Hello All,
We have ben working with AGI and AMI for
some time now. But now we need to implement voice recognition to our AGI based
apps (IVRs basically). We are currently looking at SPHINX but having
difficulties to integrate it with Asterisk.
Is there anyone having any experience in
Hi List,
I'm planning to setup and put in production a server with an [EMAIL PROTECTED]
2.8 edition and I would appreciate if someone that has done it
shares/provides some information in regards to the following questions I
have,
1. Since one of the most attractive features of the @home
Joshua Colp a écrit :
You can use the SIPAddHeader application:
SIPAddHeader(Header: Content)
Adds a header to a SIP call placed with DIAL.
Remember to user the X-header if you are adding non-standard SIP
headers, like X-Asterisk-Accountcode:. Use this with care.
Adding the wrong headers may
Hello friends,
I am trying to install asterisk. I downloaded the latest development branch
from digium thru svn. I get an error in the make which says:-
[LD] codec_gsm.o gsm/lib/libgsm.a - codec_gsm.so
[CC] codec_ilbc.c - codec_ilbc.o
make[2]: Entering directory
Why not just use FTP ? It works great plus you get a log of what happens.
- Original Message -
From: VaibhaV Sharma [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, August 01, 2006 6:52 PM
Subject:
Hi,
what about upgrading qmake?
see http://lists.digium.com/pipermail/asterisk-dev/2006-July/021599.html
Just guessing...
__
Od: [EMAIL PROTECTED]
Komu: asterisk-users@lists.digium.com
Datum: 04.08.2006 10:03
Předmět:
Hi!
I have two ISDN cards for my asterisk server, an AVM B1 (active
card) and a HFC.
I want to use the HFC card in NT mode, and the AVM B1 in TE.
Afair bristuff and vISDN doesn't support the AVM B1, so mISDN
should be my choice?
--
Marco Dieckhoff
GPG Key 0x1A6C95BA --
I am trying to do a simple agi connection to db with the guidance from http://www.voip-info.org/wiki/view/Asterisk+AGI+php Item 13 with ani.agi file, db and
extensions.conf13. another sample, ANI
Scenario - did callers call the Asterisks box and land on
the context did, Asterisks answers the
Marco Dieckhoff wrote:
I have two ISDN cards for my asterisk server, an AVM B1 (active
card) and a HFC.
I want to use the HFC card in NT mode, and the AVM B1 in TE.
Afair bristuff and vISDN doesn't support the AVM B1, so mISDN
should be my choice?
I ran a AVM C4 and HFC together for quite
Switch your echo canceller to MG2. Look for zconfig.h in zaptel and
recompile. Also experiment with echocanel=256 and adjust rxgain/txgain.
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Thanks for the info...
I was able to make if work after recompilation and
relocation of library files.
Regards,
Leonimar
--- Steve Underwood [EMAIL PROTECTED] wrote:
leonimar cape wrote:
Hi Steve,
I need to enable the Unicall channel in my asterisk
box to be able to interconnect to a
Hi Friends, We have installed "Asterisk" in our office and using it successfully. I have given public IP to our Asterisk server. We are using Snom360 5.3 softphone for communication. I tried to connect to our Asterisk server with my Snom360 5.3 softphone from my house. But, it is not
Hi.
At the moment we record our meetme conference to wav format, after which
we have a script in place converting the wav to mp3 format.
I need it to be encoded with MPEG1-layer3, so that I can play it back
via our application using the Java Media Framework, but the wave files
are encoded
Ok. I have an update! When all the problems begin (described below) the 'show
queues' command doesn't work either!! The queues have dissappeared (or asterisk
is unable to read them?)!
What the heck is going on? Why are the queues gone by themselves? When I
restart they're back.
Queues.conf in
Title: RE: [asterisk-users] Re: DUNDi with SIP
Easy, you change the fromuser=dundisip2 line to
fromuser=dundisip1. You're just matching the destination peer (dunsip2,
labpbx2.ipt.oneeighty.com) and sending the configured
fromuser.
- Brad
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
hI
any idea how to loadbalance IAX2 trafic to multiple
asteirsk
thanks
kAMRAN
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Hi Joseph,
I think you didn't made it clear. My understanding of the script you want, is
to make the process
of calling the bank's IVR system and responding to its prompts/recording via
Asterisk scripts. If
my understanding is correct, the asterisk send the DTMF needed by the prompt.
But then
Hi,
Kamran Ahmad wrote:
any idea how to loadbalance IAX2 trafic to multiple
asteirsk
Use app_random:
exten = _X.,2,Random(50:6)
exten = _X.,3,Dial(IAX2/server01/${EXTEN})
exten = _X.,4,Dial(IAX2/server02/${EXTEN})
exten = _X.,5,Goto(8)
exten = _X.,6,Dial(IAX2/server02/${EXTEN})
exten =
Open Firewallports in Office: 5060, 1-2
Edit sip.conf:
;nat
externip=xxx.xxx.xxx.xxx
externrefresh=10
localnet=192.168.0.0/255.255.0.0 ; fit to your net
nat=no
[19]
type=friend
username=19
secret=
;canreinvite=no
host=dynamic
disallow=all
;allow=g723
allow=alaw
allow=ulaw
I'm using iax2 interconnection between two asterisk (1.2.9.1) boxes,
also using jitterbuffer, because this is over cdma, codec ilbc
jitterbuffer is working fine, except some gaps in speech
I experimented with resyncthreshold, buffer size but no improvement :-(
is there any way, to force
In article [EMAIL PROTECTED],
Jan du Toit [EMAIL PROTECTED] wrote:
Hi.
At the moment we record our meetme conference to wav format, after which
we have a script in place converting the wav to mp3 format.
I need it to be encoded with MPEG1-layer3, so that I can play it back
via our
Allright. I think I've located the problem. It's reported here:
http://bugs.digium.com/view.php?id=7604
I'm not however using 'show queues'. It stops responding anyway. Maybe because
we use freepbx and flash operator panel.
Regards,
Jan
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED]
Hi,I also have Nokia N70. After seeing your mail, I also tried to configure my N70 to connect to my Asterisk. But, I am unable to find the "SIP settings" option in my mobile (Tools-Settings-Connection-SIP settings). What I have to do? Looking forward to your response. Thank
If you have zap groups defines then instead of
exten = 299,1,Macro(dialout,5,1914426,,)you would use
exten = 299,1,Macro(dialout,g0,1914426,,)
g0 or gwhatever is the group number for a list of
zap trunks.
Look at group in zapata.conf.
-- -- Steven
http://www.glimasoutheast.org
My concern there is; Does repetitively cating those devices have any negative
affect on voice, stability, etc?
--
--
Steven
http://www.glimasoutheast.org
Eric ManxPower Wieling [EMAIL PROTECTED] wrote in message news:[EMAIL
PROTECTED]
Bart Fisher wrote:
I'm trying to detect when a T1
I could be mistaken, but I thought that you had to have the text to speech
engine record a temp file first, then play that file back
durring the call.
???
--
--
Steven
http://www.glimasoutheast.org
Joey McDonald [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Greetings,
I've
Been awhile but IF memory serves...
Manually enter the boot server IP on the phone. I do not think causes
a reboot - of course this was several versions back in sofware.
Then edit a contact and press save. Every time it updates the list on
the phone, it tries to copy to the boot server.
I am using sendtext() across 2 machines that connect with IAX.
I am sending a text message to the phone.
My problem is that I do not answer the text page but after ringing 5
times the IAX connection
says it was answered.
When I generate this page local to server it works correctly. It does
I've found that when I use an IP VoIP device (phone or ATA) to the
SPA-3000/3102 (I have both) that I have the echo problems. While
I've learned to deal with it my wife won't. I have no such problem
with the POTS phone directly connected to the SPA. I have one
SPA - ATA - VoIP service and an SPA
more info, iax stats:
asterisk*CLI iax2 show netstats
LOCAL -
REMOTE
ChannelRTT Jit Del Lost % Drop OOO Kpkts
Jit Del Lost % Drop OOO Kpkts
IAX2/honzat-14
Title: RE: [asterisk-users] Re: DUNDi with SIP
Didn't
work Brad. I changed fromuser to dundisip1 from dundisip2. The first Asterisk
box sends dundisip1.
[dundisip1]
- Original Message -
From: [EMAIL PROTECTED]
To:
asterisk-users@lists.digium.com
Sent: Fri, 04 Aug 2006 05:01:11
-0300
Subject: [asterisk-users] asterisk dosenot compile
Hello friends,
I am trying to install asterisk. I downloaded the latest development
branch from digium thru
Works fine for me.
Douglas Garstang wrote:
Didn't work Brad. I changed fromuser to dundisip1 from dundisip2. The
first Asterisk box sends dundisip1.
[dundisip1]
type=friend
secret=password
insecure=very
context=global_dundi_local
host=labpbx1.ipt.oneeighty.com
qualify=no
username=dundisip1
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
[EMAIL PROTECTED] wrote:
What do you want for the TE410P
Give me a call!
or e-mail me
This is not the business list. In fact you will notice it states:
Asterisk Users Mailing List - Non-Commercial Discussion
Please stop spamming the lists.
At
I have a tellabs 2572 echo can wired and setup according to the
instructions in
http://www.voip-info.org/tiki-index.php?page=Tellabs+Hardware+Echo+Cancellers
. I added the echo can between my zaptel card and my adit 600 channel
bank. I used the straight thru and crossover cables as
Hi List,
I am not sure what this issue is. I am having a
problem where I have 2 phones that are behind NAT on the same internet
connection. The asterisk server has a public IP. Using asterisk real
time1.2.10 on CentOS 4.3 with Ztdummy. For some reason I can only get ahold of
one of the
Steve, I don't see how it could possibly work. The second system sees a
connection from dundisip1 on labpbx2, so it then looks at the dundisip2 entry
because it's matching against the host. It then looks at the username, but it's
dundisip2... which doesn't match dundisip1, and it fails the
I have basic code that see's what comes in but
nothing much more. I was going to make it that when calls come in VB looks what
comes up etc. and work from there. If you want it let me know.
- Original Message -
From:
hernany.ce
To: 'Asterisk Users Mailing List -
I'm not sure, if qualify is even supported in realtime configuration...
do you have rtpcachefriends=yes in sip.conf?
PJ
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I was wondering if other A200 users experience the same problem I have.
If someone accidentaly pulls one of the FXO cables, the driver does not
detect that and Asterisk still tries to send calls via that port. On the
other hand Digium cards immediately sense the cable disconnect and calls
are
- Original Message -
From: Steve Davies [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, July 31, 2006 6:01 AM
Subject: Re: [asterisk-users] SNOM 360
On 7/31/06, Koopmann, Jan-Peter [EMAIL PROTECTED] wrote:
- Original Message -
From: Steve Davies [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, July 31, 2006 5:57 AM
Subject: Re: [asterisk-users] SNOM 360
On 7/28/06, Dovid Bender [EMAIL PROTECTED] wrote:
Also
In the current system I'm running, qualify IS supported for realtime
peers/users, I'm using it right now.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pavel Jezek
Sent: Friday, August 04, 2006 11:15 AM
To: Asterisk Users Mailing List - Non-Commercial
Funny you mention it. The problem that i was having with the trunks was
that i put every channel in it's own group.
In amp, all the trunks where identified by g0,g1,g2,g3. We changed
that to 0,1,2,3 yesterday. Doing that it allows me to dial out on any
one trunk that i want, where before i
Not sure. I know once I register on a PC I get a file on my PC that tells it
where to go ( too tired to think of it now).
- Original Message -
From: Matt Riddell (NZ) [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent:
On 8/4/06, Dovid Bender [EMAIL PROTECTED] wrote:
The snom does seem to manage its two local ports properly though but
this cannot be hard. Worst case is that the snom needs about 128Kb/s -
Not hard on a 100Mb/s full duplex connection :)
Dovid - Have you identified where the bottleneck is
Also I forgot to put this in. I keep getting the
following:
Also 310 is plugged in to the same router in the
same location. Cant figure out why 10310 works great but 10306 wont
work.
NOTICE[9513]: chan_sip.c:11564 sip_poke_noanswer:
Peer '10306' is now UNREACHABLE! Last qualify:
0
Are you trying to set that when ever some one calls a specific DID then it
automaticly rings on an external number ?
You can do this
exten = 1234,1,Dial(SIP/[EMAIL PROTECTED])
Have sip.conf forward the call to exten 1234. So when anyone calls it they
get sent to your home phone.
-
It's ALIVEThanks Russell, that's been bugging for a couple of days now. Flite is pretty wicked, you don't' have to preprocess a text file now, you simply give it the text to speak and you're done. I first read about it here:
http://nerdvittles.com/?p=134 --joeyOn 8/3/06, Russell Bryant
[EMAIL
- Original Message -
From: Pavel Jezek [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, August 04, 2006 11:14 AM
Subject: Re: [asterisk-users] SIP/Qualify
I'm not sure, if qualify is even supported in
Paul Zimm wrote:
I get a Yellow alarm on my zaptel card (digium x100p)
I'm guessing you meant t100p?
on the echo can option 20 is 5(ESF), 60 is 1(B8ZS), and 63 is 0
You'll also need to change the c.00 (Chan 1-24) to match your
signaling. For me, channels 1-16 (FXO LS) need those
This bothers me a bit too, as I've certainly noticed this. But then, the typical phone system shouldn't be in a position where cables are accidentially unplugged very often. I've found Sangoma to be very open to suggestions though... Maybe if we bug them about it enough, they'll include it in the
On 8/4/06, Dovid Bender [EMAIL PROTECTED] wrote:
The Qos field is part of the 802.11q header, so is only available if
a VLAN has been configured.
VLAN 0 is a perfectly valid VLAN, and will cause an 802.11q packet
header to be added to all the phone traffic. This will then only work
if the
I have a DS3/T3 that was dropped into my telco closet as two coaxial
cables. A send and receive. I needed to extend it so I went to Radio
Shack to buy some barrel connectors. They did not have any but they did
have T-Connectors so I bought a couple of those.
Everything was fine up until I
Are they both being NAT'd to the same external IP?
Dovid Bender wrote:
Hi List,
I am not sure what this issue is. I am having a problem where I have 2
phones that are behind NAT on the same internet connection. The
asterisk server has a public IP. Using asterisk real time1.2.10 on
CentOS
Hi all,
I´m begginner with asterisk and i need to setup one Support Call Center.
First of all,
I want to authenticate my users in call center with AgentCallBackLogin
or something similar and the tranfer the Logged Agent to main queue.
I play with some setups from
Well, after 24 hours I'm still trying to get SIP trunking to work. I'm going in
circles and it's driving me nuts. Here's my latest sip.conf, that I have on
both systems.
system 1 sends dundisip1_in as the userid to system 2. System 2 asks for proxy
auth. System 1 sends an ACK and doesn't do
A couple of questions arising from using Jabber in svn trunk. If anyone
has any answers it would be gratefully received.
We are using Wildfire 3.0.0 (upgrading to 3.0.1 this weekend)
1) What are the pros / cons of using SASL and TLS in jabber.conf ?
We are currentl using usetls=yes and
this asks only for a password
exten = 123,1,AgentCallbackLogin(${CALLERIDNUM},${EXTEN})
hope it helps...
Guido
-Ursprüngliche Nachricht-
Von: Gleidson Antonio Henriques [mailto:[EMAIL PROTECTED]
Gesendet: Samstag, 8. April 2006 18:38
An: Asterisk-Digium
Betreff: [asterisk-users]
If you see no errors on your MX2800 for the ds3 then they are
probably not the issue.
What does the MX2800 show for T1 which do not work? If you loop
toward * does the card see itself? Loop toward GX do they see?
On Aug 4, 2006, at 11:15 AM, Steve Totaro wrote:
I have a DS3/T3 that was
Hi,
I have been fighting with this for days now. I can't seem to get festival to work with asterisk. I have followed the wiki on setting up asterisk to work with festival but no worky. I can run l the festival client and that works just fine but when I try to do it from asterisk I get the
Hi,
I've read all over that the manager conection (via sockets) isn't good
for high traffic applications with multiple manager connections at the
same time with one asterisk, the connection hangs and many other problems.
having said that, my question is:
Has anyone worked on a fairly
I am trying to record a queue conversation using mixmonitor (or
monitor). I am using dynamic queue members (not agents) so I cannot use
the agent recording facility.
Basically, the call recording stops when a local channel is used and the
user then transfers the call.
I have constructed a
I am not totally up to speed on the MX2800 but I have gone to loopback
tests and loop T1 25-28 and selected every possible selection while
watching pri debug span 1 on the console, no output at all.
Jerry Jones wrote:
If you see no errors on your MX2800 for the ds3 then they are probably
not
Dovid Bender wrote:
- Original Message - From: Steve Davies [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, July 31, 2006 6:01 AM
Subject: Re: [asterisk-users] SNOM 360
On 7/31/06, Koopmann, Jan-Peter
Dammit - meant to say that I'm using svn trunk (SVN-trunk-r38548)
Sorry!.
Julian Lyndon-Smith wrote:
I am trying to record a queue conversation using mixmonitor (or
monitor). I am using dynamic queue members (not agents) so I cannot use
the agent recording facility.
Basically, the call
The MUX will give you stats for every line running through it, DS1,
DS2, and DS3. Start there. What errors is it reporting?
Actually I did that backwards, start with DS3 then do DS1, if 3 has
issues so will everything else.
On Aug 4, 2006, at 1:04 PM, Steve Totaro wrote:
I am not totally
Can anyone help point me in the right direction? I have calls coming
into Asterisk over a PRI, all going to the same #. I need to have
asterisk route the calls to a different location based on the NPANXX
of the callerId for the inbound call. Something like
exten = 123,1,$newnumber =
note: I switched to my gmail accountI put a loopback connector on each side and the AIS indicator goes green for that interface. So the card is wired properlyOn 8/4/06, Doug Lytle
[EMAIL PROTECTED] wrote:
Paul Zimm wrote: I get a Yellow alarm on my zaptel card (digium x100p)I'm guessing you
On 8/4/06, Steve Totaro [EMAIL PROTECTED] wrote:
I am not totally up to speed on the MX2800 but I have gone to loopback
tests and loop T1 25-28 and selected every possible selection while
watching pri debug span 1 on the console, no output at all.
Jerry Jones wrote:
If you see no errors on
You can do:
exten = 123,1,GotoIf($[${CALLERID(num):0:6}=123456]?50);if cidnum is
123456 goto 50
Or you can do this:
exten = 123/_123456.,1,Goto(50);if cidnum is 123456 goto 50
exten = 123/_234567.,1,Goto(51);if cidnum is 234567 goto 51
exten = 123/,1,Goto(70);anything else or blank goto 70
exten
marvin horst wrote:
note: I switched to my gmail account
I put a loopback connector on each side and the AIS indicator goes
green for that interface. So the card is wired properly
I'm guessing that this channel bank has been functional and you now want
to add E.C?
What version of the
Where can you be
found?
Ferguson
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1.0.9What version of the software?
After 5pm EST,I can pull my test setup and verify setttings.
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Neil Cherry wrote:
I've found that when I use an IP VoIP device (phone or ATA) to the
SPA-3000/3102 (I have both) that I have the echo problems. While
I've learned to deal with it my wife won't. I have no such problem
with the POTS phone directly connected to the SPA. I have one
SPA - ATA - VoIP
astGUIclient-VICIDIAL handles volumes like that over a few manager
connections. We have done that for 3 years across multiple versions of
Asterisk and it is very stable.
The way we designed it, we have a listening daemon, a sending daemon
and a status daemon. The sending script spawns children
Hi all,
I want to call a script and have it run in the background while my
dialplan continues execution. I've tried forking my perl script but Asterisk
still halts until the script returns. Is there any way to call AGI and have the
script run in the background?
Thanks, Marc.
BJ Weschke wrote:
On 8/4/06, Steve Totaro [EMAIL PROTECTED] wrote:
I am not totally up to speed on the MX2800 but I have gone to loopback
tests and loop T1 25-28 and selected every possible selection while
watching pri debug span 1 on the console, no output at all.
Jerry Jones wrote:
If you
marvin horst wrote:
1.0.9
Marvin,
Are you using the channel bank to bring inbound analog lines to your
Asterisk setup, or are you using it to supply dial tone to your facility
for things like fax or modems?
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to
I recently had a similar problem. I had a slow database lookup, and
wanted a recording (actually, a SayDigits) to be played while that was
happening.
My solution was to execute the say digits from within the AGI script,
and not wait for the result code. The AGI script can then do the database
Ferguson, Michael wrote:
Where can you be found?
Ferguson
I am here lost in the masses.
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Hi Again,
I thought I would repost in case some people missed it. I am currently set up with asterisk 1.2.10 on Mandriva 2006 using a Mediatrix 1204 as our media gateway. Everything works well during the day. However something occursduring the evening when the system is not in use that causes
On 8/4/06, Steve Totaro [EMAIL PROTECTED] wrote:
BJ Weschke wrote:
On 8/4/06, Steve Totaro [EMAIL PROTECTED] wrote:
I am not totally up to speed on the MX2800 but I have gone to loopback
tests and loop T1 25-28 and selected every possible selection while
watching pri debug span 1 on the
Doug Lytle wrote:
Are you using the channel bank to bring inbound analog lines to your
Asterisk setup, or are you using it to supply dial tone to your
facility for things like fax or modems?
I guess this doesn't really make any difference.
As long as one side is a T1 cross over and the
I have two Asterisk Boxes - one for VM and the second
being used as a PSTN GW. I have users calling in on
the PSTN GW. When they call-in, I launch DISA and they
hear a dial-tone. They then dial a number (999)which
basically forwards them to the VM Server. The VM
server prompts them for mailbox#
Are you using the channel bank to bring inbound analog lines to your
Asterisk setup, or are you using it to supply dial tone to your facilityfor things likefax or modems?Yes, I'm using the channel bank for inbound analog lines
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BJ Weschke wrote:
It's a fairly common issue, and unfortunately, there isn't a best
practice solution that I've seen people use that isn't ugly. In a
prior job they zip tied the cables down to the connectors and this
fairly reliable.
At least on my MX2800s there is a loop for a zip tie on the
You can take a regular straight though cable and plug it into the greenand the other end into the red.You should get a yellow AIS light on
send-in and receive-in.If you don't, then your jacks are probably theissue.I get a yellow AIS light on both send-in and receive-in.
Mine is:T1 cross over from
Hi, this is my first post, so go easy on me !
Sorry if this has been covered before, I could not find an answer that
helped me.
I am trying to achieve the following :
Telco ISDN30e PRI - Asterisk with TE210P - Siemens HiPath PBX
The siemens is a legacy PBX and I am not 100% of the modules etc
probably need a crossed t1 cable
1-4
2-5
On Aug 4, 2006, at 4:20 PM, James Arscott wrote:
Hi, this is my first post, so go easy on me !
Sorry if this has been covered before, I could not find an answer that
helped me.
I am trying to achieve the following :
Telco ISDN30e PRI - Asterisk
marvin horst wrote:
I know it's wired properly because you can set mode LPb to 3 (metallic
bypass) which bypasses any processing on the echo card. After doing
this T100P was communicating with channel bank as before.
Bad card?
On the red Rcv In, have you tried swapping out the cable for
Title: Re: [asterisk-users] Asterisk and Siemens Legacy PBX
Hi, thanks I will try this tomorrow morning when the legacy PBX can be taken offline for a few hours, any suggestions on specific asterisk configuration options that I may have missed to achieve this ? I am hoping its just the cable I
Is there anyway to access the settings in asterisk.conf from the dialplan?
Something like a global ${ASTVARLIBDIR} or something like that?
~ John Williams
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Hello there,
For some reason, I haven't been able to play music on hold from within my
PHP script... Everything else seems to be fine: my script records and plays
files without any problems. I also tested MOH from my dialplan and it works
perfectly. And I can see from the console that MOH
Yes.
- Original Message -
From: Peder @ NetworkOblivion [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, August 04, 2006 12:30 PM
Subject: Re: [asterisk-users] SIP/Qualify
Are they both being NAT'd to the
Has anyone had any success getting vlan tagging to work on Aastra phones (480i or 9133i)? I enabled the tagging feature and entered my voice and data vlan IDs in the correct fields. The phone is able to communicate on its vlan but the PC connected to the back is not. I'm fairly confident its not a
Redhat 9
Asterisk - 1.2.7
TDM 400 - 1 FXO, 2 FXS
I'm using a standard residential PSTN line on my ZAP channel and
curious whether I can override the caller ID my telco has for me with
one of my choosing.
I've tried this:
exten = s-ZAP,n,Set(CALLERID(all)=My Name 999-999-999)
exten =
hugolivude wrote:
Redhat 9
Asterisk - 1.2.7
TDM 400 - 1 FXO, 2 FXS
I'm using a standard residential PSTN line on my ZAP channel and
curious whether I can override the caller ID my telco has for me with
one of my choosing.
I've tried this:
exten = s-ZAP,n,Set(CALLERID(all)=My Name 999-999-999)
The Tellabs cards I used were not configured for ESF/B8ZS when I got
them. If you have the Tellabs chassis, try connecting with a serial
connection.
Here's a copy of the manual: http://www.fnords.org/~eric/tellabs/ It's
in PDF format in 2 parts.
marvin horst wrote:
You can take a
On Fri, 2006-08-04 at 16:17 -0600, John Williams wrote:
Is there anyway to access the settings in asterisk.conf from the dialplan?
Something like a global ${ASTVARLIBDIR} or something like that?
No, there is not a direct way to do that right now. However, if you're
familiar with a scripting
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