Maybe this can help:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AMD
Mindaugas Kezys
http://www.kolmisoft.com
MOR - Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tong
Sent: Sunday, December 02, 2007
Hello:
Some time ago, in this list, I asked for a DID provider in Spain.
Somebody answers me detailing his company DID services.
I has an email problem and lost his response and his mail address.
Please, if you (the DID provider) are reading this, let me know again your
mail address because we
I guess /tmp can live in RAM, but what about eg. recording ten-twenty
WAV files to /var a day, and logs into /var/log? Do I have to worry
about the card wearing out in six months?
This is nothing really. Just make sure your using an industrial compact
flash card. These support 1-2 million
Hi,
I am trying to get a SIP extension's status without
actually making a call.
I am using sofia-sip's options example utility and
the sip clients are SJphone softphones.
From Asterisk I run the options utility and query a
sip extension at 10.215.147.240. I get:
# ./options -1 --all
I submiited to the list last night, but it never showed up. Here we go
again.
I've tried building Asterisk 1.4.15 on Solaris based on instuctions
here, http://forums.digium.com/viewtopic.php?t=5888. However, this is
the message I get. This is Solaris on X86. Any ideas?
[CC]
Dear Salvatore and Joanna,
Thank you much for both your detailed explanations.
I will surely check my firewall configuration and logs to make sure the
VoIP traffic is passing correctly.
However, I'm a bit confused as the problems that I'm experiencing are
with calls made via the
Sangoma analog
vieri,
you can get sip status with the following shell script... I named it
'sipshowpeer'... to execute, chmod 755 sipshowpeers
daveC
-- cut here -
#!/bin/sh
# sipshowpeers
#
# show current asterisk SIP peers
asterisk -r -x 'sip show
carlos,
you got further than I did... AMD didn't work at all on my release.. I
think I was using 1.4.11 at the time...
I ended up using the below
daveC
;--- amdtest (ext 13) starts here
;
; restructure this for the following conditions:
; 13 using
Hi,
try adding this in your stdtime/localtime.c
#define _POSIX_PTHREAD_SEMANTICS
#undef TM_ZONE
#undef TM_GMTOFF
if this does not work just google it, there are workaround for this problem
Thanks,
Vivek
On 12/2/07, Mike Clark [EMAIL PROTECTED] wrote:
I submiited to the list
On Sunday 02 December 2007 09:25:06 dave cantera wrote:
Vieri wrote:
I am trying to get a SIP extension's status without
actually making a call.
I am using sofia-sip's options example utility and
the sip clients are SJphone softphones.
From Asterisk I run the options utility and
On Saturday 01 December 2007 19:27:09 Philipp Kempgen wrote:
You did not make clear if you try to build on an i686 or on
a slug (as your subject says) which is not x86 but Intel
XScale.
Oh, sorry! I want to run asterisk on the slug, but compile it on my
desktop box, which is an i686. Wasn't
Thanks for the sip show peers script, Dave.
But that won't work for me.
It won't tell me whether the extension will actually
accept a call or not (eg. if DND is ON only on the
client side).
This link might clarify the problem I am facing:
--- Tilghman Lesher
[EMAIL PROTECTED] wrote:
To the original poster: OPTIONS is the right type
of request (the client
should respond exactly as the same way as if you had
sent it an INVITE).
Your next step should be to contact the author of
the softphone and impress
upon them the
If i use AMD() or the code below, now the problem is the fax machine/modem
detection and answer machine detection get detected as the same. If i need to
seperate the two how do i do that? For example, if i use AMD() to detect an
answer machine by saying any greeting exceeding 2.5 seconds is a
I also tried:
# ./options -1 -a sip:[EMAIL PROTECTED]:5072
but still received a
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.215.144.27:38102;branch=z9hG4bKFB4rQrr5aXp9H
From: sip:10.215.144.27;tag=FF1tQy74X81rm
To: sip:[EMAIL PROTECTED];tag=1639856599
Call-ID: 83259dd2-1b9e-122b-10a3-00c09f10e472
CSeq:
--- Tilghman Lesher
[EMAIL PROTECTED] wrote:
OPTIONS is the right type of request
Suppose that the user agent is not a softphone but a
gateway such as the Grandstream GXW-4008 ATA.
One of the FXS-port-connected phones of the gateway
has DND turned on.
IF I send an OPTIONS request then the UA
I tried another popular user agent: X-Lite.
I dialed *78 which in */FreePBX turns DND on AND I
pushed the DND button on the softphone.
# asterisk -vvvr
CLI database show dnd
/DND/4053 :
YES
Despite all this when I send an OPTIONS request I
always get a
Hello list,
I am trying to find a solution for interfacing a Dictaphone Freedom recorder.
Currently, 4 POTS lines interface to the recorder, and the future will have the
4 lines coming into an ABE server on PRI.
The Freedom system is using a standard amphinol connector to a punch down
block,
I have been looking forward for months to get chan_mobile working. I am
limited to using prepackaged Asterisk code, mostly Trixbox.
I have recently heard that chan_mobile is considered 'beta' and there is
no effort to move it into the main code of Asterisk. Not even for
Asterisk 1.6.
So
Fabiano Sidler wrote:
I'll probably post this again on the slug mailing list, when that seems
more appropriate for this topic.
I guess you already know this page
http://www.voip-info.org/wiki/view/Asterisk+Linksys+NSLU2
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566
On Sunday 02 December 2007 12:22:23 Robert Moskowitz wrote:
I have been looking forward for months to get chan_mobile working. I am
limited to using prepackaged Asterisk code, mostly Trixbox.
I have recently heard that chan_mobile is considered 'beta' and there is
no effort to move it into
I'm having (I think) timing issues in relation to bridged T1-T1 calls via
dynamic spans. Fax calls are intermittently working, but voice is fine. My box
has a Sangoma A400 inside it as the primary Zaptel timing source. My T1 PRIs
that are hooked to the box come in via a foneBRIDGE2 (dynamic
Hi. I am using the 'get_data' function from an AGI, and i find that
sometimes when users call in, it won't play the full gsm soundfile, and when
i try to press a number (or pound, or star), nothing will happen - it just
hangs there...
anyone else experience this?
- Dominic Son
It is not the
I have asterisk up and running on a fedora system but
having trouble accessing system via softphone (ekiga
and xlite). Im a newbie to asterisk and was looking
for some help walking through this. I imagine 10 - 15
mins would be all needed to make proper config changes
needed. Once I get this setup
Make certain that selinux, iptables and ip6tables are disabled and off.
Bryan M. Johns
Shelton | Johns
Office: 678.248.2637
FindMe: 678.229.1809
Support: [EMAIL PROTECTED]
http://www.sheltonjohns.com
On Dec 2, 2007, at 3:18 PM, James Cox wrote:
I have asterisk up and running on a fedora system
In theory, UAs that respond to OPTIONS and INVITE differently are broken.
Below is a quote from section 11.2 of RFC 3261.
The response to an OPTIONS is constructed using the standard rules
for a SIP response as discussed in Section 8.2.6. The response code
chosen MUST be the same that
On Fri, 30 Nov 2007 09:52:59 +0100, randulo [EMAIL PROTECTED]
wrote:
I have used SIP and IAX for about three years now. We don't do a lot
of traffic, but I haven't really seen a difference in quality or
dropped calls.
Sorry for jumping in, but besides ZoIPer/Idefisk, are there
IAX-capable
There are many, (i'm one of the people working for zoiper):
Look at the iaxclient homepage,
There are iaxcomm, loudhush, kiax, mediax , diax and many more,
(you could also easily make your own).
Cheers,
Zoa
Vincent wrote:
On Fri, 30 Nov 2007 09:52:59 +0100, randulo [EMAIL PROTECTED]
Hello averybody,
I'm looking the softswitch in digium website, anyone test the softswitch?
Best Regards
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I have used asterisk-1.4.14, zaptel-1.4.7, chan_ss7-1.0.0 on FC7 all
went okay. using sangoma a104dx on both machine.
I followed the write up on
http://www.voip-info.org/wiki/index.php?page=Asterisk+ss7+setup
I have the cross over cable between them.
however, wanpipe shows connected but the
--- Raj Jain [EMAIL PROTECTED] wrote:
In theory, UAs that respond to OPTIONS and INVITE
differently are broken.
Below is a quote from section 11.2 of RFC 3261.
The response to an OPTIONS is constructed using
the standard rules
for a SIP response as discussed in Section 8.2.6.
The
In the sip.conf entry assign a context.
In that context, hint the extension i.e. exten = 7302,hint,SIP/7302.
Before you get ready to dial, or whatever, do chanisavail i.e.
exten = _1,n(CheckUse),ChanIsAvail(SIP/${EXTEN:1},js)
exten = _1,n,Playback(beep)
exten =
On Dec 2, 2007 3:42 PM, Carlos Rojas [EMAIL PROTECTED] wrote:
I'm looking the softswitch in digium website, anyone test the softswitch?
Nope. No one has tested it or used it. Try the one at cisco.com.
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Thanks Richard but I think that ChanIsAvail must be
buggy (based on some user comments in the wiki,
although quite outdated).
I have the hint entry as you say (am using FreePBX and
it's already there).
But whenever I call ChanIsAvail with the s option I
always get:
${AVAILSTATUS} = 0
Ouch. Why do they almost all feel they have to use 'IAX' in their title?
Pain. While not half as bad, it's somewhat reminiscent of the iJunk
everyone seems to sell to capitalise on the iPhone phenomenon.
N.
Zoa wrote:
There are many, (i'm one of the people working for zoiper):
Look at the
I'd like to add that show hints on * CLI displays
the following for ext 4053 tested below:
4053: SIP/4053
State:IdleWatchers 0
(it should be unavailable or something, but anyway,
ChanIsAvail reports an AVAILSTATUS of 0, ie. unknown)
--- Vieri [EMAIL
I'm using 1.2.6 with the dialplan I posted so I guess the UA you are
using is just plain hosing you.
Anyway, with the queue I believe the music on hold is played to the
inbound side until the call is picked up by an agent. The queue tries
every retry seconds to get an agent for timeout
Hi,
Use orecx, voip call recording and monitoring.
www.orecx.com
Thanks Regards,
Vidura Senadeera,
Sri Lanka.
Tel - +94114520001
Mobile - +9466596
yahoo/skype Ids - vidurased
--
Message: 17
Date: Fri, 30 Nov 2007 08:58:41 +0530
From: ram [EMAIL
I appreciate the feedback. I too am not using and hope
not to use agent login. I don't know if I can apply
your dialplan because I need to distinguish whether an
agent is busy (thus average conversation time is
usually around 1-2 minutes) or has DND on (agent can
be absent for quite a while, eg.
Hello,
I recently upgraded from Asterisk 1.4.0 to 1.4.15... I am registering to
a sip provider in my sip.conf
as below
[general]
register=user:password:[EMAIL PROTECTED]/extension
Later down in my sip.conf I have the definition for that service provider
as follows
[serviceprovider]
hi, all
I want to connect asterisk with oracle database.
how to start this , that's i dont know .
any pls help me
thnks in advance
Bhrugu mehta
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To
Hello,
We've released another update to our astGUIclient-VICIDIAL suite: 2.0.4
http://astguiclient.sf.net/
The client suite runs on most modern web browsers on almost any
GUI-capable operating system, and it includes the VICIDIAL call center
suite and the astGUIclient client-side web app which
On Nov 30, 2007 11:46 PM, Thomas Balsfulland [EMAIL PROTECTED] wrote:
Hello list,
I try to setup an asterisk-server with different SIP-Peers to PSTN.
The Peer are working and configured in sip.conf:
[peer1]
type=peer
host=10.10.10.1
[peer2]
type=peer
host=10.10.10.2
Now
On Dec 3, 2007 3:12 AM, Carlos Rojas [EMAIL PROTECTED] wrote:
Hello averybody,
I'm looking the softswitch in digium website, anyone test the softswitch?
Try freeswitch.org
ram
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On Monday 03 December 2007 00:48:55 Bhrugu Mehta wrote:
I want to connect asterisk with oracle database.
You'll need to install the Oracle ODBC driver for Linux. One word of warning,
though: the ODBC driver linked against the InstantClient library has a very
nasty resource leak in the library
thnsk for giving me reply,
Bhrugu mehta
On Dec 3, 2007 12:41 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Monday 03 December 2007 00:48:55 Bhrugu Mehta wrote:
I want to connect asterisk with oracle database.
You'll need to install the Oracle ODBC driver for Linux. One word of warning,
2007/11/30, Olivier [EMAIL PROTECTED]:
Hi,
To make a long story short, I can't install any TAPI driver on my XP
platform.
A. Within Config Panel|Modems and Telephony options|Advanced parameters,
I've got a list of 7 TAPI drivers. Among them is Omniis TAPI driver for
Asterisk.
B. I can
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