I am having the same issue.
Please assist.
On Sun, Nov 1, 2009 at 1:27 PM, giancarlo lombardo
wrote:
> Dear all,
> I'm trying to setup a SIP call with XLITE using my asterisk PBX, but I have
> trouble, I see on XLITE console:
>
> Registration Error: 503 - Service unavailable.
> Someone have a tu
I have two DID numbers.
I want callers who dial 1 703 to get placed in a specific part of
IVR
I want other callers who dial 1 567 to get placed in a different
area.
How do I do this please?
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en => s,n,Hangup
>
> If you want to jump into a specific part of context, you should put a label
> near the 'n' priority where you want to jump to (eg. exten =>
> s,n(jumphere),) then specify that label into Goto()
> application.
>
> Cheers,
> //Al.
>
>
bel
> near the 'n' priority where you want to jump to (eg. exten =>
> s,n(jumphere),) then specify that label into Goto()
> application.
>
> Cheers,
> //Al.
>
>
>
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun..
part of context, you should put a label
> near the 'n' priority where you want to jump to (eg. exten =>
> s,n(jumphere),) then specify that label into Goto()
> application.
>
> Cheers,
> //Al.
>
>
>
>
> From: asterisk-users-boun.
Is this going to work:
[default]
include => stdexten
include => big10-IVR
include => cleveland-IVR
exten => _17035745353,1,Goto(big10-IVR,s,1)
exten => _15672528431,1,Goto(cleveland-IVR,s,1)
[big10-IVR]
exten => s,1,Answer()
exten => s,n,Background(dir-welcome)
;exten => s,n,WaitExten(1)
;exten
I want to match specific contexts to menus.
If users dial a number (example: 1703444) then start with context
big10-IVR
If users dial a number (example: 1567444) then start with context
cleveland-IVR
It is not working. I have played with the include statements and am close
but no cigar.
How do I check
On 11/1/09, Peter wrote:
> Try removing the include statements from the default context and see
> what happens. Also double check to make sure calls are sent to the
> default context.
>
> Peter
>
> On Nov 2, 2009, at 3:40 AM, Thomas Person wrote:
>
>> I want to match specific conte
ny errors in you configuration.
>
> Peter
>
> On Nov 2, 2009, at 4:39 AM, Thomas Perron wrote:
>
> > How do I check
> >
> > On 11/1/09, Peter wrote:
> >> Try removing the include statements from the default context and see
> >> what happens. Al
fined to reach that context and
> its extention is start extension.
> If the cleveland-IVR is based on the start extension too, the same applies.
>
>
> Besides that, it would work...(maybe not the way you expect... :-) )
>
> Regards,
> Juan
>
> Thomas Perron wrote:
>
IVR question:
Users dial my DID numbers and get connected to macros and other vectors that
guide them
to the appropriate context. Once connected to a specific context I would
like to send a text message
to their phone. Do I need a PERL script or is there something native in
Asterisk 1.6 that can
Does anyone have any success with sending a text message from
extensions.conf
to an PSTN endpoint such as a cell phone?
If so, kindly send configuration for this part. I am working on an IVR and
want
callers to get a text message at a particular part of the call, after
dialing a defined character
Will text messages work to non-SIP enpoints using your logic/code?
thank you
On Mon, Nov 9, 2009 at 8:59 PM, Matt Riddell wrote:
> On 10/11/09 12:58 PM, Thomas Perron wrote:
> > Does anyone have any success with sending a text message from
> > extensions.conf
> > to an PS
Mon, 9 Nov 2009 22:19:08 -0500
> From: thomas.per...@gmail.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] SendText
>
>
> Will text messages work to non-SIP enpoints using your logic/code?
> thank you
>
> On Mon, Nov 9, 2009 at 8:59 PM, Matt Riddell
I have two DID numbers. I want to configurate my IVR to initiate a context
1 if I dial DID 1.
If DID2 is dialed then start with context 2.
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Hi Alex,
Thank you
Tom
On Sat, Nov 21, 2009 at 10:24 AM, Alex Balashov
wrote:
> Thomas,
>
> Thomas Perron wrote:
>
> > I have two DID numbers. I want to configurate my IVR to initiate a
> > context 1 if I dial DID 1.
> > If DID2 is dialed then start with contex
thanks
On Sat, Nov 21, 2009 at 12:26 PM, Steve Edwards
wrote:
> > Thomas Perron wrote:
> >
> >> I have two DID numbers. I want to configurate my IVR to initiate a
> >> context 1 if I dial DID 1. If DID2 is dialed then start with context 2.
>
> If the DIDs a
I want to distribute a random number to each of the first 100 callers to my
IVR.
This random number will be matched to their telephone number.
Where in Asterisk can I do this? And, how?
Logic.
Call arrives.
Context for announcement begins.
You will receive a authentication code at the end of the
that is a bit heavy for me. how about some simple way to announce a random
number. using RAND. and saydigit
exten => s,1,Set(junky=${RAND(1,8)})
On Sat, Nov 21, 2009 at 7:20 PM, Steve Edwards wrote:
> On Sat, 21 Nov 2009, Thomas Perron wrote:
>
> > I want to distribute a r
How do I get to this prompt?
#!/usr/bin/php -q
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Hallo Philipp,
Wei Gehts ist Einen.
Danke.
I am in USA.
Thanks.
On Sun, Nov 29, 2009 at 8:49 PM, Philipp Kempgen
wrote:
> Thomas Perron schrieb:
>> How do I get to this prompt?
>>
>> #!/usr/bin/php -q
>>
> http://en.wikipedia.org/wiki/Shebang_%28Unix%29
>
>
I am trying to find an AGI script that runs via PHP and performs the
send text application.
Does anyone have any tools or scripts set up for this please?
If so, kindly send some info or the code that performs this action.
Thank you
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How can this scenario be implemented please?
THIS IS NOT A SEND TEXT application.
A call arrives on the IVR.
After hearing several vectors to guide the person through information
I want to confirm a transaction via email to his cell phone.
Specifically, I want to use his phone number and then app
Interesting response but I am not that saavy to follow it!
Thank you
On Sun, Dec 6, 2009 at 2:40 AM, Tzafrir Cohen wrote:
> On Sat, Dec 05, 2009 at 08:25:33PM -0500, Thomas Perron wrote:
>
>> And, then send an email to the party. Example
>>
>> 3035551...@tmobil
I am trying to use a simple tool in the Dial plan so that if the first
number does not connect the logic will go to the second and/or third.
Basically, I want the call to ring and connect to the first number
Then, if it is not answered I want another number to try to get connected
Then, if second
-what does this do please?
"subject line" .comes from where?
${the_caller_...@tmobile.net) i understand this part.
thank you
On Sun, Dec 6, 2009 at 2:40 AM, Tzafrir Cohen wrote:
> On Sat, Dec 05, 2009 at 08:25:33PM -0500, Thomas Perron wrote:
>
>>
Does anyone have a script that performs Auto Attendant / Receptionist system
If so, please send.
Thanks
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htt
I want to list 100 indiviual businesses.
and do an ivr for them specifically
some use databases so i need an agi script in .pl or php.
On Sat, Dec 12, 2009 at 7:26 PM, Doug Lytle wrote:
> Thomas Perron wrote:
>> Does anyone have a script that performs Auto Attendant / Receptionist sys
How does Fax for Asterisk work?
On Fri, Dec 18, 2009 at 7:51 PM, Leif Madsen
wrote:
> Warren Selby wrote:
>> Is the new Fax For Asterisk being released in conjunction with this
>> release?
>
> If it's not already available, then it will be available very early next week.
>
> Leif Madsen.
>
> ___
Anyone have a cookbook on configuring sendmail to work with Asterisk?
Or,a few config examples.
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I want to have Asterisk Dial individual numbers and play a recording
if each answers.
If they don't answer then the code rolls to the next number.
Should I set up a spreadsheet somewhere and load with the numbers?
Or, an AGI script?
1. Dial number 1
2. If connect, then play message
3. If conne
I want to ensure that only this extension is executed.
But, I have others that are similar.
I want:
exten => 34101,1,Answer()
exten => 34101,n,Record(34101:gsm) ; 34101 test zip code
exten => 34101,n,Playback(34101)
exten => 34101,n,Hangup
Is this correct or do I need to have each of the four
I want to play soft music in the background while the IVR passes
through various contexts.
In short, I need to mix the script with music and my pre-staged .gsm
or .wav audio.
What tools to I need to use in Asterisk to make this happen please?
exten => s,1,Answer()
;exten => s,n,system(echo "${DATE
Is there any code that I can cut/paste that will allow me to receive
an SMS text on Asterisk?
and, where can I capture the incoming text?
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asterisk-users
exten => s,1,Answer()
exten => s,n,Background(astcc-please-enter-your)
exten => s,n,Background(zip-code)
exten => s,n,WaitExten(5)
exten => s,n,Read(NUMBER,,5)
exten => s,n,SayDigits(${NUMBER})
exten => 22042,n,Dial(SIP/sipvendor/111,120,A(ginger3))
exten => 22601,n,Dial(SIP/sipvendor/1
veilen danke timm
cheers
tom
On Sun, Jan 17, 2010 at 2:10 PM, Timm Korte
wrote:
> Am 17.01.2010 18:39, schrieb Thomas Perron:
>> exten => s,1,Answer()
>> exten => s,n,Background(astcc-please-enter-your)
>> exten => s,n,Background(zip-code)
>> exten => s,n,W
I want to create a script for IVR that compiles responses, aggregates
them to a total number.
Then, run an equation based on the result.
Press 1 for X (X is a positive number 500)
Press 2 for Y (Y is a positive number 200)
Press 3 for Z (Z is a positive number 300)
Press 20 to calculate the resul
total up for current call.
then read back the number
2010/1/30 Håkon Nessjøen :
> For all calls combined, or for the current call?
>
> On Sat, Jan 30, 2010 at 2:48 PM, Thomas Perron
> wrote:
>>
>> I want to create a script for IVR that compiles responses, aggregates
&
tExten(3)
> exten => 3,1,Set(TOTAL=${MATH(${TOTAL}+300,int)})
> exten => 3,n,WaitExten(3)
> exten => 9,1,SayNumber(${TOTAL})
>
> Or something. Never used either math or saynumber before, but according to
> the documentation, something like this should work..
>
>
>
what is wrong with this please:
;exten => 4,1,WaitExten(3)
exten => 4,1,Set(TOTAL=${MATH(${TOTAL}+500,int)})
exten => 4,n,WaitExten(3)
exten => 2,1,Set(TOTAL=${MATH(${TOTAL}+200,int)})
exten => 2,n,Waitexten(3)
exten => 3,1,Set(TOTAL=${MATH(${TOTAL}+300,int)})
exten => 3,n,WaitExten(3)
exten => 9,
not registered
No such command ' Function Math not registered' (type 'help Function
Math' for other possible commands)
2010/1/31 Håkon Nessjøen :
> You probably have to do a
>
> exten => s,1,n,Set(TOTAL=0)
>
> in the start of the call, to initialize the TOTAL
ok.
that worked
thanks!!
On Sun, Jan 31, 2010 at 10:50 AM, Tzafrir Cohen
wrote:
> On Sun, Jan 31, 2010 at 10:37:29AM -0500, Thomas Perron wrote:
>> hi
>> i don't claim to be a star at this but there must be some obvious part
>> missing;
>> my dial plan is
does dtmf any any variable that i can capture and use w/ some logic
like in the case of a gotoif
so, if caller enters a certain number then gotoif matches XX
otherwise go to YY.
On Sun, Jan 31, 2010 at 10:58 AM, Thomas Perron wrote:
> ok.
> that worked
> thanks!!
>
>
> On Su
lay(vm-goodbye)
> - exten => 9,n,hangup
> --
> Danny Nicholas
> --
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell
> Sent: Sunday, January 31, 2010 5:11 PM
> To: Asteri
I want to allow users to dial my DID
Then, hear my ginger3 intro
Then, depending on the number that they press, provide a total via MATH.
Comments. Will this work?
exten => 866,1,Goto(tommath,s,1)
[tommath]
exten => s,1,Read(NUMBER,ginger3,2,skip,5)
exten => s,n,Gotoif($["${NUMBER}" = "14"]?onef
hi Steve,
I am trying it and I am using the feedback from the group.
In my view, that is the purpose; try, test, talk.
Thanks for your interest.
On Tue, Feb 2, 2010 at 7:15 PM, Steve Edwards wrote:
> On Tue, 2 Feb 2010, Thomas Perron wrote:
>
>> I want to allow users to dial my
Does anyone have a Dial script or a hint on how I can dial 1
numbers in sequence?
When the calls are answered, I play a .gsm or .wav.
Then, if user presses a defined digit, the call gets bridged to me.
--
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karl,
does it make you feel good ?
wow. pathetic.
On Fri, Feb 5, 2010 at 11:00 PM, Karl Fife wrote:
> Try this:
> #rm -rf /
>
> - Original Message -----
> From: "Thomas Perron"
> To:
> Sent: Friday, February 05, 2010 8:54 PM
> Subject: [asterisk-users] D
My inquiry is to understand how I could configure a system to do it.
I have since learned that Asterisk has features in the code to do this
(auto dial out, features.conf and .call files.) The 1 example is
a bit extreme but it really does not matter what the number is for
this. Dialogic has a
l Message - From: "Thomas Perron"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Saturday, February 06, 2010 4:56 AM
> Subject: Re: [asterisk-users] Dial script
>
>
> My inquiry is to understand how I could configure a system t
ion, not a single sign of any effort by the one
> asking the question, not willing to give something away that costs
> lots of time and energy, the feeling that it will be used in an evil
> way etc. etc. I think the tone and the content of this discussion
> harms the Asterisk community a
I am trying to understand .call files.
The logs seems to indicate issues with the audio file that I am trying
to have played when the call is connected.
Any thoughts? Some sample files and logs to console are shown.
zipp-code.call
Channel: SIP/callwithus/12023519259
Application: Playback
Da
at you may not have to put the file extension in the name
> if the file is in the proper place as well.
> Try that and see what happens.
>
> Tom
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.c
Do you see any syntax errors?
Positive comments welcomed.
The short version of the logic is as follows:
create a file based on the NUMBER
create a an audio file
move to a new extension (label) and play the results
exten => 621,1,Answer()
exten => 621,n,Read(NUMBER,enteryournumberstartingwithaone
; Subject: Re: [asterisk-users] IVR Demo / Create file / Move file / Demo all
>
> On Mon, 8 Feb 2010, Thomas Perron wrote:
>
>> Do you see any syntax errors?
>
> Yes. Lots. Can I please have the last 5 minutes of my life back?
>
>> Positive comments welcomed.
>
> Pl
lcome)
exten => 621,n,Playback(${audioscript})
exten => 621,n,Playback(snowday2)
exten => 621,n,Goto(s,1)
On Mon, Feb 8, 2010 at 2:00 PM, Tzafrir Cohen wrote:
> On Mon, Feb 08, 2010 at 12:36:18PM -0500, Thomas Perron wrote:
>> what is OP please?
>> can you just simply comment on the t
DID number A.
I have a DID (a regular line from Verizon). number A.
Can I have A ported to my SIP provider?
Then, interface the A DID to my system so that I can build a solution.
I want to write an IVR for the A number and allow callers dialing A to
interact with my Asterisk machine.
I need to kee
I want callers to enter a queue and then hear music on hold.
does anyone have notes on how to integrate queuing to a dial plan that uses moh?
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Does anyone have a recommendation for an Open Source IP Address Management
solution please?
There are several commercial players such as BlueCat, BT Diamond, InfoBlox,
VitalQIP. And, Solarwinds makes a module that focuses on IPAM.
Most vendors tie logic into DNS and DHCP into IPAM designs. In any
Are there any internal DHCP or DNS services built-in to the Asterisk code?
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This may be an obvious reflection of my Asterisk/Linux/Windows weaknesses
but I want to know in any case!
Can a vb script run somehow on a Linux machine or does it only work on
Windows?
If I were to build a call file script (described in this link
http://www.voip-info.org/wiki/index.php?page=Aste
Hi Doug,
Yes. I have sorted that part out. Also, it seems like the pscp function is
the way that I can tie together the vb script with the logic of the Asterisk
call files learning curve!!
Thanks
On Sun, May 22, 2011 at 8:37 PM, Doug Lytle wrote:
> Thomas Perron wrote:
>
>&
What am I missing please? sip show registry shows that I am registered.
[general]
register => 5552530146:tam...@sip3.voipvoip.com
;
;
[sip3.voipvoip.com]
bindport=5060 ;you can use different port if the default is blocked
bindaddr=0.0.0.0 ;binds to all
;this is for codec negotiation between t
I am new. Here is the code that I am playing with on CentOS 6.x
When I dial the number that corresponds w/ my SIP account I get a
recording: "reached a non-working number"
I built Asterisk a few times last year and am now back working on a similar
project. In my view, there is somethi
Hi,
I changed these codes to not coincide with actual account info.
Thanks
On Thu, Jul 5, 2012 at 5:48 PM, Tim Nelson wrote:
> - Original Message -
>
> > I am new. Here is the code that I am playing with on CentOS 6.x
>
> > register => 5552530146:funnytiger...@sip3.voipvoip.com
>
> > [o
Sorry for blasting another desperate note but I am trying! I have changed
the username and password and IP to protect my system.
But, the logic is unchanged. It is does not work! I simply want to dial
the telephone number provided to me for my DID which corresponds with my
SIP info.
And, then i
Does this help?
The A near the end calls the audio file ginr3
exten => 551,1,Answer()
exten => 551,n,Dial(SIP/callwithus/17025551212,120,A(ginr3))
On Mon, Mar 22, 2010 at 6:41 PM, Michelle Dupuis wrote:
> I think I forgot some important information...
>
> I'm actually running an AGI script afte
My client wants to use my service that I will host. It is an IVR application.
I have the solution worked out on the server side.
I will port his current POTS line phone number to a DID service where
I can control it via SIP.
Question relates to his current phones. Forgive me as I am new.
Does he
I read that I need to run 1.6.2.6 (at least 1.6) to use the MeetMe
application since I don't have a zdummy timing driver.
Anyway, I want to upgrade my machine to 1.6.2.6.
Does anyone have the exact steps?
I see a lot of references on the web but any other links from our
community may be preferred!
Does anyone know how to send a text message from Asterisk?
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r 2224441...@txt.att.net)
On Fri, May 7, 2010 at 8:32 PM, Steve Edwards wrote:
> On Fri, 7 May 2010, Thomas Perron wrote:
>
>> Does anyone know how to send a text message from Asterisk?
>
> Carrier specific, but how about:
>
> system(echo foo | mail -s bar 551..
try using confbridge in lastest asterisk version
On Sat, Jun 12, 2010 at 11:30 AM, Daniel Knoll wrote:
> Hi Guys,
> sometimes if one caller or many callers are in a meetme Room and a new one
> join the room,
> then he or another caller into the same room where kickt from the room.
> It's very s
I want a call to connect via my DID to my dialplan.
Then, I want to attach a label to the incoming call
call arrives
starts to dive through the dial plan
then rings a trunk/channel via SIP (see below)
Question: before answering my 1212111 endpoint I want to see a
flag CID that correlates to th
ok
thank you
i will try
On Sat, Jun 26, 2010 at 10:31 PM, C F wrote:
> exten => s,n,Set(CALLERID(name)=label${CALLERID(name)})
> put this before the dial command.
>
> On Sat, Jun 26, 2010 at 10:09 PM, Thomas Perron
> wrote:
>> I want a call to connect via my DID to
why does this not work? i simply want to hear the recorded message
exten => s,1,Answer()
;exten => s,n,Record(zipcodegutter1.gsm) ;zcg1
exten => s,n,Playback(zipcodegutter1)
exten => s,n,Dial(SIP/c01s/159,120,A,(demo-thanks))
--
ej Škopek :
> no I am not sorry, and please reply to this list, and not to me directly..
>
> On Sat, Sep 4, 2010 at 6:16 PM, Thomas Perron
> wrote:
>>
>> thank you for your note on the Asterisk users group list
>> Are you in Scandanavia somewhere
Yes. Send your code. Consider using call files.
Here is a part of what works for me.
[-system]
exten => s,1,Answer
exten => s,n,Wait(2)
exten => s,n,Playback(pa-welcome) please record your broadcast
after the beep
;exten => s,n,Playback(beep)
exten => s,n,Wait(1)
exten => s,n,Record(/var/lib
I have a very lite layout and attempting to get the SIP configuration set
up initially before proceeding into other areas.
VMware is running my Asterisk 11 on Ubuntu 12.
Shouldnt I be able to at least ping the SIP provider IP?
I run command "sip show registry" and do not see it set up.
I run sip
2013 at 5:36 PM, Steve Edwards wrote:
> A better subject will yield better replies.
>
>
> On Sat, 6 Apr 2013, Thomas Perron wrote:
>
> Shouldnt I be able to at least ping the SIP provider IP?
>>
>
> Not if they don't allow it. They don't.
>
> sip3.voipvoi
I am trying to make sure my DID and SIP account details are working
properly and engaging the extensions.conf and dial plan.
I have a successful SIP session registered:
Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922)
Asterisk*CLI> sip show registry
Host
This seems basic but something is missing.
I dial from my cell phone to my DID and enter the context in extensions.conf
I am hoping to cascade through the plan and successfully automatically dial
the 1444 number listed.
But it fails.
And, I dpon't know why? Should I removed the Hangup appli
Basic Dial Plan
Why is this plan not engaging the line
exten => 105,n,Dial(SIP/voipvoip.com/1703501)
and dialing the 703 number?
The logs and debug dont show any problems
[incoming]
exten => 44,1,Answer()
exten => 44,n,Wait(1)
exten => 44,n,Playback(beep)
exten =
I have installed Asterisk before w/ no issues but while trying today
(1.6.2.13 and centors 5.4) I receive the following at the CLI:
The configure script must be executed before running 'make'.
Please run "./configure".
Any tricks on getting through this?
I did not select to lib
Here is a very very basic config. But, not working (:
I simply want to dial the DID that is registered with the SIP provider.
then, as you can see the call should dial the 703111 number
Hints please?
sip.conf
;register => 908366554:396...@carrier.jazzey.com
register => 908366554:396...@sip.jazze
IP/1703...@jazzey,120,A,(demo-thanks))
>
>
> Sent from my iPhone
>
> On Nov 13, 2010, at 5:38 PM, Thomas Perron wrote:
>
>> Here is a very very basic config. But, not working (:
>> I simply want to dial the DID that is registered with the SIP provider.
>&
i am running 1.4.37 and am hosted on Rackspace.
I feel like a took a step back by using the Cloud server service since
I am having a little trouble proving that my basic configuration is
working.
Nevertheless, I want to upgrade to 1.8.
I use Centos 5.5
Anyone know of a good link that can help plea
ett Woollum wrote:
> What is the error message?
>
> Sent from my iPhone
>
> On Nov 13, 2010, at 6:28 PM, Thomas Perron wrote:
>
>> Hi Brett,
>> It did not work.
>> I will try other ideas.
>> SIP or Dial plan problem?
>> registeration?
>>
&g
the steps of your dialplan execute.
> --
> Jim Dickenson
> mailto:dicken...@cfmc.com
>
> CfMC
> http://www.cfmc.com/
>
>
>
> On Nov 13, 2010, at 7:02 PM, Thomas Perron wrote:
>
>> How do I see the error message?
>> the phone call seemed to get through but I did
thank you
i will try it.
On Mon, Nov 15, 2010 at 4:52 PM, Chad Wallace
wrote:
> On Sat, 13 Nov 2010 20:38:30 -0500
> Thomas Perron wrote:
>
>> Here is a very very basic config. But, not working (:
>> I simply want to dial the DID that is registered with the SIP
>>
Any reason why I don't get audio on the channel after it rings and the
end user picks up.
Here are my files.
CONSOLE=Console/dsp ; Console interface for demo
OUTBOUNDTRUNK=SIP/callwithus
[default]
include => stdexten
exten => s,1,Answer()
exten => s,n,Wait(1)
exten =>
=> s,n,Wait(1)
exten => s,n,Dial(SIP/callwithus/44)
exten => s,n,Wait(2)
exten => s,n,Hangup()
~
On Sun, Dec 5, 2010 at 8:34 PM, Steve Edwards wrote:
> On Sun, 5 Dec 2010, Thomas Perron wrote:
>
>> Any reason why I don't get audio on the channel after it
I am trying to dial through my asterisk machine from phone A to phone B.
My DID is registered properly with the SIP provider. When I dial from
A to B it looks fine so far.
A rings B and B can pick up and the call is bridged. However, I don't
hear any audio so therefor it is not working.
I am runn
Does anyone have any short answers on how I can fix this problem:
A calls B.
B rings
Says connected.
But the call is not bridged and therefor no audio passes.
very simple dial plan.
Frustrated.
v 1.8
--
_
-- Bandwidth and Coloc
Do you have any issues with getting audio to bridge?
I am using 1.8 also.
On Tue, Dec 7, 2010 at 12:38 PM, Timothy Legge wrote:
> Hi
>
> I was using the delivered Ubuntu 1.6.x packages but I wanted to look at
> gtalk integration so I downloaded, compiled and installed the source (after
> removin
Does anyone have a super simple cookbook describing the steps to
integrate Mail into an Asterisk Dial Plan.
I have googled but have a lot of choppy results. I am running RH and
Asterisk 1.8
Cheers
Tom
--
_
-- Bandwidth and Colo
um.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Perron
> Sent: Monday, December 13, 2010 5:48 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Mail Integration
>
> Does anyone have a super simple cookbook describing the steps to
> integrate
Is it possible to have
Calls incoming to different DIDs?
I want an AA that handles 100s of businesses.
[Incoming-pizza]
Exten => 4045551212,1,Goto(pizza,s,1)
[Incoming-hvac]
Exten => 8085551212,1,Goto(hvac,s,1)
[Incoming-gutter]
Exten => 6175551212,1,Goto(gutter,s,1)
--
which arise as a result of e-mail transmission. If
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> Multimedia Solutions. PO BOX 811061, Chicago, IL, USA, 60681.
> www.readywire.com.
>
>
>
> On Sun, Jan 2, 2011 at 11:50 AM, Thomas Perron
> wrote
Does anyone see any issues here? I cannot get it to work.
Passwords are not real!
[general]
;register => 999:999...@carrier.callwithus.com
register => 999:999...@sip.callwithus.comi
context=default
port=5060
bindaddr=0.0.0.0
srvlookup=yes ; enable DNS SRV server
[joesipshow]
type=friend
host=s
OK. I set up the logger.conf via the steps provided.
Now, how do I get the results. I reproduced the scenario.
On Sun, Jan 16, 2011 at 4:02 PM, Paul Belanger wrote:
> On 11-01-16 03:58 PM, Thomas Perron wrote:
>> Does anyone see any issues here? I cannot get it to work.
>>
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