Re: [asterisk-users] Can't compile DAHDI - wrong kernel source

[Chandrakant Solanki]

Hi

Following steps to do...

1] # cd /usr/src/kernels/
2] # ln -s 2.6.18-128.2.1.el5.028stab064.7-i686 2.6.18-028stab064.7

Try this 'n let me know... Hope this will work fine...


-- 
Regards,

Chandrakant Solanki

On Thu, Jul 15, 2010 at 12:00 PM, Thermal Wetland
wrote:

> On Wed, Jul 14, 2010 at 8:09 PM, Chandrakant Solanki
>  wrote:
> > Hello
> >
> > What will be your exact kernel version. Give me output "uname -a"
> command.
> >
> > --
> > Regards,
> >
> > Chandrakant Solanki
> >
>
> Thank you for the help!  Here is the output:
> [r...@ip-97-74-119-59 ~]# uname -a
> Linux ip-97-74-119-59.ip.secureserver.net 2.6.18-028stab064.7 #1 SMP
> Wed Aug 26 13:11:07 MSD 2009 i686 i686 i386 GNU/Linux
>
> -Thermal
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
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Re: [asterisk-users] Can't compile DAHDI - wrong kernel source

[Thermal Wetland]

On Wed, Jul 14, 2010 at 8:09 PM, Chandrakant Solanki
 wrote:
> Hello
>
> What will be your exact kernel version. Give me output "uname -a" command.
>
> --
> Regards,
>
> Chandrakant Solanki
>

Thank you for the help!  Here is the output:
[r...@ip-97-74-119-59 ~]# uname -a
Linux ip-97-74-119-59.ip.secureserver.net 2.6.18-028stab064.7 #1 SMP
Wed Aug 26 13:11:07 MSD 2009 i686 i686 i386 GNU/Linux

-Thermal

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Re: [asterisk-users] Can't compile DAHDI - wrong kernel source

[Chandrakant Solanki]

Hello

What will be your exact kernel version. Give me output "uname -a" command.

-- 
Regards,

Chandrakant Solanki

On Thu, Jul 15, 2010 at 6:58 AM, Thermal Wetland
wrote:

> On Wed, Jul 14, 2010 at 4:55 AM, bruce bruce  wrote:
> >
> > I am stuck with the same problem but I have used asterisk yum repository
> and it worked by itself without me worrying for kernel stuff.
> > However, I need to install speex codec and now I am stuck as it doesn't
> get picked up by the yum asterisk install somehow. I have lib speex and
> speex already installed and when doing "yum install asterisk16" I don't see
> speex in "core show translation" Is there anything specific I have to do?
> > Do I have to build from source as well?
> > -Sorry, didn't mean to hijack the thread.
> > Thanks,
> > Bruce
> > On Wed, Jul 14, 2010 at 5:08 AM, Chandrakant Solanki <
> solanki.chandrak...@gmail.com> wrote:
> >>
> >> Hi
> >>
> >> If you install rpm from any location it goes to its default location.
> >>
> >> You just go for above steps. For kernel you can go for
> http://kernel.org
> >>
> >> --
> >> Regards,
> >>
> >> Chandrakant Solanki
> >>
> >> On Wed, Jul 14, 2010 at 2:06 PM, liuxin  wrote:
> >>>
> >>> Hi.
> >>> The best easy way is:
> >>> copy kernel-devel-2.6.18-028stab064.7.rpm to /usr/src
> >>> then run rpm -ivh kernel-devel-2.6.18-028stab064.7.rpm
> >>>
> >>> 2010/7/14 Gareth Blades 
> 
>  Thermal Wetland wrote:
>  > I have a virtual server with godaddy but can not compile DAHDI as it
>  > complains that I do not have the correct kernel source.
>  >
>  > The package installed is - kernel-devel-2.6.18-164.11.1.el5.i686:
>  > Package kernel-devel-2.6.18-164.11.1.el5.i686 already installed and
>  > latest version
>  > Nothing to do
>  >
>  > uname -a returns:
>  > Linux ip-XXX-XXX-XXX-XXX.ip.secureserver.net
>  >  2.6.18-028stab064.7
> #1
>  > SMP Wed Aug 26 13:11:07 MSD 2009 i686 i686 i386 GNU/Linux
>  >
>  > When I try to compile DAHDI it fails with:
>  > make[2]: Leaving directory
>  >
> `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware'
>  > You do not appear to have the sources for the 2.6.18-028stab064.7
> kernel
>  > installed.
>  >
>  > Is there a way to trick DAHDI to use the installed kernel?
>  >
>  > Thanks for the help!
>  >
>  > --
>  > -Thermal
>  >
> 
>  What kernel versions do you have installed?
> 
>  If you are currently running an older kernel but installed a newer
>  kernel and sources but havent rebooted to activate the new one yet
> then
>  it may still be trying to locate the source for the older running
> kernel.
> 
> 
> 
>
>
> I was able to download the rpm's and install them:
>
> [r...@ip-97-74-119-59 src]# rpm -ivh
> ovzkernel-2.6.18-128.2.1.el5.028stab064.7.i686.rpm
> warning: ovzkernel-2.6.18-128.2.1.el5.028stab064.7.i686.rpm: Header V3
> DSA signature: NOKEY, key ID a7a1d4b6
> Preparing...###
> [100%]
> package ovzkernel-2.6.18-128.2.1.el5.028stab064.7.i686 is
> already installed
>
> [r...@ip-97-74-119-59 src]# rpm -ivh
> ovzkernel-devel-2.6.18-128.2.1.el5.028stab064.7.i686.rpm
> warning: ovzkernel-devel-2.6.18-128.2.1.el5.028stab064.7.i686.rpm:
> Header V3 DSA signature: NOKEY, key ID a7a1d4b6
> Preparing...###
> [100%]
> package ovzkernel-devel-2.6.18-128.2.1.el5.028stab064.7.i686
> is already installed
>
> [r...@ip-97-74-119-59 src]# cd -
> /usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0
> [r...@ip-97-74-119-59 dahdi-linux-complete-2.3.0.1+2.3.0]# make all
> make -C linux all
> make[1]: Entering directory
> `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux'
> make -C drivers/dahdi/firmware firmware-loaders
> make[2]: Entering directory
>
> `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware'
> make[2]: Leaving directory
>
> `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware'
> You do not appear to have the sources for the 2.6.18-028stab064.7
> kernel installed.
> make[1]: *** [modules] Error 1
> make[1]: Leaving directory
> `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux'
> make: *** [all] Error 2
>
> The directories in /usr/src/kernels is:
> [r...@ip-97-74-119-59 kernels]# ls -l
> total 51328
> drwxr-xr-x 20 root root 4096 Jul 14 18:04
> 2.6.18-128.2.1.el5.028stab064.7-i686
> drwxr-xr-x 19 root root 4096 Jul 13 20:25 2.6.18-164.11.1.el5-i686
> drwxrwxr-x 19 root root 4096 Feb 23  2007 linux-2.6.18.8
>
> I tried to install the kernel from source but couldn't find the exact
> kernel, I installed linux-2.6.18.8 as I was the closest.
>
> Both of the directories in /usr/src/kernels/ have the -i686 suffix, is
> that the issue?
>
> --
> -Thermal
>
> --
>

Re: [asterisk-users] SKYPE - Authenticate incoming call automatically

[Neeraj Chand]

Hi All, 

After getting licences for Skype for asterisk a while ago I finally got
around to setting up a server with two channels and setting up a bcp on
the skype end. 

The idea behind this is the following: 

Users can dial into the PBX, get authenticated and only after
authentication get access to internal PBX extensions. 

I CAN do this with a PIN, no sweat, but from a user perspective it
becomes a bit clunky, i.e. password to remember, security in terms of
pin leaks, multiple passwords for users, etc. 

I was wondering if there was a way I could extract the "FROM - USER" and
assign it to a variable, then do a lookup of that username in a database
using ODBC to decide whether to allow or disallow access. 

NOTE: The bit I need help with is extracting the "FROM - USER" the rest
of the stuff I've done already / before.  

Thanks, 

Neeraj. 

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[asterisk-users] Get channel name of originated channel

[Deepesh D]

Hello,

I am using asterisk manager interface (http) for originating calls.
How can I get the name of the channel which is created by originate? I
want to use this channel for other manager commands like Atxfer,
Monitor, Hangup etc.

If I do action=originate, channel=SIP/200  then it creates a channel
like 'SIP/200-0865ff80' which I can see in the asterisk console using
"core show channels verbose".  Now if I want to transfer this call I
have to use action=Atxfer, channel=SIP/200-0865ff80 for which I need
the channel name. Is there any way to get this channel name or set the
channel name during originate? On what basis does asterisk assign
channel names, is it random?

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Re: [asterisk-users] power outage

[C F]

On Wed, Jul 14, 2010 at 5:03 AM, liuxin  wrote:
> Hi,
> probably a misconfiguration or you havent plugged the cable in yet.

OMG you are right, I forgot to plug in the cable. Hey but wait which
cable you talking about?

>
> 2010/7/14 C F 
>>
>> It has nothing to do with the D-channel, however you will never know
>> if the B-channels work if the D-channel is down. D-channel is what
>> allows the B-channels to work, and is the first place to troubleshoot.
>> If something is screwed up with the the symptom you'll get is a
>> non working PRI, the way to check it is by means of seeing if the
>> D-channel synced up or not.
>>
>>
>> On Mon, Jul 12, 2010 at 2:17 AM, Justin Case
>>  wrote:
>> > What would the power have to do with the D Channel ? Isn't which channel
>> > used a logical setting (as opposed to physical). I am not saying your
>> > wrong
>> > I am just trying to understand why it happens.
>> >
>> > On Mon, Jul 12, 2010 at 7:56 AM, C F  wrote:
>> >>
>> >> I have found that sometimes shutting down the machine waiting a full
>> >> minute while the power cable is unplugged then restarting can fix such
>> >> problems if it's power related.
>> >>
>> >> On Fri, Jul 9, 2010 at 12:42 PM, Jerry Geis 
>> >> wrote:
>> >> > I have a TE205P that has been working fine for 2 years.
>> >> > power outage yesterday took out my everything for over an hour.
>> >> >
>> >> > Everything has come back up except the PRI. My provider has checked
>> >> > it
>> >> > to the box
>> >> > and says everything looks good on their end.
>> >> >
>> >> > I get this message:
>> >> > [Jul  9 12:40:32] WARNING[13709] chan_dahdi.c: No D-channels
>> >> > available!
>> >> > Using Primary channel 24 as D-channel anyway!
>> >> >
>> >> > ztcfg -vvv
>> >> >
>> >> > Zaptel Version: 1.4.12.1
>> >> > Echo Canceller: MG2
>> >> > Configuration
>> >> > ==
>> >> >
>> >> > SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)
>> >> >
>> >> > Channel map:
>> >> >
>> >> > Channel 18: Clear channel (Default) (Slaves: 18)
>> >> > Channel 19: Clear channel (Default) (Slaves: 19)
>> >> > Channel 20: Clear channel (Default) (Slaves: 20)
>> >> > Channel 21: Clear channel (Default) (Slaves: 21)
>> >> > Channel 22: Clear channel (Default) (Slaves: 22)
>> >> > Channel 23: Clear channel (Default) (Slaves: 23)
>> >> > Channel 24: D-channel (Default) (Slaves: 24)
>> >> >
>> >> > 7 channels to configure.
>> >> >
>> >> > and show status gives me condition RED of course.
>> >> >
>> >> > How do I find out whats wrong here?
>> >> >
>> >> > Jerry
>> >> >
>> >> > --
>> >> > _
>> >> > -- 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
>> >> >
>> >> > asterisk-users mailing list
>> >> > To UNSUBSCRIBE or update options visit:
>> >> >   http://lists.digium.com/mailman/listinfo/asterisk-users
>> >> >
>> >>
>> >> --
>> >> _
>> >> -- 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
>> >>
>> >> asterisk-users mailing list
>> >> To UNSUBSCRIBE or update options visit:
>> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
>> >
>> >
>> > --
>> > _
>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>> >               http://www.asterisk.org/hello
>> >
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>> > To UNSUBSCRIBE or update options visit:
>> >   http://lists.digium.com/mailman/listinfo/asterisk-users
>> >
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>               http://www.asterisk.org/hello
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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asterisk-users ma

Re: [asterisk-users] How to calculate number of speakers needed for PAGING and INTERCOM coverage area?

[C F]

I'm happy to hear it worked out so well with so little. :)

On Wed, Jul 14, 2010 at 12:39 AM, bruce bruce  wrote:
> Thanks for the input guys. For other refrence, a CyberData Voip Amplifier
> which supplies 10 Watt to each of the two bogen 30 Watt speakers did the job
> for a 35, square feet warehouse with environmental noise level of
> slightly higher than standard but not those of industrial.
> Only two speakers and done deal. Though I know that three speaker would have
> been the perfect solution but 4 would cover every single little corner and
> be an overkill.
> -Bruce
>
> On Tue, Jul 13, 2010 at 8:47 PM, C F  wrote:
>>
>> I agree with horns you'll usually get better coverage. I have done
>> this in the past with 5 speakers for a 30k sq ft warehouse very good
>> coverage. Using bogen horns. This was for a 300ft by 100ft warehouse.
>> Starting at 30 ft of the 300ft wall at the 50ft line off the 100ft
>> side I installed a horn every 60ft alternating facing one north and
>> the other south, which ended up 3 facing one way and 2 the other. You
>> can get double horn speakers which will face 2 sides. I wouldn't mount
>> them on the wall specifically not so low as fork lifts and what not
>> will damage them.
>>
>>
>> On Mon, Jul 12, 2010 at 2:17 PM, bruce bruce  wrote:
>> > Well, these are horn speakers with 30 Watt which will receive 10 Watt
>> > only
>> > from Amplifer. I am not connecting them to ceiling so maybe 10 feet off
>> > the
>> > ground. I guess my coverage would be better???
>> > Based on your calculations for for 40k sqfeet that would be 33 speakers.
>> > I
>> > think that's way too much of an overkill.
>> > thanks,
>> > Bruce
>> >
>> > On Mon, Jul 12, 2010 at 1:05 AM, C F  wrote:
>> >>
>> >> In my experience using height for radius works, for example if you
>> >> have a 20 ft high ceiling then the coverage for one speaker would be
>> >> 40 ft diameter circle (around 1200 sq ft). Of course overlapping 5 ft
>> >> has never killed anyone, but this really depends on the power of the
>> >> speaker, I usually deal with 70v speakers tapped at 16 or 8 watts
>> >> depending on how many speakers I put on one amplifier and the output
>> >> wattage of that amplifier.
>> >>
>> >>
>> >>
>> >> On Fri, Jul 9, 2010 at 2:29 AM, bruce bruce 
>> >> wrote:
>> >> > Hi Guys,
>> >> > I am looking to buy a 25 Watt output CyberData VoIP amplifier and to
>> >> > use
>> >> > 2
>> >> > Bogen sp308a speakers with it for a 40, 000 squar feet area and 21
>> >> > feet
>> >> > height. Is that enough? Is there calculator online I can use to
>> >> > determine
>> >> > the number of speakers needed? I guess these speakers go in chain so
>> >> > I
>> >> > am
>> >> > not sure if the full capacity of the speaker (30 watt) will be used.
>> >> > I appreciate your advice.
>> >> > Thanks,
>> >> > Bruce
>> >> > --
>> >> > _
>> >> > -- 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
>> >> >
>> >> > asterisk-users mailing list
>> >> > To UNSUBSCRIBE or update options visit:
>> >> >   http://lists.digium.com/mailman/listinfo/asterisk-users
>> >> >
>> >>
>> >> --
>> >> _
>> >> -- 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
>> >>
>> >> asterisk-users mailing list
>> >> To UNSUBSCRIBE or update options visit:
>> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
>> >
>> >
>> > --
>> > _
>> > -- 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
>> >
>> > asterisk-users mailing list
>> > To UNSUBSCRIBE or update options visit:
>> >   http://lists.digium.com/mailman/listinfo/asterisk-users
>> >
>>
>> --
>> _
>> -- 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] Can't compile DAHDI - wrong kernel source

[Thermal Wetland]

On Wed, Jul 14, 2010 at 4:55 AM, bruce bruce  wrote:
>
> I am stuck with the same problem but I have used asterisk yum repository and 
> it worked by itself without me worrying for kernel stuff.
> However, I need to install speex codec and now I am stuck as it doesn't get 
> picked up by the yum asterisk install somehow. I have lib speex and speex 
> already installed and when doing "yum install asterisk16" I don't see speex 
> in "core show translation" Is there anything specific I have to do?
> Do I have to build from source as well?
> -Sorry, didn't mean to hijack the thread.
> Thanks,
> Bruce
> On Wed, Jul 14, 2010 at 5:08 AM, Chandrakant Solanki 
>  wrote:
>>
>> Hi
>>
>> If you install rpm from any location it goes to its default location.
>>
>> You just go for above steps. For kernel you can go for http://kernel.org
>>
>> --
>> Regards,
>>
>> Chandrakant Solanki
>>
>> On Wed, Jul 14, 2010 at 2:06 PM, liuxin  wrote:
>>>
>>> Hi.
>>> The best easy way is:
>>> copy kernel-devel-2.6.18-028stab064.7.rpm to /usr/src
>>> then run rpm -ivh kernel-devel-2.6.18-028stab064.7.rpm
>>>
>>> 2010/7/14 Gareth Blades 

 Thermal Wetland wrote:
 > I have a virtual server with godaddy but can not compile DAHDI as it
 > complains that I do not have the correct kernel source.
 >
 > The package installed is - kernel-devel-2.6.18-164.11.1.el5.i686:
 > Package kernel-devel-2.6.18-164.11.1.el5.i686 already installed and
 > latest version
 > Nothing to do
 >
 > uname -a returns:
 > Linux ip-XXX-XXX-XXX-XXX.ip.secureserver.net
 >  2.6.18-028stab064.7 #1
 > SMP Wed Aug 26 13:11:07 MSD 2009 i686 i686 i386 GNU/Linux
 >
 > When I try to compile DAHDI it fails with:
 > make[2]: Leaving directory
 > `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware'
 > You do not appear to have the sources for the 2.6.18-028stab064.7 kernel
 > installed.
 >
 > Is there a way to trick DAHDI to use the installed kernel?
 >
 > Thanks for the help!
 >
 > --
 > -Thermal
 >

 What kernel versions do you have installed?

 If you are currently running an older kernel but installed a newer
 kernel and sources but havent rebooted to activate the new one yet then
 it may still be trying to locate the source for the older running kernel.





I was able to download the rpm's and install them:

[r...@ip-97-74-119-59 src]# rpm -ivh
ovzkernel-2.6.18-128.2.1.el5.028stab064.7.i686.rpm
warning: ovzkernel-2.6.18-128.2.1.el5.028stab064.7.i686.rpm: Header V3
DSA signature: NOKEY, key ID a7a1d4b6
Preparing...    ### [100%]
    package ovzkernel-2.6.18-128.2.1.el5.028stab064.7.i686 is
already installed

[r...@ip-97-74-119-59 src]# rpm -ivh
ovzkernel-devel-2.6.18-128.2.1.el5.028stab064.7.i686.rpm
warning: ovzkernel-devel-2.6.18-128.2.1.el5.028stab064.7.i686.rpm:
Header V3 DSA signature: NOKEY, key ID a7a1d4b6
Preparing...    ### [100%]
    package ovzkernel-devel-2.6.18-128.2.1.el5.028stab064.7.i686
is already installed

[r...@ip-97-74-119-59 src]# cd -
/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0
[r...@ip-97-74-119-59 dahdi-linux-complete-2.3.0.1+2.3.0]# make all
make -C linux all
make[1]: Entering directory
`/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux'
make -C drivers/dahdi/firmware firmware-loaders
make[2]: Entering directory
`/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware'
make[2]: Leaving directory
`/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware'
You do not appear to have the sources for the 2.6.18-028stab064.7
kernel installed.
make[1]: *** [modules] Error 1
make[1]: Leaving directory
`/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux'
make: *** [all] Error 2

The directories in /usr/src/kernels is:
[r...@ip-97-74-119-59 kernels]# ls -l
total 51328
drwxr-xr-x 20 root root 4096 Jul 14 18:04
2.6.18-128.2.1.el5.028stab064.7-i686
drwxr-xr-x 19 root root 4096 Jul 13 20:25 2.6.18-164.11.1.el5-i686
drwxrwxr-x 19 root root 4096 Feb 23  2007 linux-2.6.18.8

I tried to install the kernel from source but couldn't find the exact
kernel, I installed linux-2.6.18.8 as I was the closest.

Both of the directories in /usr/src/kernels/ have the -i686 suffix, is
that the issue?

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Re: [asterisk-users] Hosted PBX in the UK

[Steve Kennedy]

On Wed, Jul 14, 2010 at 10:27:13PM +0100, Wipe_Out wrote:

>Might be off topic but I thought it would be a good place to ask.. I am
>investigating switching to a hosted PBX and dumping my old Asterisk box
>thats been running in my office for the last few years.. The few I have
>found seem very expensive..

There's several (some being on this list)

Gradwell.com cone to mind
You could also look at pibix.com who are in early stages

Steve

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Re: [asterisk-users] Distinctive ring for INTERNAL calls only? How to do it?

[Ira]

At 03:05 PM 7/14/2010, you wrote:
>Thanks for the input but that won't be good because people are not 
>going to remember two extensions for one person.

That's why there's a dialplan. But the piece I'm unsure of is how the 
second SIP address handles more than one call.

Ira 


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Re: [asterisk-users] Hosted PBX in the UK

[Ishfaq Malik]

We do hosted VoIP

www.pack-net.co.uk

Contact me off list for more details if it sounds right for you

Ish

On 14/07/10 22:27, Wipe_Out wrote:
> Hi,
>
> Might be off topic but I thought it would be a good place to ask.. I 
> am investigating switching to a hosted PBX and dumping my old Asterisk 
> box thats been running in my office for the last few years.. The few I 
> have found seem very expensive..
>
> Can anyone point me to any VoIP PBX hosts in the UK?
>
> TIA

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Re: [asterisk-users] DAHDI Outdial To Cell Phone Playing Music

[Alec Davis]

Call progress (is only experimental), relies on defined ring tones, coloured
ring (music) messes this up.
 
in chan_dahdi.conf
 
callprogress=no

busydetect=yes
busycount=4

and possibly if your incoming analog lines support it.
 
answeronpolarityswitch=yes
hanguponpolarityswitch=yes


  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deric Page
Sent: Thursday, 15 July 2010 9:36 a.m.
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] DAHDI Outdial To Cell Phone Playing Music



Using Asterisk 1.6.1.14 and dahdi 2.2.0.2+2.2.0.

We're placing outbound calls over an analog line. Some of these calls are
going to cell phones that play music rather than providing a standard ring.
As a result, the Dial command sometimes returns a DIALSTATUS of CHANUNAVAIL
and sometimes it returns BUSY. The problem is that this is happening on
calls that are being answered.

Has anyone else run into this problem and if so, is there a solution?

Thanks.

-- 

Deric Page

deric.p...@nisc.coop

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[asterisk-users] sip message to ip 330 or 550 phones

[Jerry Geis]

Is it possible to send a test message to the IP 330 or 550 polycom 
phones with asterisk?

Thanks,

Jerry

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Re: [asterisk-users] Distinctive ring for INTERNAL calls only? How to do it?

[bruce bruce]

Thanks for the input but that won't be good because people are not going to
remember two extensions for one person.

The sip header should be able to carry alert_info to internal extensions
really easily. Anyone else got a thought?

Thanks again,

On Wed, Jul 14, 2010 at 5:44 PM, Ira  wrote:

> At 11:44 AM 7/14/2010, you wrote:
> >Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra
> >phones, how can one receive distinctive ring tones for INTERNAL calls
> ONLY?
>
> It's ugly, but you could give the phone two different SIP IDs and
> give those different ringtones.
>
> Ira
>
>
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Re: [asterisk-users] Distinctive ring for INTERNAL calls only? How to do it?

[Ira]

At 11:44 AM 7/14/2010, you wrote:
>Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra 
>phones, how can one receive distinctive ring tones for INTERNAL calls ONLY?

It's ugly, but you could give the phone two different SIP IDs and 
give those different ringtones.

Ira 


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[asterisk-users] DAHDI Outdial To Cell Phone Playing Music

[Deric Page]

Using Asterisk 1.6.1.14 and dahdi 2.2.0.2+2.2.0.

We're placing outbound calls over an analog line. Some of these calls
are going to cell phones that play music rather than providing a
standard ring. As a result, the Dial command sometimes returns a
DIALSTATUS of CHANUNAVAIL and sometimes it returns BUSY. The problem is
that this is happening on calls that are being answered.

Has anyone else run into this problem and if so, is there a solution?

Thanks.

-- 
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deric.p...@nisc.coop

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[asterisk-users] Hosted PBX in the UK

[Wipe_Out]

Hi,

Might be off topic but I thought it would be a good place to ask.. I am
investigating switching to a hosted PBX and dumping my old Asterisk box
thats been running in my office for the last few years.. The few I have
found seem very expensive..

Can anyone point me to any VoIP PBX hosts in the UK?

TIA
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[asterisk-users] realtime music on hold

[Jonas Kellens]

Hello list,

using asterisk 1.4.30.

When setting up the MySQL table 'musiconhold' as described in 
http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf , 
what is the meaning of the fields :


  `*digit*` char(1) NOT NULL default '',
  `*sort*` varchar(16) NOT NULL default '',

and what are there default values ?!


What is the default value of :

  `*format*` varchar(16) NOT NULL default '',



Kind regards,

Jonas.
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[asterisk-users] Distinctive ring for INTERNAL calls only? How to do it?

[bruce bruce]

Hi Everyone,

Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra phones,
how can one receive distinctive ring tones for INTERNAL calls ONLY?

Even though FreePBX Inbound has an option for Alert_INFO but that doesn't
work when the call comes into an IVR or Queue. The calls has to go directly
to extension for external ringtone to be different. So, I am looking for
internal calls ringtones to be different rather than external call
ringtones.

Anyone has got this working?

Thanks,
Burce
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Re: [asterisk-users] Dahdi Echo canceller setup

[Ira]

At 11:23 AM 7/14/2010, you wrote:
>Is there something I need to do with HPEC to make sure the
>dahdi_genconf generates a proper system.conf or is there somewhere
>else I show tell asterisk to use HPEC?

Well, Moments later I found  /etc/dahdi/genconf_parameters which 
seems to solve the problem.

Sorry for the bother.

Ira 


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Re: [asterisk-users] Asterisk core dumping on SendFax with FFA

[Kevin P. Fleming]

On 07/14/2010 01:16 PM, Ilmars Knipšis wrote:
>Hello again!
> 
> Just info what we discovered if anybody gets the same problem.
> The reason is fax file (.tiff) resolution.
> If you try to improve fax quality by raising resolution then * crashes 
> with core dump.

This has already been fixed in recent releases of FFA; there was a bug
previously where the module would cause Asterisk to crash if a document
to be sent could not be queued (for one of many reasons).

-- 
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Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com & www.asterisk.org

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[asterisk-users] Dahdi Echo canceller setup

[Ira]

Hi

I have a TDM400 and 4 channels of HPEC. I don't use the POTs lines 
much so I didn't realize it wasn't working. This morning I was 
watching the console and noticed that the echo canceller didn't load 
when a call came in. /etc/dahdi/system.conf showed mg2 for all 4 
channels. I changed them all to hpec and then restarted dahdi and 
Asterisk and suddenly HPEC was working.

Is there something I need to do with HPEC to make sure the 
dahdi_genconf generates a proper system.conf or is there somewhere 
else I show tell asterisk to use HPEC?

Ira


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[asterisk-users] Asterisk core dumping on SendFax with FFA

[Ilmars Knipšis]

   Hello again!

Just info what we discovered if anybody gets the same problem.
The reason is fax file (.tiff) resolution.
If you try to improve fax quality by raising resolution then * crashes 
with core dump.

Best,
-- 

*Ilmars*

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Re: [asterisk-users] Where should I look for MWI settings if Aastra phones don't do it?

[bruce bruce]

Thanks for the input guys. I don't use .xml files for Aastra. Everything is
done on the UI.

#voicemail show users:

*ContextMbox  User  Zone   NewMsg*
*|default007   Alex 2*
*default2100  Peter1*
*
*
This system is using FreePBX, so I checked the Device and Users in asterisk
tables and they have "default" for voicemail setup which I think is right. I
can also see the msg.txt in a folder that has new voicemail waiting.
However, I am not sure about privileges though. Here is it:

[r...@elastix INBOX]# ls -la
total 284
drwxrwxr-x 2 asterisk asterisk   4096 Jul 14 09:57 .
drwxrwxr-x 8 asterisk asterisk   4096 Jul 13 11:57 ..
*-rw-rw-r-- 1 asterisk asterisk282 Jul 12 18:46 msg.txt*
-rwxrwxr-x 1 asterisk asterisk  83244 Jul 12 18:46 msg.wav
-rwxrwxr-x 1 asterisk asterisk   8510 Jul 12 18:46 msg.WAV
*-rw--- 1 asterisk asterisk261 Jul 14 09:57 msg0001.txt*
-rwx-- 1 asterisk asterisk 150124 Jul 14 09:57 msg0001.wav
-rwx-- 1 asterisk asterisk  15270 Jul 14 09:57 msg0001.WAV

Thanks,
Bruce

On Wed, Jul 14, 2010 at 12:25 PM, Steve Johnson  wrote:

> On Wed, Jul 14, 2010 at 10:04 AM, bruce bruce  wrote:
> > Hi Guys,
> > Running Asterisk v1.4.26 (Elastix flavour) and have Aastra 9480i, 6757i,
> and
> > 6730i, but none of them indicate the voic-email. Where should I look for
> > trouble to find the root issue for MWI?
>
> (1) Check from the CLI> voicemail show users
>
> to ensure that the proper mailboxes have been set up and there is new
> mail in them.  If this is not right, check the voicemail.conf entry
> for this mailbox.
>
> (2) Check the phone device configuration (in sip.conf) to ensure that
> the phone has a mailbox=xxx entry.
>
> for example:
>
> ;entry in sip.conf for extension 115
> [115]
> context=yourcontext
> mailbox=115
> ...
>
> Restart asterisk if you've made changes and re-test.
>
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Re: [asterisk-users] beeping during call

[Tzafrir Cohen]

On Wed, Jul 14, 2010 at 09:27:29AM -0700, Steve Casto wrote:
> Asterisk 1.4.32
> dahdi-2.3.0.1
> Centos 5.5
> Digium TE420
> CAC channel bank (2)
> Cisco RVS4000 router
> Cox 50 Mbps/ 5 Mbps cable modem
> Flowroute provider
> codac G-711
> 90 % CPU idle
> callwaiting=no
> 
> When there are 10-15 or more calls up the farend hears a callwaiting 
> like beep every 3 to 6 sec. the duration of this "beep" is very short 
> and would be no problem if it didn’t happen every few seconds, some 
> callers think they are being recorded. It doesn't mater if the call is 
> incoming or outgoing. Call quality is fine.
> 
> All calls:
> Local User -> channel bank -> TE420 -> Asterisk box -> Cox cable modem 
> -> Flowroute

https://issues.asterisk.org/view.php?id=17529

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Re: [asterisk-users] Where should I look for MWI settings if Aastra phones don't do it?

[Steve Johnson]

On Wed, Jul 14, 2010 at 10:04 AM, bruce bruce  wrote:
> Hi Guys,
> Running Asterisk v1.4.26 (Elastix flavour) and have Aastra 9480i, 6757i, and
> 6730i, but none of them indicate the voic-email. Where should I look for
> trouble to find the root issue for MWI?

(1) Check from the CLI> voicemail show users

to ensure that the proper mailboxes have been set up and there is new
mail in them.  If this is not right, check the voicemail.conf entry
for this mailbox.

(2) Check the phone device configuration (in sip.conf) to ensure that
the phone has a mailbox=xxx entry.

for example:

;entry in sip.conf for extension 115
[115]
context=yourcontext
mailbox=115
...

Restart asterisk if you've made changes and re-test.

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Re: [asterisk-users] Where should I look for MWI settings if Aastra phones don't do it?

[Gareth Blades]

bruce bruce wrote:
> Hi Guys,
> 
> Running Asterisk v1.4.26 (Elastix flavour) and have Aastra 9480i, 6757i, 
> and 6730i, but none of them indicate the voic-email. Where should I look 
> for trouble to find the root issue for MWI?
> 
> Thanks,
> 

For each extension in sip.conf I have :-
mailbox=mail...@context
subscribemwi=no

I also have the following set as a global setting for all phones 
(aastra.cfg on the tftp server)
^sip explicit mwi subscription: 1


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[asterisk-users] beeping during call

[Steve Casto]

Asterisk 1.4.32
dahdi-2.3.0.1
Centos 5.5
Digium TE420
CAC channel bank (2)
Cisco RVS4000 router
Cox 50 Mbps/ 5 Mbps cable modem
Flowroute provider
codac G-711
90 % CPU idle
callwaiting=no

When there are 10-15 or more calls up the farend hears a callwaiting 
like beep every 3 to 6 sec. the duration of this "beep" is very short 
and would be no problem if it didn’t happen every few seconds, some 
callers think they are being recorded. It doesn't mater if the call is 
incoming or outgoing. Call quality is fine.

All calls:
Local User -> channel bank -> TE420 -> Asterisk box -> Cox cable modem 
-> Flowroute

thanks
Steve Casto


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Re: [asterisk-users] Chanspy - Meetme

[Steve Edwards]

> On 07/12/2010 05:36 PM, Xavier wrote:
> 
> I've got a question about chanspy and meetme. I'd like to transfer all 
> the persons involved in a chanspy (the guy spying, the guy that is spied 
> and the guy that is speaking to the spied one -> total: 3) in a 
> conference room. Is there a way to do it quickly without especially 
> knowing each channels ? It's a bit tricky to know and remember each 
> channels, no ?

> On 07/14/2010 04:36 PM, Russell Bryant wrote:
> 
> You may not need to do this at all.  ChanSpy (in Asterisk 1.6.2, at 
> least) has a barge mode that allows the spying channe l to speak to both 
> parties.  There is also the ability to enable DTMF key presses to swap 
> between spy, whisper, and barge modes.

On Wed, 14 Jul 2010, Xavier wrote:

> I totally agree with the barge mode but for future evolution, what about 
> if there is more than 3 people ?

I wonder if you would have more success starting with meetme. I did this 
for an adult chat application many years ago.

Each agent sat in a meetme waiting for a caller to join them. Before the 
caller was active, a "whisper" was played only to the agent so they knew 
the "theme" of the call.

A supervisor could join the meetme without notifying the agent or the 
caller and could join in, kick the caller or kick the agent and take 
control of the call.

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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[asterisk-users] Where should I look for MWI settings if Aastra phones don't do it?

[bruce bruce]

Hi Guys,

Running Asterisk v1.4.26 (Elastix flavour) and have Aastra 9480i, 6757i, and
6730i, but none of them indicate the voic-email. Where should I look for
trouble to find the root issue for MWI?

Thanks,
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Re: [asterisk-users] Chanspy - Meetme

[Xavier]

 I totally agree with the barge mode but for future evolution, what 
about if there is more than 3 people ?


On 07/14/2010 04:36 PM, Russell Bryant wrote:

- Original Message -

On 07/12/2010 05:36 PM, Xavier wrote:
I've got a question about chanspy and meetme.
I'd like to transfer all the persons involved in a chanspy (the guy
spying, the guy that is spied and the guy that is speaking to the
spied one ->  total: 3) in a conference room.
Is there a way to do it quickly without especially knowing each
channels ? It's a bit tricky to know and remember each channels, no ?

You may not need to do this at all.  ChanSpy (in Asterisk 1.6.2, at least) has 
a barge mode that allows the spying channel to speak to both parties.  There is 
also the ability to enable DTMF key presses to swap between spy, whisper, and 
barge modes.

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Re: [asterisk-users] Can't compile DAHDI - wrong kernel source

[bruce bruce]

I am stuck with the same problem but I have used asterisk yum repository and
it worked by itself without me worrying for kernel stuff.

However, I need to install speex codec and now I am stuck as it doesn't get
picked up by the yum asterisk install somehow. I have lib speex and speex
already installed and when doing "yum install asterisk16" I don't see speex
in "core show translation" Is there anything specific I have to do?

Do I have to build from source as well?

-Sorry, didn't mean to hijack the thread.

Thanks,
Bruce

On Wed, Jul 14, 2010 at 5:08 AM, Chandrakant Solanki <
solanki.chandrak...@gmail.com> wrote:

> Hi
>
> If you install rpm from any location it goes to its default location.
>
> You just go for above steps. For kernel you can go for http://kernel.org
>
> --
> Regards,
>
> Chandrakant Solanki
>
>
> On Wed, Jul 14, 2010 at 2:06 PM, liuxin  wrote:
>
>> Hi.
>> The best easy way is:
>> copy kernel-devel-2.6.18-028stab064.7.rpm to /usr/src
>> then run rpm -ivh kernel-devel-2.6.18-028stab064.7.rpm
>>
>> 2010/7/14 Gareth Blades 
>>
>>  Thermal Wetland wrote:
>>> > I have a virtual server with godaddy but can not compile DAHDI as it
>>> > complains that I do not have the correct kernel source.
>>> >
>>> > The package installed is - kernel-devel-2.6.18-164.11.1.el5.i686:
>>> > Package kernel-devel-2.6.18-164.11.1.el5.i686 already installed and
>>> > latest version
>>> > Nothing to do
>>> >
>>> > uname -a returns:
>>> > Linux 
>>> > ip-XXX-XXX-XXX-XXX.ip.secureserver.net
>>> > >
>>> 2.6.18-028stab064.7 #1
>>> > SMP Wed Aug 26 13:11:07 MSD 2009 i686 i686 i386 GNU/Linux
>>> >
>>> > When I try to compile DAHDI it fails with:
>>> > make[2]: Leaving directory
>>> >
>>> `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware'
>>> > You do not appear to have the sources for the 2.6.18-028stab064.7
>>> kernel
>>> > installed.
>>> >
>>> > Is there a way to trick DAHDI to use the installed kernel?
>>> >
>>> > Thanks for the help!
>>> >
>>> > --
>>> > -Thermal
>>> >
>>>
>>> What kernel versions do you have installed?
>>>
>>> If you are currently running an older kernel but installed a newer
>>> kernel and sources but havent rebooted to activate the new one yet then
>>> it may still be trying to locate the source for the older running kernel.
>>>
>>>
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>>
>>
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>
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Re: [asterisk-users] 1.6.2: Using hints on multiple parking lots

[Russell Bryant]

- Original Message -
> How do I specify to which parking lot the hints refer to?
> 
> For exemple, on 1.4 I use this to see whether a call is parked in 800:
> 
> exten => 800,hint,park:8...@parkedcalls
> 
> But on 1.6 I have multiple parking lots working apparently
> sucessfully. How do I build the hint for parkinglot1 and parkingloit2
> so that my phone , which is subscribing to 800, only see parkinglot1
> and NOT parkinglot2?
> 
> I tried the obvious answer
> 
> exten => 800,hint,park:8...@parkinglot1
> 
> but that didnt seem to do anything.

Instead of parkinglot1, use the name of the configured context for that parking 
lot.  For example, if you set up the parking lot in features.conf as:

[parkinglot1]
context = parkedcalls_custom
parkpos=800-850

the hint should be:

exten => 800,hint,park:8...@parkedcalls_custom

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Re: [asterisk-users] Chanspy - Meetme

[Russell Bryant]

- Original Message -
> On 07/12/2010 05:36 PM, Xavier wrote:

> I've got a question about chanspy and meetme.
> I'd like to transfer all the persons involved in a chanspy (the guy
> spying, the guy that is spied and the guy that is speaking to the
> spied one -> total: 3) in a conference room.
> Is there a way to do it quickly without especially knowing each
> channels ? It's a bit tricky to know and remember each channels, no ?

You may not need to do this at all.  ChanSpy (in Asterisk 1.6.2, at least) has 
a barge mode that allows the spying channel to speak to both parties.  There is 
also the ability to enable DTMF key presses to swap between spy, whisper, and 
barge modes.

--
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Digium, Inc.  |  Engineering Manager, Open Source Software
445 Jan Davis Drive NW   -Huntsville, AL 35806  -  USA
jabber: rbry...@digium.com-=-skype: russell-bryant
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Re: [asterisk-users] Recording from g729 to wav means transcoding ?

[Gordon Henderson]

On Wed, 14 Jul 2010, Jonas Kellens wrote:

> On 07/14/2010 03:41 PM, Gordon Henderson wrote:
>> It's the default codec used in DECT phones. I trialled it for a while for
>> some backhaul applications - the users didn't notice anything different
>> and CPU overhead seemed very low, but I've since gone back to alaw. It
>> does save 32Kb/sec per call though.
>>
>> Gordon
>>
> I'm using G711 also and was looking for a codec that takes less bandwidth.
>
> If you say your users did not notice anything in sound quality, then
> this G726-codec looks good.
>
> And you went back to G711 because you did not had to care about the
> bandwidth ?!

It's not that I don't care about bandwidth, rather than in most cases it's 
not a big issue. Most of my clients have at least 600Kb/sec upload speed 
and the number of concurrent calls they make is small. There are 
exceptions though and in those places we have better networking...

Gordon

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Re: [asterisk-users] Chanspy - Meetme

[Zeeshan Zakaria]

I found this requirement very interesting because it is challenging, needs
some serious thinking on how to do it, but it is certainly possible. My idea
would be to record the sip channels which are involved in the spying process
and use a dynamic feature, pressing which would generate a conference and
redirect all of them to this conference room. The hardesst part here is to
figure out the sip channels. All this is not simple to do, and one needs to
be a dialplan expert to do it.

Zeeshan A Zakaria

--
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www.trashinternetexplorer.com

On 2010-07-14 6:11 AM, "Xavier"  wrote:

 No one have, at least, an idea ?



On 07/12/2010 05:36 PM, Xavier wrote:
>
> Hi guys,
>
> I've got a question about chanspy and meetm...

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Re: [asterisk-users] BLF with Realtime

[Zeeshan Zakaria]

On asterisk 1.4 using real-time, subscribecontext field never worked for me
and I have to add the hints in extensions.conf. But once there, they work
just fine.

Zeeshan A Zakaria

--
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On 2010-07-14 9:12 AM, "Ishfaq Malik"  wrote:

On 14/07/10 12:17, Danny Dias wrote:
>
> Hello Asterisk community,
>
> I'm trying to use BLF with As...
Hi

I was working on this myself a few weeks ago so here's a few tips, some you
may already know.

Changes required to the sip.conf
notifyringing should be set to "yes"
both rtcachefriends and rtupdate set to "yes"
limitonpeers must be set to "yes"

Changes to the sip table
new columns
call-limit int (I think this can be call_limit as well which means you would
not always have to put the col name in`` quotes)
subscribecontext varchar(80)

the subscribecontext has to have the same context name in it that the hint
commands are under (the local context for the sip peers...)

I could see no way of putting the hint commands into the extensions table
and BLF working, so they need to go into the extensions.conf but if your
realtime setup is similar to ours this is written by a shell script that is
running on a cron. You can change the shell script to recognise when the
required context is matched, let it write the 'switch => Realtime/@' line
and then get it to write the necessary 'exten => ,hint,SIP/'
lines to the extensions.conf, i.e. something like this

appenddynamic()
{
  mysql -u dbusername -pdbpassword -h dbhostname dbname -se "select
distinct(context) from extensions" | grep -v '^context$' | while read
context; do
echo "[$context]" >> $ASTCONF
echo "switch => Realtime/@" >> $ASTCONF
if [ $context = "target_context" ]
then
echo "exten => 100,hint,SIP/100"
echo "exten => 101,hint,SIP/101"
fi
echo "" >> $ASTCONF
  done
}


I really hope that makes some sense!

Ish
 --
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Software Developer
PackNet Ltd

Office:   0161 660 3062

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Re: [asterisk-users] Recording from g729 to wav means transcoding ?

[Jonas Kellens]

On 07/14/2010 03:41 PM, Gordon Henderson wrote:
> It's the default codec used in DECT phones. I trialled it for a while for
> some backhaul applications - the users didn't notice anything different
> and CPU overhead seemed very low, but I've since gone back to alaw. It
> does save 32Kb/sec per call though.
>
> Gordon
>
I'm using G711 also and was looking for a codec that takes less bandwidth.

If you say your users did not notice anything in sound quality, then 
this G726-codec looks good.

And you went back to G711 because you did not had to care about the 
bandwidth ?!


Jonas.

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Re: [asterisk-users] Unable to open pseudo device

[Tzafrir Cohen]

On Wed, Jul 14, 2010 at 10:45:33AM +0800, Malvin Rito wrote:
> Thanks for the reply. There is no folder dahdi under /dev folder. I cannot
> also find /udev.d on /etc folder.
> 
> Under /dev folder I only see /dev/zap/pseudo.

What version of Asterisk is it?

-- 
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Re: [asterisk-users] Recording from g729 to wav means transcoding ?

[Gordon Henderson]

On Wed, 14 Jul 2010, Jonas Kellens wrote:

> On 07/14/2010 01:39 PM, Gordon Henderson wrote:
>> And it's nice to have a choice of vendors to buy G729 from now too.
>> Doesn't help on weedy hardware though.
>
> I thought you could only buy licenses from Digium ? Can you install
> other G729-licenses on Asterisk ?

Sure.

http://www.howlertech.com/products/howlets/

More flexable licensing than Digium.

Or if you're in a place that doesn't honour software patents, (or don't 
care) then there is a publicly avalable unlicensed one avalable too.

> I need the MixMonitor-application so if I want to record the
> G729-audiostream, I need licenses (the question is how much).

I'd like to think it was no more than one per channel... However there's 
probably 2 transcodes going on concurrently, one for audio in and one for 
audio out, so who knows how it works at that level.

> Isn't G726 also a quality codec ? (http://www.ozvoip.com/voip-codecs/)

It's the default codec used in DECT phones. I trialled it for a while for 
some backhaul applications - the users didn't notice anything different 
and CPU overhead seemed very low, but I've since gone back to alaw. It 
does save 32Kb/sec per call though.

Gordon

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Re: [asterisk-users] BLF with Realtime

[Ishfaq Malik]

On 14/07/10 12:17, Danny Dias wrote:

Hello Asterisk community,

I'm trying to use BLF with Asterisk Realtime, i've been searching for
some info but nothing seems to be clear, can anyone help me eith some
ideas to make this work ok?

I'va my dialplan with Realtime

Thanks in advance

   

Hi

I was working on this myself a few weeks ago so here's a few tips, some 
you may already know.


Changes required to the sip.conf
notifyringing should be set to "yes"
both rtcachefriends and rtupdate set to "yes"
limitonpeers must be set to "yes"

Changes to the sip table
new columns
call-limit int (I think this can be call_limit as well which means you 
would not always have to put the col name in`` quotes)

subscribecontext varchar(80)

the subscribecontext has to have the same context name in it that the 
hint commands are under (the local context for the sip peers...)


I could see no way of putting the hint commands into the extensions 
table and BLF working, so they need to go into the extensions.conf but 
if your realtime setup is similar to ours this is written by a shell 
script that is running on a cron. You can change the shell script to 
recognise when the required context is matched, let it write the 'switch 
=> Realtime/@' line and then get it to write the necessary 'exten => 
,hint,SIP/' lines to the extensions.conf, i.e. something 
like this


appenddynamic()
{
  mysql -u dbusername -pdbpassword -h dbhostname dbname -se "select 
distinct(context) from extensions" | grep -v '^context$' | while read 
context; do

echo "[$context]" >> $ASTCONF
echo "switch => Realtime/@" >> $ASTCONF
if [ $context = "target_context" ]
then
echo "exten => 100,hint,SIP/100"
echo "exten => 101,hint,SIP/101"
fi
echo "" >> $ASTCONF
  done
}


I really hope that makes some sense!

Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062
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Re: [asterisk-users] How to pass through supported 100rel

[Kevin P. Fleming]

On 07/14/2010 05:15 AM, kawanobe tomohito wrote:
> 
> 
> hello
> 
> I want to know how to pass through 100rel header.
> and I hope that asterisk PRACK to UAS.(RFC3262 behavior)

Asterisk is not a proxy; it does not 'pass through' headers, or any
other portion of SIP requests and responses. Asterisk is a B2BUA UA, so
the two SIP dialogs involved in a 'call' are completely separate.

Asterisk does not have any support for 100rel or PRACK.

-- 
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
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Re: [asterisk-users] Recording from g729 to wav means transcoding ?

[Jonas Kellens]

On 07/14/2010 01:39 PM, Gordon Henderson wrote:
> And it's nice to have a choice of vendors to buy G729 from now too.
> Doesn't help on weedy hardware though.
>
> Gordon
>

I thought you could only buy licenses from Digium ? Can you install 
other G729-licenses on Asterisk ?

I need the MixMonitor-application so if I want to record the 
G729-audiostream, I need licenses (the question is how much).

Isn't G726 also a quality codec ? (http://www.ozvoip.com/voip-codecs/)


Jonas.

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Re: [asterisk-users] MAC Address prefixes of Voip equipment

[Benny Amorsen]

Frank Church  writes:

> Is there a database of MAC address prefixes used the common VoIP
> devices. I see the Linksys Sipura devices state with 00:0E.
>
> Does the same apply to other Linksys VoIP equipment?
>
> Is there some way VoIP equipment allow themselves to be identified by
> requesting data from some ports?

With Snom you can actually find the specific phone model from the MAC
address. Unfortunately this information isn't published anywhere.
Perhaps there would be community interest in maintaining a database for
the various vendors?


/Benny


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Re: [asterisk-users] Recording from g729 to wav means transcoding ?

[Gordon Henderson]

On Wed, 14 Jul 2010, Jonas Kellens wrote:

> On 07/14/2010 08:55 AM, Gordon Henderson wrote:
>> On Tue, 13 Jul 2010, Paul Belanger wrote:
>>
>>> On Tue, Jul 13, 2010 at 7:41 AM, Jonas Kellens  
>>> wrote:
>>>
 I have no licenses and I want to avoid transcoding all together.

>>> For terminating a call into Asterisk, you need g729 licenses.  It is
>>> that simple.
>>>
>> The sounds package is avalable in g729 formats and voicemail can write in
>> g729 format too without transcoding from a g729 source. So what else is
>> there if you're not interfacing with PSTN hardware or using meetme?
>>
> And how about Monitor() to record conversations ?? The only format in
> which you can choose to record is wav|wav49|gsm.

Hm. you're right, and I've just checked the code:


 snprintf(tmp,sizeof(tmp) - 1,"%s/%s.%s",urlprefix,fname_base,
 ((strcmp(format,"gsm")) ? "wav" : "gsm"));

That appears to force the filename into .wav or .gsm.

I can understand why MixMonitor might not be able to handle g729 as they 
presumably have to be transcoded to a format that can be mixed then 
transcoded back to g729, but not a simple Monitor which is writing 2 
files...

So pure g729 is possible if you're not using

   MeetMe
   Monitor
   MixMonitor
   PSTN

Although you should be able to use PSTN if it's local to the site and not 
being sent via the Internet - so a local phone which understands both G711 
and G729 ought to work with local PSTN and a VoIP trunk in G729

And it's nice to have a choice of vendors to buy G729 from now too. 
Doesn't help on weedy hardware though.

Gordon

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[asterisk-users] BLF with Realtime

[Danny Dias]

Hello Asterisk community,

I'm trying to use BLF with Asterisk Realtime, i've been searching for
some info but nothing seems to be clear, can anyone help me eith some
ideas to make this work ok?

I'va my dialplan with Realtime

Thanks in advance

-- 
Saludos
Danny Dias
SkypeID: danny.dias1

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[asterisk-users] How to pass through supported 100rel

[kawanobe tomohito]

hello

I want to know how to pass through 100rel header.
and I hope that asterisk PRACK to UAS.(RFC3262 behavior)

 _  
  
_
Hotmailがサクサク動く！ フィッシング詐欺も怖くない
http://o.jp.msn.com/ie8/

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[asterisk-users] Silence RTP

[kawanobe tomohito]

hello

I found silence RTP packet from Asterisk in early dialog.
I want to know reason and how to solve.

RTP packet
 80 00 40 22 00 0c 74 58 06 98 eb 44 ff ff ff ff ..@"..tX...D
0010 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 
0020 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 
0030 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 
0040 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 
0050 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 
0060 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 
0070 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 
0080 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 
0090 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 
00a0 ff ff ff ff ff ff ff ff ff ff ff ff 

_   
  
_


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Re: [asterisk-users] Chanspy - Meetme

[Xavier]

 No one have, at least, an idea ?

On 07/12/2010 05:36 PM, Xavier wrote:

Hi guys,

I've got a question about chanspy and meetme.
I'd like to transfer all the persons involved in a chanspy (the guy 
spying, the guy that is spied and the guy that is speaking to the 
spied one -> total: 3) in a conference room.
Is there a way to do it quickly without especially knowing each 
channels ? It's a bit tricky to know and remember each channels, no ?


Thanks in advance
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Re: [asterisk-users] Asterisk + Hylafax + Iiaxmodem - Outbound number.

[Marta Silva]

Thank you for your response Doug,

Please move the thread as you think appropriate (but please tell me
how/where to join the mailling list (as this is the only one I have
subscribed).

I have 2 physical Fax machines connected to the GXW and people will be
sending faxes old fation from thembut instead of having an analog line
direct out, I have the PBX which does the actual deliveryso, I still
need to have diferent customised numbers when faxing out from each one of
them.

I have configured the (ttyIAX) modems with the specific  valuesI was
hoping I could choose which one to use for each of the sip clients (defined
on the GXW box), and this way, select the outbound numberif not, how can
I do this?

Thank you very much.

Regards,

- Marta
On 13 July 2010 13:01, Doug Lytle  wrote:

> Marta Silva wrote:
> >  Hi there,
> > Thank you for your response. So I can use the ModemSetOriginCmd
> > command to assign the outbound number on the iaxmodem, but how do I
> > choose which modem to use for my specific sip client (GXW-4004), as I
> > have 2 faxes connected to my GXW box?
> >
>
> Why would you need to choose a specific modem?  I believe everything can
> be set from the command line, when generating your fax.  Granted, I
> haven't read the complete thread.
>
> If you have specific HylaFAX+ or iaxmodem questions, it'd probably be
> better to move this thread to either of those lists.
>
> 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] Can't compile DAHDI - wrong kernel source

[Chandrakant Solanki]

Hi

If you install rpm from any location it goes to its default location.

You just go for above steps. For kernel you can go for http://kernel.org

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On Wed, Jul 14, 2010 at 2:06 PM, liuxin  wrote:

> Hi.
> The best easy way is:
> copy kernel-devel-2.6.18-028stab064.7.rpm to /usr/src
> then run rpm -ivh kernel-devel-2.6.18-028stab064.7.rpm
>
> 2010/7/14 Gareth Blades 
>
>  Thermal Wetland wrote:
>> > I have a virtual server with godaddy but can not compile DAHDI as it
>> > complains that I do not have the correct kernel source.
>> >
>> > The package installed is - kernel-devel-2.6.18-164.11.1.el5.i686:
>> > Package kernel-devel-2.6.18-164.11.1.el5.i686 already installed and
>> > latest version
>> > Nothing to do
>> >
>> > uname -a returns:
>> > Linux 
>> > ip-XXX-XXX-XXX-XXX.ip.secureserver.net
>> > >
>> 2.6.18-028stab064.7 #1
>> > SMP Wed Aug 26 13:11:07 MSD 2009 i686 i686 i386 GNU/Linux
>> >
>> > When I try to compile DAHDI it fails with:
>> > make[2]: Leaving directory
>> >
>> `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware'
>> > You do not appear to have the sources for the 2.6.18-028stab064.7 kernel
>> > installed.
>> >
>> > Is there a way to trick DAHDI to use the installed kernel?
>> >
>> > Thanks for the help!
>> >
>> > --
>> > -Thermal
>> >
>>
>> What kernel versions do you have installed?
>>
>> If you are currently running an older kernel but installed a newer
>> kernel and sources but havent rebooted to activate the new one yet then
>> it may still be trying to locate the source for the older running kernel.
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>   http://www.asterisk.org/hello
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] power outage

[liuxin]

Hi,
probably a misconfiguration or you havent plugged the cable in yet.

2010/7/14 C F 

> It has nothing to do with the D-channel, however you will never know
> if the B-channels work if the D-channel is down. D-channel is what
> allows the B-channels to work, and is the first place to troubleshoot.
> If something is screwed up with the power the symptom you'll get is a
> non working PRI, the way to check it is by means of seeing if the
> D-channel synced up or not.
>
>
> On Mon, Jul 12, 2010 at 2:17 AM, Justin Case
>  wrote:
> > What would the power have to do with the D Channel ? Isn't which channel
> > used a logical setting (as opposed to physical). I am not saying your
> wrong
> > I am just trying to understand why it happens.
> >
> > On Mon, Jul 12, 2010 at 7:56 AM, C F  wrote:
> >>
> >> I have found that sometimes shutting down the machine waiting a full
> >> minute while the power cable is unplugged then restarting can fix such
> >> problems if it's power related.
> >>
> >> On Fri, Jul 9, 2010 at 12:42 PM, Jerry Geis 
> wrote:
> >> > I have a TE205P that has been working fine for 2 years.
> >> > power outage yesterday took out my everything for over an hour.
> >> >
> >> > Everything has come back up except the PRI. My provider has checked it
> >> > to the box
> >> > and says everything looks good on their end.
> >> >
> >> > I get this message:
> >> > [Jul  9 12:40:32] WARNING[13709] chan_dahdi.c: No D-channels
> available!
> >> > Using Primary channel 24 as D-channel anyway!
> >> >
> >> > ztcfg -vvv
> >> >
> >> > Zaptel Version: 1.4.12.1
> >> > Echo Canceller: MG2
> >> > Configuration
> >> > ==
> >> >
> >> > SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)
> >> >
> >> > Channel map:
> >> >
> >> > Channel 18: Clear channel (Default) (Slaves: 18)
> >> > Channel 19: Clear channel (Default) (Slaves: 19)
> >> > Channel 20: Clear channel (Default) (Slaves: 20)
> >> > Channel 21: Clear channel (Default) (Slaves: 21)
> >> > Channel 22: Clear channel (Default) (Slaves: 22)
> >> > Channel 23: Clear channel (Default) (Slaves: 23)
> >> > Channel 24: D-channel (Default) (Slaves: 24)
> >> >
> >> > 7 channels to configure.
> >> >
> >> > and show status gives me condition RED of course.
> >> >
> >> > How do I find out whats wrong here?
> >> >
> >> > Jerry
> >> >
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Re: [asterisk-users] Can't compile DAHDI - wrong kernel source

[Chandrakant Solanki]

Hi

Check your kernel version using *uname -r *and then try to download tar.gz
setup for that version.

And extract it into /usr/src/kernels directory , then try to compile.


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On Wed, Jul 14, 2010 at 1:46 PM, Gareth Blades
wrote:

> Thermal Wetland wrote:
> > I have a virtual server with godaddy but can not compile DAHDI as it
> > complains that I do not have the correct kernel source.
> >
> > The package installed is - kernel-devel-2.6.18-164.11.1.el5.i686:
> > Package kernel-devel-2.6.18-164.11.1.el5.i686 already installed and
> > latest version
> > Nothing to do
> >
> > uname -a returns:
> > Linux ip-XXX-XXX-XXX-XXX.ip.secureserver.net
> >  2.6.18-028stab064.7 #1
> > SMP Wed Aug 26 13:11:07 MSD 2009 i686 i686 i386 GNU/Linux
> >
> > When I try to compile DAHDI it fails with:
> > make[2]: Leaving directory
> >
> `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware'
> > You do not appear to have the sources for the 2.6.18-028stab064.7 kernel
> > installed.
> >
> > Is there a way to trick DAHDI to use the installed kernel?
> >
> > Thanks for the help!
> >
> > --
> > -Thermal
> >
>
> What kernel versions do you have installed?
>
> If you are currently running an older kernel but installed a newer
> kernel and sources but havent rebooted to activate the new one yet then
> it may still be trying to locate the source for the older running kernel.
>
>
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Re: [asterisk-users] Can't compile DAHDI - wrong kernel source

[liuxin]

Hi.
The best easy way is:
copy kernel-devel-2.6.18-028stab064.7.rpm to /usr/src
then run rpm -ivh kernel-devel-2.6.18-028stab064.7.rpm

2010/7/14 Gareth Blades 

> Thermal Wetland wrote:
> > I have a virtual server with godaddy but can not compile DAHDI as it
> > complains that I do not have the correct kernel source.
> >
> > The package installed is - kernel-devel-2.6.18-164.11.1.el5.i686:
> > Package kernel-devel-2.6.18-164.11.1.el5.i686 already installed and
> > latest version
> > Nothing to do
> >
> > uname -a returns:
> > Linux 
> > ip-XXX-XXX-XXX-XXX.ip.secureserver.net
> > >
> 2.6.18-028stab064.7 #1
> > SMP Wed Aug 26 13:11:07 MSD 2009 i686 i686 i386 GNU/Linux
> >
> > When I try to compile DAHDI it fails with:
> > make[2]: Leaving directory
> >
> `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware'
> > You do not appear to have the sources for the 2.6.18-028stab064.7 kernel
> > installed.
> >
> > Is there a way to trick DAHDI to use the installed kernel?
> >
> > Thanks for the help!
> >
> > --
> > -Thermal
> >
>
> What kernel versions do you have installed?
>
> If you are currently running an older kernel but installed a newer
> kernel and sources but havent rebooted to activate the new one yet then
> it may still be trying to locate the source for the older running kernel.
>
>
> --
> _
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Re: [asterisk-users] Can't compile DAHDI - wrong kernel source

[Gareth Blades]

Thermal Wetland wrote:
> I have a virtual server with godaddy but can not compile DAHDI as it 
> complains that I do not have the correct kernel source.
> 
> The package installed is - kernel-devel-2.6.18-164.11.1.el5.i686:
> Package kernel-devel-2.6.18-164.11.1.el5.i686 already installed and 
> latest version
> Nothing to do
> 
> uname -a returns:
> Linux ip-XXX-XXX-XXX-XXX.ip.secureserver.net 
>  2.6.18-028stab064.7 #1 
> SMP Wed Aug 26 13:11:07 MSD 2009 i686 i686 i386 GNU/Linux
> 
> When I try to compile DAHDI it fails with:
> make[2]: Leaving directory 
> `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware'
> You do not appear to have the sources for the 2.6.18-028stab064.7 kernel 
> installed.
> 
> Is there a way to trick DAHDI to use the installed kernel?
> 
> Thanks for the help!
> 
> -- 
> -Thermal
> 

What kernel versions do you have installed?

If you are currently running an older kernel but installed a newer 
kernel and sources but havent rebooted to activate the new one yet then 
it may still be trying to locate the source for the older running kernel.


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Re: [asterisk-users] Recording from g729 to wav means transcoding ?

[Jonas Kellens]

On 07/14/2010 08:55 AM, Gordon Henderson wrote:
> On Tue, 13 Jul 2010, Paul Belanger wrote:
>
>> On Tue, Jul 13, 2010 at 7:41 AM, Jonas Kellens  
>> wrote:
>>  
>>> I have no licenses and I want to avoid transcoding all together.
>>>
>> For terminating a call into Asterisk, you need g729 licenses.  It is
>> that simple.
>>  
> The sounds package is avalable in g729 formats and voicemail can write in
> g729 format too without transcoding from a g729 source. So what else is
> there if you're not interfacing with PSTN hardware or using meetme?
>
> Gordon
>

And how about Monitor() to record conversations ?? The only format in 
which you can choose to record is wav|wav49|gsm.

Jonas.

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