want to setup Asterisk+a2billing for over 10,000 extensions for voip
resale. Has anyone done this before. What are the hardware requirements and
challenges?
James
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- James Mutuku listmut...@gmail.com wrote:
Hellos,
I want to setup Asterisk+a2billing for over 10,000 extensions for voip resale.
Has anyone done this before. What are the hardware requirements and challenges?
James
If you're asking that sort of question, you probably shouldn't be
Why not use OpenSIPS or Kamailio in stateful mode?
You will need to look at how media is handled though, but a SIP proxy
will work easily.
On 13/05/2009, Adrian Marsh adrian.ma...@ubiquisys.com wrote:
Hi David,
Thanks for the reply. That's pretty much what I've already tried, but
with no
Recently I noted few recording link with # sign on it.
like 223345#-all.gsm
and all those voice files are NOT available for download.
I tried changing the file name in the mysql db and removed # but still its
not available.
What could be the reason for # and why its NOT available for
On Thu, 14 May 2009, David @ULC wrote:
Recently I noted few recording link with # sign on it.
Asterisk creates links?
like 223345#-all.gsm
and all those voice files are NOT available for download.
Asterisk downloads files?
I tried changing the file name in the mysql db and removed #
You should send that to the Business list, not
users.
From: asterisk-users-boun...@lists.digium.com
This could be a nice opportunity for users with a high volume of SIP traffic
terminating in the US:
Collecting dip fees on outbound phone calls - fees that would otherwise go
to the local phone company.
With all the recent fees and surcharges, the cost of wholesale telecom and
dialer traffic
Hello,
I have a strange situation with an SPA3102 FXO/FXS device. I'm in
situation that when i receive a call from PBX line I must forward the
calls to 2 VoIP numbers.
Right now i have the following settings: (S0:1...@gw1). I want to
forward at 1020 too. I tested (S0:1010|1...@gw1) and doesn't
Date: Thu, 14 May 2009 01:59:58 +0300
From: Catalin S. jonsonpla...@gmail.com
Reply-To: Asterisk Developers Mailing List asterisk-...@lists.digium.com
To: Asterisk Developers Mailing List asterisk-...@lists.digium.com
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
good news, i just made my isdn device ring! ok, after it ring, any
timout then hangup up the chan, but a ringing from chan_dahdi via
bri_net_ptmp - isdn_device was possible.
to made this happen i made some very crude hacks inside libpri, but i
hope the next days i can offer a patch that offer
Howdy,
How do i perform a lookup from a remote odbc database in the asterisk dialplan?
I can do it with mysql but not sure of commands for odbc connection.
Cheers!!!
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Hi
I am in the middle of move a small business over from legacy PABX + PSTN
lines to VOIP infrastructure.
I borrowed a spa9000 to place between the PABX and the PSTN lines. I
have had this going for a while (5 months) and it has been working fine
(some issues with echo and other minor things),
On Wed, May 13, 2009 at 11:53 AM, Barry L. Kline blkl...@attglobal.net wrote:
If I insert a Monitor() prior to dialing the outbound call, I get no
audio in the recording and the caller hears no audio. Occasionally it
works (perhaps 1 out of 5 times) but most of the time the caller can't
hear
I think you have your line types mixed up - FXS is for phones, FXO is
for lines.
An analogue passthorugh setup _is_ doable, just not overly recommended.
PaulH
Alex Samad wrote:
Hi
I am in the middle of move a small business over from legacy PABX + PSTN
lines to VOIP infrastructure.
I
David Backeberg wrote:
I don't know why recording is breaking your calls. My guess is
something is screwed up with your PRI configuration. Are you getting
alarms in your logs from dahdi?
Not a peep, either with or without using the monitor command. I've
been using this system for around
On Wednesday 13 May 2009 17:55:41 carl Lougher wrote:
Howdy,
How do i perform a lookup from a remote odbc database in the asterisk
dialplan?
I can do it with mysql but not sure of commands for odbc connection.
See func_odbc.conf for examples. You'll also need to setup res_odbc.conf, as
this
Ok, I'm still rather new to Asterisk, and I'm sure there is a simple fix
here, but I can't see it.
Client with a small system, AsteriskNow 1.4, 10 Polycom IP330 phones. Has
been up and running flawlessly for about a year.
This morning I logged in to make a couple of extension changes, and
On Thu, May 14, 2009 at 12:17:47PM +1000, Paul Hales wrote:
I think you have your line types mixed up - FXS is for phones, FXO is
for lines.
sorry why do you think that, I have 3 fxs + 1 fxo (my understanding is
that a attached fxs presents internally as a fxo
I have a pstn line attached to
Alex Samad wrote:
On Thu, May 14, 2009 at 12:17:47PM +1000, Paul Hales wrote:
I think you have your line types mixed up - FXS is for phones, FXO is
for lines.
sorry why do you think that, I have 3 fxs + 1 fxo (my understanding is
that a attached fxs presents internally as a fxo
I
Have you tried plugging analog phones into the FXS ports in the Asterisk
box?
That should let you know what the Asterisk is really doing with it's FXS
ports.
PaulH
Alex Samad wrote:
On Thu, May 14, 2009 at 12:17:47PM +1000, Paul Hales wrote:
I think you have your line types mixed up -
On Thu, May 14, 2009 at 03:18:28PM +1000, Paul Hales wrote:
Alex Samad wrote:
On Thu, May 14, 2009 at 12:17:47PM +1000, Paul Hales wrote:
I think you have your line types mixed up - FXS is for phones, FXO is
for lines.
sorry why do you think that, I have 3 fxs + 1 fxo (my
On Thu, May 14, 2009 at 03:31:18PM +1000, Paul Hales wrote:
Have you tried plugging analog phones into the FXS ports in the Asterisk
box?
good ideal, but trying to find an old style phone the site has a
commander PABX with digital handsets. I will see if I can track one down
:)
A
FXO channels shuld have FXS signalling, and FXS channels shuld have FXO
signalling, so:
# FXO channels are 1,2,3
fxsks=1,2,3
# FXS channel is 4
fxoks=4
sorry why do you think that, I have 3 fxs + 1 fxo (my understanding is
that a attached fxs presents internally as a fxo
I have a
Alex Samad wrote:
On Thu, May 14, 2009 at 03:31:18PM +1000, Paul Hales wrote:
Have you tried plugging analog phones into the FXS ports in the Asterisk
box?
good ideal, but trying to find an old style phone the site has a
commander PABX with digital handsets. I will see if I can
Alex Samad wrote:
On Thu, May 14, 2009 at 03:18:28PM +1000, Paul Hales wrote:
What happens if you make a call in from the old fax line and send that
over to the old PABX? Does that work OK?
not sure what you are asking here. I have checked an incoming call
through the FXO(PSTN)
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