Re: [asterisk-users] DAHDI and Oslec

[Christopher Harrington]

It's called "echo" in the kernel configs.


On Tue, Feb 26, 2013 at 11:09 AM, Doug Lytle  wrote:

> >>I encountered the same. Turns out, you need to disable OSLEC in your
> codec .config, and delete the modules in your /lib/modules/[kernel
> version]/drivers/staging directory, and then >> (in your kernel sources)
> >> make && make modules && make modules install
>
> There was nothing listed in /usr/src/linux/.config for OSLEC
> There were no modules in
> /lib/modules/3.6.9-custom-3.6.9/kernel/drivers/staging for OSLEC
>
> Doug
>
>
> --
> Ben Franklin quote:
>
> "Those who would give up Essential Liberty to purchase a little Temporary
> Safety, deserve neither Liberty nor Safety."
>
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Re: [asterisk-users] DAHDI and Oslec

[Christopher Harrington]

On Tue, Feb 26, 2013 at 10:38 AM, Doug Lytle  wrote:
>
> But, when trying to set my E.C. to oslec, I get:
>
> Feb 26 11:21:37 indyvoip modprobe: FATAL: Error inserting
> dahdi_echocan_oslec
> (/lib/modules/3.6.9-custom-3.6.9/dahdi/dahdi_echocan_oslec.ko): Unknown
> symbol in module, or unknown parameter (see dmesg)
> Feb 26 11:21:37 indyvoip kernel: [ 1395.262334] dahdi_echocan_oslec:
> Unknown symbol oslec_create (err 0)
> Feb 26 11:21:37 indyvoip kernel: [ 1395.262348] dahdi_echocan_oslec:
> Unknown symbol oslec_update (err 0)
> Feb 26 11:21:37 indyvoip kernel: [ 1395.262365] dahdi_echocan_oslec:
> Unknown symbol oslec_free (err 0)
> Feb 26 11:21:41 indyvoip modprobe: FATAL: Error inserting
> dahdi_echocan_oslec
> (/lib/modules/3.6.9-custom-3.6.9/dahdi/dahdi_echocan_oslec.ko): Unknown
> symbol in module, or unknown parameter (see dmesg)
> Feb 26 11:21:41 indyvoip kernel: [ 1398.799227] dahdi_echocan_oslec:
> Unknown symbol oslec_create (err 0)
> Feb 26 11:21:41 indyvoip kernel: [ 1398.799241] dahdi_echocan_oslec:
> Unknown symbol oslec_update (err 0)
> Feb 26 11:21:41 indyvoip kernel: [ 1398.799258] dahdi_echocan_oslec:
> Unknown symbol oslec_free (err 0)
>
> And dmesg shows:
>
> [ 1395.262334] dahdi_echocan_oslec: Unknown symbol oslec_create (err 0)
> [ 1395.262348] dahdi_echocan_oslec: Unknown symbol oslec_update (err 0)
> [ 1395.262365] dahdi_echocan_oslec: Unknown symbol oslec_free (err 0)
> [ 1398.799227] dahdi_echocan_oslec: Unknown symbol oslec_create (err 0)
> [ 1398.799241] dahdi_echocan_oslec: Unknown symbol oslec_update (err 0)
> [ 1398.799258] dahdi_echocan_oslec: Unknown symbol oslec_free (err 0)
>
>

I encountered the same. Turns out, you need to disable OSLEC in your codec
.config, and delete the modules in your /lib/modules/[kernel
version]/drivers/staging directory, and then (in your kernel sources)
make && make modules && make modules install

The dahdi tools actually contains an out-of-tree version that worked out of
the box.


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Re: [asterisk-users] Calendar: cert mismatch

[Christopher Harrington]

On Mon, Feb 25, 2013 at 3:18 PM, Phil Daws  wrote:

> Hi,
>
> Am testing out the iCal functionality but when changing the URL am faced
> with the following warning:
>
> [Feb 25 20:55:20] WARNING[6234] res_calendar_icalendar.c: Unable to
> retrieve iCalendar 'dummycal' from '
> https://webmail.domain.com/home/u...@domain.com/Calendar/': Server
> certificate verification failed: certificate issued for a different
> hostname, issuer is not trusted
>
> Has a cert been cached somewhere ?
>
> Have you actually verified that visiting https://[whatever] doesn't
generate a certificate validity warning in your browser?

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Re: [asterisk-users] exten => h,n,AGI(generateCall.php,${NEXT})

[Christopher Harrington]

On Wed, Feb 20, 2013 at 9:23 AM, Mahendra Dobariya <
mahendra_mahen...@hotmail.com> wrote:

>  File: /etc/asterisk/extensions.conf
> [call]
> exten => call,1,Answer
> exten => call,n,Playback(hello-world)
> exten => call,n,Hangup()
>
> exten => h,1,Set(NEXT=$[${NEXT}+1])
> exten => h,n,AGI(generateCall.php,${NEXT})
>

Try
exten =>
h,n,AGI(/usr/bin/php,/usr/share/asterisk/agi-bin/generateCall.php,${NEXT})


> exten => h,n,Hangup()
>




> AGI Rx << Could not open input file: 1
>
>
This is indicating that, for whatever reason, php is seeing "1" as argv[1],
not the name of your script file. I reproduced this by making a php shebang
that looks like
#!/usr/bin/php 1

Not sure why, though. The above should be a workaround for now.

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Re: [asterisk-users] Asterisk SMS()

[Christopher Harrington]

On Tue, Feb 19, 2013 at 10:12 AM, Nicholas Johnson wrote:

> On Feb 19, 2013, at 10:41 AM, Christopher Harrington wrote:
>
> On Tue, Feb 19, 2013 at 9:14 AM, Nicholas Johnson wrote:
>
>> All,
>>   I'm trying to send an SMS directly from asterisk but it doesn't seem to
>> be working.  The SMS() function does create an outgoing file but doesn't
>> deliver the SMS.  Can anyone help me to understand how SMS() works.  Thanks.
>>
>> extensions.conf example:
>>
>> same => n,SMS(hello,a,17654307001,"hello nick")
>>
>>
>>
> Let's start out by figuring out what hardware you have. Is Asterisk
> connected to the PSTN? What is physically delivering the SMS to the
> carriers?
>
> Also, when I run `core show application SMS` it talks about some software
> called smsq. Are you running that software?
>
> Thanks for the help.  Right now I'm running asterisk on a raspberry pi
> using a phone number from flowroute.  Is using a company like flowroute the
> same as connecting to the PSTN?  Also i've tried to install smsq but I
> couldn't find any good documentation to get it setup properly.  So no, I'm
> not using smsq.
>
>
I'm not well informed, but it appears that you need to (at a minimum)
provide some sort of interface to connect to a hardware interface for this
process. Google "ETSI ES 201 912".

Having looked at Flowroute, they don't appear to mention SMS anywhere on
their website, so I am going to go out on a limb and say that they will not
provide what you need to send an SMS.


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Re: [asterisk-users] Asterisk SMS()

[Christopher Harrington]

On Tue, Feb 19, 2013 at 9:14 AM, Nicholas Johnson  wrote:

> All,
>   I'm trying to send an SMS directly from asterisk but it doesn't seem to
> be working.  The SMS() function does create an outgoing file but doesn't
> deliver the SMS.  Can anyone help me to understand how SMS() works.  Thanks.
>
> extensions.conf example:
>
> same => n,SMS(hello,a,17654307001,"hello nick")
>
>
>
Let's start out by figuring out what hardware you have. Is Asterisk
connected to the PSTN? What is physically delivering the SMS to the
carriers?

Also, when I run `core show application SMS` it talks about some software
called smsq. Are you running that software?

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Re: [asterisk-users] Dialplan / check / tool

[Christopher Harrington]

On Mon, Feb 18, 2013 at 10:54 AM, Steve Edwards
wrote:

> ) If the AGI is [...] a compiled executable (C, Fortran, Cobol, assembler,
> etc.)


I'd like to see an AGI written using Fortran or Cobol.


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Re: [asterisk-users] ControlPlayback unable to play MixMonitor files, Premature EOF

[Christopher Harrington]

On Sun, Feb 17, 2013 at 6:59 PM, Mehmet Avcioglu wrote:

>
> I am recording wav files
>MixMonitor(/path/filename.wav,b)
> When I try to play them back, I get no audio, dialplan continues to the
> next line as if the sound file is 0 seconds long
>
Using Asterisk 1.8.19.
>
I've encountered the same problem recently with Asterisk 11. You might try
using StopMixMonitor prior to playback.


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Re: [asterisk-users] Dial failed due to trunk reporting BUSY - giving up

[Christopher Harrington]

On Saturday, February 16, 2013, Muhammad wrote:

> Hi
> this message give me when I calling a number than actually not busy:
> "Dial failed due to trunk reporting BUSY - giving up"
>
> max channel is unlimited and sometimes it dial number ok but most of the
> time it gives me this error.
>
> Please inform me how can solve this problem.
>
>
> thanks
>

Version, hardware, configuration... we can't read your mind.


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Re: [asterisk-users] auto install all required dependences for asterisk.

[Christopher Harrington]

On Fri, Feb 15, 2013 at 1:50 PM, Mahendra Dobariya <
mahendra_mahen...@hotmail.com> wrote:

> but it did not works..
>
>
Unfortunately, this is completely useless for anyone to help you. You will
need to actually tell us what isn't working and include output from your
console.

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Re: [asterisk-users] How to install in /usr/local/sbin instead of /usr/sbin ? [SOLVED]

[Christopher Harrington]

On Tue, Feb 12, 2013 at 7:43 AM, Olivier  wrote:

>
> Using the commands bellow, I could install in /usr/local/sbin
> ./configure --prefix=/usr/local
> make
> make install
>
> Thanks for sharing this.
>
>
Just for reference, this is standard among tools compiled using ./configure
; make ; make install . Most software works this way.

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Re: [asterisk-users] Asterisk calls between 2 private networks

[Christopher Harrington]

Digium phones, which (as far as I can tell with my experience) do not
support VPN yet.


On Thu, Feb 7, 2013 at 11:57 AM, Justin Killen <
jkil...@allamericanasphalt.com> wrote:

> Or if it's just a couple phones, you might be able to setup a vpn
> connection directly on the phone itself - have it vpn into 'HQ' and get an
> address on that network.  I'm not sure which phones you're using though or
> what phones support that setup.
>
> Justin Killen
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen
> Sent: Thursday, February 07, 2013 9:55 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk calls between 2 private networks
>
> I don't see how that would really solve anything - instead of the server
> sending the 192.168.x.x packets onto the local network, it will send them
> up toward the internet and get black-holed.  What probably makes more sense
> would be to switch the subnet on one of the networks, AND put up a vpn
> between them, adding the routes for the private networks to cross thru the
> tunnels.
>
> Justin Killen
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Frank
> Sent: Thursday, February 07, 2013 9:49 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Cc: Eric Wieling
> Subject: Re: [asterisk-users] Asterisk calls between 2 private networks
>
> I thought about that.
> I will give it a shot tonight and will post back my results in here.
> Thanks
>
> On 2/7/13 12:39 PM, Eric Wieling wrote:
> > The easiest thing to is renumber one of the networks so they are not
> using the same address block.
> >
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Frank
> > Sent: Thursday, February 07, 2013 12:27 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] Asterisk calls between 2 private networks
> >
> > AJS,
> >
> > That is a solution that I am envisaging.
> > But I would really love to try to work out with my issue first. It will
> allow me to deploy more phones in separates buildlings in the future. If I
> do the IAX solution, it means that for every building, I need a box..
> > Which I would like to prevent.
> >
> >
> >
> > On 2/7/13 10:46 AM, A J Stiles wrote:
> >> On Thursday 07 February 2013, Frank wrote:
> >>> My apologies if this topic was already discussed in the past.
> >>>
> >>> Here is my scenario:
> >>> Network A - 192.168.1.0
> >>> 1 Asterisk
> >>> 1 Digium phone
> >>> Router does NAT from the public IP to asterisk, and forward ports
> >>> 5060tcp/udp and 10k-20k udp
> >>>
> >>> Network B - 192.168.1.0
> >>> 1 Digium phone, registering to the public IP of network A
> >>>
> >>>
> >>> My SIP.CONF has:
> >>> nat=yes
> >>> localnet=192.168.1.0/255.255.255.0
> >>> externaddr=public_ip_of_network_a
> >>> directmedia=no
> >>
> >> My  (lazy)  solution to this problem was to throw hardware at it .
> >>
> >> Bearing in mind that Asterisk will run on just about any old scrapper
> >> (or even a Raspberry Pi, if you feel so inclined),  there's little
> >> point even trying to send SIP over the Internet.  Just have an
> >> Asterisk box at each end, and then you only need a much
> simpler-to-configure IAX trunk between the two.
> >> The routers at each end then just need one port -- UDP 4569 --
> >> forwarded to the Asterisk box  (if it isn't configured as the default
> DMZ machine).
> >>
> >>
> >
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Re: [asterisk-users] problem

[Christopher Harrington]

On Wed, Feb 6, 2013 at 9:30 AM, brahim abidar wrote:

> *
> error: cannot seek `/dev/sda'.
> error: cannot seek `/dev/sda'.
> error: cannot seek `/dev/sda'.
> /usr/sbin/grub-probe: error: cannot seek `/dev/sda'.
> *
>

I hope your hard drive is not /dev/sda. Although this could also be a
configuration issue.


> *
> dpkg: error processing grub-pc (--configure):
>  subprocess installed post-installation script returned error exit status 1
> *
>

And this indicates you're trying to install grub.

If you're installing grub, you're probably starting from scratch on a new
disk. You should look up a tutorial on how to test your hard drive, but an
abbreviated tutorial is to get a Live DVD distro, burn it, boot off of it,
and before mounting any partitions on /dev/sda, run badblocks -nvs /dev/sda
on it. Let that run for however long it needs to; this will test all of the
sectors on your drive to ensure they're working correctly. If it outputs
any errors or starts writing out lots of numbers, you have a disk problem.

Now, on to Asterisk: If your box is blank, you should just install a
distribution designed for Asterisk:
http://www.freepbx.org/freepbx-distro
http://www.elastix.org/index.php/en/downloads/main-distro.html

If this is an existing system, it sounds like you have a serious
configuration issue or hardware problem, so you should head to the forums
for whatever distro you're using (looks like some Debian variant).

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Re: [asterisk-users] sip register peer (the quest for near 100% availability)

[Christopher Harrington]

On Thu, Jan 31, 2013 at 9:45 AM, joachim  wrote:

>
> >
>>
>>  If you have no NAT or dynamic IP in your network, you can just remove
>> the registration process and assign to each peer its IP address.
>>
>>
>  This is the answer. If 100% availability is critical, your IP addresses
> shouldn't be changing anyway, so take the registration process out
> entirely.
>
>   This advice is not valid for android / iphones though.
>

That's absurd. Why would you use a battery-powered smartphone if you are
trying to have 100% availability?


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Re: [asterisk-users] OT - Chan-mobile -Bluetooth dongle on remote LAN workstation

[Christopher Harrington]

On Thu, Jan 31, 2013 at 4:15 AM, Olivier  wrote:

> What I had in mind is to use someone's cellphone as a presence detector.
> Let me explain:
> - as the first thing you take along when leaving a room or location, is
> your own cellphone, why not use chan_mobile and a bluetooth dongle on your
> on PC (as you're not supposed to be within bluetooth range from an asterisk
> server ;-)) to advertise you're away from your desk
>
> - it seems that chan_mobile is not up to expectations for voice delivery
> but would it remain the same for presence detection, if may call it this
> way ?
>
>
What you're trying to do would probably be better served by interfacing
with AMI and firing an Originate command (
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerAction_Originate)
to trigger your dialplan and update presence state. Or use existing
software to accomplish the same thing locally on the PC by updating a
user's XMPP status, and have Asterisk subscribe to that.

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Re: [asterisk-users] sip register peer (the quest for near 100% availability)

[Christopher Harrington]

On Thu, Jan 31, 2013 at 3:08 AM, Leandro Dardini  wrote:

> 2013/1/31 Ishfaq Malik 
>
>> On Wed, 2013-01-30 at 14:10 -0700, Carlos Alvarez wrote:
>> > On Wed, Jan 30, 2013 at 12:05 PM, XBrian  wrote:
>> > Thanks - I was hoping there was some silver bullet to use out
>> > there. Thanks
>> > anyway.
>> >
>
> If you have no NAT or dynamic IP in your network, you can just remove the
> registration process and assign to each peer its IP address.
>
>
This is the answer. If 100% availability is critical, your IP addresses
shouldn't be changing anyway, so take the registration process out
entirely.

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Re: [asterisk-users] Frames with invalid timing info

[Christopher Harrington]

At a command prompt (not at the Asterisk CLI), if you run

dahdi_tool

and hit F1, what does it say?

This is what I see: http://i.imgur.com/je7qRHa.png




On Fri, Jan 25, 2013 at 8:20 AM, Richard Kenner  wrote:

> I'm now getting these errors:
>
> [Jan 25 09:19:01] WARNING[29877]: abstract_jb.c:284 ast_jb_put:
> DAHDI/i1/2128518396-ba7 received frame with invalid timing info:
> has_timing_info=1, len=0, ts=426891164, src=RTP
> [Jan 25 09:19:01] WARNING[29877]: abstract_jb.c:284 ast_jb_put:
> DAHDI/i1/2128518396-ba7 received frame with invalid timing info:
> has_timing_info=1, len=0, ts=426891174, src=RTP
>
> even *without* any transcoding.
>
> Suggestions?
>
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Re: [asterisk-users] How to assign the button on the IP Phone

[Christopher Harrington]

On Thu, Jan 24, 2013 at 12:11 PM, bilal ghayyad  wrote:

> Both: SPA and 7900. let us say 7942. How?
>
>
Googled cisco 7942 soft keys, first result:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmesoftk.html

This is pretty off-topic, by the way.


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Re: [asterisk-users] two steps when calling from web!

[Christopher Harrington]

On Wed, Jan 23, 2013 at 10:14 AM, Danny Nicholas  wrote:

> Originate is the answer here.  Let’s say your X-lite is SIP/100 and you’re
> dialing 555-1212.  From the x-lite you dial 555-1212 and Asterisk does a
> dial command to execute the call.  From the web, we “originate” the call
> from SIP/100 to 555-1212.  Asterisk makes sure SIP/100 is available then
> dials the call.
>
> sendcommand( Action => 'Originate',
>
>Channel => "SIP/100",
>
>Exten => 5551212,
>
>Context => 'default',
>
>priority => 1,
>
>Number => 5551212
>
>);
>
> I use this in my office with Apache 1.X and 2.X.
>


He's already doing an originate invocation. From his email:

fputs($oSocket, "Action: originate\r\n");
fputs($oSocket, "Channel: $channel\r\n");
fputs($oSocket, "WaitTime: $waitTime\r\n");
fputs($oSocket, "CallerId: $callerId\r\n");
fputs($oSocket, "Exten: $number\r\n");
fputs($oSocket, "Context: $context\r\n");
fputs($oSocket, "Priority: $priority\r\n\r\n");

-Chris


> 
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Christopher
> Harrington
> *Sent:* Wednesday, January 23, 2013 9:42 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] two steps when calling from web!
>
> ** **
>
> On Wed, Jan 23, 2013 at 1:09 AM, Muhammad 
> wrote:
>
> -1 in normal way, when I type the number in softphone, it call the number
> and show me just "End" bottom.
>
> 2- when I calling the number through the web, it show me "Answer" bottom
> and I have to click answer to calling then number. it is 2 steps to calling
> from web.
>
> ** **
>
> ** **
>
> For Asterisk, there is no way to bring a device in on a call unless
> Asterisk dials out to it first. That device needs to accept the
> Asterisk-originated call as if a normal call were incoming.
>
> ** **
>
> When I was referring to headers, I was talking about SIP headers that
> allow many hardware SIP phones to go into what is effectively an intercom
> mode, not requiring an explicit answer function. I don't know (off of the
> top of my head) how to set SIP headers from the AMI originate action, but I
> suppose there probably is some way to do it. Then question then becomes
> whether or not your softphone supports it.
>
> ** **
>
> Otherwise, there may be an option to configure your softphone to simply
> automatically answer all incoming calls.
>
> ** **
>
> --
> -Chris Harrington
>
> ACSDi Office: 763.559.5800
>
> Mobile Phone: 612.326.4248
>
> ** **
>



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Re: [asterisk-users] two steps when calling from web!

[Christopher Harrington]

On Wed, Jan 23, 2013 at 1:09 AM, Muhammad wrote:

> -1 in normal way, when I type the number in softphone, it call the number
> and show me just "End" bottom.
> 2- when I calling the number through the web, it show me "Answer" bottom
> and I have to click answer to calling then number. it is 2 steps to calling
> from web.
>
>
For Asterisk, there is no way to bring a device in on a call unless
Asterisk dials out to it first. That device needs to accept the
Asterisk-originated call as if a normal call were incoming.

When I was referring to headers, I was talking about SIP headers that allow
many hardware SIP phones to go into what is effectively an intercom mode,
not requiring an explicit answer function. I don't know (off of the top of
my head) how to set SIP headers from the AMI originate action, but I
suppose there probably is some way to do it. Then question then becomes
whether or not your softphone supports it.

Otherwise, there may be an option to configure your softphone to simply
automatically answer all incoming calls.

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Re: [asterisk-users] Capture queue agent drop and put caller back in queue

[Christopher Harrington]

My suggestion of Nagios was only to avoid re-implementing the "detect
device has gone out to lunch" code. Obviously the part interacting with AMI
is still custom and always will be.


On Tue, Jan 22, 2013 at 4:15 PM, Danny Nicholas  wrote:

> A qualify value that low would be a resource hog (some phones can't even
> re-register in 10 seconds).  The Nagios solution would require a custom
> shell, so it would less needy to make the shell be a daemon independent of
> either.
>
> -Original Message-
> From: Eric Wieling [mailto:ewiel...@nyigc.com]
> Sent: Tuesday, January 22, 2013 4:12 PM
> To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial
> Discussion; Danny Nicholas
> Cc: Benny Amorsen
> Subject: RE: [asterisk-users] Capture queue agent drop and put caller back
> in queue
>
> Using qualify=10 ?
>

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Re: [asterisk-users] Capture queue agent drop and put caller back in queue

[Christopher Harrington]

Does that, then, solve Mitch's original issue? I only proposed Nagios
because nobody actually responded to his question.


On Tue, Jan 22, 2013 at 4:12 PM, Eric Wieling  wrote:

> Using qualify=10 ?
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher
> Harrington
> Sent: Tuesday, January 22, 2013 5:11 PM
> To: Danny Nicholas
> Cc: Asterisk Users Mailing List - Non-Commercial Discussion; Benny Amorsen
> Subject: Re: [asterisk-users] Capture queue agent drop and put caller back
> in queue
>
> How do you propose that Asterisk determines that the endpoint has vanished
> off the network without waiting for a 10-90 second timeout period?
>
>
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Re: [asterisk-users] Capture queue agent drop and put caller back in queue

[Christopher Harrington]

How do you propose that Asterisk determines that the endpoint has vanished
off the network without waiting for a 10-90 second timeout period?


On Tue, Jan 22, 2013 at 4:07 PM, Danny Nicholas  wrote:

> Not doubting how quickly Nagios can respond, but if the Nagios solution is
> going to place a call using Asterisk, wouldn’t it be just as efficient (or
> more so) to depend on Asterisk? 
>
> **
>
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Re: [asterisk-users] Capture queue agent drop and put caller back in queue

[Christopher Harrington]

On Tue, Jan 22, 2013 at 3:01 PM, Benny Amorsen wrote:

>
> Can a Nagios-based solution provide quicker failover than the 90 seconds
> provided by sip timers or the 10-30 seconds provided by rtptimeout?
>

Certainly; Nagios can detect missed ping responses with a granularity of
single seconds.


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Re: [asterisk-users] Asterisk, Digium phones, and voicemail.

[Christopher Harrington]

On Tue, Jan 22, 2013 at 2:27 PM, Frank  wrote:

> Hi all,
>
> I registered my Digium D70 using a name ("D70") instead of a number.
> Is there a way to program Asterisk (or the phone?) so when I press the
> MSGS button, it automatically goes to the correct voicemail, with or
> without asking me for a password ?
>
>
You don't need to use DPMA if you're provisioning the phones via XML. It's
up to you.

View https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=21463877and
search for "voicemail" on the page. That parameter sets what the phone
automatically dials; set that value to the extension for that phone's
voicemail (as per Danny's email).


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Re: [asterisk-users] Capture queue agent drop and put caller back in queue

[Christopher Harrington]

On Mon, Jan 21, 2013 at 12:03 PM, Mitch Claborn wrote:

> How can I accomplish my goal?


Since nobody seems to have come up with an Asterisk-specific solution, it
sounds like the real approach here is something more generic.

You can set up Nagios to fire off an event if it detects endpoints or
infrastructure are suddenly dead. In particular, Nagios could launch a
program written for this purpose, passing the endpoints it detects are
missing, and that program could then query Asterisk via AMI about the call
IDs each endpoint is a participant in, then do a forced-transfer to a
dedicated queue that announces the failure condition to the caller. This
AMI could also conveniently remove the dead endpoints from the existing
queues (including the failover queue).

The wildcard here is obviously the program itself, which would be slightly
complex. It would have to have intimate knowledge about your
infrastructure. It's unlikely something like this currently exists that you
can just drop-in, but I'm sure there's someone on this list that would be
willing to sell you the labor to write it.

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Re: [asterisk-users] two steps when calling from web!

[Christopher Harrington]

On Tue, Jan 22, 2013 at 1:57 AM, Muhammad wrote:

> Dear All.
>
> When I calling a number from web, my softphone show me "Answer" and
> "Decline" bottoms, and then I have to click Answer to call the number. it
> seems it is two step to calling the number. If I type the number direct to
> my client softphone, it calls directlly the number without show me to
> choose Answer to calling.
>
>
We'll need to know more about the software you're using. That is, what is
it called, what version is it, and so on.



> My source code is in AMI fsocket open to make call from web. how can I
> call direct to the number?
>
>
There may be headers you can pass to cause your softphone to automatically
answer the incoming call. By default, most physical SIP phones allow a sort
of "intercom" mode that requires no user action before the SIP call is
brought up. I don't know if your software would support the same mechanism.


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Re: [asterisk-users] Google voice with no voice

[Christopher Harrington]

On Mon, Jan 21, 2013 at 9:59 PM, Frank  wrote:

> Actually, the funny thing is that it works randomly.
>

This may be due to the fact that voice.google.com actually resolves to a
range of IP addresses. When you set up your firewall, it may not be
including all of the possible resolutions for voice.google.com...

voice.l.google.com. 300 IN A 74.125.225.36
voice.l.google.com. 300 IN A 74.125.225.46
voice.l.google.com. 300 IN A 74.125.225.33
voice.l.google.com. 300 IN A 74.125.225.32
voice.l.google.com. 300 IN A 74.125.225.41
voice.l.google.com. 300 IN A 74.125.225.38
voice.l.google.com. 300 IN A 74.125.225.35
voice.l.google.com. 300 IN A 74.125.225.39
voice.l.google.com. 300 IN A 74.125.225.40
voice.l.google.com. 300 IN A 74.125.225.34
voice.l.google.com. 300 IN A 74.125.225.37

(ie 74.125.225.32-41 and 74.125.225.46)

Since these are short TTL values (the 300 means 5 minutes) there may be a
brief period where your devices and your firewall agree, before one or both
change their mind about the IP address behind that hostname.





> I just tried out of the blue calling from D70 through Google Voice to a
> cell phone, and it worked. I hung up, redial, and no audio at all.
>
>
> On 1/21/13 10:38 PM, Frank wrote:
>
>> Greetings all,
>>
>> I was reading the documentation tonight, and decided to try Google voice
>> with my asterisk.
>>
>> I was able to setup iksemel, connect to google using jabber, and connect
>> to google voice using gtalk.
>>
>>
>> Here is my physical configuration:
>>
>> Digium D70 <-- private network 192.168.1.x --> Airport express <-->
>> Internet <--> Asterisk with public IP
>>
>> My asterisk has the following ports open:
>> 5060 tcp/udp from my Airport Express public IP and from voice.google.com
>> 10,000:20,000 from my Airport Express public IP and from voice.google.com
>>
>> My issue is that when I place a call with google voice, I have no audio
>> path at all in both way.
>>
>> When a call is received on google voice (and sent to the D70), if I pick
>> up, nothing happen, and the caller still hear the ringing tone.
>>
>>
>>
>> My D70 is setup as follow in the sip.conf:
>> [D70]
>> type=friend
>> nat=yes
>> qualify=yes
>> directmedia=no
>> host=dynamic
>> secret=takapoum
>> disallow=all
>> allow=ulaw
>> context=LocalSets
>> mailbox=D70@default
>>
>>
>> my gtalk.conf is setup as follow:
>> [general]
>> bindaddr=0.0.0.0
>> allowguest=yes
>>
>> [guest]
>> disallow=all
>> allow=ulaw
>> context=gtalk_incoming
>> connection=asterisk
>>
>>
>>
>> and finally, the interesting parts in my extensions.conf are setup as
>> follow:
>> ;Dialing out on google voice:
>> exten => _1zxxzxx,1,Dial(Gtalk/**asterisk/+${EXTEN}@voice.**
>> google.com )
>>  same => n,Hangup()
>>
>> ;Google voice incoming
>> [gtalk_incoming]
>> exten => r...@gmail.com,1,Verbose(0, Incoming gtalk from
>> ${CALLERID(all)})
>>  same => n,Answer()
>>  same => n,Wait(2)
>>  same => n,Dial(SIP/D70)
>>  same => Hangup()
>>
>>
>> I would appreciate if anyone could give me a hint about the audio path.
>> This is a project that we I will try to setup in a small fire
>> department, and before I try it, I would like to make sure that my
>> Digium phones will be able to get full audio path behind private networks.
>>
>> Thanks a ton for the help !
>>
>> --
>> __**__**_
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>>
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>>
>
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Re: [asterisk-users] Capture queue agent drop and put caller back in queue

[Christopher Harrington]

On Mon, Jan 21, 2013 at 12:03 PM, Mitch Claborn wrote:

> Shouldn't asterisk somehow know when the agent disappears?
>
Asterisk will only notice that the agent is gone when a timeout has
occurred. When you're pulling out the Ethernet cable, there's no
opportunity for any equipment to signal to Asterisk that something has
changed.


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Re: [asterisk-users] Getting UDPTL (SIP): Transmission error: Resource temporarily unavailable

[Christopher Harrington]

Can you be more specific about your Asterisk version? 10.xx.yy ?

Sounds like some sort of resource leak.


On Tue, Jan 15, 2013 at 3:02 PM, Ahmed Munir wrote:

> Hi,
>
> I configured Asterisk 10 for inbound fax, for couple of weeks I didn't see
> any issues until today. The setup  I configured for inbound fax is quite
> simple i.e. Cisco Voice GW sends the fax calls to Asterisk using T.38
> protocol and later Asterisk stores/forwards the fax to specific end user.
>
> The configuration I made in sip.conf for enabling T38 is listed below;
>
> t38pt_udptl = yes,fec,maxdatagram=400
> faxdetect = t38
>
> And in udptl.conf, I just uncommented 'use_even_ports = yes
> ;' and rest of it set as default.
>
>
> Here is the error I'm usually seeing in Asterisk side;
>
> [Jan 15 14:13:28] NOTICE[20514] udptl.c: UDPTL (SIP/10.3.22.6-0ad6):
> Transmission error to 10.3.22.6:18428: Resource temporarily unavailable
>
> If this notice comes, it occurs repeatedly unless I need to restart the
> asterisk service. For some reason it also effect the V-GW.
>
> Please advise what is the reason that I'm getting this message and how can
> I avoid it?
>
>
> --
> Regards,
>
> Ahmed Munir Chohan
>
>
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Re: [asterisk-users] Playing music through VoIP handsets while on hook

[Christopher Harrington]

On Thu, Jan 10, 2013 at 7:31 PM, chris  wrote:

> I've seen this implemented on polycom phones where a secondary extension
> is on the phone that is setup to auto answer and they have something on the
> PBX side that is configured to call some or all of the secondary extensions
>

Wow, that seems wildly bandwidth inefficient. Is it possible to do
multicast VoIP?

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Re: [asterisk-users] Your thoughts and opinions on Asterisk 11 for production use

[Christopher Harrington]

Nope, we're using Digium D40's.


On Thu, Jan 10, 2013 at 8:52 AM, Julian Lyndon-Smith wrote:

> are you using cisco 79xx phones ?
>

Nope, we're using Digium D40's.


>
> We had a similar problem. Upgrading the sip firmare to 8.12 fixed it for
> us.
>
> FWIW we're using 11 in a call centre, with 25k+ call attempts per day.
> Rock solid. Not a single crash since Oct 15
>
> Julian
>
>
> On 10 January 2013 14:25, Christopher Harrington  wrote:
> > On Thu, Jan 10, 2013 at 8:18 AM, Danny Nicholas 
> wrote:
> >>
> >> I don’t presently have 11 in production, but in each case where I’ve put
> >> 11 in on top of 10.X the process has been relatively seamless, so I
> expect
> >> my 10.X boxes will go to 11.X sometime this year.
> >
> >
> > Upgrading from 10.x to 11.x silently broke phone-to-phone call transfers
> for
> > us. Haven't had time to investigate it more thoroughly, so we rolled
> back to
> > 10.
> >
> >
> > --
> > -Chris Harrington
> > ACSDi Office: 763.559.5800
> > Mobile Phone: 612.326.4248
> >
> >
> > --
> > _
> > -- 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
>
>
>
> --
> Julian Lyndon-Smith
> IT Director, Dot R Limited
>
> "I don’t care if it works on your machine!  We are not shipping your
> machine!”
>
> The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg
>

Top-and-bottom posting just to irritate the zealots.

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Re: [asterisk-users] Call Disconnected by Caller or Agent

[Christopher Harrington]

On Thu, Jan 10, 2013 at 8:23 AM, RSCL Mumbai  wrote:

> Hello,
>
> Can asteriskCDR logs tell me if a call was disconnected by the caller
> or the Agent ?
>
> My call flow is as follows:
> Caller Dials a DID >> Inbound Routes > Play Greeting > Call Enter
> Queue > Call sent to Dynamic Logged-in Agent(s)
>
> Thank you in advance.
>

Take a look at this article if you haven't already.

https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=20185363

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Re: [asterisk-users] Your thoughts and opinions on Asterisk 11 for production use

[Christopher Harrington]

On Thu, Jan 10, 2013 at 8:18 AM, Danny Nicholas  wrote:

> I don’t presently have 11 in production, but in each case where I’ve put
> 11 in on top of 10.X the process has been relatively seamless, so I expect
> my 10.X boxes will go to 11.X sometime this year.
>
> **
>

Upgrading from 10.x to 11.x silently broke phone-to-phone call transfers
for us. Haven't had time to investigate it more thoroughly, so we rolled
back to 10.


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Re: [asterisk-users] DIDForSale spam

[Christopher Harrington]

On Wed, Jan 9, 2013 at 12:16 PM, Matthew J. Roth  wrote:

> I suspect that they are using this
> list to harvest email addresses and think they should be called out on
> this poor business practice if that is the case.
>

Perhaps email cnighswon...@foundations.edu, whose previous emails have
included references to DIDForSale, implying that he or she is either a
customer or an employee.

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Re: [asterisk-users] Paging unit suggestions

[Christopher Harrington]

On Mon, Jan 7, 2013 at 2:10 PM, Doug Lytle  wrote:

> I'm looking for suggestions on a IP based amp or similar that could drive
> the current speakers?  I was envisioning a unit that would register as a
> SIP extension then would handle auto-answer that I could send a sound file
> to.
>
>
Seems like a RaspberryPi would be a great candidate for this, assuming you
keep the amplifier.


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Re: [asterisk-users] Moving User Agent To Remote Location

[Christopher Harrington]

On Thu, Jan 3, 2013 at 2:21 PM, Nick Khamis  wrote:

> [Dec 12 15:35:54] WARNING[1736]: app_dial.c:2198 dial_exec_full:
> Unable to create channel of type 'SIP' (cause 20 - Unknown)
>

Can you check that the registration is happening correctly? Try `sip show
peers` or `sip show peer [destination phone]`. Usually "cause 20" means the
peer isn't registered.

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Re: [asterisk-users] asterisk seg fault 1.4.43

[Christopher Harrington]

What version of ALSA do you have installed? 1.0.26 is current (
http://alsa-project.org/main/index.php/Main_Page ) and it looks like the
crash is in there.


On Wed, Jan 2, 2013 at 10:32 AM, Jerry Geis  wrote:

> I finally got it to happen again.
>
> #0  0x00296f96 in __memcpy_ia32 () from /lib/libc.so.6
> #1  0x0002 in ?? ()
> #2  0x4d44fa0e in snd_pcm_area_copy () from /usr/lib/libasound.so.2
> #3  0x4d44ff09 in snd_pcm_areas_copy () from /usr/lib/libasound.so.2
> #4  0x4d4620f4 in snd_pcm_mmap_read_areas () from /usr/lib/libasound.so.2
> #5  0x4d454bd0 in snd1_pcm_read_areas () from /usr/lib/libasound.so.2
> #6  0x4d4624e4 in snd_pcm_mmap_readi () from /usr/lib/libasound.so.2
> #7  0x4d44bbe5 in _snd_pcm_readi () from /usr/lib/libasound.so.2
> #8  0x4d44d2d3 in snd_pcm_readi () from /usr/lib/libasound.so.2
> #9  0xb7496575 in alsa_read (chan=0x830ac00) at chan_alsa.c:711
> #10 0x0808b658 in __ast_read (chan=0x830ac00, dropaudio=0) at
> channel.c:2411
> #11 0x0808d325 in ast_read (c0=0xb750eb68, c1=0x830ac00,
> config=0xb6f2acdc, fo=0xb6f29dac, rc=0xb6f29da8) at channel.c:2720
> #12 ast_generic_bridge (c0=0xb750eb68, c1=0x830ac00, config=0xb6f2acdc,
> fo=0xb6f29dac, rc=0xb6f29da8) at channel.c:4647
> #13 ast_channel_bridge (c0=0xb750eb68, c1=0x830ac00, config=0xb6f2acdc,
> fo=0xb6f29dac, rc=0xb6f29da8) at channel.c:4989
> #14 0xb74f2fad in ast_bridge_call (chan=0xb750eb68, peer=0x830ac00,
> config=0xb6f2acdc) at res_features.c:2281
> #15 0xb6f6df63 in dial_exec_full (chan=0xb750eb68, data= out>, peerflags=0xb6f2ae4c, continue_exec=0x0)
> at app_dial.c:1894
> #16 0xb6f703c6 in dial_exec (chan=0xb750eb68, data=0xb6f2cebc) at
> app_dial.c:1942
> #17 0x080d2d9b in pbx_exec (c=0xb750eb68, con=,
> context=0xb750ece8 "smvoice-pa",
> exten=0xb750ed38 "s", priority=8, label=0x0, callerid=0xb7510b30
> "501", action=E_SPAWN) at pbx.c:550
> #18 pbx_extension_helper (c=0xb750eb68, con=,
> context=0xb750ece8 "smvoice-pa",
> exten=0xb750ed38 "s", priority=8, label=0x0, callerid=0xb7510b30
> "501", action=E_SPAWN) at pbx.c:1893
> #19 0x080d432f in ast_spawn_extension (c=0xb750eb68) at pbx.c:2367
> #20 __ast_pbx_run (c=0xb750eb68) at pbx.c:2461
> #21 0x080d5e3e in pbx_thread (data=0xb750eb68) at pbx.c:2688
> #22 0x08107e6b in dummy_start (data=0xb750f4a8) at utils.c:856
> #23 0x003c1a49 in start_thread () from /lib/libpthread.so.0
> #24 0x002fe63e in clone () from /lib/libc.so.6
>
>
>
>
> This is from the gdb "where" command.
> I am just calling into the box and using the ALSA channel for audio. This
> is VERY hard to re-create
> but it does happen.
>
>
> jerry
>
>
>
> --
> __**__**_
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   
> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>



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Re: [asterisk-users] Top Posting

[Christopher Harrington]

On Sun, Dec 30, 2012 at 2:54 PM, Benny Amorsen wrote:

> Gergo Csibra  writes:
>
> > Complaining about top posting on a list where's no moderation,
> > no sanction if somebody top posting is pointless.
>
> There is a sanction. People like me will score top posters lower and
> soon not see their posts at all.
>
>
Let me point out that there are users of this mailing list (such as myself)
who are completely unaware of what you're talking about.


> It is often a quick way to see if it is worth responding to someone. If
> they top post, nothing of value is likely to come out of the
> conversation.
>

That kind of attitude is unlikely to yield dividends in the long term.


>
> So by all means, everybody who wants to, keep top posting.
>
>
I probably will, from time to time. Not always, but as Gmail evolves as a
service, they seem to be making this style of conversation more and more
difficult. Inline replies and bottom-posting are nearly impossible to do
nicely on an iPhone. Outlook  – as mentioned elsewhere in this thread –
isn't helping here either.

But a thinly veiled "I'll take my ball and go home" reaction isn't
productive for either you or the communities you participate in.

-- 
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Re: [asterisk-users] asterisk seg fault 1.4.43

[Christopher Harrington]

On Fri, Dec 28, 2012 at 9:36 AM, Jerry Geis  wrote:

>  If you got the segfault, then recompiled with debug info, then the
> addresses in the segfault are no longer relevant to the binary you have.
>
> Are you getting the segfault repeatedly? If so, then just wait for it to
> happen again with the debug info and work with that, and/or the core file
> that Asterisk creates.
>
> If you can't reproduce the segfault, unfortunately that is just the way of
> things sometimes. Could be a random bit-flip in a stick of memory.
>
> In either case, if that segfault line was generated by a non-debug asterisk
> build, it isn't really useful to you.
>
>  I was able with gdb to set a break point at the address of the
> instruction pointer:
>
>  ip 00296f96
>
> This is in memcpu_iax32, which is called from sip_alloc() which is called 
> from transmiter_register()
>
> Yes, but that probably isn't where it was in the original binary. Now that
you've recompiled it including debug info, you have a different binary,
with different addresses. Chances are slim that this is actually where the
crash occurred.

Reproduce the segfault with the new debug binary.


> ??
>
> Jerry
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>



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Re: [asterisk-users] asterisk seg fault 1.4.43

[Christopher Harrington]

On Fri, Dec 28, 2012 at 7:59 AM, Jerry Geis  wrote:

> I got the above seg fault on 1.4.43, I have recompiled with debug info.
> Is there any way to take the above and "locate" where in the code that is
> ???
>
>
If you got the segfault, then recompiled with debug info, then the
addresses in the segfault are no longer relevant to the binary you have.

Are you getting the segfault repeatedly? If so, then just wait for it to
happen again with the debug info and work with that, and/or the core file
that Asterisk creates.

If you can't reproduce the segfault, unfortunately that is just the way of
things sometimes. Could be a random bit-flip in a stick of memory.

In either case, if that segfault line was generated by a non-debug asterisk
build, it isn't really useful to you.


> Any other thoughts?
>
> Jerry
>
>
> --
> __**__**_
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   
> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>



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Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through

[Christopher Harrington]

On Thu, Dec 27, 2012 at 3:45 PM, Eric Wieling  wrote:

> sip.conf settings:
> directmedia=yes
>
>
I know you've said you tried it both ways, but consensus seems to be that
directmedia needs to be =no when using UDPTL.


-- 
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Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through

[Christopher Harrington]

Last thing to check, just for sanity's sake:

t38pt_udptl=yes in sip.conf? It defaults to off.




On Thu, Dec 27, 2012 at 12:32 PM, Eric Wieling  wrote:

> It does not appear to make any difference.  Calls are still failing.
>
> -Original Message-
> From: Christopher Harrington [mailto:ch...@acsdi.com]
> Sent: Thursday, December 27, 2012 1:20 PM
> To: Eric Wieling
> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38
> Pass-through
>
> True, but it should bypass Asterisk when possible for SIP streams and may
> solve your problem.
>
>
> On Thu, Dec 27, 2012 at 12:16 PM, Eric Wieling  wrote:
>
>
> We have directrtpsetup=no because the comments in the sample
> config indicates it does not work in all situations.
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher
> Harrington
> Sent: Thursday, December 27, 2012 1:13 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran
> T.38 Pass-through
>
> directrtpsetup=yes in sip.conf?
>
>
>
> On Thu, Dec 27, 2012 at 12:09 PM, Eric Wieling 
> wrote:
>
>
> We have set directmedia=yes as well as directmedia=no.
>  There is no NAT involved.
>
>
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini
> Sent: Thursday, December 27, 2012 1:08 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] $100 Bounty:
> Level3/Asterisk/Adtran T.38 Pass-through
>
> Have you configured the canreinvite=yes in sip peer?
>
> I am currently off work for two days, but a 100% fail
> means a configuration problem for sure.
>
>
> Leandro
>
>
> 2012/12/27 Eric Wieling 
>
>
> We are offering $100 (paid via paypal or check) to
> the first person who assists us in successfully sending and receiving faxes
> in the setup described below.  Offer expires Dec 31.  We are a direct
> customer of Level 3, there is no other carrier involved.
>
> What we want to work:
>
> Level 3 T.38 TN <-> MSX/Nextone SBC <->
> Asterisk 1.8.18.1 <-> Adtran NetVanta w/POTS and T.38 support.
>
> When we replace Asterisk with Kamailio faxes work
> fine.  When we put Asterisk there instead, then faxes fail nearly 100% of
> the time.
>
> I see the switch to T.38 in the Adtran debug logs.
>   We can originate and terminate T.38 calls in Asterisk using
> SendFax/ReceiveFax using T.38 so I'm assuming I have my udptl.conf and
> sip.conf settings correct.
>
>
>
> --
>
> _
> -- Bandwidth and Colocation Provided by
> http://www.api-digital.com --
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> webinar every Thurs:
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>
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> --
>
> _
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> http://www.api-digital.com --
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> every Thurs:
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>
>
>
>
>
> --
> -Chris Harrington
>
> ACSDi Office: 763.559.5800
> Mobile Phone: 612.326.4248
>
>
>
>
>
>
> --
> -Chris Harrington
>
> ACSDi Office: 763.559.5800
> Mobile Phone: 612.326.4248
>
>


-- 
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Re: [asterisk-users] Vxml record voice parameter

[Christopher Harrington]

If VoiceGlue is the software making the recording, then that's where you
need to look for support.

Try https://github.com/voiceglue/voiceglue/issues and
http://www.voiceglue.org/mailing-list/ .


On Thu, Dec 27, 2012 at 12:27 PM, ulvi cesur  wrote:

> I am using voiceglue to record voice.
>
> VXML  :
>
>  dtmfterm="true">
> 
>  namelist="rec" method="post">
> 
> 
>
> I can take rec parameter but it is not file. rec is
> audio/basic:len(123123):p0x5a6e6241.
>
> 2012/12/26 Christopher Harrington 
>
>> On Tue, Dec 25, 2012 at 8:57 AM, ulvi cesur  wrote:
>>
>>> Hi, I am working on vxml to record voice. I have trouble with getting
>>> url of recorded voice. I want to save and I am using java to get record
>>> parameter from url and it returns a string which is
>>> audio/basic:len(123123):p0x5a6e6241, but I want to get file object or
>>> base64 string with parameter or to relate returning string with path in
>>> asterisk server, are there any way to do this?
>>>
>>>
>> How are you recording the audio in Asterisk? ChanSpy, Voicemail, etc?
>>
>>
>> --
>> -Chris Harrington
>> ACSDi Office: 763.559.5800
>> Mobile Phone: 612.326.4248
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
>
>


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Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through

[Christopher Harrington]

True, but it should bypass Asterisk when possible for SIP streams and may
solve your problem.


On Thu, Dec 27, 2012 at 12:16 PM, Eric Wieling  wrote:

> We have directrtpsetup=no because the comments in the sample config
> indicates it does not work in all situations.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher
> Harrington
> Sent: Thursday, December 27, 2012 1:13 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38
> Pass-through
>
> directrtpsetup=yes in sip.conf?
>
>
>
> On Thu, Dec 27, 2012 at 12:09 PM, Eric Wieling  wrote:
>
>
> We have set directmedia=yes as well as directmedia=no.  There is
> no NAT involved.
>
>
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini
> Sent: Thursday, December 27, 2012 1:08 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran
> T.38 Pass-through
>
> Have you configured the canreinvite=yes in sip peer?
>
> I am currently off work for two days, but a 100% fail means a
> configuration problem for sure.
>
>
> Leandro
>
>
> 2012/12/27 Eric Wieling 
>
>
> We are offering $100 (paid via paypal or check) to the
> first person who assists us in successfully sending and receiving faxes in
> the setup described below.  Offer expires Dec 31.  We are a direct customer
> of Level 3, there is no other carrier involved.
>
> What we want to work:
>
> Level 3 T.38 TN <-> MSX/Nextone SBC <-> Asterisk
> 1.8.18.1 <-> Adtran NetVanta w/POTS and T.38 support.
>
> When we replace Asterisk with Kamailio faxes work fine.
>  When we put Asterisk there instead, then faxes fail nearly 100% of the
> time.
>
> I see the switch to T.38 in the Adtran debug logs.   We
> can originate and terminate T.38 calls in Asterisk using SendFax/ReceiveFax
> using T.38 so I'm assuming I have my udptl.conf and sip.conf settings
> correct.
>
>
>
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>
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>
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>
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>


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Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38 Pass-through

[Christopher Harrington]

directrtpsetup=yes in sip.conf?


On Thu, Dec 27, 2012 at 12:09 PM, Eric Wieling  wrote:

> We have set directmedia=yes as well as directmedia=no.  There is no NAT
> involved.
>
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini
> Sent: Thursday, December 27, 2012 1:08 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] $100 Bounty: Level3/Asterisk/Adtran T.38
> Pass-through
>
> Have you configured the canreinvite=yes in sip peer?
>
> I am currently off work for two days, but a 100% fail means a
> configuration problem for sure.
>
>
> Leandro
>
>
> 2012/12/27 Eric Wieling 
>
>
> We are offering $100 (paid via paypal or check) to the first
> person who assists us in successfully sending and receiving faxes in the
> setup described below.  Offer expires Dec 31.  We are a direct customer of
> Level 3, there is no other carrier involved.
>
> What we want to work:
>
> Level 3 T.38 TN <-> MSX/Nextone SBC <-> Asterisk 1.8.18.1 <->
> Adtran NetVanta w/POTS and T.38 support.
>
> When we replace Asterisk with Kamailio faxes work fine.  When we
> put Asterisk there instead, then faxes fail nearly 100% of the time.
>
> I see the switch to T.38 in the Adtran debug logs.   We can
> originate and terminate T.38 calls in Asterisk using SendFax/ReceiveFax
> using T.38 so I'm assuming I have my udptl.conf and sip.conf settings
> correct.
>
>
>
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>
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>
>
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[asterisk-users] Presence Registration on the D40

[Christopher Harrington]

So I'm working with our Digium D40's and we're not using DPMA.

This video ( http://www.youtube.com/watch?v=zcuocp01pfM#t=35s ) shows
presence information being displayed in the Contacts application. Obviously
the video is showing DPMA in play. Is it possible to enable this
functionality without it? Is this status information only available on
higher-end Digium phones?

In the contacts XML data, I am supplying the appropriate parameters:


but I am not seeing the icons shown in the video at all. On the Asterisk
CLI, I can run:
etc*CLI> core show hint 2003
   2003@default : SIP/charrington_desk
 State:IdleWatchers  0
1 hint matching extension 2003

and Watchers is always 0 for all extensions.

Is there a separate way I can test subscribing for presence information? I
don't even know, at this point, if it's the phones or Asterisk.

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Re: [asterisk-users] Vxml record voice parameter

[Christopher Harrington]

On Tue, Dec 25, 2012 at 8:57 AM, ulvi cesur  wrote:

> Hi, I am working on vxml to record voice. I have trouble with getting url
> of recorded voice. I want to save and I am using java to get record
> parameter from url and it returns a string which is
> audio/basic:len(123123):p0x5a6e6241, but I want to get file object or
> base64 string with parameter or to relate returning string with path in
> asterisk server, are there any way to do this?
>
>
How are you recording the audio in Asterisk? ChanSpy, Voicemail, etc?


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Re: [asterisk-users] What is the maximum number of meetme's allowed?

[Christopher Harrington]

Can you paste the output of

ps -p `pgrep asterisk` u

?


On Mon, Dec 24, 2012 at 11:35 PM, Deepesh D  wrote:

> I am using 64-bit Linux OS. Also before starting asterisk I have set
> the ulimit to a higher value.
>
> When this happened there was no calls in the system. There was only
> about 160 Meetme conferences, and in each Meetme there was only one
> channel.
>
> On Mon, Dec 24, 2012 at 11:36 PM, Johan Wilfer  wrote:
> > 2012-12-24 16:13, Deepesh D skrev:
> >> Hello,
> >>
> >> What is the maximum number of meetme's allowed by asterisk.
> >>
> >> On my server with an 8 GB memory, I start getting the following error
> >> after 150-160 meetme's are created
> >>
> >> WARNING[3485]: app_meetme.c:1820 conf_run: Unable to open DAHDI pseudo
> >> channel: Cannot allocate memory
> >>
> >> At this time the server still has about 6 GB of free memory. I even
> >> tried this on a server with higher memory, it gives the same result.
> >>
> >> I am using asterisk 1.4.44.
> >
> >
> > You have probably run out of file descriptors. Try
> > ulimit -n 8192
> > before starting asterisk (or in the safe_asterisk-script or the
> > init.d-script).
> >
> > I think this is per default 1024 on debian, and if you use sip + meetme
> > you will hit the limit with about 150 concurrent calls.
> >
> > --
> > Johan Wilfer
> >
> > JT Technologies & Telecommunications AB
> > Jabber: jo...@jttech.se | Web: www.jttech.se
> >
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Re: [asterisk-users] Congestion() forcing PRI channels to be not available

[Christopher Harrington]

You probably already know this, but 1.4x is very old (released in 2006) and
is officially end-of-life.

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

You might get more help or better behavior by updating to a newer more
current version of Asterisk, such as 1.8 which will be receiving bug fixes
into October 2014.


On Wed, Dec 19, 2012 at 3:47 PM, James Lamanna  wrote:

> Hi,
> I have a PSTN Asterisk box that's connected to other dialplan PBXes
> through IAX2.
>
> Recently this box was upgraded to 1.4.44 with the latest DAHDI version.
> I've noticed that if one of the dialplan PBXes calls Congestion(), the PRI
> will return ISDN code 34 (as its supposed to do).
> However, the issue is that subsequent calls into that PRI channel are
> immediately responded by a Code 44 (channel not available) even though
> there is no live call on the channel.
>
> Has anyone else experienced this behavior? Its a pretty crippling behavior
> since all of our channels eventually become unresponsive until a 'dahdi
> restart' is issued.
>
> Thanks.
>
> -- James
>
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Re: [asterisk-users] Asterisk 11 originate errors

[Christopher Harrington]

I've observed the same behavior. This is what happens when you close the
socket before logoff is completed. You need to wait until the logoff action
is completed before closing the socket (your fclose() call). Alternately,
use a proxy or a daemon that will sit between your script and the AMI and
keep the socket open.


On Wed, Dec 12, 2012 at 12:44 PM, Faheem  wrote:

> Hi,
> I'm getting errors while originating a call through AMI.
> [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite()
> returned error: Broken pipe
> [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite()
> returned error: Broken pipe
> [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite()
> returned error: Broken pipe
> Asterisk version 11.0.1
> OS: CentOS release 5.8 (Final)
>
> //manager.conf settings
> [faheem]
> secret =f@xx
> permit=127.0.0.1/255.255.255.255
> read =
> system,call,log,verbose,agent,user,config,dtmf,reporting,cdr,dialplan
> write = system,call,agent,user,config,command,reporting,originate
>
> ///AMI script
> 
> $sys_ip = "127.0.0.1";
> $User_str = "faheem";
> $Secret_str = "f@h33m112xx";
> $phoneNumb = 1234;
> $dialNumb =  4567;
> $spoofNumb = 786;
> $context = "x-x";
>
> $oSocket = fsockopen($sys_ip, 5038, $errnum, $errdesc) or die("Connection
> to host failed");
> fputs($oSocket, "Action: login\r\n");
> fputs($oSocket, "Username: $User_str\r\n");
> fputs($oSocket, "Secret: $Secret_str\r\n\r\n");
> fputs($oSocket, "Events: off\r\n\r\n");
> fputs($oSocket, "Action: originate\r\n");
> fputs($oSocket, "Channel: SIP/testTrunk/$phoneNumb\r\n");
> fputs($oSocket, "Exten: $dialNumb\r\n");
> fputs($oSocket, "Context: $context\r\n");
> fputs($oSocket, "Priority: 1\r\n\r\n");
> fputs($oSocket, "Timeout: 1\r\n");
> fputs($oSocket, "CallerId: $spoofNumb\r\n");
> fputs($oSocket, "Async: false\r\n");
> fputs($oSocket, "Action: Logoff\r\n\r\n");
> echo "originate executed";
> fclose($oSocket);
>
> ?>
>
>
> Can any one please help me over it.
> Thank you!
>
> Muhammad Faheem
>
>
>
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Re: [asterisk-users] ODBC Connection Problem

[Christopher Harrington]

On Mon, Dec 10, 2012 at 10:52 AM, Steven Howes wrote:

> On 10 Dec 2012, at 16:13, Christopher Harrington wrote:
> Hostname address is RFC1918, he'll probably be ok ;)
>
>
Private subnet or not, that's a social engineering and recon target. If all
it takes is a Google search for this guy's name and "password", that's
dangerous.


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Re: [asterisk-users] Is there an issue with 11.0.2 and registration

[Christopher Harrington]

On Mon, Dec 10, 2012 at 8:24 AM, Jerry Geis  wrote:

> When I start up and do a "sip show peers" all devices are on and show an
> IP Address.
> After some time "sip show peers" shows two devices of my 12 as
> (Unspecified).
>

When you say "two", is it two every time? The same two? Is there something
different about the two that show this behavior? There isn't enough
information in your message.


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Re: [asterisk-users] ODBC Connection Problem

[Christopher Harrington]

On Mon, Dec 10, 2012 at 5:23 AM, Chandrakant Solanki <
solanki.chandrak...@gmail.com> wrote:

> Password= c3podb@2012


In case you didn't realize you were sending this out publicly to a publicly
archived and searchable list, you might want to change that password now.

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Re: [asterisk-users] callerid not received from dahdi

[Christopher Harrington]

>From the last time you sent this to the list, here's the response from Richard
Mudgett ...

> my scenario is below
>
> analog phone (10 to 99)--> pbx-->(77)asterisk>
> jitsi(2000)
>
> i have analog telephone interface numbered 77 attached with asterisk
> and
> other sip user is 2000 on jitsi.
>
> I can call from any number from 10 to 99(in intercom) on 77 and ivr
> response will come then i can typed 2000# and call go to 2000 named
> user
> in asterisk.
>
> Now my problem is when i am calling from 10 to 99 (any number) this
> number
> should display to sip 2000's user. But its not showing to user. Its
> shows
> asterisk@my_asterisk_server_ip.
>
> my config. as follow
>
> extension.conf
>
> exten => s,1,Goto(phrase-menu,s,1)
>
> [phrase-menu]
>
> exten => s,1,Answer()
> exten => s,2,Wait(1)
> exten => s,3,Read(PHRASEID,/var/lib/asterisk/sounds/custom/soip)
> exten => s,4,Wait(2)
> exten => s,5,Set(CALLERID(num,CID)=${CALLERID})

Remove the CID option.  It does nothing in this case because
it does not apply.  The CID option here only applies to reading
not writing.  Please re-read the documentation for CALLERID().

> exten => s,6,Dial(SIP/${PHRASEID},40,tT)
> exten => h,1,Hangup()
>
>
> and in chan_dahdi.conf
>
> ; General options
> [channels]
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> threewaycalling=yes
> transfer=yes
> echocancel=yes
> echocancelwhenbridged=yes

> cidsignalling=dtmf
> cidstart=polarity
> callerid=asreceived

> rxgain=0.0
> txgain=0.0
> ;FXO Modules
> group=1
> echocancel=yes
> signalling=fxs_ks
> context=default
> channel=1-20
>
> #include dahdi-channels.conf

>From your description, the link between the pbx and (77)asterisk
is analog.  Analog can only pass caller id information in one
direction.  It looks like you have it setup to pass caller id
from the pbx to (77)asterisk.  Is the pbx even sending caller id?
Is it sending it in the form you have configured in Asterisk?
(dtmf, polarity start, dtmfcidlevel=???)


On Sun, Dec 9, 2012 at 11:42 PM, Harish Mandowara <
asteriskhelp2...@gmail.com> wrote:

> my scenario is below
>
> analog phone (10 to 99)--> pbx-->(77)asterisk> jitsi(2000)
>
> i have analog telephone interface numbered 77 attached with asterisk and
> other sip user is 2000 on jitsi.
>
> I can call from any number from 10 to 99(in intercom) on 77 and ivr
> response will come then i can typed 2000# and call go to 2000 named user
> in asterisk.
>
> Now my problem is when i am calling from 10 to 99 (any number) this number
> should display to sip 2000's user. But its not showing to user. Its 
> showsasterisk@my_asterisk_server_ip 
> .
>
> my config. as follow
>
> extension.conf
>
> exten => s,1,Goto(phrase-menu,s,1)
>
> [phrase-menu]
>
> exten => s,1,Answer()
> exten => s,2,Wait(1)
> exten => s,3,Read(PHRASEID,/var/lib/asterisk/sounds/custom/soip)
> exten => s,4,Wait(2)
> exten => s,5,Set(CALLERID(num,CID)=${CALLERID})
> exten => s,6,Dial(SIP/${PHRASEID},40,tT)
> exten => h,1,Hangup()
>
>
> and in chan_dahdi.conf
>
> ; General options
> [channels]
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> threewaycalling=yes
> transfer=yes
> echocancel=yes
> echocancelwhenbridged=yes
> cidsignalling=dtmf
> cidstart=polarity
> callerid=asreceived
> rxgain=0.0
> txgain=0.0
> ;FXO Modules
> group=1
> echocancel=yes
> signalling=fxs_ks
> context=default
> channel=1-20
>
> #include dahdi-channels.conf
>
>
> any help
>
> thanks..
>
>
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Re: [asterisk-users] Asterisk not starting (illegal instruction core dumped)

[Christopher Harrington]

Build for the wrong processor type? Wrong arch? Kernel binary format
support?


On Tue, Nov 27, 2012 at 6:05 AM, Adolphus Enaboifo <
adolphus.enabo...@osenkorp.com> wrote:

> Hi List members,
> Thanks for the support so far as I try to install and test my first
> asterisk system.
> I was able to finally install asterisk-1.8.18.0 with libpri-1.4.13 and
> dahdi-linux-complete-2.6.1+2.6.1 according to the instructions given in
> the online documentation (asterisk the definitive guide).
> But while trying to start asterisk with the following command
> "/usr/sbin/asterisk -cvvv" or "/usr/sbin/asterisk -c" I get the message
> "Illegal instruction (core dumped)"
> Kindly advice on what to do.
>
> thanks
>
> Adolphus Enaboifo
>
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Re: [asterisk-users] SIP password probe

[Christopher Harrington]

It's an open source project. Pay a programmer or make the modification
yourself and submit a patch.


On Sat, Nov 24, 2012 at 4:51 PM, Ron Wheeler  wrote:

> I looking through my logs, I found that people where probing my SIP
> accounts looking for passwords.
> Asterisk was helping them out by processing hundreds of requests per
> minute.
> I did a bit of Googling and this seems to be a frequent knock against
> Asterisk's security.
>
> It would seem pretty simple to add a configuration setting to sip.conf to
> delay the response to a bad account or password.
>
> There is a half measure to confuse the probe by sending the same error
> return for either error.
> It appears that many people have complained that this should be the
> default setting only changed if your are debugging a problem.
>
> There is no reason for a working system to ever have bad passwords so this
> is clearly an attack in almost every case.
>
> A simple delay would solve the problem for most people who use reasonable
> passwords.
>
> I had to install fail2ban which is a PITA but thanks to someone's clear
> recipe, I was able to get it working.
>
> I hope that this can be worked into a release soon.
>
> Ron
>
> --
> Ron Wheeler
> President
> Artifact Software Inc
> email: rwhee...@artifact-software.com
> skype: ronaldmwheeler
> phone: 866-970-2435, ext 102
>
>
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Re: [asterisk-users] Conf into a call in progress

[Christopher Harrington]

On Sun, Nov 18, 2012 at 11:32 AM, Michael  wrote:

> Gentlemen,
>
> So, from your answers I understand that I have 2 options:
> 1. AMI "Redirect" command
> 2. Asterisk command "ChannelRedirect"
>
> I'm inclined to prefer the 2nd option, as we've never used AMI, but I
> don't know if it can be web-initiated.
>
>
If you're unfamiliar with the AMI, I would strongly suggest becoming
familiar with it. We use PHP with a socket connection to the asterisk AMI
and it works fantastically. This is precisely the kind of thing the AMI was
meant to do.



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Re: [asterisk-users] "Simple" failover configuration

[Christopher Harrington]

On Thu, Nov 15, 2012 at 8:59 AM, Chris Nighswonger <
cnighswon...@foundations.edu> wrote:

> At present I have two hardware identically freepbx/asterisk boxes. The
> mysql db on one is slaved to the other and all config files are
> rsync'd once every 24 hours (we have few configuration changes).
>
> We use Polycom 321/331/550/650 phones, and I notice that these phones
> can be configured with two SIP servers.
>
> Would the simplest approach to failover be to just configure my
> primary asterisk server as the first SIP server and my backup as the
> second?
>
> Unless your Polycom phones automatically detect that the primary Asterisk
server has returned after an outage, you will likely end up with a
partition, won't you?


> Kind Regards,
> Chris
>
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Re: [asterisk-users] problem on LDAP (Invalid credential)

[Christopher Harrington]

Posting your email again after 12 hours is not going to make anyone more
likely to help you. Please don't do that.


On Thu, Nov 8, 2012 at 11:39 AM, Samira Hosseini
wrote:

>
>
>
> Hello all,
>
> I am going to register asterisk sip users through active directory
> accounts LDAP (that is a separated server with ip : 192.168.11.17)
> So I have followed the below link as well:
>
> https://wiki.asterisk.org/wiki/display/AST/LDAP+Realtime+Driver
>
> http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html
>
> http://ensiwiki.ensimag.fr/index.php/Asterisk's_external_configuration_(LDAP)
>
>
>
>
>
> Server:192.168.14.90  => asterisk
> server:192.168.11.17 =>  ActiveDirectory
> Finally, this is my configuration file :
>
> [root@PBX ~]# telnet 192.168.11.17 389
> Trying 192.168.11.17...
> Connected to 192.168.11.17 (192.168.11.17).
> Escape character is '^]'.
>
> [_general]
> host=192.168.11.17; LDAP host
> port=389
> protocol=3   ; Version of the LDAP protocol to use; default is 3.
> url=ldap://192.168.11.17:389
> basedn=dc=example,dc=com
> ;User=cn=,dc=example,dc=com
> ;User=cn=join_lan,dc=example,dc=com
> ;User=cn=sa_hosseini,dc=rasana,dc=ir
> User=cn=lan,cn=technical,cn=xyz,cn=join_lan,dc=example,dc=com
> Pass=123456
> ---
> vim /etc/asterisk/extconfig.conf
> sipusers => ldap,"dc=example,dc=com",sip
> 
> vim /etc/asterisk/sip.conf
> [general]
> callevents=yes
> rtcachefriends=yes
>
>
> but i got the follwoing error :
>
>
>
> PBX*CLI> module reload res_config_ldap.so
> -- Reloading module 'res_config_ldap.so' (LDAP realtime interface)
>   == Parsing '/etc/asterisk/res_ldap.conf':   == Found
> [Nov  8 09:38:06] WARNING[8687]: res_config_ldap.c:1750 ldap_reconnect:
> bind failed: Invalid credentials
> [Nov  8 09:38:06] WARNING[8687]: res_config_ldap.c:1598 reload: Couldn't
> establish connection to your directory server. Check debug.
>   == LDAP RealTime driver reloaded.
>
> Then i have registered with user:join_lan;pass:123456 domain:192.168.14.90
> and get the following error on CLI:
> Verbosity is at least 15
> [Nov  8 09:41:42] NOTICE[8674]: chan_sip.c:25005 handle_request_register:
> Registration from '"join_lan"' failed for '
> 192.168.19.21:38968' - No matching peer found
> [Nov  8 09:41:42] NOTICE[8674]: chan_sip.c:25005 handle_request_register:
> Registration from '"join_lan"' failed for '
> 192.168.19.21:38968' - No matching peer found
>
>
>
>
>
>
>
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Re: [asterisk-users] Incoming Fax to Recipient using OCR

[Christopher Harrington]

On Tue, Nov 6, 2012 at 1:50 PM, Roy Abshire  wrote:

> I have fax working but since most people and services don't know how to
> Fax to Extensions,
> I installed tesseract to convert the Fax to Text.
>
> I really only need the First Page converted and will tell Faxers to make
> sure they put To: Name on the cover page.
>
> tesseract is converting the entire fax fine but its unnecessary and extra
> time to convert the entire fax.
>
> I searched and can't find anything on how to tell it just to do the first
> page.  Does anyone have any ideas?



If you're passing a TIFF file to tesseract, you can pass it through
imagemagick first to pop off the first "page". This really seems off-topic
for Asterisk.

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Re: [asterisk-users] play wav file

[Christopher Harrington]

On Mon, Nov 5, 2012 at 10:52 AM, Jerry Geis  wrote:

> I converted the wave file to 8K, mono and it doesn't sound very good, I am
> also
> using 1.4.43 and ulaw,alaw,gsm allowed.
>
>
This has been covered just recently, try searching for "mp3" on the mailing
list.

What format will give me the best sounding output and how do I get that?
> Do I need somethink like g722?
>
>
Keep in mind that you are going to be using codecs and hardware that are
optimized for speech, so anything that isn't speech is not going to sound
good. In that case, "best" is really going to depend on what the content is
and will probably require you to simply test all of the permutations and
find the one that sounds the "least bad".

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Re: [asterisk-users] not hear the busy playtone

[Christopher Harrington]

On Thu, Nov 1, 2012 at 10:25 AM, Jerry Geis  wrote:

> second phone calls in and I detect the Console/dsp is busy, and i try to
> use
> playtones(busy) and I hear nothing. (see below)
>
>
I experienced a similar issue in the past, where Asterisk and DAHDI seemed
to disagree about my zone. In any case, try using Congestion() instead of
PlayTones(busy).



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Re: [asterisk-users] asterisk and sip web client

[Christopher Harrington]

On Sun, Oct 28, 2012 at 2:48 PM,  wrote:

> Hello guys,
> I would like to use asterisk with a html sip web client.
>
> What asterisk version or particular question are required?
>
>
If you're starting without any pre-existing configuration, it would be
smart to use the current stable release of Asterisk. Ultimately, however,
nearly all stable releases of Asterisk are appropriate for SIP, like nearly
all models of cars are appropriate for driving on roads. This "HTML SIP web
client", is it something you're developing? Or something you have found
existing? What are you trying to accomplish?


> Thanks,
> Regards
>
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Re: [asterisk-users] high capacity analog <-> sip gateway

[Christopher Harrington]

On Thu, Oct 25, 2012 at 4:09 PM, Carlos Alvarez wrote:

> I always advocate throwing out old analog phones as they will be a pain,
> but understand if you absolutely cannot.  Just keep in mind you can get a
> decent VoIP phone for $60 that is very likely to be nicer than what they
> have now and do much more.
>
>
Out of curiosity, would you mind sharing that with us?


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Re: [asterisk-users] i extension not triggering

[Christopher Harrington]

On Thu, Oct 25, 2012 at 11:24 AM, Tony Mountifield wrote:

> The 'i' extension is not used when entering a context. You can only enter
> a context (with Dial(), Goto(), etc), at an extension that exists. If it
> doesn't exist, the context cannot be entered.
>

So it sounds like what he's really looking for is the 's' extension, maybe?

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Re: [asterisk-users] i extension not triggering

[Christopher Harrington]

On Thu, Oct 25, 2012 at 10:42 AM, Danny Nicholas  wrote:

> Based on the output below, DockPhone is expecting to be reached with a
> dialstring of 444.  If you change 444 to ZXX, the problem should go away.
>
> The point is that he's trying to trigger the invalid extension dialplan.
Not that he's trying to reach 444.

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Re: [asterisk-users] as soon as Phone rings I'm disconnected yet phone rings two more times‏

[Christopher Harrington]

On Tue, Oct 23, 2012 at 7:54 PM, Mitchell Johnson
wrote:

>
> One of the things I'm trying to do it to connect my 8x8 DTA 310 terminal
> adapter onto my asterisk.
>

What version of Asterisk are you using?

[Oct 18 16:27:46] NOTICE[1513]: chan_sip.c:23352 handle_request_invite:
> Call from '' (172.16.200.1:65451) to extension '5000' rejected because
> extension not found in context 'default'.
>

Did you mean to include this notice in your email? It indicates a dialplan
problem.


> -- Executing [5000@pstn-incoming:1] Dial("SIP/172.16.200.1-0006",
> "SIP/5000,20|p") in new stack
>

The pipe has been deprecated in more recent versions of Asterisk, make sure
this isn't related to your issue.

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Re: [asterisk-users] Can't get Lua Pattern Matching to work

[Christopher Harrington]

On Tue, Oct 23, 2012 at 2:34 PM, Danny Nicholas  wrote:

> Nope – see page 138 of the Asterisk manual – N matches 2-9 and X matches
> 0-9 so the N excludes numbers starting with 0 or 1.
>
> **
>
Ah, sorry, I was thrown off by you suggesting _NXX. which wouldn't have
matched either. So Cody needs _ZXX as the pattern.


>  **
>
> On Tue, Oct 23, 2012 at 2:18 PM, Danny Nicholas  wrote:
>
> **
>
> _NXX is only going to match a 3 digit number.  I think you need _NXX.  For
> this case.
>
>
> --
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Re: [asterisk-users] Can't get Lua Pattern Matching to work

[Christopher Harrington]

On Tue, Oct 23, 2012 at 2:18 PM, Danny Nicholas  wrote:

> _NXX is only going to match a 3 digit number.  I think you need _NXX.  For
> this case.
>
> **
>
Wouldn't _NXX match 107? That's what he's saying isn't working.


>  **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Cody Harris
> *Sent:* Tuesday, October 23, 2012 2:17 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* Re: [asterisk-users] Can't get Lua Pattern Matching to work
>
>

> Shouldn't _NXX match 107?**
>
> **
>
>

-- 

> **
>
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Re: [asterisk-users] Voicemail to text for Asterisk

[Christopher Harrington]

On Mon, Oct 22, 2012 at 3:05 PM, Lefteris Zafiris  wrote:

> If you are able to find a reliable way of chopping speech samples in
> segments no bigger
> than 20 seconds based on silence detection, so words wont be cut in half,
> you might come
> up with something very similar to Google Voice transcription service.
>

Unfortunately Google's transcription is vastly improved by its context
comprehension (for instance, understanding that the word "phone" is likely
to be followed by words like "call" or "number") and chopping up the audio,
even between words, will reduce that context data for the transcriber.

Good luck, anyway.

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Re: [asterisk-users] Voicemail to text for Asterisk

[Christopher Harrington]

On Mon, Oct 22, 2012 at 2:16 PM, Carlos Alvarez wrote:

> A customer has asked us to provide that feature.  I know there are a few
> methods and products out there, but I haven't paid attention in a while.
>  It is for about 300 users, and we'll consider open as well as paid-for
> products.  We would prefer to pay for supported products as the cost will
> be passed on to the customer and they are willing to pay for quality.  Do
> not want any complex scripting screwing around with third parties and such.
>  Your ideas welcome.
>
> All automated solutions -- paid or free -- are terrible. The technology
simply does not exist at this point at a level that is acceptable to most
customers. If quality is paramount, you are better off doing the
transcription in-house with a human.


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Re: [asterisk-users] Tell apart between network disruption and asterisk restart via AMI

[Christopher Harrington]

On Fri, Oct 19, 2012 at 12:31 PM, Alex Villací­s Lasso <
a_villa...@palosanto.com> wrote:

> I have a program that connects to the Asterisk Manager Interface through
> port 5038 on a remote machine. Suppose I get a TCP disconnection on my
> program. The program will then attempt to reconnect to the AMI and will
> eventually succeed. Is there a way to check whether the disconnection was
> caused by a network disruption, or an Astersk restart/crash? In other
> words, is the Asterisk process I contacted now the same as the one I was
> connected before, or is it a different one? The reason I want to know is
> that I have a cache of information that is costly to parse (scales linearly
> with the number of extensions) and I want to know how to realize that the
> information is now stale.
>
>
In the CLI, you can run `core show settings` which will tell you the
startup time of the server.


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Re: [asterisk-users] Problems with AGI and existing channel

[Christopher Harrington]

On Thu, Oct 18, 2012 at 6:33 AM, Magnus Löfqvist  wrote:

> Hi,
>
> ** **
>
> “Asterisk 1.8.10.0-1digium1~squeeze built by pbuilder @ nighthawk on a
> x86_64 running Linux on 2012-03-08 23:05:09 UTC”
>
> ** **
>
> We have some problem when running a AGI script (build with PHP), existing
> channels (all of them) gets a “hickup” and then continues.
>

You say all of your connected channels experience an audio glitch? Sounds
like PHP is briefly consuming all of your CPU or RAM and causing Asterisk
to fail to meet timing demands.

Whatever your PHP CLI interface is (mine is just `php` but yours may be
different), run

time php < /dev/null

The returned time should be very, very small. Mine was "real0m0.018s".

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Re: [asterisk-users] Connecting Skype to Asterisk

[Christopher Harrington]

On Fri, Oct 12, 2012 at 1:44 PM, Philip Bennefall  wrote:
> Hi all,
>
> I have an Asterisk PBX under development, that I would like to link to a
> Skype account if possible. The idea is that people would call a particular
> Skype username, and be redirected to my SIP and through that to Asterisk. Is
> this doable? I have looked around and saw the Skype for Asterisk driver, but
> of course that has been discontinued. Are there any other options? I would
> prefer not to have to go through the regular PSTN telephone network but
> directly from Skype to Asterisk via SIP. If you have any tips on how to
> configure my sip.conf to get this working, this would also be highly
> appreciated.
>

It looks like this is what you want:
http://www.skype.com/intl/en/business/skype-connect/


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Re: [asterisk-users] Digium D40 phones and Caller ID

[Christopher Harrington]

On Fri, Oct 12, 2012 at 11:47 AM, Matthew Jordan  wrote:
> After loading, [peer01] shows up as a known SIP endpoint.  Calling
> peer01 displays the same caller ID as before, i.e., either 101/D40 01 or
> 101/foo.  Note that none of this using the DPMA either.
Here's something important that I think I was missing. The CallerID
construct seems to be working, but only after the call is answered.
(In my testing, I didn't think to pick up the line, just ending the
call from the source if I didn't see the CID.)

Turning on SIP debugging, I can see the PAI header being sent in the
invite packet being sent to the D40 upon initiating (prior to
answering). Does this shed any light on the issue?

>> I don't know if that is necessarily true; the phones were new in box
>> but were not purchased from Digium or an authorized reseller.
>>
>
> Did they fall off the back of a truck? :-)
>
> Call anyway.  If you purchased said phones in a legitimate manner, they
> should be able to help you.
They were acquired from a company that unexpectedly imploded,
actually. I'm trying to avoid calling as that kills my ability to
multitask on my other projects; naturally I'm the "do everything" guy
at my company.

> (And, as previously suggested, you may want to make sure you've upgraded
> to the latest firmware, just to rule out any solved problems).
They are absolutely all on 1.1.0.

-- 
-Chris Harrington
ACSDi Office: 763.559.5800
Mobile Phone: 612.326.4248

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Re: [asterisk-users] Digium D40 phones and Caller ID

[Christopher Harrington]

On Fri, Oct 12, 2012 at 9:59 AM, Matthew Jordan  wrote:
> On 10/11/2012 05:39 PM, Christopher Harrington wrote:
>> First post to this mailing list. I'll keep it brief: My D40 phones
>> don't show the "name" component of CALLERID.
>> It only displays the number. This includes calls originating from PSTN
>> with their own CID already set, and calls
>> where we've specifically filled in this data. Changing the destination
>> of my test extension to a softphone (zoiper
>> in this case) correctly displays the information. sip.conf already
>> contains sendrpid=pai.
>>
>> From what I can tell, this appears to be a Digium phone limitation. Or
>> am I missing something crucial?
>>
>
> No, the D40s display the name.
>
> Using the following configuration in sip.conf:
I'm using users.conf, so my questions will mostly pertain to that. I
apologize for what I'm sure are some dumb questions up ahead here.

>
> [peer01]
> type = peer
Is "type=peer" strictly necessary? I don't know how they're currently
being specified from users.conf, is that possible to specify in
users.conf? I was under the impression that peers specified in
users.conf would be type=friend.

> secret = 
> callerid = "D40 01" <101>
> host = dynamic
My hosts are manually specified (ie they do not register), that
shouldn't matter, correct?

> sendrpid = pai
I have this specified in the general section of sip.conf. Does this
need to be specified per-peer?

> disallow = all
> allow = ulaw
> allow = g722
>
> And extensions.conf:
>
> exten => 101,1,NoOp()
> same => n,Set(CALLERID(name)=foo)
> same => n,Dial(SIP/101)
> same => n,Hangup()
This is effectively what I've done with my test extension. I've tried
both CALLERID(all)=... and CALLERID(name)=...

>
> Shows the following on the D40:
> 101
> foo
>
> If I remove the CALLERID function call, the D40 shows:
> 101
> D40 01
>
> Note that this is using the 1.1.0 firmware.  I imagine there is a
Yep, 1.1.0.0.

> configuration issue somewhere.  You may want to provide your entire
It occurs to me that you're probably using DPMA, and I am not. That's
probably where this issue is.

> configuration, or - since you have purchased phones from Digium - call
> technical support.  They should be able to help you resolve this issue.
I don't know if that is necessarily true; the phones were new in box
but were not purchased from Digium or an authorized reseller.

>
> --
> Matthew Jordan
> 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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-- 
-Chris Harrington
ACSDi Office: 763.559.5800
Mobile Phone: 612.326.4248

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Re: [asterisk-users] Recommendation for extension mapping on inbound T1 line

[Christopher Harrington]

On Fri, Oct 12, 2012 at 9:10 AM, Mitch Claborn  wrote:
> Converting this customer from a MiTel system to asterisk. Discovered that
> the inbound calls from the T1 are going to extension 366.  (This was mapped
> in the MiTel for some arcane purpose.)  The dial plan I am currently using
> is shown below.  When loading the dial plan, I get this warning:
>
>  WARNING[5004]: pbx_config.c:1561 pbx_load_config: The use of '_.' for an
> extension is strongly discouraged and can have unexpected behavior.  Please
> use '_X.' instead at line 331 of extensions.conf
>
> Question: Do I need to worry about this warning?
>From what I've seen, _X. will always match any extension starting with
a digit, whereas _. matches everything, including things other than
incoming calls (like hangups and timeouts for instance).

In any case, your question is already answered in the diagnostic
output: that pattern is strongly discouraged. So yes, you need to
worry about it. If you didn't, they wouldn't warn you about it.

>
> I'm a little leery of just using 366 in the dialplan, since the company we
> are dealing with is a little flaky.
>
>
> [from-pstn]
> exten =>s,1,NoOp(pstn call from ${CALLERID(all)} exten=${EXTEN})
>   same =>n,Goto(MainMenu,s,1)
> exten =>_.,1,NoOp(pstn call from ${CALLERID(all)} exten=${EXTEN})
>   same =>n,Goto(MainMenu,s,1)
>
>
>
> --
>
> Mitch
>
>
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-- 
-Chris Harrington
ACSDi Office: 763.559.5800
Mobile Phone: 612.326.4248

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[asterisk-users] Digium D40 phones and Caller ID

[Christopher Harrington]

First post to this mailing list. I'll keep it brief: My D40 phones
don't show the "name" component of CALLERID.
It only displays the number. This includes calls originating from PSTN
with their own CID already set, and calls
where we've specifically filled in this data. Changing the destination
of my test extension to a softphone (zoiper
in this case) correctly displays the information. sip.conf already
contains sendrpid=pai.

>From what I can tell, this appears to be a Digium phone limitation. Or
am I missing something crucial?

--
-Chris Harrington
ACSDi Office: 763.559.5800
Mobile Phone: 612.326.4248

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