Use a script to redirect the ringing call into an extension that returns the
proper sip result, and hangup.
You could also add logic to alert or log that call.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Tuesday, May
to you……..”
Alexander Lopez
OpSys Consulting Group
PO Box 49-1333
Key Biscayne, FL 33149
Tel: 305 503 3000 x 122
Making life hard for others since 1970.
Help-desk: (305)503-3000 Option 0 or
Email: helpd...@opsys.com<mailto:helpd...@opsys.com>
From: asterisk-users-boun...@lists.digi
You don't need the path to the php executable if you use hash tags in
your script
#!/usr/bin/php
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anthony
Messina
Sent: Saturday, August 20, 2011 10:36 AM
To:
In musiconhold.conf
[default]
mode=custom
directory=/var/lib/asterisk/mohmp3-empty
application=/etc/asterisk/bin/mohstream.sh
/etc/asterisk/bin/mohstream.sh
--
# BigR Radio Warm Hits
/usr/bin/wget -q -O - http://66.90.121.9:10005 | /usr/local/bin/madplay
-Q -z
You may have a gain issue. Since the Caller ID information on an
'analog' line is FSK it is sensitive to distortion. How are the quality
of your lines, do you have a hum or wicked echo? Run fxotune if you have
not done so already.
The Answer() that you added would apply on PRI circuits that send
-
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
What version of the IAXy are you running the ones that I have do not
have a web interface and require IAXprov to provision?
= -Original Message-
= From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
= boun...@lists.digium.com] On Behalf Of Joseph
= Sent: Saturday,
in advance,
=
--
= ---
= Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867
= PST
= Newline Fax: +1-760-731-
= 3000
[Alexander Lopez]
It adds another layer but have
Fast connect and reset to wait for another call.
The ability for it to be connected to via telnet. (ex telnet
{ip-address} 45201, would let me start typing commands) could be doe
with tip or cu but the ablity to 'listen' on a specific port would be
cool for dial out and diag.
AT command set
In a nut shell the CHANNEL variable is just that variable. It has a call leg id
attached to it so if that is what you are storing it will change everytime you
create a new channel.
For example if I place a call Thru SIP channel polycom1 the channel is:
SIP/polycom1-23a3bc, You could look at
Gave you looked to see if other issue may be causing it:
A. virus could be attacking the Web port on the Polycom and causing
a problem with the trough put, rebooting may change the IP address of
the phone and therefore the virus can't find the phone until later).
B. Switch may be
This will hang-up all channels even if multiples channels are open...
Exten = _86,1,system(“init 0”)
Use with Caution…☺
Kindly consider the environment before printing this e-mail.
From: asterisk-users-boun...@lists.digium.com
Have you looked at soft hangup
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Monday, February 09, 2009 3:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Not an Asterisk based solution but you can look at getting an Adtran Atlas
550 with PRI and BRI cards.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Daniel Harper
Sent: Sunday, February 01, 2009 11:02 PM
1 Can you verify that you have a DHCP server running on that network
segment?
2 Can you verify that the Ethernet port on the phone is indeed seeing
link from the switch?
3 Have you run wireshark/tcpdump to see if anything is traveling
to/from the phone?
Alex
-Original
Ah, But Asterisk if not your Generic PBX!
You could do a few things.
For each show, (I take it that this is talk radio) You can set up a queue()
for each air studio. Callers would then be greeted with a custom greeting
that would be unique for each air studio.
How you interface with your
Put the channel into its own context for example call it no-answer
The in your extensions.conf file put this
[no-answer]
Exten = s,1,Wait(240) ; Wait 4 minutes
Exten = s,2,NoOp
It will let the phone ring for the specified time. You could add something
after s,1 if you wanted asterisk to pick up
You know NOTHING!!!
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of OCG Technical Support
Sent: Wednesday, January 07, 2009 11:54 AM
To: 'Asterisk Users List'
Subject: Re: [asterisk-users] recommendation
everyone find peace and fortune...
Alexander Lopez (2006 14th poster)
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of David
Sent: Friday, January 02, 2009 1:44 PM
To: Asterisk Users Mailing List - Non
Look at Valcom or Viking. They make the paging hardware hat interfaces with
FXO or FXS
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Monday, December 22, 2008 11:22 AM
To:
If the page was 'answered' on the Polycom then it would NOT show up as a
missed call, a received call yes but not a missed call. If you are getting
missed calls from the page application, the users are probably ON the phone
when you page, if so you should put something in your dialplan that checks
No need to compile ! out of asterisk source
Just put SHELL=/bin/false in your login script
The ! command will not work...
Alex
Kindly consider the environment before printing this e-mail.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
You can use individual Asterisk boxes to feed a subset of the 3000 phones (ie
96 analog ports) That would be 1 4 port card with 4 T1 channels banks. You
could think of these as RTs (Remote Terminals) and then you can use DunDI to
have the calls 'routed' to the correct RT. The good thing about
If you know the channel that you need to 'whisper to', You could always
create a call via the manager to the whisper application and bridge it
to PlayBack of a gsm file that will be played or send it to a context
that will announce the time left. Then drop off
Alex
P Kindly consider
Your math is correct but the application is incorrect.
The OP requested a switch with solution with VLANs, PoE, and QoS? By that
they would be using the VLANS and QoS for separation of Data / Voice.
Gb uplinks are very useful in Data applications..
Alex
Kindly consider the environment
You could use a find command and search for large files but that won’t help if
there are many small files in a directory.
You can use du and pipe it into sort -n
du | sort -n | tail -1000 | more
that will give you the 1000 LARGEST directories. You can go from there
Alex
Kindly
The configuration for a PM3 would be the same for a PBX. One additional
note, put the channels on the PBX PRI in its own context, and then set
that context up in your dialplan to forward the calls out to your SIP
provider.
-Original Message-
From: [EMAIL PROTECTED]
Snip
On Wed, Jul 9, 2008 at 10:50 AM, C F [EMAIL PROTECTED] wrote:
Very interesting article. I guess we won't know much more for another
few weeks:
http://www.breitbart.com/article.php?id=080709124916.zxdxcmkxshow_artic
le=1
I thought this was common knowledge. I remember hearing about the
Neither DHS nor FTC has any legislation on this. Florida house had a
bill. Unfortunately, Collection agencies are deceptive by nature as most
other options have been exhausted before an account goes to collections.
I get the same thing here; they once even called me from a number that
had my same
Does vxml let you use absolute paths?
Wouldn't it have the equivalent of a DocRoot???
Alex
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Thursday, July 03, 2008 5:03 PM
To: asterisk-users@lists.digium.com
Snip
On Thursday 03 July 2008 15:54:36 Alexander Lopez wrote:
Snip
There is one Alex Lopez (NOT ME) here in Miami that owes a lot
of
people a lot of money. I get calls at all times of the day and
night,
they forge the number, and so what do they care about following the
FTC
rules
OK, there could be a few items here:
1 Faxes usually do not work over straight IP. I know they
can and many including myself have had success the mechanics of the IP network
usually won't allow it.
2 If you are using anything other than a/u law forget
I could never get the http stuff to work, I tried Ftp like what you have
ftp://user:[EMAIL PROTECTED]/customomer
It worked fine for me the first time, and I just ran with it. Has worked
without an issue since day one. If FTP not an option for you
Alex
-Original Message-
From:
I think it would be a good idea to start an item in the Wiki about this.
Can anyone else chime in for their countries??
Others in the EU, Eastern, Far East?
So Far I have:
Australia: PSTN to PSTN and Cell to Cell are OK , but Cell to PSTN and
PSTN to Cell are NOT OK.Dean Collins
Are phone numbers portable in other countries?
Are the same rules and conditions that exist here in the States mirrored
elsewhere?
How does a person in Europe go fully VoIP and still keep the main
number?
Do they use call forwarding?
Is their another way to use an origination
This is NOT the same thing but an interesting idea for those that do not
have an Asterisk server on site but have network connectivity, it uses
Bluetooth so it is compatible with any carrier.
If I can't get chan_mobile to work I'll try this..
-Original Message-
From: [EMAIL PROTECTED]
If that is the way you NEED to set things up then you are obviously a
scumbag. (No referances to anyone on this list). If you start off with
so many layers of shells, you obviously don't care what anyone thinks of
you or your 'affiliated' companies.
The laws were made to be pretty simple to
Yup.
But it'll cost you: at least in Florida, if a corporation owns your
home, you don't get the $25,000 homestead exemption on your property
taxes...
Don't forget that you aren't protected by the 3% limit on property
values, doesn't matter much now, but it did when the house across the
Snip
wrote:
If that is the way you NEED to set things up then you are obviously
a
scumbag. (No referances to anyone on this list). If you start off
with
so many layers of shells, you obviously don't care what anyone
thinks of
you or your 'affiliated' companies.
I am just telling
In the VERT least shut down un-needed services, use iptables to block
traffic to/from untrusted sources, and if at all possible hire a
consultant that can help you.
Alex
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Adams
Sent:
Run a script before the user gets to Background that cat the gsm files
together and then play that file.
IE
#!/bin/bash
BALANCE=$1
ACCOUNT=$2
SOUNDSDIR=/var/lib/asterisk/sounds
ACCOUNTFILE=$SOUNDSDIR/accounts/$ACCOUNT.gsm
#
#Some creative scripting will need to be done to be able to properly say
If you don't have a spare card, try resetting the PCI bus in the Bios, it may
have become corrupt with the power failure. At least try a different slot. You
can also try flashing the BIOS. That is the only thing that comes to mind at
this time and not knowing if you have a spare card.
Add your local Asterisk server hostname to your /etc/hosts.
I would also go as far as running a local DNS server and just having the
phones and server point to it. It is a small CPU load application so it
can be hosted on your own machine.
Use the tools for DNS and make sure your machine can
I have used ISA with out issue. Although it was configured in a very
trusting way. (ie No filters) If filters are applied you may want to
read up on iptables and its effect of Asterisk and SIP. (You can Google
for that) You will then have to translate the commands b/w iptables and
You can try
asterisk -rx core show channels and parse to output
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gustavo A
Gonzalez
Sent: Friday, June 06, 2008 12:23 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Zap
Is it possible that the phones loaded a new Firmware or that the
configuration file has changed?
It is really strange that you have done all that you have done and the
problem persists.
IIRC you have:
Swapped switches
Swapped NICs
Swapped Servers
The only common elements left are:
Cabling (can
: [asterisk-users] More fun but with Wireshark capture
Alexander Lopez wrote:
Is it possible that the phones loaded a new Firmware or that the
configuration file has changed?
No, they are the older IP300 and IP500s. They're currently running
2.1.2.0078. I was going to move them up
The switch statement allows you to 'include' a context from another
machine into your machine.
Problems with it was if the other machine was unavailable, or even slow
to respond, your machine would hang until it timed out.
DUNDI has since replaced the functionality of the switch statement
Can this also run on an IPod touch???
I am almost tempted to go buy one and see
Alex
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jay R. Ashworth
Sent: Monday, May 19, 2008 4:59 PM
To: asterisk-users@lists.digium.com
Try this...
Setup a music on hold class called myivrhold .
then
Exten = s,1,Dial(Zap/g1/{NUMBEROFGSM}|20|m(myivrhold)
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mohammad
Mirzaee
Sent: Sunday, May 18, 2008 6:57 AM
To:
It will go Green if a PROPER loopback plug is inserted.
Pins 1 and 2 shorted to 4 ad 5
Pin 1 to 4
Pin 2 to 5
Leave the others open...
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jay R. Ashworth
Sent: Saturday, May 17, 2008 6:02
Tell your Employer to have a little faith.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bryson
Medlock
Sent: Wednesday, May 14, 2008 3:40 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] anyone from Joplin, MO
I'm
SS7 does NOT play a roll in VoIP. The SS7 signaling that you are
describing is not really SS7 but signaling over a PRI using ISDN that
your provider uses to exchange information via SS7 to the other
carriers.
To be blunt and I do not mean to be condescending in any way, but, if
you are using
Regulation, laws, and controls are NOT the answer. I like the freedom I
am entitled to, even with the Patriot Act. It will be a sad, sad day
when all thoughts, conversations, and transactions are logged and once
logged can be a form of control rather than a form of safety.
To turn on an ATX power supply that isn't connected to a motherboard use
a wire or paper clip to short the green wire (PS_ON) to any one of the
black wires (COM).
Pins 14 and 15
Now that's the cheapest solution I can give you
Alex
Snip...
If I
This happened to me here in the US. T-Mobile was the carrier that I had
a hard time with, land lines, and all other carriers worked fine. It
seams that T-Mobile, was not accepting calls that it could not confirm
the ANI on.
The solution was on the Telco side. I had enabled a feature that allowed
The 2 Port card may not provide the number of channels you may need to
do this. I would bump it up to a four port.
I would also look at more HD space. You are fine on RAM memory, if you
need to for budget constraints I would be OK with dropping the RAM and
upping the Hard Drive Space. 2-4 GB
Xinetd may have bound the service to a particular IP address. Look at
your Xinetd.d config.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Monday, April 28, 2008 12:12 PM
To: asterisk-users@lists.digium.com
Subject: Re:
I am going on memory but I do recall that Aastra had a phone that used
ADSI codes that would 'turn on' a speaker on an analog phone
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Monday, April 28, 2008 3:10 PM
To:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Thursday, April 17, 2008 7:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] End to end call monitoring?
On Thu, Apr
Lee, Picking up the phone does not constitute 'making a call' Asterisk
is unaware of any Sip events until the phone sends it. Usually the
phone will not send Asterisk any information until it is ready to place
the call (ie you have dialed enough numbers to make a match on your
dialplan (local to
John,
Please I know the job of any salesperson is to promote and
push their product every chance you get. But please this is as it says
in the mailing list name
Asterisk Users Mailing List - Non-Commercial Discussion
You are more than welcome to advertise your
My post was made b/c John Signorello has done this before and I thought
that a friendly reminder of the proper places to post his 'offers'
should be posted.
This is the one that came to mind when I composed the email reply:
in-dial).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Alexander Lopez
Sent: Friday, 18 April 2008 1:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUDE! was RE:Dialplan
Jorge is correct you will not get the information need via FXO/FXS
unless you program the Mitel to do DTMF inband. It is possible but a
cludge of a fix at best. We have successfully integrated several Mitel
SX200 and SX2000 switches via the PRI (preferred) or T1 using EM_Wink
(works but you have
Look at the ChanSpy Application. It would be the easiest to implement
and it also allows the trainee to speak to the support person without
the customer knowing.
You can also use on-demand recording or simply record ALL calls
(legality and disclosure to calling parties are outside the scope of
Use call file to call out to the Alarm Panel and them put it in a
context that would do this:
[alarm-keepup]
exten = s,1,Answer
exten = s,2,SendDTMF(1)
exten = s,3,Wait(15)
exten = s,4,Goto(s,2)
You did not specify if you needed to do anything other than send the
digit to the alarm panel. If you
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Charlie Farinella
Sent: Thursday, December 27, 2007 11:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] No SMDI interfaces are available
How are the calls being transferred from Box A to Box B?
On what box is the receptionist registered too?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel
Cole
Sent: Tuesday, December 11, 2007 9:00 PM
To:
A few questions for you:
Where is your DNS Server for your LAN located by using the 172.17.x.x
address I suppose there is more to your network than two segments,
(Asterisk may drop connections if it has a problem with DNS)
How are your Polycom phones configured? Are they using a ftp/tftp
Concatenate the files into one larger file, in the order you want them
to play
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Joel Hill
Sent: Wednesday, September 26, 2007 7:01 PM
To: Asterisk Users Mailing List - Non-Commercial
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Paul
Sent: Friday, September 21, 2007 10:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] OT: Samsung Sprint CDMAoIP
Does it switch back
Snip headers
On 9/20/07, Jason Parker [EMAIL PROTECTED] wrote:
C F wrote:
AFAIK, the calls are free when you use it thru that device. Sprint
however charges $15 a month per phone or $30 for family plan.
While I
agree that sprint should pay me for this, it's not as bad.
T-mobile on
Snip
Subject: Re: [asterisk-users] Interesting Conference Request - Can
this
be done ?
Dovid B wrote:
Hi List,
I have a client that has an interesting request. He wants to have
people call in to a conference room and all be able to talk however
they should not hear each other. There should
Or you can skip the scripts and use the Page() Application.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Tim Panton
Sent: Friday, September 14, 2007 10:35 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Use a MeetMe room
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Vincent Sweeney
Sent: Friday, August 17, 2007 9:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk Channel as
That is wrong on so many levels.
You may want to take the time to install hylafax+iaxmodem, it offers
error correction and has many more features that offset the time
required to install...
Alex
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rizwan
Hisham
Sent: Tuesday,
In the top directory of your asterisk source in the doc dir there is a
file that explains channel variables.
From that file:
${UNIQUEID} * Current call unique identifier
BEWARE the UNIQUEID can be repeated do not use this as a primary index
on your databse.
-Original
You must be in Miami!
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore
- Aspendora
Sent: Wednesday, July 04, 2007 9:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Suing Dell||Dull
Add an Answer and add a m option to your dial command. They will hear
your music on hold until you answer.
Alex
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of GNUbie
Sent: Wednesday, June 27, 2007 12:18 PM
To:
If I understand your problem correctly you need to set ANI/CALLERID on a
peer by peer basis.
You can use the accountcode variable in the sip.conf file and set that
to the DID or you can use another variable.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
The only way I have ever seen any SIP and/or Network configurations is
from the Enterprise server management screen. If you purchased the 8800
thru a participating carrier, RIM offers a single user express license
for free (with purchase) Google for Free BlackBerry Express and that
should give
Cross posted from -users to -dev
I was looking at adding this functionality in last night.
I saw that in app_queue when a call is bridged it determines hold time.
Using the following:
holdtime = abs((now - qe-start) / 60);
and for queue.log the following:
(long) (callstart - qe-start)
My
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Gordon Henderson
Sent: Wednesday, April 11, 2007 8:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Re: Play audio and continue to next
'Your extension' would only use the bandwidth if it is off-site. If your
phone ('extension') is on the LAN then it 'should' not touch the T1.
Furthermore, the XO product is not compatible with Asterisk unless you
do as you say and connect the FXO or T1 port to your asterisk server.
You will still
Put this in the incoming context for that number called.
Exten = s,1,Wait(1)
Exten = s.2.System(mail -s 'Smitty called from ${CALLERID(all)'
[EMAIL PROTECTED])
Exten = s,3,Congestion
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
What do you mean by 'non-standard' IP block?
Is the Asterisk machine behind a NAT, or are only your clients?
Did you look at the nat setting sin sip.conf?
Do you have a static public address that can be routed to the Asterisk
box?
From: [EMAIL
It is a HUGE workaround but in concept it should work.
You will need to build completion confirmation into your script as you
will always get a success code from the manager.
Action: Originate
Application: System
Data: /path/to/script
Channel: Local/[EMAIL PROTECTED]
Context: dummy
Exten: 2
: [asterisk-users] Re: System from AMI
Alexander Lopez wrote:
It is a HUGE workaround but in concept it should work.
You will need to build completion confirmation into your script as
you
will always get a success code from the manager.
Action: Originate
Application: System
Data: /path
Exten = alarm,1,System(/usr/local/bin/sendalarm.sh|[EMAIL PROTECTED])
Or
Exten = alarm,1,AGI(sendalarm)
/usr/local/bin/sendalarm
#!/bin/sh
Mail -s Alarm condition on PBX $1 /dev/null
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf
You are better off running a small AGI script and calling the Dialplan
functions from there.
You can always start musiconhold, process, and return to dial plan.
Alex
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Rilawich Ango
Sent:
Is your Cisco device a Cisco router if so make sure you have no sip
fixup.
The Cisco may be fudging the SIP headers.
Alex
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of AL Daei
Sent: Sunday, January 14, 2007 11:57 AM
To:
More like a ID-10-T error.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bryan M.
Johns
Sent: Wednesday, January 10, 2007 11:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Directory
It can be configured and DOES work with ZAP channels.
If you are looking to use IP based devices your Mileage may vary from
Hybrid to Sherman Tank.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeronimo
Romero
Sent: Monday, December 11,
I think puck.nether.net may still have a
txt file with the CO broken down by NPA-NXX. You can then look at the carrier
and know if it is Cell/LandLine.
You can also X-ref the CO-list and get
Lat/Long and or simply the zipcode to help you locate the caller. Not
perfect but unless the
Use the AGI I sent. It looks like the email did not put a CR
correctly.
Run it from the commandline and see if you get output.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jon Weisman
Sent: Saturday, October 14, 2006 12:45 PM
To:
System(echo $RANDOM)
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jon Weisman
Sent: Friday, October 13, 2006
12:53 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [asterisk-users] Generate
Random Numbers in dialplan
Hi All,
You may need to wrap it in an AGI.
Like so:
Createrandnum.agi:
#!/bin/sh
RANDNUM=`echo $RANDOM$RANDOM | cut -c1-5'` echo SET VARIABLE
asteriskrandom $RANDNUM \\\n
Call it with:
Exten = s,1,AGI(createrandom)
your should then have the variable ${ASTERISKRANDOM} in your channel
snip..
I am going to reply inline as you asked
many questions
I have two questions.
Sure, you do!!
First I am running a t400p with three fxo
ports signalling fxs (inbound CO lines). I have six polycom 501's. The problem
is the amount of time the call setup takes. I have done this
Try running the echo test from both the house side and the co (outside)
side. That will let us know where the problem is.
Post results.
Alex
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Barry Fawthrop
Sent: Monday, October 02,
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