Sebastian,
There are many reasons why someone would want the DIDs provided by one
provider and outbound calls to go out via 1,2 3, or more providers.
In one of my installs I have a situation where local calls are placed
via a local telco switch but LD calls go out via a voip provider. The
On 03/13/2014 01:13 PM, Ron Wheeler wrote:
-1
Prefer top posting.
Easy to see if I want to scroll down to see if it is something
interesting to me.
I get a lot of e-mails each day and scrolling wastes too much time.
But if you have a solution to a problem that I raise, please feel free
to
Hello,
CentOS 6.x and Asterisk 11.x
I have an interesting, to me at least, situation. Using a Polycom
501(also tried with X-Lite). I have set up Asterisk to accept
registration from the Polycom and it registers successfully but then
withing 30 seconds on the CLI I get the message that the Polycom
or
individual sip peer settings.
Nick Olsen
Network Operations
(855) FLSPEED x106
*From*: John Millican j...@millican.us
*Sent*: Thursday, January 02, 2014 10:50 AM
*To*: Asterisk Users Mailing List - Non-Commercial
/2014 10:51 AM, Nick Olsen wrote:
Make sure you have nat=yes in your sip.conf either under globals or
individual sip peer settings.
Nick Olsen
Network Operations
(855) FLSPEED x106
*From*: John Millican j
On 12/04/2013 11:00 AM, Paul Belanger wrote:
On 13-12-04 10:19 AM, CDR wrote:
Digium is 100% lost in the map. If they would come up with a Paid
version of Asterisk, one that would use the .NET framework in Windows,
something simple to install, they could go public on the product.
Linux has a
Hello,
I have an OpenVox A400E02 (2FXO) in a box running Debian 6.0.2 running
Asterisk 1.8.6.0. I have to POTS line on it from Verizon in Virginia,
USA. Whenever I place a call to one of the two lines I get a red alam
and then it clears and repeats this till I hang up. There is no caller
Hello,
I am using a perl script to pull call info from a DB and place calls via
telnet and AMI, all on local machine of course. My problem is that I
need to capture any response from the carier, such as this taht appears
in the CLI:
[Mar 1 12:55:50] == Using SIP RTP CoS mark 5
[Mar 1
Hello,
I am using a mix of Call files and AMI telnet from a perl app to place
calls. I sometimes get this in the CLI:
-- Attempting call on sip/551234@providerfor 1@mycontext:1
(Retry 1)
[Feb 27 13:47:07] == Using SIP RTP CoS mark 5
[Feb 27 13:47:07] -- Got SIP response 503 No
On 12/2/2011 12:44 PM, Steve Edwards wrote:
On Fri, 2 Dec 2011, Jim Lucas wrote:
How is using Fail2Ban less resource intensive then me writing (by
hand) iptable rules?
It depends on how you define resources and how much of those resources
you have.
Gordon (based on my understanding of his
On 11/29/2011 12:48 PM, C F wrote:
On Mon, Nov 28, 2011 at 10:57 AM, Tom Browningttbrown...@gmail.com wrote:
On Sun, Nov 27, 2011 at 8:47 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
Linux has excellent built-in subsystems to control firewalling and so on
without resorting to
On 11/28/2011 3:35 PM, Danny Nicholas wrote:
If you put a gun to my head I would say to stay with Centos 5 and
either 1.4.42 or 10.0.0-rc2. 10.0.0-rc2 removes a feature that was
killing me in 1.4, but if you aren't doing IVR stuff, you can stay
with what you know. Another thing to
Thanks to all for the responses. Boss calls overseas a lot and has an
unlimited data plan, so this coupled with the rates that we get for
our VoIP calls it is much cheaper than what Verizon charges.
JohnM
On 10/11/2011 1:29 AM, Jeremy Kister wrote:
On 10/10/2011 10:08 PM, Andres wrote:
I
Hello all,
Does anyone know of a good free/inexpensive 3G SIP client for the
iPhone? If anyone is using one that works good for them could you
please let me know.
Thank You,
JohnM
--
_
-- Bandwidth and Colocation Provided by
On Tue, Aug 16, 2011 at 4:42 AM, john Millican j...@millican.us
mailto:j...@millican.us wrote:
On 8/15/2011 5:48 PM, john Millican wrote:
Hello,
Asterisk 1.4.38
Linux version 2.6.9-89.31.1.EL CentOS
Trying to get variables into a dial plan from AMI
Hello,
Asterisk 1.4.38
Linux version 2.6.9-89.31.1.EL CentOS
Trying to get variables into a dial plan from AMI. I have tried all
sorts of combinations,entering them after making a connection to ami
through telnet, of the many available examples on voip-info.org such as:
Action: Originate
On 8/15/2011 5:48 PM, john Millican wrote:
Hello,
Asterisk 1.4.38
Linux version 2.6.9-89.31.1.EL CentOS
Trying to get variables into a dial plan from AMI. I have tried all
sorts of combinations,entering them after making a connection to ami
through telnet, of the many available examples
On 7/28/2011 11:31 AM, Bruce B wrote:
Hmmm, if alwaysauthreject is already breaking RFC rules then why not
break another rule for the greater good? It would only add another layer
of security.
Maybe: *alwaysregreject=yes*
*
*
*To drop SIP packets for both unauthorized registers and anonymous
Hello all,
Just hoping to get some opinions from folks that have actually used the
Rhino R4FXO-EC. Looking for user experiences, good or bad. This looks
like a nice unit and I have a need for exactly this config, 4FXO and EC
TIA,
JohnM
--
Hello all,
I have a perl script that updates a M$ SQL DB based on an ivr that is
run on asterisk.
When it runs as a normal agi, it works great.
when run as a DeadAGI it does not work.
When i execute the script from h channel withDeadAGI and agi debug on i get:
[2010-12-20 01:08:54] --
Hello,
I originally thought I should post to the biz list but I am not looking
for offers of DID's, I am looking for actual user
experiences/information on obtaining a DID for an Office I am working
with in Hyderabad, India.
Can anyone offer recommendations based on personal experience of where
John Millican wrote:
Hello,
I have a situation where a remote worker dials in to the asterisk server,
enters
the secret code, then dials out via Disa on a PRI. This was all working
great
until this morning. Now the calls is placed out, connected but there is no
voice from/to either
Hello,
I have a situation where a remote worker dials in to the asterisk server, enters
the secret code, then dials out via Disa on a PRI. This was all working great
until this morning. Now the calls is placed out, connected but there is no
voice from/to either phone. This is what shows on the
David Gibbons wrote:
snip
Customers in Europe all have mobile phones, while senders in North
America rarely have them ( they have answering machines, though ).
/snip
What planet/year are you/your clients living on/in? I don’t know anyone
who doesn’t have a mobile. Maybe it’s just
Joe Greco wrote:
Sorry, I can't resist.
How do I join the Mail List Nazi Corp? Do I have to be invited, or can I
just self appoint myself? Asking neophyte questions are objected to by
some, top posting by those who blast others, etc.
How about leaving member chastisement to the
Hello All,
I have a need for a wireless solution and have been looking at the
Aastra 57i CT phone that have the wireless handset with them. Aastra
says they will cover up to 300,000 square feet.
I am finding this hard to accept. I was also wondering about the
secure WDCT cordless technology
2009, John Millican wrote:
The manager wants to be able to spy on agents who dial through the PBX
from their homes. Currently the agents dial the main number, use the
secret code to get to authenticate and DISA, and then dial back out
for their sales calls. I have chanspy working great
Hello all,
OS OpenSuSE 10.3
* ver 1.4.26.2
zaptel ver. 1.12
Digium TE122
I have a request for remote users to be able to dial through the system
so that the sales managers can barge/chanspy on the sales force.
I have the DISA part working with authentication(rather straight
forward) but what I
Steve Edwards wrote:
On Tue, 29 Sep 2009, John Millican wrote:
I have a request for remote users to be able to dial through the system
so that the sales managers can barge/chanspy on the sales force. I have
the DISA part working with authentication(rather straight forward) but
what I can
Steve Edwards wrote:
On Tue, 29 Sep 2009, John Millican wrote:
I have a request for remote users to be able to dial through the system
so that the sales managers can barge/chanspy on the sales force. I have
the DISA part working with authentication(rather straight forward) but
what I can
C. Savinovich wrote:
What about if I use the browser from my cellular phone?
CS
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pascal Bruno
Sent: Wednesday, September 16, 2009 10:21 PM
To: Asterisk Users Mailing List
OK this is the RTFM question of the day but I need a sanity check.
I have 2 Astra 51i phone and a linksys PAP2 on a single DSL connection.
2 Aastra 51i-|
|-NAT on dsl moden--(Internet)--Asterisk
PAP2t|
The DSL modem/router which has QOS set for the
Hello all,
I have a need to be able to use the originate AMI command to dial out to
the PSTN, have that person answer and then have the second PSTN
connection dialed out.
I have tried to use:
Action: Originate
Channel: sip/number@provider
Context: default
Exten: othernumber
Priority: 1
in sip.conf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Millican
Sent: Thursday, May 28, 2009 1:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users
Justin Phelps wrote:
I'm replacing a SoundPoint IP 600 with a SoundPoint IP 650.
I attempted to simply reuse the existing config files for the old phone
on the new phone, but the new phone would lock up on the 4th digit when
attempted to dial out certain numbers. So, I downloaded the
Justin Phelps wrote:
digitmap
dialplan.digit
map=[2-9]11T|[*]xxT|891xxxT|[1-7]xxxT|8[0-46-8]xxT|8500T|851xxxT|9,1[2-9]xT|9,xxxT
dialplan.digitmap.timeOut=3|3|3|3|3|3|3|3|3/
Do the above changes look in line with common practice JohnM?
Short Answer:
They do.
Longer answer,
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Tuesday, April 07, 2009 6:21 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Fwd: [asterisk-r2] MFCR2 With Unicall -
Jon Pounder wrote:
Cary Fitch wrote:
The problem has two prongs - first we are in control of our own
landlines and can use asterisk to screen whatever crap we wish before
disturbing a real user or allowing a vm to get stored (but it would be
nice not to have to).
The other issue is we
Shaun Ruffell wrote:
John Millican wrote:
Well,
lsmod | grep hisax returns nothing
plain lsmod:
Module Size Used by
dahdi_dummy22472 0
dahdi 215776 1 dahdi_dummy
crc_ccitt 18944 1 dahdi
af_packet 57100 2
Hello all,
Ok it is Sunday afternoon and I am going crazy. I have been running in
circles so long that I can't think straight. As an example, I sent this
message to the wrong address the first try, AAAGGH. I have
Asterisk 1.6.0.6 and DAHDI Tools Version - 2.1.0.2,
DAHDI Version:
Shaun Ruffell wrote:
John Millican wrote:
# /etc/init.d/dahdi start
Loading DAHDI hardware modules:
wctdm: modprobe wctdm
What is the output of the 'dmesg' command at this point?
No hardware timing source found in /proc/dahdi, loading dahdi_dummy
Running dahdi_cfg: /usr
Shaun Ruffell wrote:
John Millican wrote:
Shaun Ruffell wrote:
John Millican wrote:
# /etc/init.d/dahdi start
Loading DAHDI hardware modules:
wctdm: modprobe wctdm
What is the output of the 'dmesg' command at this point?
All I see in dmesg is:
dahdi: Telephony Interface
Mike wrote:
Hi,
I`ve been toying with an Aastra phone (9143i) wondering if it could be a
good alternative to to the more expensive Polycom phones.
One thing which I can't figure out, although it certainly looks simple,
is to update the firmware though FTP (not TFTP). I have set
Chris Bagnall wrote:
I would suggest using OpenSIPS with Asterisk and bypass IAX all together for
this
particular application.
If the users in question are often in hotels abroad, something like this may
not solve the problem - I've noticed quite a few hotels are now blocking SIP
Ira wrote:
At 01:36 PM 7/30/2008, you wrote:
Nhadie wrote:
Hi
How cn i define in GotoIfTime from day 1 extending to day 2?
e.g July 30 2200 up to July 31 0200
I'm thinking like this: GotoIfTime(22:00-02:00|*|30-31|jul?test,s,1)
GotoIfTime(22:00-23:59|*|30-31|jul?test,s,1)
Doug Lytle wrote:
John Koenig wrote:
exten=s,1,set(CALLERID(all)= null)
exten=s,n,Dial(${ARG1})
Just a guess.
exten = s,1,Set(CALLERID(all)= null 0)
exten = s,n,SetCallerPres(prohib)
exten = s,n,Dial(${ARG1})
Doug
I believe you need to use:
exten = s,1,Set(CALLERID(all)=)
To
what I am sending to my VoIP terminating node?
John
John Millican wrote:
Doug Lytle wrote:
John Koenig wrote:
exten=s,1,set(CALLERID(all)= null)
exten=s,n,Dial(${ARG1})
Just a guess.
exten = s,1,Set(CALLERID(all)= null 0)
exten = s,n,SetCallerPres(prohib)
exten = s
Hello,
Asterisk 1.4.21.1
Well it seems like my month for questions. I have a situation where the
CallerID num shows as [EMAIL PROTECTED](the ip of the asterisk
box) on calls to any of the internal phones. This prevents the
ability to dial out from the missed call list. I have not been able to
Hello All,
I have a request that I have not been able to figure out as yet. I need
to be able to play a beep when a call is transfered via attended transfer.
This is exactly what is in the bug tracker at:
http://bugs.digium.com/view.php?id=3819
Has any one found a way, elegant ot otherwise, to
Hello,
Asterisk version 1.4.21.1
Can anybody tell me what I am doing wrong or why the Read application
does not accept the # key as input? My read statement:
exten = s,n,Read(uchoice|thankyouforcalling|3||1|1);
In the prompt thankyouforcalling it says press pound for a company
directory along
Tilghman Lesher wrote:
On Wednesday 09 July 2008 09:08:50 John Millican wrote:
Can anybody tell me what I am doing wrong or why the Read application
does not accept the # key as input? My read statement:
exten = s,n,Read(uchoice|thankyouforcalling|3||1|1);
In the prompt thankyouforcalling
Hello All,
Asterisk 1.4.20.1
SuSE 10.3
I have been building a dial plan and have run into some questions that I
have not been able to answer on Voip-info or google. I am trying to use
either Read or Background to gather user input to an IVR in a Macro. I
need to be able to branch based on the
Hello all,
I was just asked a question from a client that I have in regards to
TTY/TDD telecommunications device for the deaf. I have read on
voipinfo at http://www.voip-info.org/wiki/view/tdd+mode that back in Dec
2006 this was in alpha stage in Asterisk. There does not (in my limited
searching)
Tilghman Lesher wrote:
On Tuesday 01 April 2008 05:14:25 Pete Kay wrote:
I am hoping someone can help me out on this. I want to be able to
interrupt MOH every X seconds after the DIAL command is executed. The
interrupt greeting is something like please wait while we transfer your
call. How
Shane D wrote:
Try this:
exten = 1000,1,Answer()
exten = 1000,2,Wait(2)
exten = 1000,3,VoiceMailMain()
You do not specify the mailbox number in the call to the application.
You only specify the number to VoiceMail()
HTH,
Shane
On 1/31/08, Drew Gibson [EMAIL PROTECTED] wrote:
John
to the phone it may not be feasible.
Having said that any suggestions will be appreciated. I know I could
use an ATA and a PSTN Phone from wally world, but this will not fit the
project or the need.
Thanks,
JohnM
begin:vcard
fn:John Millican
n:;John Millican
adr:;;PO Box 9;Wentworth;NH;03282;US
Tzafrir Cohen wrote:
On Sun, Jan 27, 2008 at 11:27:16AM -0500, John Millican wrote:
Hello All,
This may be a little OT for the list but it seems to be to be the place
to get the best answer. I have looked at the many Skype/Yahoo phones out
there and none seem to be what I am looking for.
I
Gordon Henderson wrote:
On Sun, 27 Jan 2008, John Millican wrote:
Tzafrir Cohen wrote:
On Sun, Jan 27, 2008 at 11:27:16AM -0500, John Millican wrote:
Hello All,
This may be a little OT for the list but it seems to be to be the place
to get the best answer. I have looked at the many Skype
it takes
a full minute respond after the pass phrase is typed in. Could this be
related or am I just grasping at straws?
Any Ideas?
--
JohnM
begin:vcard
fn:John Millican
n:;John Millican
adr:;;PO Box 9;Wentworth;NH;03282;US
email;internet:[EMAIL PROTECTED]
title:Director of Technology
tel;work:603
Doug wrote:
At 14:54 1/10/2008, John Millican wrote:
Hello all,
I know this has been discussed before but I am not finding the thread on
voip-info or site:lists.digium.com through google.
I have a site with * ver. 1.4.15 (started out as 1.4.2 or so) running on
openSuSE 10.2, Dual core AMD
at or where to go (keep it
clean ;-) please) would be greatly appreciated.
Thanks in advance
JohnM
begin:vcard
fn:John Millican
n:;John Millican
adr:;;PO Box 9;Wentworth;NH;03282;US
email;internet:[EMAIL PROTECTED]
title:Director of Technology
tel;work:603-764-9163
x-mozilla-html:FALSE
On Wednesday December 19 2007 6:09 pm, Tzafrir Cohen wrote:
On Wed, Dec 19, 2007 at 02:47:39PM -0800, Steve Edwards wrote:
This only works because you are closed to the alternative. The
alternative (verb-noun) works fine for the above referenced applications
and many more. Do you want to
See Inline
On Tuesday December 11 2007 2:20 pm, Jonn R Taylor wrote:
The old x100p cards where 5 volt pci cards. I had this same problem and it
was the type of pci slot that I had the card plugged into.
Jonn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Hello
I have a setup where i have 2 asterisk servers connected over the public
internet with plenty of bandwidth, NAT on one side only. If i use IAX
between the two *'s dtmf is flawless. If I use SIP, DTMF detection is around
30% or less. I have an exten to dial into and check DTMF:
exten
On Monday November 12 2007 9:38 am, Eric Jacksch wrote:
Hello all,
We're using a lot of the linksys phones, and while user feedback is
generally positive, the speakerphone leaves a bit to be desired.
For those of you using the polycom desk phones, how do you find the
built-in speakerphone?
On Monday November 12 2007 1:50 pm, Doug wrote:
At 08:38 11/12/2007, Eric Jacksch wrote:
Hello all,
We're using a lot of the linksys phones, and while user feedback is
generally positive, the speakerphone leaves a bit to be desired.
For those of you using the polycom desk phones,
Hello All,
I have a setup of ABE on rPath linux,Sangoma A101D, and a T-1 line (Not PRI)
which is all happily coexisting and all lights are green.
The T-1 comes in from the world into a Shark Box which splits the T into
384K data and 6 channels voice. The data side is working great. The voice
Hello All,
I am looking at doing some video conferencing with SIP. I was hoping to get
some early pointers from any one that is currently doing this. I have been
all over goggle and voip-info and there is a ton of anecdotal information
but, I was hoping for more specifics of what people are
sniped and moved to below for readability
John Millican wrote:
Hello All,
I am looking at doing some video conferencing with SIP. I was hoping to
get some early pointers from any one that is currently doing this. I
have been all over goggle and voip-info and there is a ton of anecdotal
On Saturday October 13 2007 12:09 pm, Lee Jenkins wrote:
Steve Edwards wrote:
On Sat, 13 Oct 2007, Lee Jenkins wrote:
I have been using axVoice.com for some about 9 month to a year now and
their service is pretty damn good. For home users they have unlimited
plan for around 22.00-24.00
On Saturday October 13 2007 12:47 pm, Doug Lytle wrote:
John Millican wrote:
On Saturday October 13 2007 12:09 pm, Lee Jenkins wrote:
Be sure to read the fine print as most of the unlimited plans do
actually have a limit on usage (even the ones I offer). Some are out in
the open some
as it gets and would get my
vote.
JohnM
--
John Millican
Senior Partner
Director of Technology
Sentinel Communications
PO Box 9
Wentworth, NH 03282
Phone (603) 764-9163
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users
Hello All,
I have a requirement to setup a predictive dialer for a customers call center.
I am asking for pros and cons of the different dialers available for
Asterisk. If you are going to send marketing material send it to my e-mail
directly please and not to the list. I was hoping to get
On Wednesday August 08 2007 8:28 am, Tim Johnson wrote:
Hi Drew. Thanks for the tips. My Line 1 works as I'd like it to, and I
could be wrong, but I don't think changing the dialplan there will help. I
really just want to be able to dial local phone calls (7 digits) and have
it go out the
On Wednesday August 08 2007 12:10 pm, Mike wrote:
I can be a bit slow sometimes, but you said that it's not possible, and on
the other hand told me to write my own function (which appears to
contradict the first statement).
Your example of the use of a function is exactly what I need (Create
On Wednesday August 01 2007 5:49 pm, Douglas Garstang wrote:
Don't know about the 320, but we provisioned the 301's. They're
provisioning is basically the same as the 501's and 601's. What problems
are you having?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
On Tuesday July 31 2007 4:44 pm, Joe acquisto wrote:
. . .
Even if you can find non-original-artist recordings of such music, the
*compositions* are registered with BMI and ASCAP, and you'll need
blanket licenses to play them. (Well, if you only wanted one or two
tracks, you might
On Monday July 23 2007 9:26 am, Matt wrote:
Hi,
Has anyone here ever used a Polycom IP 4000 Soundstation SIP
Conference Phone with asterisk? If so, how well does it work and how
does it sound?
I have one at a customer site and they are very happy with it. Works well,
sound quality is
Hello All,
I have Asterisk 1.4.5 running on a SuSE 10.3 x86_64 2.6.18.2-34
I upgraded from 1.4.2 to 1.4.5 on sunday the 24 of june and since have been
getting:
Internal RTCP NTP clock skew detected: lsr=1402479300, now=1402675136,
dlsr=196500 (2:998ms), diff=664
I see an entry in Mantis that
On Thursday June 28 2007 1:19 pm, Jared Smith wrote:
On 6/28/07, John Millican [EMAIL PROTECTED] wrote:
Would i be correct in assuming that if i pull a copy of
1.4.5 from digium this weekend that this message will go away?
No... you'd have to pull the latest code from the 1.4 branch using
and the hold works on the ones
that have been tested. We are not on the latest firmware yet though. I will
be testing that tomorrow.
John M
--
John Millican
Director of Technology
Sentinel Communications
PO Box 9
Wentworth, NH 03282
Phone (603) 764-9163
On Sunday April 15 2007 5:48 am, JNA wrote:
Is there a way to make it so you do not have to dial 9 by default to dial a
outside number? I would like it if we could just dial the number any
pointers?
In a number of my ATA's and IP Phones I have a delay in the pattern match so
that if the user
On Thursday October 12 2006 4:15 pm, Dave Cotton wrote:
On Thu, 2006-10-12 at 21:30 +0200, Csibra Gergo wrote:
Thursday, October 12, 2006, 6:58:57 PM, Tim wrote:
I've read alot of comments on the SPA-3000, many if not all saying they
had echo issues, but I've not seen anyone comment on
. If this is not an option, I'm also open to devices that will fail
over to GSM to make the emergency call. I apologize if this topic has
already been covered before.
-brandon
Sipura 3000 or 3102 to start with I am sure there are others
--
John Millican
Senior Partner
Director of Technology
Sentinel
On Friday September 01 2006 9:27 am, ram wrote:
Hi
how can i do balance anouncement by using asterisk
take example, i have table balance , user name 9, balance 200$
user dial *98 or what ever, then i need anouce his balance is 200$, by
reading from that row
any clues how can i achive
On Friday September 01 2006 10:19 am, ram wrote:
Hi
thanks for the quick reply
any documents to read to achive this
or any examples would be great to read
Ram
On 9/1/06, John Millican [EMAIL PROTECTED] wrote:
On Friday September 01 2006 9:27 am, ram wrote:
Hi
how can i do
On Saturday August 26 2006 11:15 am, Matt Riddell (IT) wrote:
John Millican wrote:
Hello all,
I am trying to test if the length of a dialed number is greater than 7.
When i use:
exten = 1,n,GoToIf($[${LEN(${numdial})}7]?dialout:nodial);
and I dial an 11 digit number i.e. 1 800 xxx
Hello all,
I am trying to test if the length of a dialed number is greater than 7. When
i use:
exten = 1,n,GoToIf($[${LEN(${numdial})}7]?dialout:nodial);
and I dial an 11 digit number i.e. 1 800 xxx
i get this in the console:
Executing GotoIf(SIP/xxx-xxx-xxx-xxx-006ca720, 0?dialout:nodial)
straight through SIP.
Any suggestions?
Thanks!
-Ken
Ken,
Just a hunch but it may be the space in the dial string between the and the :
Your string:
9,:xxx :@gw0|424 :@gw0)
corrected:
9,:xxx:@gw0|424:@gw0)
as I said just a guess.
--
John Millican
Senior Partner
Director
to speech.)
##
I have used this (on a very low call volume obviously) on as low end a machine
as PII 400 with 512 meg ram.
Hope this helps
--
John Millican
Senior Partner
Director of Technology
Sentinel Communications
PO Box 9
Wentworth, NH 03282
(603) 764-9163
Instead of Background() use Read(). this will allow for any number of digits.
example:
exten = 1234, 1, Read(var_to_use|prompt_name|number_of_digits_to_accept);
;then use a goto based on the value of var_to_use.
exten = 1234,2 GoToIf($[${var_to_use} = 1]?new_exten,1:3);
this way you are
that is on a cable
connection that receives the call over IP and then dials out to a voip
provider? How do any fxo devices come into this picture? How does a
zap channel come into this picture?
John Millican wrote:
Doug,
The interface that i dial to is at my Service provider and am not sure
what
. You mentioned you have an SPA-3000 in your inventory.
That is what I use here and I do not load or use zap or pri modules. I use
the 3000 as my fxo/fxs via sip on my local network. I have no cards in my
computer. You could do the same for testing of your problem.
Doug
On Tue, 20 Jun 2006, John
Warren,
My suggestion for testing would be just use ethernet hand off to the asterisk
from the Cisco. You could bypass the Cisco but then you would need a T-1 card
for the asterisk box and they are not cheap. I believe there are valid
arguments for both choices though and ultimately should be
setup?
W
John Millican wrote:
Warren,
My suggestion for testing would be just use ethernet hand off to the
asterisk from the Cisco. You could bypass the Cisco but then you would
need a T-1 card for the asterisk box and they are not cheap. I believe
there are valid arguments for both choices
. Worth a try I guess. There are some rfc8322 issues that apparently
will be addressed with a rewrite in the next makor version release.
Doug
On Mon, 19 Jun 2006, John Millican wrote:
Doug, I read that post and unfortunately it was not a solution. I do not
believe it has to do with interstate
Doug,
The interface that i dial to is at my Service provider and am not sure what
they are using. I dial out of my * box to a service provider number which is
answerd by an asterisk box that I have at another location on a high speed
cable connection, that box then dials the numberI ultimately
Shaun,
I believe that there are 2 models of the WRT54GP2 as there was/is with the
PAP2's one that is set for VONAGE and one that is not, typically referred to
as the WRT54GP2-NA
John M
On Monday June 19 2006 3:37 pm, Shaun wrote:
I'm looking for somehting like the standard house hold
Matt,
Thank you very much!
I am currently running 1.2.7.1 but will be upgrading to 1.2.9.1 this week. I
will try toneduration=200 first and let you/list know how well it worked.
I read in zapata.conf.sample where it says:
How long generated tones (DTMF and MF) will be played on the channel
for dial-up and 2 plain
vanilla local only lines.
John
On Monday June 19 2006 5:28 pm, Doug Crompton wrote:
Is the PAP2 an ethernet connected device to * ? I was wondering why you
were using zap if it were not an internal card?
Doug
On Mon, 19 Jun 2006, John Millican wrote:
Doug,
The interface
1 - 100 of 163 matches
Mail list logo
