[asterisk-users] Homer SipCapture

[Bryan Anderson]

Is anyone using Homer from sipcapture.org or anything like it for capture
sip traffic for debuging?  If so what are your experiences.

Thanks
-Bryan Anderson
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Re: [asterisk-users] GotoIf($["${CALLERID(number)}

[Joseph]

On 06/14/13 18:43, Noah Engelberth wrote:

-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Joseph
Sent: Friday, June 14, 2013 2:25 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] GotoIf($["${CALLERID(number)}

I'm trying to to to "dial1" if caller id match:
but dial plan execute  220,n(dial1) regardless

exten => 220,n,GotoIf($["${CALLERID(number)}" = "7804792668"]?dial1)
exten =>
220,n(dial1),Dial(${sales_support}&${accounting}&${family},25,m(penguin)w
)
exten => 220,n,

I was under impression that if condition is met it will jump to  "n(label)" no
comma in between
but dial plan is executing it regardless.

--
Joseph

--


You're correct in that GotoIf($[CONDITION]?label) will jump to the indicated 
label if the condition is true.

However, without being told to go somewhere else when the condition is false, 
Asterisk will keep going at the next priority of the current extension.

So, since your dialplan is something like:

exten => 220,3,GotoIf(...)
exten => 220,4(dial1),Dial(...)
exten => 220,5,...

When Asterisk sees the Goto as true, it jumps to the label dial1, which happens 
to be priority 4 (or whatever it actually is).  When Asterisk sees the Goto as 
false, it's not being told to do anything, so it moves on from priority 3, to 
the next available priority -- which happens to be 4.

If you want Asterisk to terminate the call when the condition of your Goto 
isn't matched, then you could change your dialplan to something like...

exten => 220,n,GotoIf($["${CALLERID(number)}" = "7804792668"]?dial1:hangup,1)
exten => 
220,n(dial1),Dial(${sales_support}&${accounting}&${family},25,m(penguin)w
)
exten => 220,n,...

exten => hangup,1,Hangup()

Or, you could of course jump anywhere else and do anything else with the call 
you want to.


Thank you,

Noah Engelberth
MetaLINK Technologies


Thanks Noah, you are correct; I knew I'm missing some logic here.

The correct way of doing it was to have only one line pointing to internal 
existing extension:
exten => 220,n,GotoIf($["${CALLERID(number)}" = "7804792668"]?internal,222,1)

and if the condition is not true, the dial plan will keep continue on.

--
Joseph

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Re: [asterisk-users] asterisk fax in debian

[jg]

Let 123456 be your fax number, iaxmodem0 the account name of your IAX 
modem (configured in iax.conf), then in extensions.conf you would have 
something like


[from-pstn...]

exten => 123456,1,Verbose(1,Incoming fax...)
same => n,Dial(IAX2/iaxmodem0,40)
same => n,Hangup()

and for outgoing facsimiles in context [fax-out]

[fax-out]
exten => _X.,1,Verbose(1,Outgoing fax...)
same => n,Dial(...${FILTER(0-9,${EXTEN})},40)
same => n,Hangup()

Email sending is configured inside Hylafax.

jg

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Re: [asterisk-users] asterisk fax in debian

[Steve Totaro]

On Thu, Jun 13, 2013 at 12:29 PM, vortex  wrote:

> Hello. i am running debian 6 with asterisk 11.4. The system has exim4 to
> send to email the voicemails.
> i would like to get rid of the analog fax machine and use asterisk to
> send/receive faxes.
> I do have a PSTN line with a SPA3102 adapter to interface it to asterisk.
> The number of the PSTN line is dedicated to faxing only. So i would like to:
> -receive faxes to asterisk and then send it as PDFs to an email address
> -Send from my PC a fax directly.
>
> is there any guide on how to do that since i got lost with all of it?
>
>
>
I would go with HylaFAX.  FAX is an art with any VoIP solution.  The best
art I have done and seen turned out to use HylaFAX.

Thanks,
Steve Totaro
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Re: [asterisk-users] I need a second opinion on a new phone systemdeployment

[Terry Brummell]

Another option instead of 2 servers dedicated as PRI gateways is to use
AudioCodes Mediant 1000 or 2000 gateways.  Either of them will also
failover to a backup proxy if the primary proxy (server) is offline.
Probably much cheaper than the kick ass box you plan to build + PRI
card(s).

I'm not affiliated either, but we do place them in our 911 call centers.
They have analog gateways as well for FXO & FXS devices.

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Totaro
Sent: Friday, June 14, 2013 3:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] I need a second opinion on a new phone
systemdeployment

 

http://red-fone.com 
might be a good place look and see if other ideas pop up.  They have
good products.  I am not affiliated with them, just a happy user on a
couple of deployments.  

 

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Re: [asterisk-users] I need a second opinion on a new phone system deployment

[Steve Totaro]

http://red-fone.com  might be
a good place look and see if other ideas pop up.  They have good products.
 I am not affiliated with them, just a happy user on a couple of
deployments.


On Fri, Jun 14, 2013 at 11:43 AM, Nunya Biznatch
wrote:

> Howdy All,
>They say opinions are like belly buttons, everybody has one. (that's
> the "clean" version of the saying). So I'm asking for yours. I hope you see
> it as a fun exercise.
>
> I'm designing a phone system from the ground up. Will be about 1000-1300
> seats mixed 80/20 VoIP/Analog. 58-acre campus environment with 23
> buildings. Userbase is emergency services organization, 24/7/365 operation.
> Down time is not an option, but "blips" are acceptable. Repair time is
> immediate. We need failover for the failover essentially. However, money is
> a major factor, so I have to do it all for nothing. So here's what I'm
> thinking. Please throw in your 2 cents.
>
> Network will be separate for phones. Fiber infrastructure available
> between buildings as well as copper. Internet access will be limited to a
> single administrative console on a temporary basis, and then only when
> remote 3rd party support is required. Access for 3rd party support will be
> supervised through remote access tools such as VNC, GoToMeeting, etc...
> etc... System will have zero access to local data network. This means all
> ancillary support servers such as DHCP, DNS, NTP, FTP, etc...etc... will be
> specific to the phone system. Yes, I know some responders at this time will
> become fixated on me gaining this connectivity. It ain't gonna happen. It's
> not an option. Period, end of story. These are the parameters I must work
> within. Trying to "fix" that will be a non-starter.
>
> The phone system will upgrade an existing TDM-based system. Mitel SX2000
> with NuPoint Voicemail. This will not be a dump-trunk replacement. I expect
> at least a one to two-year transition, meaning we will have time to find
> problems,  work bugs, and learn over time, with minimized impacts. It also
> means we'll be supporting two systems for some time.
>
> PBX is 97% serving your basic phone on the desk. Nothing special.
> Customers expect the usual list of features. There will be a goodly number
> of hints required for BLF on maybe 150 phones. There is one office of about
> 30 phones in a call-center environment that will need that service. They
> would be considered low volume (but don't tell them that).
>
> My Skills... I am not a Linux kung fu master, but I have built and managed
> my share of Linux servers on mutiple Linux flavors. I am a DCAA, having
> been through formal training, and have been playing with Asterisk for
> years, but always in fits and spurts and never in a live environment so I
> am by no means a kung fu master there either. I have started dabbling with
> virtualizations via XEN, but I am not comfortable enough with it to go live
> this first round. I can see myself implementing it in about three years
> once we're totally comfortable with what we have, so I can then have time
> to get that skill sorted. I was a network engineer for the US no3. telecom
> for a number of years, 10-years in comm-electronics in the military before
> that. Telecom my entire career. I've got the kung-fu to handle the network
> side of the house, and having administrated multiple PBXs for decade-plus,
> I've got the concepts down.
>
> No plans to build databases for things like directories, etc... I'm not
> greatly confident in those skills, and to date, haven't found anything that
> really stands out that would make me require that. You may think otherwise,
> so please chime in. I say that, but at the same time I recognize I may
> require a GUI interface once fully deployed to allow lower-skilled people
> to follow the motions to complete simple moves, adds, and changes. I'm
> fighting the uphill battle that is the "GUI is new, CLI is old" mentality.
>
> System will use G.722 for VoIP Phones.
>
> So there's the groundwork. Here's the hardware plan.
>
> Plan is to build my own servers following industry standards (ATX) and
> using industry standard equipment. Why? Spares? Whether redundant or not, I
> will still have spares for the most common elements on the shelf so
> equipment can be returned to service as quickly as possible. This will also
> allow me to be comfortable with more "basic server" configurations and help
> keep cost down. For example, Servers with single power supplies vs. dual.
> Also, components will be standardized for all equipment to aid in supply
> requirements.
>
> First the layout.
>
> 2-servers acting as gateways. Each handling 2 PRIs for outside trunks.
> They'll also handle the analog ports. Failover will be in the form of
> degraded trunk access if one should fail, but the second will be able to
> support services in degraded fashion.
>
> 2-servers acting as VoIP PBX. A primary and a spare. Meaning one will be
> capable of han

Re: [asterisk-users] I need a second opinion on a new phone system deployment

[Daniel Tryba]

On Fri, Jun 14, 2013 at 09:43:29AM -0600, Nunya Biznatch wrote:
> System will use G.722 for VoIP Phones.
[...]
> 2-servers acting as gateways. Each handling 2 PRIs for outside
> trunks.

So why use g722? Just use your local g711 law and thus avoid the
transcoding impact to/from the PSTN and calls between the voip and analog
users.

And why would you seperate the PBX and PRI machines? Those few extra
channels don't really matter.


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Re: [asterisk-users] asterisk fax in debian

[vortex]

I did install hylafax and iaxmodem. Everything installed correctly and 
the services do run OK. The problem is the sip.conf and extensions.conf. 
I do not know how to link the PSTN number 123456 to work as fax and then 
send as email.



Hi. I have seen these kind of instructions but there i lost it.
here is what i got.
Asterisk has a spa3102 to interface the PSTN line. It works smoothly 
and i got in/outgoing calls. i do have the codec to g711alaw (since i 
am in Europe). on the fxs port of the spa3102 i had the fax machine up 
to now and it worked.
in the asterisk i installed spandsp and recompiled asterisk. i have 
losted though with the config to receive/send faxes.








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Re: [asterisk-users] ILEC Interconnect: Basic MUX: M13 vs DCS: VT1.5 vs DS3

[Nick Khamis]

Hello James, thank you so much for your response!

On 6/14/13, James Cloos  wrote:
> If they will do atm over oc-n, perhaps that would work better.

Yes they will do atm over oc-n only not sure if they will ring or spur it...

> Ie, a perm virt circ for SS7 and as-needed vc's for ulaw.

I know you're a busy guy, can you please expand on this kindly :).

>
> Atm oc-n cards with linux sw support are widely available, according to
> goog.
>
> libss7 and and ast *might* need a bit of patching to work with it, but it
> shoudn't take too much.

Will goog some more!

>
> Sip/rtp over private ptp ethernet is an option with at least some of the
> ILECs.
>
> They may call it virtual-pri or some such.
>
> Of course, if they are installing an actual sonet ring, and not just a
> spur, that can have built-in redundancy, depedning on physical routing.

We are trying to position ourselves as facility based virtual PRI service
provider here, and would like to put something of our own together vs.
resell another LEC's red ribbon product.

>
> -JimC
> --
> James Cloos  OpenPGP: 1024D/ED7DAEA6
>

PS I love your website!!!

Kinid Regards,

Nick.

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Re: [asterisk-users] GotoIf($["${CALLERID(number)}

[Noah Engelberth]

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Joseph
> Sent: Friday, June 14, 2013 2:25 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] GotoIf($["${CALLERID(number)}
> 
> I'm trying to to to "dial1" if caller id match:
> but dial plan execute  220,n(dial1) regardless
> 
> exten => 220,n,GotoIf($["${CALLERID(number)}" = "7804792668"]?dial1)
> exten =>
> 220,n(dial1),Dial(${sales_support}&${accounting}&${family},25,m(penguin)w
> )
> exten => 220,n,
> 
> I was under impression that if condition is met it will jump to  "n(label)" no
> comma in between
> but dial plan is executing it regardless.
> 
> --
> Joseph
> 
> --

You're correct in that GotoIf($[CONDITION]?label) will jump to the indicated 
label if the condition is true.

However, without being told to go somewhere else when the condition is false, 
Asterisk will keep going at the next priority of the current extension.

So, since your dialplan is something like:

exten => 220,3,GotoIf(...)
exten => 220,4(dial1),Dial(...)
exten => 220,5,...

When Asterisk sees the Goto as true, it jumps to the label dial1, which happens 
to be priority 4 (or whatever it actually is).  When Asterisk sees the Goto as 
false, it's not being told to do anything, so it moves on from priority 3, to 
the next available priority -- which happens to be 4.

If you want Asterisk to terminate the call when the condition of your Goto 
isn't matched, then you could change your dialplan to something like...

exten => 220,n,GotoIf($["${CALLERID(number)}" = "7804792668"]?dial1:hangup,1)
exten => 
220,n(dial1),Dial(${sales_support}&${accounting}&${family},25,m(penguin)w
)
exten => 220,n,...

exten => hangup,1,Hangup()

Or, you could of course jump anywhere else and do anything else with the call 
you want to.


Thank you,

Noah Engelberth
MetaLINK Technologies

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[asterisk-users] GotoIf($["${CALLERID(number)}

[Joseph]

I'm trying to to to "dial1" if caller id match:
but dial plan execute  220,n(dial1) regardless

exten => 220,n,GotoIf($["${CALLERID(number)}" = "7804792668"]?dial1)
exten => 
220,n(dial1),Dial(${sales_support}&${accounting}&${family},25,m(penguin)w)
exten => 220,n, 


I was under impression that if condition is met it will jump to  "n(label)" no 
comma in between
but dial plan is executing it regardless.

--
Joseph

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Re: [asterisk-users] blocking spammer by callerID "name"

[Greg Woods]

On Thu, 2013-06-13 at 18:29 -0600, Joseph wrote:
> When I play:
> exten => s,n,Background(welcome)
> and press extension "1" the system will not jump to this extension 
> immediately, there is a few sec. pause.

Mine looks like this:

exten => s,1,Answer()
exten => s,2,Set(TIMEOUT(digit)=1)
exten => s,3,Background(greeting)
exten => s,n,WaitExten(10)
exten => s,n,Goto(3)

I think setting the digit timeout does the trick for me. (My greeting
file says at the end, "stay on the line and this menu will repeat", and
of course it does until the caller hangs up.

--Greg



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Re: [asterisk-users] ILEC Interconnect: Basic MUX: M13 vs DCS: VT1.5 vs DS3

[James Cloos]

If they will do atm over oc-n, perhaps that would work better.

Ie, a perm virt circ for SS7 and as-needed vc's for ulaw.

Atm oc-n cards with linux sw support are widely available, according to goog.

libss7 and and ast *might* need a bit of patching to work with it, but it
shoudn't take too much.

Sip/rtp over private ptp ethernet is an option with at least some of the ILECs.

They may call it virtual-pri or some such.

Of course, if they are installing an actual sonet ring, and not just a
spur, that can have built-in redundancy, depedning on physical routing.

-JimC
-- 
James Cloos  OpenPGP: 1024D/ED7DAEA6

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[asterisk-users] I need a second opinion on a new phone system deployment

[Nunya Biznatch]

Howdy All,
   They say opinions are like belly buttons, everybody has one. (that's 
the "clean" version of the saying). So I'm asking for yours. I hope you 
see it as a fun exercise.


I'm designing a phone system from the ground up. Will be about 1000-1300 
seats mixed 80/20 VoIP/Analog. 58-acre campus environment with 23 
buildings. Userbase is emergency services organization, 24/7/365 
operation. Down time is not an option, but "blips" are acceptable. 
Repair time is immediate. We need failover for the failover essentially. 
However, money is a major factor, so I have to do it all for nothing. So 
here's what I'm thinking. Please throw in your 2 cents.


Network will be separate for phones. Fiber infrastructure available 
between buildings as well as copper. Internet access will be limited to 
a single administrative console on a temporary basis, and then only when 
remote 3rd party support is required. Access for 3rd party support will 
be supervised through remote access tools such as VNC, GoToMeeting, 
etc... etc... System will have zero access to local data network. This 
means all ancillary support servers such as DHCP, DNS, NTP, FTP, 
etc...etc... will be specific to the phone system. Yes, I know some 
responders at this time will become fixated on me gaining this 
connectivity. It ain't gonna happen. It's not an option. Period, end of 
story. These are the parameters I must work within. Trying to "fix" that 
will be a non-starter.


The phone system will upgrade an existing TDM-based system. Mitel SX2000 
with NuPoint Voicemail. This will not be a dump-trunk replacement. I 
expect at least a one to two-year transition, meaning we will have time 
to find problems,  work bugs, and learn over time, with minimized 
impacts. It also means we'll be supporting two systems for some time.


PBX is 97% serving your basic phone on the desk. Nothing special. 
Customers expect the usual list of features. There will be a goodly 
number of hints required for BLF on maybe 150 phones. There is one 
office of about 30 phones in a call-center environment that will need 
that service. They would be considered low volume (but don't tell them 
that).


My Skills... I am not a Linux kung fu master, but I have built and 
managed my share of Linux servers on mutiple Linux flavors. I am a DCAA, 
having been through formal training, and have been playing with Asterisk 
for years, but always in fits and spurts and never in a live environment 
so I am by no means a kung fu master there either. I have started 
dabbling with virtualizations via XEN, but I am not comfortable enough 
with it to go live this first round. I can see myself implementing it in 
about three years once we're totally comfortable with what we have, so I 
can then have time to get that skill sorted. I was a network engineer 
for the US no3. telecom for a number of years, 10-years in 
comm-electronics in the military before that. Telecom my entire career. 
I've got the kung-fu to handle the network side of the house, and having 
administrated multiple PBXs for decade-plus, I've got the concepts down.


No plans to build databases for things like directories, etc... I'm not 
greatly confident in those skills, and to date, haven't found anything 
that really stands out that would make me require that. You may think 
otherwise, so please chime in. I say that, but at the same time I 
recognize I may require a GUI interface once fully deployed to allow 
lower-skilled people to follow the motions to complete simple moves, 
adds, and changes. I'm fighting the uphill battle that is the "GUI is 
new, CLI is old" mentality.


System will use G.722 for VoIP Phones.

So there's the groundwork. Here's the hardware plan.

Plan is to build my own servers following industry standards (ATX) and 
using industry standard equipment. Why? Spares? Whether redundant or 
not, I will still have spares for the most common elements on the shelf 
so equipment can be returned to service as quickly as possible. This 
will also allow me to be comfortable with more "basic server" 
configurations and help keep cost down. For example, Servers with single 
power supplies vs. dual. Also, components will be standardized for all 
equipment to aid in supply requirements.


First the layout.

2-servers acting as gateways. Each handling 2 PRIs for outside trunks. 
They'll also handle the analog ports. Failover will be in the form of 
degraded trunk access if one should fail, but the second will be able to 
support services in degraded fashion.


2-servers acting as VoIP PBX. A primary and a spare. Meaning one will be 
capable of handling the load of the entire system, and the other will 
pickup when the other dies, an active/passive cluster. Will also take 
care of voicemail. Use of heartbeat, pacemaker, etc... etc...


2-servers for support services. DNS, DHCP, FTP, NTP, etc... 
etc...Basically, everything the phones need to run plus system 
monitoring via something like Nagios.


1-Desktop for

Re: [asterisk-users] SIGTRAN Integration

[Mitul Limbani]

There is no open source solution for SIGTRAN yet.

If you come across one, do let everyone here know about it.

You can however request some time restricted demo from Netfors or LeibICT.

Mitul

On Friday, June 14, 2013, Nick Khamis wrote:

> Hello Mitul,
>
> Thank you so much for your response. During the testing phase
> we would like to employ an open source solution, and wanted
> to know what people have had success with, given the different
> user part etc..
>
> On a side note, anyone know of service providers offering SIGTRAN?
>
> Kind Regards,
>
> Nick.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>


-- 
Regards,
Mitul Limbani,
Chief Architech & Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967121
Cell: +91-9820332422
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Re: [asterisk-users] SIGTRAN Integration

[Nick Khamis]

Hello Mitul,

Thank you so much for your response. During the testing phase
we would like to employ an open source solution, and wanted
to know what people have had success with, given the different
user part etc..

On a side note, anyone know of service providers offering SIGTRAN?

Kind Regards,

Nick.

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Re: [asterisk-users] SIGTRAN Integration

[Mitul Limbani]

I think you need a SIGTRAN stack from Netfors or LeibICT.

Mitul

On Friday, June 14, 2013, Nick Khamis wrote:

> Hello Everyone,
>
> I was wondering how SIGTRAN/SS7oIP can fit into our currently 100% SIP
> model.
> We are looking to interconnect with the PSTN world, and our supplier
> has given us
> a few options. We can either do this over traditional PRIs, A-Links or
> the SS7IP new.
>
> I am really interested in SIGTRAN, and was wondering how some of you
> have integrated
> it into your architecture. Can Asterisk handle SS70IP or do we have to
> put a yate or squire
> server at the end of that connection.
>
> Kind Regards,
>
> Nick.
>
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-- 
Regards,
Mitul Limbani,
Chief Architech & Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967121
Cell: +91-9820332422
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[asterisk-users] SIGTRAN Integration

[Nick Khamis]

Hello Everyone,

I was wondering how SIGTRAN/SS7oIP can fit into our currently 100% SIP model.
We are looking to interconnect with the PSTN world, and our supplier
has given us
a few options. We can either do this over traditional PRIs, A-Links or
the SS7IP new.

I am really interested in SIGTRAN, and was wondering how some of you
have integrated
it into your architecture. Can Asterisk handle SS70IP or do we have to
put a yate or squire
server at the end of that connection.

Kind Regards,

Nick.

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Re: [asterisk-users] Executing Stored Procedure using ODBC MSSQL

[Grant Bagdasarian]

Note, that writing CDRs using ODBC to a MSSQL database does work. So I don't 
know why this doesn't.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Grant Bagdasarian
Sent: Friday, June 14, 2013 2:43 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Executing Stored Procedure using ODBC MSSQL

Hello,

I'm trying to execute a stored procedure on a MSSQL Server from the dial plan, 
but it's not working. I'm getting the following error: Unable to execute 
query

Asterisk has been compiled with UnixODBC, and I've done the necessary 
configurations in func_odbc, res_odbc and odbc.ini.

Has anyone done this before with success?
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[asterisk-users] Executing Stored Procedure using ODBC MSSQL

[Grant Bagdasarian]

Hello,

I'm trying to execute a stored procedure on a MSSQL Server from the dial plan, 
but it's not working. I'm getting the following error: Unable to execute 
query

Asterisk has been compiled with UnixODBC, and I've done the necessary 
configurations in func_odbc, res_odbc and odbc.ini.

Has anyone done this before with success?
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Re: [asterisk-users] Light-weight voice recognition for IVR

[Thorsten Göllner]

Hi,

some month ago we installed a VoiceRec-Module from Vestec 
(https://www.vestec.com/) on Asterisk 11.x. It works so far and you will 
find examples for your dialplan. It should be ok for your needs.


-Thorsten-

Am 13.06.2013 23:19, schrieb asterisk users:

Hello list,

'Just wondering if anyone can point to a very light-weight and easy to 
incorporate into Asterisk (v. 11.x) to handle a minimal set of 
responses, like:

   0 - 9
   yes
   no
   (maybe * and # for some people)

The idea is that within an IVR menu, the caller could respond by 
speaking to the typical IVR options, like:


For Archie, press or say 1 now
For Veronica, press or say 2 now
For Jughead, press or say 3 now
(etc.)

You have selected option 2 for Veronica, press 1 or say "yes" if 
this is correct.


If a voice response was received (not a DTMF key press) indeterminate, 
some status would be useful (beyond just a timeout).


It would be great if this was simple to code into the dialplan, much 
like like the current background/wait model for keypresses. Low cost 
or free would be nice too!


Thanks for any suggestions.


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Re: [asterisk-users] Codec Negotiation problem

[research]

Hi Matt

Thanks for your response. I have tried with two GXV3175 with same result.
Let me dig deep on this to find out the route cause

Sam
Matthew Jordan wrote:
> On Thu, Jun 13, 2013 at 12:04 PM,  wrote:
>
>> Hi there
>>
>> I have asterisk 10.11.1 which seems to have problem negotiating codec.
>>
>> Scenario: SIP PHONE1 (XLite) extension 1003, allowed codecs alaw, h263p
>> and SIP phone2 (Grandstream GXV3175) extension 1004, allowed codec alaw,
>> h263p. I have tried similar combination of codecs and SIP phone but when
>> making a video call, it report "Peer doesn't provide video". It seems
>> Asterisk is failing to set capability correct. Both codecs are enabled
>> on
>> the SIP Phones
>>
>>
> 
>
> The 200 OK response from the called XLite phone is declining the video
> stream:
>
> <--- SIP read from UDP:10.10.10.129:48464 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK368135b0;rport=5060
> Contact: 
> To: "SAM";tag=0c90cc0c
> From: ;tag=as24914503
> Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA
> CSeq: 102 INVITE
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> SUBSCRIBE, INFO
> Content-Type: application/sdp
> Supported: replaces, eventlist
> User-Agent: X-Lite release 4.5.2 stamp 70142
> Content-Length: 234
>
> v=0
> o=- 13015615910543193 2 IN IP4 10.10.10.129
> s=X-Lite 4 release 4.5.2 stamp 70142
> c=IN IP4 10.10.10.129
> t=0 0
> m=audio 53188 RTP/AVP 8 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sendrecv
> m=video 0 RTP/AVP 115
> <->
> --- (12 headers 10 lines) ---
> Found RTP audio format 8
> Found RTP audio format 101
> Found audio description format telephone-event for ID 101
> Capabilities: us - (alaw|h263p), peer -
> audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
>
> Note that the port for the video stream is set to 0.
>
> Asterisk is doing the correct thing: it notes that the answer to its offer
> declined the video stream, so it disables video for the call between the
> two endpoints.
>
> Matt
>
> --
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> Digium, Inc. | Engineering Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
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