Re: [asterisk-users] 100% CPU after upgrade. (Solved)

[Mike Diehl]

I had meant to post a follow up to this, but just... didn't.  Sorry.

Anyway, I had made a silly change to my safe_asterisk script that caused it to 
start asterisk in the background, but also with a console.  This caused 
asterisk to try to write to a non-existent console tty.

Dumb mistake on my part.

Hope this helps someone else.

Mike.


On Thursday, April 06, 2017 10:28:03 AM you wrote:
> On Thu, Apr 6, 2017 at 10:20 AM, Mike Diehl  wrote:
> > I found it!
> >
> > I had customized the safe_asterisk script and managed to slip in a -c on 
the asterisk command line.
> >
> > So, when I ran strace on the running process, I saw a bunch of messages 
indicating an invalid IOCTL on file handle 1, which is always STDOUT.  A 
background process shouldn't be writing to STDOUT, so I knew I had dorked 
something up.
> >
> > I appreciate your time.
> >
> 
> Thanks so much for letting me know.  Would you mind posting this
> resolution publicly so that anybody following it can learn from what
> happened?
> 
> Best wishes,
> Matthew Fredrickson
> 
> > Mike.
> >
> >
> >
> > On Tuesday, April 04, 2017 09:18:26 AM you wrote:
> >> On Mon, Apr 3, 2017 at 4:45 PM, Mike Diehl  
wrote:
> >> > Those are all rational questions, so here we go:
> >> >
> >> > We upgraded from 11.x, though the system was a backup server, so it was 
never
> >> > actually used.
> >> >
> >> > The system is a 2.4Gh quad-core Xenon with 4G of RAM, so it should have 
plenty
> >> > of power for what I'm asking it to do.  The system is configured via RT 
using
> >> > a local Mysql database.
> >> >
> >> > We only use the native SIP channel driver at this time.
> >> >
> >> > I honestly don't see any reason for this server to eat 100% of it's 
cpu, and
> >> > am hesitant to roll it out to production until I understand why it is.
> >>
> >> I don't either.  Is there any Asterisk logging that indicates
> >> something that might be going on?  If you can't see anything, try
> >> increasing the core debug level and core verbose level (core set
> >> verbose 10, core set debug 10) at the Asterisk CLI and see if you get
> >> anything more out of logging to see what's going on.
> >>
> >>
> >
> > --
> > Mike Diehl
> > Diehlnet Communications, LLC.
> > Sales: (800) 254-6105
> > Support: (505) 903-5700
> > Fax: (505) 903-5701
> >
> 
> 
> 
> 

-- 
Mike Diehl



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

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ** in extensions.conf

[Tech Support]

Is ** also defined in features.conf?
Thanks;
John

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Wednesday, April 26, 2017 05:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ** in extensions.conf

On Wed, 26 Apr 2017, Jerry Geis wrote:

> dialplan show testing-sip
>   '**' =>   1. Noop(Testing)  
> [pbx_config]
> 2. Playback(demo-congrats)
> [pbx_config]
> 
> Looks like its there.
> 
> if I do ** "Dial" it works, but if I do "New Call" ** then "Dial" it 
> does not work. Weird. How do I get it to work for both cases. (glad I 
> tried the other)

I never use 'New Call' -- just 'Dial' and 'Redial,'

I suspect you'll need to fiddle with the Polycom dialplan. As soon as I press 
the first '*' my Poly sends the INVITE.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 https://www.linkedin.com/in/steve-edwards-4244281


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

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] SIP and Voice on different nets

[Artem Chekulaev]

​I have connection with two networks (by VoIP provider setup)
1 - 10.10.10.0/24 = SIP
2 - 10.10.11.0/24 = Voice

How to tell Asterisk send / receive voice traffic not on SIP network. When
I look into dumps, I see Asterisk trying to use SIP net for voice

Unfortunately, I _need_ to use two networks instead of one​
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ** in extensions.conf

[Steve Davies]

On Wed, 26 Apr 2017 at 20:29 Jerry Geis  wrote:

> I just tried this in my extensions.conf
>
> exten => **,1,Noop(Testing)
> exten => **,n,Playback(demo-congrats)
>
> Did a reload... and the above does not happen.
> I created as 12 instead of the ** and that works fine.
>
> Is there anyway to get the ** to work?  I also am using a polycom phone if
> that affects things. I'm using asterisk 13.15.0
>
> Thanks
>
> Jerry
>
>
On a Polycom handset, dialling '**' will by default be translated into '+'
before it is dialled. You could:

1) dial *..pause..* which will overcome that AFAIK.
2) Configure call.InternationalDialing.enabled="0" on the handset to stop
it.
3) Put a pattern of _[+],1,... into your dialplan.

That would be by guess anyway :)
Steve
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SIP and Voice on different nets

[Joshua Colp]

On Thu, Apr 27, 2017, at 09:10 AM, Artem Chekulaev wrote:
> ​I have connection with two networks (by VoIP provider setup)
> 1 - 10.10.10.0/24 = SIP
> 2 - 10.10.11.0/24 = Voice
> 
> How to tell Asterisk send / receive voice traffic not on SIP network.
> When
> I look into dumps, I see Asterisk trying to use SIP net for voice
> 
> Unfortunately, I _need_ to use two networks instead of one​

Both the chan_sip and chan_pjsip modules have a "media_address" option
which can be used to specify the address to place in the SDP for media.
In the case of chan_pjsip there is also a "bind_rtp_to_media_address"
option which can be used to guarantee that RTP leaves from that same
address as well.

Cheers,

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SIP and Voice on different nets

[Artem Chekulaev]

Yes, Voice = RTP

Using chan_sip

2017-04-27 15:32 GMT+03:00 Dovid Bender :

> By voice do you mean RTP? Are you using chan_sip or pjsip?
>
>
> On Thu, Apr 27, 2017 at 8:10 AM, Artem Chekulaev 
> wrote:
>
>> ​I have connection with two networks (by VoIP provider setup)
>> 1 - 10.10.10.0/24 = SIP
>> 2 - 10.10.11.0/24 = Voice
>>
>> How to tell Asterisk send / receive voice traffic not on SIP network.
>> When I look into dumps, I see Asterisk trying to use SIP net for voice
>>
>> Unfortunately, I _need_ to use two networks instead of one​
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> 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 --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> 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 --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SIP and Voice on different nets

[Dovid Bender]

By voice do you mean RTP? Are you using chan_sip or pjsip?


On Thu, Apr 27, 2017 at 8:10 AM, Artem Chekulaev  wrote:

> ​I have connection with two networks (by VoIP provider setup)
> 1 - 10.10.10.0/24 = SIP
> 2 - 10.10.11.0/24 = Voice
>
> How to tell Asterisk send / receive voice traffic not on SIP network. When
> I look into dumps, I see Asterisk trying to use SIP net for voice
>
> Unfortunately, I _need_ to use two networks instead of one​
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> 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 --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SIP and Voice on different nets

[Dovid Bender]

Seems I responded the same time as Josh. Follow what he has suggested.


On Thu, Apr 27, 2017 at 8:41 AM, Artem Chekulaev  wrote:

> Yes, Voice = RTP
>
> Using chan_sip
>
> 2017-04-27 15:32 GMT+03:00 Dovid Bender :
>
>> By voice do you mean RTP? Are you using chan_sip or pjsip?
>>
>>
>> On Thu, Apr 27, 2017 at 8:10 AM, Artem Chekulaev 
>> wrote:
>>
>>> ​I have connection with two networks (by VoIP provider setup)
>>> 1 - 10.10.10.0/24 = SIP
>>> 2 - 10.10.11.0/24 = Voice
>>>
>>> How to tell Asterisk send / receive voice traffic not on SIP network.
>>> When I look into dumps, I see Asterisk trying to use SIP net for voice
>>>
>>> Unfortunately, I _need_ to use two networks instead of one​
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> Check out the new Asterisk community forum at:
>>> https://community.asterisk.org/
>>>
>>> New to Asterisk? Start here:
>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> 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 --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> 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 --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> 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 --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] TDM400P takes too long to ring

[Ryan, Travis]

I am getting caller id fine on the phones and console, but not sure about the 
formatting your are talking about. It always just worked for me in the past. Is 
there something I can easily see to know if I’m not setting it right?

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Duffett
Sent: Thursday, April 27, 2017 2:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] TDM400P takes too long to ring

If you are trying to detect caller ID, and it is being supplied by the telco in 
the format you have configured in /etc/chan_dahdi.conf then this should not 
cause a delay. Are you actually seeing the caller ID being displayed on the 
ringing phones?

If, however, the telco is not supplying caller ID info, or it is being supplied 
in a format that you have not set up for, this is likely the cause of the delay 
(looking for caller ID).

All the best,

David

On 27 April 2017 at 12:48, Ryan, Travis 
> wrote:
Hey all,

I have a setup with two analog lines coming into and Asterisk 13 box with a 
TDM400P and it takes a lot of rings before asterisk takes over. I’ve traced 
this same box on two different analog providers so it probably isn’t a problem 
with them.

I DO have callerid enabled and not sure I can turn it off (if they will let 
me). Any other ideas of making it ring through faster?

By the time my internal phones get rang the customer has heard upwards of 7 
rings. Some customers hang up thinking no one will answer.

Also, I have fax detection off in my dialplan.

Thanks,
Travis

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

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users



--
[Image removed by sender. Digium logo]
David Duffett
Digium, Inc. · Director, Worldwide Asterisk Community
6 Landscape Close · Weston on the Green · Bicester · Oxfordshire OX25 3SX · UK
direct/fax: +1 256 428 6119 · mobile: +44 7722 442236
twitter: dduffett · linkedin: 
www.linkedin.com/in/davidduffett
Check us out at: http://digium.com · 
http://asterisk.org
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] TDM400P takes too long to ring

[David Duffett]

If you are trying to detect caller ID, and it is being supplied by the
telco in the format you have configured in /etc/chan_dahdi.conf then this
should not cause a delay. Are you actually seeing the caller ID being
displayed on the ringing phones?

If, however, the telco is not supplying caller ID info, or it is being
supplied in a format that you have not set up for, this is likely the cause
of the delay (looking for caller ID).

All the best,

David

On 27 April 2017 at 12:48, Ryan, Travis  wrote:

> Hey all,
>
>
>
> I have a setup with two analog lines coming into and Asterisk 13 box with
> a TDM400P and it takes a lot of rings before asterisk takes over. I’ve
> traced this same box on two different analog providers so it probably isn’t
> a problem with them.
>
>
>
> I DO have callerid enabled and not sure I can turn it off (if they will
> let me). Any other ideas of making it ring through faster?
>
>
>
> By the time my internal phones get rang the customer has heard upwards of
> 7 rings. Some customers hang up thinking no one will answer.
>
>
>
> Also, I have fax detection off in my dialplan.
>
>
>
> Thanks,
>
> Travis
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
[image: Digium logo]
*David Duffett*
Digium, Inc. · Director, Worldwide Asterisk Community
6 Landscape Close · Weston on the Green · Bicester · Oxfordshire OX25 3SX ·
UK
direct/fax: +1 256 428 6119 · mobile: +44 7722 442236
twitter: dduffett · linkedin: www.linkedin.com/in/davidduffett
Check us out at: http://digium.com · http://asterisk.org

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

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] TDM400P takes too long to ring

[Ryan, Travis]

Hey all,

I have a setup with two analog lines coming into and Asterisk 13 box with a 
TDM400P and it takes a lot of rings before asterisk takes over. I've traced 
this same box on two different analog providers so it probably isn't a problem 
with them.

I DO have callerid enabled and not sure I can turn it off (if they will let 
me). Any other ideas of making it ring through faster?

By the time my internal phones get rang the customer has heard upwards of 7 
rings. Some customers hang up thinking no one will answer.

Also, I have fax detection off in my dialplan.

Thanks,
Travis
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users