Re: [asterisk-users] How to detect fake CallerID? (8xx?)

[Andrew Latham]

On Wed, May 10, 2017 at 10:11 AM, Steve Edwards 
wrote:

> I have a 'time and attendance' application. Think janitorial or security
> kind of thing where an employee goes from location to location.
>
> They're supposed to 'clock in' when they get to a site using a phone at
> that site to prove they're there.
>
> Some employees have discovered 'fake caller ID' services can be used to
> say they're on site when they are not.
>
> How can I detect a fake CallerID? The INVITE looks the same to me.
>
> If I have the employees call an 8xx number, can I ask my SIP provider to
> include more headers to show the real ANI? What would that service be
> called?
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> https://www.linkedin.com/in/steve-edwards-4244281
>
>
For dangerous material sites a call back was used. They call in and get a
code, the system calls back and asks for the code. Convoluted yes, the call
back was all that was really needed to thwart the fraud. A simple RFID pad
setup could be built to use low usage GSM plan to tag in the RFID on site.
But this is beyond the scope of telephony.

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Re: [asterisk-users] Script to Program Snom Phones

[Andrew Latham]

On Thu, Apr 9, 2015 at 12:23 PM, Derek Andrew derek.and...@usask.ca wrote:

 SNOM phones can be configured using files on a TFTP server.

 On Thu, Apr 9, 2015 at 11:14 AM, jg webaccounts...@jgoettgens.de wrote:


   Does anyone know how to program Snom phones using a Mac addresses like
 what is done with the Ciscos. I have about 50 extensions to be programmed
 and I am hoping there is a way on Asterisk to program extensions on the
 snom phones. Please assist.


  What do you mean with 50 extensions? Snom phones allow to define a
 directory, where you can export and import a simple text file. There might
 also be a way to automate this using one of the provisioning methods.

 jg




 --
 Copyright 2015 Derek Andrew (excluding quotations)

 +1 306 966 4808
 University of Saskatchewan
 Peterson 120; 54 Innovation Boulevard
 Saskatoon,Saskatchewan,Canada. S7N 2V3
 Timezone GMT-6

 Typed but not read.



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HTTP is the prefered method of provisioning. You can see
http://wiki.snom.com/Settings/setting_server and even the dynamic tools
baked into Asterisk at
https://wiki.asterisk.org/wiki/display/AST/Phone+Provisioning+in+Asterisk


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Re: [asterisk-users] recommendations for RJ-11 surge supressors?

[Andrew Latham]

On Thu, Jun 27, 2013 at 10:34 AM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
 On Thursday 27 June 2013, Eric Cooper wrote:
 I'd like to protect my expensive Digium FXO cards from spikes on my
 three incoming PSTN lines.  Does anyone have any recommendations?

 Does your telco not fit surge suppressors to the NTE as a matter of standard
 practice?  Perhaps we are spoiled in the UK .

 --
 AJS

 Answers come *after* questions.

APC sells a modular solution that has rack mount or wall mount
options. ProtectNet is the product line.
http://www.apc.com/products/family/index.cfm?id=145


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Re: [asterisk-users] Optimizing Asterisk Environment

[Andrew Latham]

On Sat, Mar 23, 2013 at 12:06 PM, Joshua Colp jc...@digium.com wrote:
 Nick Khamis wrote:

 Oh no secret. Some things I do is increase the ulimit size. I was
 wondering if there was a way to increase allocated memory. I have been
 reading about a -p option but when I start asterisk using asterisk -p
 -10 it does not accept it but asterisk -p 10 works fine. Not sure
 if that was the intended new value.

 Also, I  just want to mention I am not trying to break any records.
 Just would like to get a ~200 concurrent call stable environment using
 G729 out of our setup.


 Are you transcoding? If so then that is where most of your CPU is going, and
 the only option to make it go further is to use a hardware transcoding
 solution.

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

+1 on hardware card.  There are various other tools, even a network
based encoding solution.  Offloading to hardware can show you how
stable/strong your system might already be.

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Re: [asterisk-users] Optimizing Asterisk Environment

[Andrew Latham]

On Sat, Mar 23, 2013 at 3:21 PM, Nick Khamis sym...@gmail.com wrote:
 Hello Gentlemen,

 Thank you so much for your responses. We have been working on a
 SIP/RTP Proxy+Asterisk in backed by MySQL for a few weeks. Everything
 is working nicely I am pleased to say. And will be making some
 donations for G729 licenses etc.. (it's the least we can do to support
 the cause).

 Speaking about transcoding cards. We are functioning 100% on SIP using
 u/alaw and eventually G729. Some typical observations being great
 performance when not using G729 :)...
 Is there any transcoding happening when using only G729 and no other
 codec? We tried disallow=all and allow=g729 and judging by the CPU
 load 260% there seems to be...

 I hope this is not a silly question, but if we force the DID reseller
 to send only G729 encoded media, our asterisk server only allows G729,
 and finally for termination most sip trunk providers have g729 in
 there list of supported codecs, would there still be transcoding
 happening on our * box? I hope this is not as silly question as I
 think

 To answer your question, we also tried with only ulaw and alaw and we
 seem to be stuck on exactly 101 peak. Is there a limit setting
 hidden in one of the *.conf files?

 We let sipp run for almost 3 hours on our box, from another local
 computer using the following command:

 extensions.conf

 exten = 1002,1,Answer
 exten = 1002,n,Goto(demo,s,1)
 exten = 1002,n,Hangup

 ./sipp -sn uac -d 1 -s 1002 test.example.com -l 200 -mp 5606:


 And we got the following results: http://pastebin.com/J0YCprCb

 At 9.4 cps 96963 calls were executed with 0 failed calls. Where is the
 concurrent call figure in this tool? Please forgive me still getting
 use to it :).

 In regards to hardware transcoding cards for SIP protocol. Please let
 us know of some digium solutions. Again, we would love to support the
 cause.

 Nick.

 On 3/23/13, Andrew Latham lath...@gmail.com wrote:
 On Sat, Mar 23, 2013 at 12:06 PM, Joshua Colp jc...@digium.com wrote:
 Nick Khamis wrote:

 Oh no secret. Some things I do is increase the ulimit size. I was
 wondering if there was a way to increase allocated memory. I have been
 reading about a -p option but when I start asterisk using asterisk -p
 -10 it does not accept it but asterisk -p 10 works fine. Not sure
 if that was the intended new value.

 Also, I  just want to mention I am not trying to break any records.
 Just would like to get a ~200 concurrent call stable environment using
 G729 out of our setup.


 Are you transcoding? If so then that is where most of your CPU is going,
 and
 the only option to make it go further is to use a hardware transcoding
 solution.

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

 +1 on hardware card.  There are various other tools, even a network
 based encoding solution.  Offloading to hardware can show you how
 stable/strong your system might already be.

 --
 ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~

Are you recording calls?  If so that is a transcode if you are using
WAV or other.

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Re: [asterisk-users] Asterisk authentication on LDAP (SSHA and SHA passwords)

[Andrew Latham]

On Sun, Mar 10, 2013 at 11:37 AM, Paulo Victor Fernandes da Silva
paulovictorsi...@gmail.com wrote:
 hello guys,

   I'm working on a federal university at Brasil, we already have an openLdap
 with all users and this base is used to authenticate several services like
 email, vpn, wireless (RADIUS), and we have also Shibboleth providing SSO.

  During my studies of Asterisk, i see a lot of people talking about the
 incapacity of asterisk (more precisely because of SIP) to authenticate
 against a ldap that uses password encrypted for anything other than MD5.

  I like to know if exist any how to use Asterisk + Ldap (using SSHA and SHA
 passwords). It can be achieved in some how?

 PS: Sorry for my bad english.

 Best Regards,
 Paulo V.

Paulo

I was looking at that code a month or so ago.  It should be possible
to update res_config_ldap.c to use SHA instead of MD5 when talking to
the OpenLDAP server.  It is also possible, and a good idea. to
maintain a separate password/secret object(MD5/SHA) for Asterisk/PBX
to mitigate any toll fraud.  Keep in mind that the password could be
deployed over HTTPS configuration and be a combination of account info
(typically MAC address of UA).  Mass deployment is key in such an
infrastructure.  Also take the time to catalog the user
devices/software devices that support SHA for direct LDAP directory
look up.

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Re: [asterisk-users] Conf Bridge

[Andrew Latham]

On Thu, Jan 17, 2013 at 3:02 PM, Bryant Zimmerman brya...@zktech.com wrote:
 Hey all.

 RE: Conf Bridge.

 I am looking into a project that would need 8 to 10 thousand parties in a
 single conference.
 Most would be on mute but 5 to 6 would be presenters.

 Is the new conf bridge solid enough to handle this kind of load?
 Any ideas on hardware projections?

 If not 8 to 10 thousand how many would be realistic?

 If not asterisk any other suggestions.

 Thanks for any input.

 zktech

If most are on mute, then have them call into a stream of the actual conference.

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Re: [asterisk-users] Need Help

[Andrew Latham]

On Thu, Jan 17, 2013 at 3:05 PM, Joe Ruffolo j...@mrkgroup.com wrote:

 Hi all! In need of some serious help. We currently run Trixbox and Cent Os
 on a 2u server for our small business’s phones system.



 We are using some Polycom Soundpoint IP phones. The whole thing came
 crashing down over the Holidays and as of right now that’s about



 all we have working right now are the phones. The reason I joined this
 list is because I was hoping to get our external paging  intercom system
 back up and running



 (it runs off of a sound card but cant get it all configured correctly) and
 to be honest I have no clue where to start. I’ve tried reading some online
 guides but nothing.







 Joe Ruffolo

 Director of Operations

 801 N State St Unit C

 Elgin, Il. 60123

 847-468-1700v

 847-468-0717f

 j...@mrkgroup.com

 www.mrkgroupltd.com

Trixbox forums are here http://fonality.com/trixbox/ as this list will
not have the help you might need.


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Re: [asterisk-users] Mail list settings?

[Andrew Latham]

On Thu, Jan 17, 2013 at 6:32 PM, Bryant Zimmerman brya...@zktech.com wrote:
 Hey all

 For some reason the mailing list is sending all messages from the sending
 party.
 This makes it less than ideal when responding; as selecting reply goes to
 the person and not the list.
 Can we have it set back to the old way please?

 Thanks Andrew for pointing this out to me.

 Bryant

I just checked back over the list emails and Bryant's email appears to
be unique in this problem.  I assume it is a simple issue somewhere.
List admins?


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Re: [asterisk-users] Asterisk Support from Digium

[Andrew Latham]

On Sat, Nov 3, 2012 at 2:16 PM, Danny Dias ing.diasda...@gmail.com wrote:
 Hello,

 I wonder if Digium provides support for Asterisk OpenSource versions as an
 anual fee or something?

 For example, if i download Asterisk 1.8.X (Certified or not...) can i buy
 support from Digium to maintain and help on possible future problems in my
 configuration?

 Thanks

Yes

Please review 
http://www.digium.com/en/supportcenter/custom-communications-solutions/
for more information.


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Re: [asterisk-users] Asterisk error message so uncommon, not even Google knows abuot it

[Andrew Latham]

On Fri, Oct 19, 2012 at 11:28 AM, Eric Wieling ewiel...@nyigc.com wrote:
 I'm setting up a test server with a Digium TE122 and am getting the following 
 error on the console, spewing as fast as it can.  Does anyone have any idea 
 what this error might be?

 [Oct 19 11:24:53] NOTICE[2076]: chan_dahdi.c:3108 my_handle_dchan_exception: 
 PRI got event: Event 59 (59) on D-channel of span 2


You have two D channels, why?  Some more info would help, like configs
and where the PRIs are coming from.

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Re: [asterisk-users] asterisk installation under a single directory

[Andrew Latham]

On Mon, Oct 15, 2012 at 3:07 AM, sudeep melekar
sudeep.meleka...@gmail.com wrote:
 hello,
 i want to install asterisk 1.8 in a single directory myasterisksetup
 i.e after asterisk installs it put some of it's installation files in
 different directories
 e.g /var/log/asterisk
 /var/run/asterisk
 and many more

 i want all this installation files to be under my  directory
 myasterisksetup

 can any one provide me step by step installation of asterisk in a single
 directory
 i m completely new to asterisk
 so any help would be appreciated
 --

 regards

 Sudeep S M

Give Live Asterisk a try.  Its in the contrib/scripts directory.

http://svn.asterisk.org/svn/asterisk/trunk/contrib/scripts/live_ast


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Re: [asterisk-users] Parameterize asterisk config files

[Andrew Latham]

On Tue, Oct 2, 2012 at 8:04 PM, Steve Edwards asterisk@sedwards.com wrote:
 On Tue, 2 Oct 2012, Mitch Claborn wrote:

 I'd like to be able to use the same config files in CVS and have the
 differences resolved at run time, based on host name of the asterisk server.

 Another idea would be to write a simple perl or other program to
 pre-process the files and put some markers in the files themselves.


 I don't use CVS, old bad habits die hard :)

 I use a preprocessor. Specifically:

 http://git.dyne.org/freej/plain/lib/javascript/config/preprocessor.pl

 (Not where I got it from, but it's the same file.)

 because it had enough features and because my production hosts already have
 Perl so I didn't have to add yet another scripting language.

 This preprocessor allows you to do '#if HOSTNAME==v0' where HOSTNAME is a
 shell environment variable or it can be defined on the command line.

 You can also define variables in an 'include' file

 It will do a whole lot more, but 'if' and substitution were the only
 features I needed at the time. If I could have deciphered 'm4' I might have
 used that, but the sendmail.mc files look too damn ugly to maintain.

 I hacked in CURDATE and CURTIME as 'pre-defined variables'.

 Here's a sample of one of my files:

 #
 #   Filename:   /source/src/${PROJECT}/manager.conf.pre
 #
 #   Version:001
 #
 #   Edit date:  2008-12-02
 #
 #   Facility:   Asterisk
 #
 #   Abstract:   Define connections to the manager interface.
 #
 #   Environment:Asterisk
 #
 #   Author: Steven L. Edwards
 #
 #   Modified by
 #
 #   000 2008-10-17  SLE Started documenting.
 #   001 2008-12-02  SLE Preprocessorize.
 #expand ; Created by makefile on __CURDATE__ at __CURTIME__
 #expand ; from __FILE__

 [general]
 enabled = yes
 port= 5038
 #if HOSTNAME==v0
 bindaddr= 127.0.0.1
 #else
 bindaddr= 0.0.0.0
 #endif

 [@AMI_USERNAME@]
 deny= 0.0.0.0/0.0.0.0
 #if HOSTNAME==v0
 permit  = 127.0.0.1/255.255.255.255
 #else
 permit  = 192.168.0.0/255.255.255.0
 #endif
 read= all
 secret  = @AMI_SECRET@
 write   = all

 ; (end of /etc/asterisk/manager.conf)
 # (end of /source/src/${PROJECT}/manager.conf.pre)

 This gets munged by my makefile so deployment consists of 'make rsync;
 make config'

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


While we are at it, GIT, Python Fabric and sed balance out most of my
deployment needs.  There are other moving parts but those are my own
design...


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Re: [asterisk-users] 'Training mode'

[Andrew Latham]

On Fri, Sep 28, 2012 at 5:27 PM, Adam Moffett adamli...@plexicomm.net wrote:
 I was asked today if we could somehow have a trainee on the phone with a
 supervisor conferenced in, but somehow have it so anything the supervisor
 says is only heard by the trainee and not the customer.

 Is there a feature like that?


 --

Yup, pretty standard stuff

https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_ChanSpy


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Re: [asterisk-users] How do you convert your prompts to an asterisk-friendly format?

[Andrew Latham]

On Tue, Aug 28, 2012 at 10:39 AM, Johan Wilfer li...@jttech.se wrote:
 Hi,

 I've used the shells-script at the end of this email to generate 8khz mono
 wave-files for asterisk from a 144 khz recording.

 The script does two things: resample  normalize the audio volume.

 Anyone like to share their recommendations / scripts for doing this
 conversion? I've just converted to 8khz wave, should I convert to something
 else?

 For the googler in the future this is my current script (which I hope to
 improve):

 BASEDIR=`dirname $0`
 PROMPTDIRS=dir1 dir2
 for dir in ${PROMPTDIRS}
 do
   src=${BASEDIR}/recordings/prompts/${dir}
   dst=${BASEDIR}/generated/prompts/8khz/${dir}
   for i in ${src}/*.wav; do sox $i  -V -r 8000 -c 1 -q -s \
 ${dst}/$(basename $i .wav).wav vol 0.8; done

   normalize-audio -a -20dBFS ${dst}/*
 done


 --
 Johan Wilfer

 JT Technologies  Telecommunications AB
 Jabber: jo...@jttech.se | Phone: +46 31 3809100


Try this to test with
http://www.digium.com/en/products/ivr/audio-converter.php and compare
your output first...

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Re: [asterisk-users] How do you convert your prompts to an asterisk-friendly format?

[Andrew Latham]

On Tue, Aug 28, 2012 at 11:00 AM, Johan Wilfer li...@jttech.se wrote:
 2012-08-28 16:44, Andrew Latham skrev:

 On Tue, Aug 28, 2012 at 10:39 AM, Johan Wilfer li...@jttech.se wrote:

 Hi,

 I've used the shells-script at the end of this email to generate 8khz
 mono
 wave-files for asterisk from a 144 khz recording.


 Try this to test with
 http://www.digium.com/en/products/ivr/audio-converter.php and compare
 your output first...


 Interesting. Didn't know about this. It's good for testing, but I would like
 to automate it. Is the source-code open or available?



 --
 Johan Wilfer

 JT Technologies  Telecommunications AB
 Jabber: jo...@jttech.se | Phone: +46 31 3809100


Yep, check out repotools for that
http://svn.asterisk.org/svn/repotools/sound_tools/scripts/


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Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?

[Andrew Latham]

On Mon, Aug 20, 2012 at 10:52 AM, Joshua Colp jc...@digium.com wrote:
 - Original Message -
 Joshua

 Can you copy and past into a wiki page for everyone's benefit?  Maybe
 https://wiki.asterisk.org/wiki/display/~jcolp/WebRTC_Demo_Setup or
 like page would be good.

 If this thread has taught me anything it's that there needs to be a complete 
 wiki page, just copying/pasting what I'm saying here isn't enough. It's on my 
 list. I won't call it a demo setup though... since it won't actually work yet.

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

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Agreed, but we need something and a place for comments.  The wiki is
great because we can rename and move things when they are no longer
relevant to our needs.

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Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?

[Andrew Latham]

On Mon, Aug 20, 2012 at 2:58 PM, Juan Castro jcas...@instant.com.br wrote:
 Hoo-hah. It registers. Progress!

 Now... media. Or not.

 On Mon, Aug 20, 2012 at 12:51 PM, Joshua Colp jc...@digium.com wrote:
 - Original Message -
 
  The complete URL to use is http://asterisk IP address or
  host:8088/ws
 
  Note the /ws at the end. WebSocket support is only available there.
  Doing otherwise would have required core HTTP server changes,
  which I wanted to avoid. Depending on what you are testing with
  you may need to change it slightly to add that in.

 Well, I did the following changes in sipml5 and now I get a Bad
 Request on REGISTER, instead of 404. Clearly, I'm still missing
 something. Here are the changes I made:

 You are probably getting hit by a bug in Asterisk 11 that has been fixed.

 It's noted here in the wiki page I'm working on: 
 https://wiki.asterisk.org/wiki/display/~jcolp/Asterisk+WebRTC+Support along 
 with a work around via configuration.

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

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 --
 Juan Carlos Castro y Castro
 Instant Solutions - Telefonia Gerando Resultado
 http://www.instant.com.br
 Principais capitais: 4063-6100
 Demais regiões: (11)4063-6100

 --

Juan

Matt just opened
https://issues.asterisk.org/jira/browse/ASTERISK-20267 to document
some of this.  Feel free to pipe in.

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Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?

[Andrew Latham]

On Fri, Aug 17, 2012 at 2:45 PM, Juan Castro jcas...@instant.com.br wrote:
 I see no indication of how to do this in sip.conf, and when I start
 Asterisk, it doesn't wait on port 80.

 Greetings,

 --
 Juan Carlos Castro y Castro
 Instant Solutions - Telefonia Gerando Resultado
 http://www.instant.com.br
 Principais capitais: 4063-6100
 Demais regiões: (11)4063-6100

 --

Websocket support is being actively worked on.  HTTP support should be
enabled in manager.conf and http.conf first.

--- manager.conf ---
[general]
enabled = yes
webenabled = yes

--- http.conf ---
[general]
enabled=yes
bindaddr=0.0.0.0
bindport=8088

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Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?

[Andrew Latham]

On Fri, Aug 17, 2012 at 4:01 PM, Joshua Colp jc...@digium.com wrote:
 - Original Message -
 On Fri, Aug 17, 2012 at 2:45 PM, Juan Castro jcas...@instant.com.br
 wrote:
  I see no indication of how to do this in sip.conf, and when I start
  Asterisk, it doesn't wait on port 80.
 

 Websocket support is being actively worked on.  HTTP support should
 be
 enabled in manager.conf and http.conf first.

 Hola!

 The above will get the HTTP server portion going, but here's some other items:

 1. transport=ws must be added to the peer/friend/user in sip.conf
 2. avpf=yes must be set for that peer/friend/user as well.

 Depending on what you are testing with this can get you a little further.

 If you are using Chrome things will not quite work, yet. While they have made 
 considerable progress becoming compliant with the ICE specification (SDP is 
 now almost proper) it seems as though their STUN implementation is still not 
 there yet. Completely valid packets sent by the library we use just seem to 
 be ignored.

 Patience is a virtue really as things are still evolving.

 As well I will be working on a wiki page that will describe this stuff in 
 detail. I was holding off until things were a bit more there but as people 
 are at least trying it shall appear soon.

 Cheers,

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

 --
 ___

Joshua

Can you copy and past into a wiki page for everyone's benefit?  Maybe
https://wiki.asterisk.org/wiki/display/~jcolp/WebRTC_Demo_Setup or
like page would be good.

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Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?

[Andrew Latham]

On Fri, Aug 17, 2012 at 4:07 PM, Andrew Latham lath...@gmail.com wrote:
 On Fri, Aug 17, 2012 at 4:01 PM, Joshua Colp jc...@digium.com wrote:
 - Original Message -
 On Fri, Aug 17, 2012 at 2:45 PM, Juan Castro jcas...@instant.com.br
 wrote:
  I see no indication of how to do this in sip.conf, and when I start
  Asterisk, it doesn't wait on port 80.
 

 Websocket support is being actively worked on.  HTTP support should
 be
 enabled in manager.conf and http.conf first.

 Hola!

 The above will get the HTTP server portion going, but here's some other 
 items:

 1. transport=ws must be added to the peer/friend/user in sip.conf
 2. avpf=yes must be set for that peer/friend/user as well.

 Depending on what you are testing with this can get you a little further.

 If you are using Chrome things will not quite work, yet. While they have 
 made considerable progress becoming compliant with the ICE specification 
 (SDP is now almost proper) it seems as though their STUN implementation is 
 still not there yet. Completely valid packets sent by the library we use 
 just seem to be ignored.

 Patience is a virtue really as things are still evolving.

 As well I will be working on a wiki page that will describe this stuff in 
 detail. I was holding off until things were a bit more there but as people 
 are at least trying it shall appear soon.

 Cheers,

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

 --
 ___

 Joshua

 Can you copy and past into a wiki page for everyone's benefit?  Maybe
 https://wiki.asterisk.org/wiki/display/~jcolp/WebRTC_Demo_Setup or
 like page would be good.

 --
 ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~

s/past/paste/

oops

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Re: [asterisk-users] Websockets on Asterisk 11 and SipML5

[Andrew Latham]

On Tue, Aug 14, 2012 at 1:20 PM, James Mortensen
james.morten...@a-cti.com wrote:
 mailsvb mailsvb at gmail.com writes:



 Hi,

 I was facing the very same issue and created a ticket...


 https://issues.asterisk.org/jira/browse/ASTERISK-20221

 best regards,
 Sven2012/8/13 James Mortensen james.mortensen at a-cti.com
 Andrew Latham lathama at gmail.com writes:
 
  On Mon, Aug 13, 2012 at 2:58 PM, James Mortensen

  james.mortensen at a-cti.com wrote:
   Hello,
  
   I'm trying to register a user using sipml5 on Asterisk 11. I followed the
   instructions here:
   http://thr3ads.net/asterisk-users/2012/08/1972342-Asterisk-Websockets
  
   I added transport=ws to my sip.conf file:
  
   [3002]
   username=3002
   secret=X
   host=dynamic
   type=friend
   context=test
   disallow=all
   allow=g729
   ;allow=all ; Allow codecs in order of preference
   allow=ilbc
   allow=silk8
   allow=gsm
   transport=ws
  
  
   I also modified the sipml5 library so that the URL looks like this:
   ws://example.org:8088/ws with the /ws at the end, as instructed.
  
   Now, where I get confused is here:
  
   You will need to change sipml5 to use http://hostname or IP address of
  
   Asterisk:8088/ws as the URL. WebSocket is only available on the /ws
 path.
  
  
   Did Joshua mean to say ws:// instead of http://?  Because I'm not aware 
   of
   WebSockets working with http protocols, only ws protocols. Is there
   something I'm missing here?
  
  
  
   The error that I'm getting in the sipml5 client is:  Disconnected: 
   Failed
   to connet to the server  And that typo is not mine.
  
  
  
  
   On the server, here is what I see from a tcpdump. The port appears to be
   open, but I'm not convinced that Asterisk is actually listening for
   WebSocket traffic:
  
  
  
  
   tcpdump -v port 8088
  
  
  
  
   18:57:03.051712 IP (tos 0x0, ttl 243, id 21320, offset 0, flags [DF],
 proto
   TCP (6), length 60)
   static-50-43-101-83.bvtn.or.frontiernet.net.63036 
   ip-10-168-151-65.us-west-1.compute.internal.omniorb: Flags [S], cksum
 0x4f7a
   (correct), seq 4055598050, win 14600, options [mss
   1380,sackOK,nop,nop,nop,nop,nop,nop,nop,nop,nop,nop,nop,nop,nop,nop],
 length
   0
   18:57:03.051758 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto 
   TCP
   (6), length 40)
   ip-10-168-151-65.us-west-1.compute.internal.omniorb 
   static-50-43-101-83.bvtn.or.frontiernet.net.63036: Flags [R.], cksum
 0xeaf4
   (correct), seq 0, ack 4055598051, win 0, length 0
  
  
  
   Is there something else I'm missing?  Please let me know what additional
   information you need from me.
  
   Thank you!
  
   --
   James Mortensen
  
 
  Look to see if the /ws is showing in an http show status
 
  '''
  *CLI http show status
  HTTP Server Status:
  Prefix:
  Server Enabled and Bound to 0.0.0.0:8088
 
  Enabled URI's:
  /httpstatus = Asterisk HTTP General Status
  /phoneprov/... = Asterisk HTTP Phone Provisioning Tool
  /amanager = HTML Manager Event Interface w/Digest authentication
  /uploads = HTTP POST mapping
  /arawman = Raw HTTP Manager Event Interface w/Digest authentication
  /manager = HTML Manager Event Interface
  /rawman = Raw HTTP Manager Event Interface
  /static/... = Asterisk HTTP Static Delivery
  /amxml = XML Manager Event Interface w/Digest authentication
  /mxml = XML Manager Event Interface
  /ws = Asterisk HTTP WebSocket
 
  Enabled Redirects:
/ = /static/admin.html
  *CLI
  '''
 
 Hi Andrew,
 I uncommented enabled=yes in http.conf and now see the /ws = Asterisk HTTP
 WebSocket. I also modified bindaddr=0.0.0.0 as it was previously 127.0.0.1.  
 I
 can connect and I do see the following output in my Chrome NET tab:
 Request URL:ws://example.org:8088/ws
 Request Method:GET
 Status Code:101 Switching Protocols
 Request Headersview source
 Connection:Upgrade
 Host:example.org:8088
 Origin:http://local:
 Sec-WebSocket-Extensions:x-webkit-deflate-frame
 Sec-WebSocket-Key:fazgtURy132RAFXGRiT9TA==
 Sec-WebSocket-Protocol:sip
 Sec-WebSocket-Version:13
 Upgrade:websocket
 (Key3):00:00:00:00:00:00:00:00
 Response Headersview source
 Connection:Upgrade
 Sec-WebSocket-Accept:fQA1LFnbYFSxFYAr7Ls1Keh54KY=
 Sec-WebSocket-Protocol:sip
 Upgrade:websocket
 (Challenge Response):00:00:00:00:00:00:00:00:00:00:00:00:00:00:00:00
 However, the Asterisk server dies afterwards and must be restarted. The
 /var/log/messages file has no helpful information; I was tailing it as I made
 one of my connect attempts.
 If it helps, I have a local Asterisk 11 setup in verbose mode, and I did see
 the
 following warning message when trying to connect to it instead:
 *CLI [Aug 13 13:17:39] WARNING[567]: res_http_websocket.c:533
 websocket_callback: WebSocket connection from '127.0.0.1:53845' could not be
 accepted - no protocols out of 'sip' supported
 Also, here is what I see in the Chrome NET tab:  (I hope this doesn't confuse
 the problem. Keep in mind that these are 2 separate Asterisk 11

Re: [asterisk-users] Websockets on Asterisk 11 and SipML5

[Andrew Latham]

On Mon, Aug 13, 2012 at 2:58 PM, James Mortensen
james.morten...@a-cti.com wrote:
 Hello,

 I'm trying to register a user using sipml5 on Asterisk 11. I followed the
 instructions here:
 http://thr3ads.net/asterisk-users/2012/08/1972342-Asterisk-Websockets

 I added transport=ws to my sip.conf file:

 [3002]
 username=3002
 secret=X
 host=dynamic
 type=friend
 context=test
 disallow=all
 allow=g729
 ;allow=all ; Allow codecs in order of preference
 allow=ilbc
 allow=silk8
 allow=gsm
 transport=ws


 I also modified the sipml5 library so that the URL looks like this:
 ws://example.org:8088/ws with the /ws at the end, as instructed.

 Now, where I get confused is here:

 You will need to change sipml5 to use http://hostname or IP address of

 Asterisk:8088/ws as the URL. WebSocket is only available on the /ws path.


 Did Joshua mean to say ws:// instead of http://?  Because I'm not aware of
 WebSockets working with http protocols, only ws protocols. Is there
 something I'm missing here?



 The error that I'm getting in the sipml5 client is:  Disconnected: Failed
 to connet to the server  And that typo is not mine.




 On the server, here is what I see from a tcpdump. The port appears to be
 open, but I'm not convinced that Asterisk is actually listening for
 WebSocket traffic:




 tcpdump -v port 8088




 18:57:03.051712 IP (tos 0x0, ttl 243, id 21320, offset 0, flags [DF], proto
 TCP (6), length 60)
 static-50-43-101-83.bvtn.or.frontiernet.net.63036 
 ip-10-168-151-65.us-west-1.compute.internal.omniorb: Flags [S], cksum 0x4f7a
 (correct), seq 4055598050, win 14600, options [mss
 1380,sackOK,nop,nop,nop,nop,nop,nop,nop,nop,nop,nop,nop,nop,nop,nop], length
 0
 18:57:03.051758 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto TCP
 (6), length 40)
 ip-10-168-151-65.us-west-1.compute.internal.omniorb 
 static-50-43-101-83.bvtn.or.frontiernet.net.63036: Flags [R.], cksum 0xeaf4
 (correct), seq 0, ack 4055598051, win 0, length 0



 Is there something else I'm missing?  Please let me know what additional
 information you need from me.

 Thank you!

 --
 James Mortensen


Look to see if the /ws is showing in an http show status

'''
*CLI http show status
HTTP Server Status:
Prefix:
Server Enabled and Bound to 0.0.0.0:8088

Enabled URI's:
/httpstatus = Asterisk HTTP General Status
/phoneprov/... = Asterisk HTTP Phone Provisioning Tool
/amanager = HTML Manager Event Interface w/Digest authentication
/uploads = HTTP POST mapping
/arawman = Raw HTTP Manager Event Interface w/Digest authentication
/manager = HTML Manager Event Interface
/rawman = Raw HTTP Manager Event Interface
/static/... = Asterisk HTTP Static Delivery
/amxml = XML Manager Event Interface w/Digest authentication
/mxml = XML Manager Event Interface
/ws = Asterisk HTTP WebSocket

Enabled Redirects:
  / = /static/admin.html
*CLI
'''

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Re: [asterisk-users] TDM400P: Lifetime Replacement

[Andrew Latham]

On Sun, May 6, 2012 at 12:42 PM, Greg Woods g...@gregandeva.net wrote:
 I have a Digium TDM400P card that appears to have died. The first noted
 symptoms were that dahdi would fail to reload on boot. On closer
 inspection, the card looks totally dead; no lights on at all. I have
 tried moving it to a different PCI slot, and removing the other PCI card
 (a 3com 10/100 NIC) completely.  I have not tried removing the PCI-E
 graphics card, of course, because I can't boot the system without it,
 but that is unlikely to be fruitful anyway.

 So the questions are: first, what is the expected lifetime of one of
 these cards? It just passed its 5th birthday. Is that as long as it
 could be expected to last?

 Second, since the parts of this card are very expensive, I am wondering
 if these symptoms likely mean that the main board of the card is dead,
 but the FXS and FXO modules might still be good. In that case, I could
 just get a new main card and move the modules to the new main card. The
 problem is that I can't find any TDM400P cards anywhere, all I can find
 are TDM410P's. Will the modules I have (assuming they are still good)
 work with a TDM410P?

 Last question: the TDM410 card is available in PCI and PCIx1 forms. I do
 have a free PCIx1 slot. Is there any advantage in one over the other?

 Thanks,
 --Greg

Sounds like you did a kernel update and did not rebuild DAHDI.

I have never witnessed a failed Tormenta or Digium card.  I have read
about them.

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Re: [asterisk-users] Asterisk generating backtrace

[Andrew Latham]

On Wed, Mar 21, 2012 at 3:04 PM, Jonas Kellens jonas.kell...@telenet.be wrote:
 Hello,

 when generating backtrace I get following output :

 [root@sip ~]# gdb -se asterisk -ex bt full -ex thread apply all bt
 --batch -c core.sip-2012-03-21T10\:57\:29+0100  /root/backtrace.txt
 asterisk: No such file or directory.

 warning: no loadable sections found in added symbol-file system-supplied DSO
 at 0x7fff00799000


 What am I doing wrong ?


 Kind regards,
 Jonas.

maybe you need to quote the core file name with those chars.  Try
renaming the core file to something simple

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Re: [asterisk-users] fallback to default extension

[Andrew Latham]

On Wed, Mar 21, 2012 at 8:27 AM, Paolo Supino paolo.sup...@gmail.com wrote:
 Hi

  I was asked by our development departement to setup asterisk in a
 manner that if someone calls an extension in the department that was
 was only configured, but a handset was never attached to it to fall
 back to a default extension. For example: Someone calls extension
 2408, but there's no phone attached to 2408 it should fall back and
 ring at 2400..

 How do I setup asterisk to find out if there's a phone attached to an
 internal number if not ring another extension?

 TIA
 Paolo

Just add a dial(SIP/2400) at a later priority or any of the other many
ways.  Assuming 2400 is you operator then set the var and drop to the
operator. Verify your options to you dial syntax and any std-exten
setups.

Priority numbers
https://wiki.asterisk.org/wiki/display/AST/Contexts%2C+Extensions%2C+and+Priorities


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Re: [asterisk-users] 8-span TE820 card and interrupts

[Andrew Latham]

On Wed, Mar 21, 2012 at 8:45 AM, Tony Mountifield t...@softins.co.uk wrote:
 Over the years I have experienced a few interrupt issues when using some
 of the Digium E1/T1 cards with Zaptel drivers, and usually resolved them
 by disabling USB devices in the motherboard BIOS settings.

 Now more and more systems are coming without PS/2 connections, so USB is
 needed for the keyboard or KVM.

 I never knew whether these conflict issues were down to the design of the
 card, the motherboard, the Zaptel drivers or the kernel.

 I need soon to build an 8-span E1 system using the Digium TE820 PCIe card,
 and want to know whether I am likely to have to solve similar issues, or
 if they are now history with newer kernels and DAHDI instead of Zaptel.

 I would be interested in any comments from anyone with experience in this
 area. Also, can anyone easily tell me in which version of DAHDI support
 for the TE820 was introduced? (If not, I'm happy to go and search SVN)

 Finally, does anyone have a feel for how much CPU power would be required
 to run Meetme with DAHDI mixing if all 240 channels were active in various
 conferences? (Yes, I know about ConfBridge, but my application currently
 needs to use MeetMe).

 Cheers
 Tony
 --
 Tony Mountifield
 Work: t...@softins.co.uk - http://www.softins.co.uk
 Play: t...@mountifield.org - http://tony.mountifield.org

By design PCIe has dedicated channels for each port and interrupts are
not an issue. There are some vendors that are moving all legacy
devices to a USB based controllers in addition.  PCIe is not like PCI.
 PCIe is serial in nature, not parallel.

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Re: [asterisk-users] fallback to default extension

[Andrew Latham]

On Wed, Mar 21, 2012 at 3:10 PM, Paolo Supino paolo.sup...@gmail.com wrote:
 H Andrew

 Your solution is the simplest I received and so I tried implementing
 it only to discover that it doesn't work as expected...






 TIA
 Paolo

snip

Check your Dial() options... Verify your options to you dial syntax
and any std-exten setups.


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Re: [asterisk-users] res_http_post.so questions

[Andrew Latham]

On Mon, Feb 6, 2012 at 8:24 PM, Josh mojo1...@privatedemail.net wrote:
 In short - is this module essential for the running of Asterisk? What is its
 function? Is there a help/list where I could find a description of what it
 does? Thanks!

The primary goal was to upload audio for IVRs in the Asterisk GUI.

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Re: [asterisk-users] Asterisk 1.8 warns for lines starting with # in /etc/dahdi/system.conf

[Andrew Latham]

On Thu, Dec 22, 2011 at 2:33 PM, Olivier oza_4...@yahoo.fr wrote:
 Hi,

 Testing 1.8.8.0 (with Dahdi 2.5.0.2 and asterisk-gui 2.1.0-rc1), I'm
 seeing this on my console:

 WARNING[25363]: config.c:1208 process_text_line: Unknown directive '#'
 at line 1 of /etc/asterisk/../dahdi/system.conf

 This warning is repeated for every line starting with  a # char.
 Shall I care ?
 Suggestions ?

 (To me, /etc/asterisk/../dahdi/system.conf belongs to Dahdi, not to Asterisk)

 Cheers

These are only warnings.  You can remove the comments if you want just
to stop the warnings.

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Re: [asterisk-users] Asterisk 1.8 warns for lines starting with # in /etc/dahdi/system.conf

[Andrew Latham]

On Thu, Dec 22, 2011 at 2:48 PM, Shaun Ruffell sruff...@digium.com wrote:
 On Thu, Dec 22, 2011 at 06:33:44PM +0100, Olivier wrote:
 Hi,

 Testing 1.8.8.0 (with Dahdi 2.5.0.2 and asterisk-gui 2.1.0-rc1), I'm
 seeing this on my console:

 WARNING[25363]: config.c:1208 process_text_line: Unknown directive '#'
 at line 1 of /etc/asterisk/../dahdi/system.conf

 This warning is repeated for every line starting with  a # char.
 Shall I care ?
 Suggestions ?

 (To me, /etc/asterisk/../dahdi/system.conf belongs to Dahdi, not to Asterisk)

 I could be mistaken, but I don't think, by default, any of the
 configuration files should be including /etc/dahdi/system.conf.

 Where is that include coming from in your configuration?

 In mine it only shows up in some comments:

  $ grep dahdi\/system\.conf /etc/asterisk/*
  /etc/asterisk/chan_dahdi.conf:; /etc/dahdi/system.conf. This sets the tone 
 zone by number.
  /etc/asterisk/chan_dahdi.conf:; /etc/dahdi/system.conf).

 --
 Shaun Ruffell
 Digium, Inc. | Linux Kernel Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at: www.digium.com  www.asterisk.org

Shaun

During a normal install from sources with the GUI, you get

dahdi_guiread.conf:#include ../dahdi/system.conf


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Re: [asterisk-users] Asterisk 1.8 warns for lines starting with # in /etc/dahdi/system.conf

[Andrew Latham]

On Thu, Dec 22, 2011 at 3:42 PM, Kevin P. Fleming kpflem...@digium.com wrote:
 On 12/22/2011 12:02 PM, Shaun Ruffell wrote:

 On Thu, Dec 22, 2011 at 02:54:05PM -0300, Andrew Latham wrote:

 On Thu, Dec 22, 2011 at 2:48 PM, Shaun Ruffellsruff...@digium.com
  wrote:


 I could be mistaken, but I don't think, by default, any of the
 configuration files should be including /etc/dahdi/system.conf.

 Where is that include coming from in your configuration?

 In mine it only shows up in some comments:

  $ grep dahdi\/system\.conf /etc/asterisk/*
  /etc/asterisk/chan_dahdi.conf:; /etc/dahdi/system.conf. This sets the
 tone zone by number.
  /etc/asterisk/chan_dahdi.conf:; /etc/dahdi/system.conf).


 Shaun

 During a normal install from sources with the GUI, you get

 dahdi_guiread.conf:#include ../dahdi/system.conf


 Ahh, ok. Then that explains it. Thanks


 That's very broken. /etc/dahdi/system.conf is not an Asterisk configuration
 file, it doesn't follow the same syntax, and there's no reason whatsoever
 for Asterisk to be reading it.

 What is the GUI? There are lots of GUIs for Asterisk.

 --
 Kevin P. Fleming
 Digium, Inc. | Director of Software Technologies
 Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming

 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at www.digium.com  www.asterisk.org


This is the GUI that I am referring to.
http://svn.asterisk.org/svn/asterisk-gui/  Some people use it,  as it
is very light.  I have many patches for it.

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Re: [asterisk-users] Suppress -- Remote UNIX connection message

[Andrew Latham]

On Wed, Dec 21, 2011 at 12:03 PM, Bryant Zimmerman brya...@zktech.com wrote:
 We have written some monitoring and stat collection scripts that use
 asterisk -rx command  The script runs once a min and logs data and posts
 any critical notifications.  Everything is working well with this method but
 we get the -- Remote UNIX connection / disconnect message once a min and we
 would like to suppress it. Is it possible without reducing the verbose
 logging level.

 Thanks

 Bryant


from http://svn.asterisk.org/svn/asterisk/trunk/configs/asterisk.conf.sample

;hideconnect = yes

; Hide messages displayed when a remote console; connects and disconnects.

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Re: [asterisk-users] Digium TE205P leds flash red on startup

[Andrew Latham]

On Thu, Dec 15, 2011 at 11:05 AM, Vieri rentor...@yahoo.com wrote:
 Hi,

 I have a new Digium TE205P 2-span E1 card I just installed on a server.

 As soon as I boot the machine, the card's leds flash red (ports 1 and 2) - 
 even when in the BIOS.

 That's not good, right?

 I don't have another machine to test at the moment but would like to know 
 what to expect.
 I have several single-span E1 cards and when the machine boots, their leds 
 are off until the kernel module is loaded.

 What could be the problem with my TE205P? Could it be damaged (brand new) or 
 is it more likely to be a PCI-BIOS issue?

 Thanks,

 Vieri

That is normal expected behavior.  The card is in a red alarm state
that just means there is no link.  Its fine.

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Re: [asterisk-users] Best PBX for Call Centers?

[Andrew Latham]

On Thu, Dec 15, 2011 at 3:39 PM, Carlos Alvarez car...@televolve.com wrote:

 On Thu, Dec 15, 2011 at 11:33 AM, Tarek Sawah tareksa...@hotmail.com
 wrote:


 Hello List,
 I have customer with a 40 Agents call center. and is looking to install a
 PBX switch that can serve those agents.
 As per my experience i suggested Asterisk as i have tested it with Call
 Centers, however he has been advised not to use it although his provider is
 using Asterisk to send him calls. He has been advised to use Sippy which
 they claim is more stable than Asterisk.


 More stable?  We have Asterisk servers that have run for many years without
 being unstable.  There's a pair of them in a colo facility that are 7
 years old and haven't been touched in at least two years.  Just how many
 more years do you need to be more stable?

 Asterisk isn't perfect, but done right it's quite stable.  IMO, the people
 giving you advice just don't know how to do it right.


 --
 Carlos Alvarez
 TelEvolve
 602-889-3003

I fully support the statement by Carlos.  Planing, engineering and
other factors can make almost any software stable.  Experience is key.


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Re: [asterisk-users] What version to upgrade to...?

[Andrew Latham]

On Mon, Dec 12, 2011 at 12:26 PM, Danny Nicholas da...@debsinc.com wrote:
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
 Sent: Monday, December 12, 2011 8:27 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] What version to upgrade to...?

 2011/12/12, Mike Diehl mdi...@diehlnet.com:
 Hi all,

 I have 2 servers running 1.6.2.9 and I'm about to build a third server.
 This
 suggests the possibility of doing a rolling upgrade of all of my servers.

 This brings up the question of what version to install and upgrade to.
 I don't have many upgrade opportunities, so I'd like to get as much
 bang for my buck.  Since I've applied some custom patches to my 1.6,
 I'd also like to get to a new enough version that my patches would be
 useful to the community.

 Should I go to 1.8.x?  Or all the way up to 10.x?  This is a
 production system and I can't afford to be testing code.

 --

 Take care and have fun,
 Mike Diehl.



 I'm roughly wondering the same thing.

 If I may add, I read few weeks ago, that Asterisk's SNMP features required
 asterisk to run as root. If any of  asterisk 1.8 or 10 version could solve
 this limitation, that would convince to dive in that one.

 I'm wondering if the bind 161 as root statement is a mis-statement or if
 not, maybe somebody like Tzafir can explain why since none of the other
 Asterisk binds require root access (this message is still in 10.0-rc3).


Any port under 1024 is a reserved system port and normally can only be
opened by root.  161 is under 1024, thus root.  You can run snmp on
other ports if you really want to.

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Re: [asterisk-users] Walkie talkie to sip phone interface

[Andrew Latham]

On Wed, Nov 30, 2011 at 6:20 PM, Ferdinand Babas ba...@cfht.hawaii.edu wrote:
 Hi All,

 I've been trying to find a solution that would allow our sip phones to
 communication with walkie talkies.  Our setup is that we have sip phones
 setup in 2 locations, headquarters and dome.  We can communication from
 headquarters and dome through sip phones, but within the dome we have
 technicians that use walkie talkies to communicate as they go about their
 work.  Our hope is to allow 2 way communications from our sip phones at
 headquarters (or within the dome) with our technicians using their walkie
 talkies as they are working throughout the dome.  Not sure if this is
 possible but I would appreciate any suggestions.

 Thanks,

 Ferdinand

Yes there are interfaces for POTS, SIP, H323 and others.   People
could help if they knew what type of signaling or vendor you are
using. A google search gave me this.

http://www.motorola.com/web/Business/Products/Two-way%20Radio%20Infrastructure/Gateways/MOTOBRIDGE%20Interoperable%20IP%20Solution/_Documents/MotoBridgeSS_Final.pdf


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Re: [asterisk-users] Limit monthly calls by context

[Andrew Latham]

On Wed, Nov 16, 2011 at 10:46 AM, Hans Goossen goos...@planet.com.py wrote:
 Hello group,

 I have this situation:

 I have several contexts with a few extensions each one. I need to give every 
 context a limited quantity of minutes they can use. All the extensions in the 
 context will share the same bag of minutes. Meaning ext 101 use 1900 mins, 
 ext 102 60 mins and ext 40 mins.
 The limit must be monthly.

 I guess some billing solution can do the trick, but I think it's too much 
 for that little. I don't need any other feature.

 I was thinking something like checking the CDR before make the call, I know 
 it may permit some extra minutes to be used, but it really doesn't need to 
 be that exact. A couple of extra minutes won't hurt.

 Ideas, suggestions ?

 Hans Goossen
 Investigación  Desarrollo
 Planet S.A.
 http://www.pla.net.py

You can use the DB[1] to add a table with user and seconds. Then use
the start and end seconds to do this.  I know there are many ways of
doing this with AGI, Manager, Realtime, etc...

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Function_DB

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Re: [asterisk-users] Issues with Xorcom astribanks when running Ubuntu 11.10 (oneiric) as a quemu/libvirt guest

[Andrew Latham]

On Sat, Oct 29, 2011 at 3:14 PM, Eric van der Vlist v...@dyomedea.com wrote:
 Hi,

 Xorcom astribanks get initialized straight on when using Ubuntu 11.10
 packages but I am having a hard time to get the same result running in a
 qemu/libvirt image.

 The first difficulty is that astribanks devices get different usb device
 ids during their initialisation process, requiring hot plug support.

 I have figured out how to solve this issue using the technique described
 in this post :
 http://www.blogs.uni-osnabrueck.de/rotapken/2011/04/11/how-to-auto-hotplug-usb-devices-to-libvirt-vms/

 That doesn't seem to be enough and the initialisation fails with a
 status 1:

 Oct 28 18:58:19 asterisk-rg 'xpp_fxloader'[1006]: Trying to find what to
 do for product e4e4/1160/101, device /dev/bus/usb/001/004
 Oct 28 18:58:19 asterisk-rg 'xpp_fxloader'[1010]: Loading firmware
 '/usr/share/dahdi/USB_FW.hex' into '/dev/bus/usb/001/004'
 Oct 28 18:58:23 asterisk-rg 'xpp_fxloader'[1024]: Trying to find what to
 do for product e4e4/1161/101, device /dev/bus/usb/001/005
 Oct 28 18:58:34 asterisk-rg
 'xpp_fxloader'[1035]: /usr/sbin/astribank_tool failed with status 1

 Seeing that Xorcom requires USB 2.0 and that the current versions of
 libvirt and qemu in Ubuntu 11.10 emulate USB 1.10 in guests, I have
 installed Boris Derzhavets' packages:
 https://launchpad.net/~bderzhavets/+archive/seabios163 and updated my
 host definition to emulate USB 2.0 but I still have the same issue.

 Have I missed something?

 Thanks,

 Eric


Try Xorcom's great support  Tzafrir posted the solution to this a
few months ago, search the list for it.

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Re: [asterisk-users] nvfaxdetect in 10.0

[Andrew Latham]

On Tue, Oct 18, 2011 at 6:21 PM, Danny Nicholas da...@debsinc.com wrote:
 Hi gang,

     We are moving our 1.4 asterisk with DAHDI over to 10.0 with
 SIP.  Everything is going nicely except that I can’t get NV_FAXDETECT to
 compile properly into 10.0.  Because of this, I will have to have my
 receptionist manually transfer incoming faxes.  Any suggestions?



 Thanks in Advance

 Danny Nicholas

Lacking info but guess

# echo faxdetect = yes  /etc/asterisk/sip.conf

might help


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Re: [asterisk-users] asterisk hardware

[Andrew Latham]

On Tue, Oct 4, 2011 at 6:06 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
 On Fri, Sep 30, 2011 at 10:50:12AM -0400, Adam Moffett wrote:
  Is there any reason not to run Asterisk on an Intel Atom board?

 Only if it's not strong enough. Note that Atom may mean some different
 things. So consider taking various reports with a few grains of salt.

 --
               Tzafrir Cohen
 icq#16849755              jabber:tzafrir.co...@xorcom.com
 +972-50-7952406           mailto:tzafrir.co...@xorcom.com
 http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

Like Tzafrir says keep an eye out for what Atom is.  The 1.6-1.8 ghz
processor is powerful enough for simple servers but some of the
supporting chipsets and hardware may not be.  My personal suggestion
is the 
http://www.supermicro.com/products/motherboard/ATOM/ICH9/X7SPE-HF-D525.cfm
which also has an IPMI onboard.

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Re: [asterisk-users] Beep file with Record

[Andrew Latham]

On Tue, Oct 4, 2011 at 11:37 AM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
 This is my complete CLI logging

 -- Executing [s@ serviceline:93] Record(CAPI/ISDN1#02/318647615-37, 
 /var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,60) in 
 new stack
 [Oct  4 16:19:38] WARNING[13370]: file.c:644 ast_openstream_full: File beep 
 does not exist in any format
 [Oct  4 16:19:38] WARNING[13370]: file.c:950 ast_streamfile: Unable to open 
 beep (format 0x8 (alaw)): No such file or directory
 [Oct  4 16:19:38] WARNING[13370]: app_record.c:281 record_exec: 
 ast_streamfile failed on CAPI/ISDN1#02/318647615-37

 In de Conf file I use the following command:
 exten = 
 s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
 exten = s,n,Record(${A_serviceline_file}.wav,0,60)


 -Oorspronkelijk bericht-
 Van: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] Namens Danny Nicholas
 Verzonden: 04-10-2011 16:30
 Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Onderwerp: Re: [asterisk-users] Beep file with Record

 Usually this message is received because you did something like
 playback(beep.gsm) or playback(beep.wav) instead of playback(beep).  It is
 (IMO) somewhat confusing because you have to do record(foo.gsm) but you have
 to playback using playback(foo).

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Arjan Kroon |
 Mobillion
 Sent: Tuesday, October 04, 2011 9:21 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Beep file with Record

 Hi,

 I'm using the functionality Record in asterisk 1.8.5.
 But when I want to record something I get the following error message:
 file.c:644 ast_openstream_full: File beep does not exist in any format

 Could anybody tell me where I have to place the beep.gsm file?
 I already tried the following directories:
        /var/lib/asterisk/sounds/beep.gsm
        /var/lib/asterisk/sounds/recordings/beep.gsm

 Regards,

 Arjan Kroon

Beep is called from
http://svn.asterisk.org/svn/asterisk/trunk/apps/app_record.c and it
looks fine a first glance.  Are you using the language prefix?

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Re: [asterisk-users] Overlap SIP dialing

[Andrew Latham]

On Thu, Sep 8, 2011 at 9:38 AM, Kevin P. Fleming kpflem...@digium.com wrote:
 On 09/07/2011 11:06 AM, Daniel Tryba wrote:

 The aim of the quest for overlap dialing is to let the user enter a
 number at their own pace but immediatly dial when all digits are
 received (just like plain old ISDN does). My trunk is a bunch of E1 PRIs
 in overlap mode. The following just works for any SIP client (without
 overlap dialing):
 exten =  _X.,1,Answer()
 exten =  _X.,n,Dial(${TRUNK})

 Unless I'm mis-remembering, this was the point of adding the '!' dialplan
 match character. If you use _X!, and you have your SIP endpoints configured
 to send an INVITE as soon as the user has entered two digits (and you have
 no other patterns in the context that could match), then the dialplan will
 match against that and initiate a Dial() on your ISDN PRI. Since the number
 is not yet complete, the SETUP message on the PRI won't result in the call
 proceeding, and as the user of the phone presses additional digits they'll
 be sent to Asterisk as DTMF, bridged over to chan_dahdi, and it will send
 them as INFORMATION messages rather than as DTMF digits, because it knows
 the outbound call is still in 'dialing' state.

 However, this is still going to 'mess with CDRs' as you put it, because the
 only switch in the network that knows the complete number that was dialed is
 the PSTN switch that your PRI is connected to. It seems possible that
 chan_dahdi could 'update' the EXTEN on the current channel as the additional
 digits are dialed so that the CDR contains the complete number, but I have
 no idea whether it does or not.

 --
 Kevin P. Fleming
 Digium, Inc. | Director of Software Technologies
 Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at www.digium.com  www.asterisk.org

Exactly Kevin.  I remember now that I was using it for my
http://etel.wiki.oreilly.com/wiki/index.php/Asterisk_Man_in_the_Middle
in some setup/testing.

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Re: [asterisk-users] Overlap SIP dialing

[Andrew Latham]

On Thu, Sep 8, 2011 at 11:21 AM, Olle E. Johansson o...@edvina.net wrote:

 8 sep 2011 kl. 17:17 skrev Kevin P. Fleming:

 Honestly, I'm not really sure that there is a practical solution here. ISDN 
 overlap dialing was intended for 'dumb' phones, and SIP phones aren't 'dumb' 
 :-)

 That's a quote that goes to my quote storage layer.

 /O ;-)
 --

I want a t-shirt   SIP phones aren't 'dumb' :-)
Overlap dialing has very limited use, however I found it helpful when
testing integration with other PBX/VM/PSTN connections.

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Re: [asterisk-users] Overlap SIP dialing

[Andrew Latham]

On Wednesday, September 7, 2011, Olle E. Johansson wrote:


 7 sep 2011 kl. 15:59 skrev Daniel Tryba:

  Looking at the history of the list I don't expect any answer but lets
  try anyway:
 
  Does anybody use overlap dialing from SIP devices to asterisk? Does
  anybody have a working example?

 To add to your question: Does anyone have a phone that supports this
 properly?

 /O


Yup, I have a few...  http://wiki.snom.com/Settings/overlap_dialing


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Re: [asterisk-users] Overlap SIP dialing

[Andrew Latham]

On Wed, Sep 7, 2011 at 10:28 AM, Olle E. Johansson o...@edvina.net wrote:

 7 sep 2011 kl. 16:20 skrev Andrew Latham:



 On Wednesday, September 7, 2011, Olle E. Johansson wrote:

 7 sep 2011 kl. 15:59 skrev Daniel Tryba:

  Looking at the history of the list I don't expect any answer but lets
  try anyway:
 
  Does anybody use overlap dialing from SIP devices to asterisk? Does
  anybody have a working example?

 To add to your question: Does anyone have a phone that supports this 
 properly?

 /O

 Yup, I have a few...  http://wiki.snom.com/Settings/overlap_dialing


 Great. Haven't seen this - thank you.

 The whole concept is interesting. Suppose the call forks and one UA answers 
 with 484, another with 486 and another with 180 ringing. What are you 
 supposed to do? I think there's a problem with the RFC 3261 here and don't 
 know if it's been clarified.

 Now - in the case of Asterisk if we call out to two devices from the dialplan 
 and one responds with 484 and another with 180 ringing - what happens in 
 Asterisk?

 /O


In the past (2004/2005) I have dealt with this and hoot and holler* systems...

* http://en.wikipedia.org/wiki/Hoot-n-holler

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Re: [asterisk-users] USB or Ethernet based FXO device ?

[Andrew Latham]

On Wed, Aug 31, 2011 at 11:49 AM, Carlos Chavez cur...@telecomabmex.com wrote:
 On Wed, 2011-08-31 at 17:03 +0200, Marco Signorini wrote:
 Hi.

 I was following this thread. We normally use Patton SmartNode SN4112
 series to interface to FXO ports. But I'm looking for something
 different for a future setup.
 Xorcom USB channel banks seems something quite interesting. Is there
 anyone that could/would share experiences using that?
 We need to replace an old PBX interfaced to 50 FXS and 8 BRI ISDN in
 Italy.
 My concern is about reliability of USB
 Any success stories with it? Tips and tricks?


 Gilles wrote:
  On Tue, 30 Aug 2011 10:41:30 -0500, Carlos Chavez
  cur...@telecomabmex.com wrote:
 
     Actually Xorcom makes USB channelbanks of up to 32 FXO/FXS ports.
  
 
  Thanks for the tip. It looks like the smallest option is 8 FXO ports:
 
  www.xorcom.com/telephony-interfaces/astribank-models.html
 
 
        We use them a lot for high density analog lines and extensions.  The
 only thing to keep in mind is to always connect the units in a
 predetermined order to the USB ports so you do not mess up your
 configuration.  Apart from that they are really easy to use since the
 drivers are included in the standard dahdi distribution.


 --
 Telecomunicaciones Abiertas de México S.A. de C.V.
 Carlos Chávez Prats
 Director de Tecnología
 +52-55-91169161 ext 2001

I am sure that Tzafrir can pipe in here.  There is an method of
setting the ID of each astribank to keep them in order.  Ask Xorcom
for more info.

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Re: [asterisk-users] Bind SIP over TCP port in asterisk 1.4.42.

[Andrew Latham]

On Thu, Aug 25, 2011 at 9:26 AM, Catalin S. jonsonpla...@gmail.com wrote:
 Hello,
 I need to listen on tcp 5060 on my actual asterisk 1.4.42. I tried in
 sip.conf at
 [general] section the following options:
 transport=tcp
 tcpenable=yes
 tcpbindaddr=0.0.0.0
 but after all that changes i still not see tcp port raised up. Did somebody
 had the same problem and had some solutions?
 Thank you very much.
 Jonson.
 --

I looked  TCP + Transport are listed in
http://svn.asterisk.org/svn/asterisk/branches/1.4/channels/chan_sip.c
but not in 
http://svn.asterisk.org/svn/asterisk/branches/1.4/configs/sip.conf.sample

try
transport=TCP

Beware, some systems use SIP(not encrypted) over TCP on port 5061,
which is not really wrong, just not what the standards say.


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Re: [asterisk-users] Polycoms rebooting themselves

[Andrew Latham]

On Wed, Aug 17, 2011 at 6:01 PM, Mike Diehl mdi...@diehlnet.com wrote:
 Hi all,

 I've got a customer with 10 Polycom 335's and the latest(ish) firmware.  For
 the most part, things are working well.

 However, about once a day, a given phone will just reboot.  They don't do it
 all at once, and they don't do it along any pattern that I can discern.

 I've got a tcpdump running against one of the phones on my server, but so far,
 it's not rebooted, so I've got nothing to look at.

 Any other ideas?


 --

 Take care and have fun,
 Mike Diehl

POE Switch is running close to its limit, some older Polycom phones do
not adjust the POE usage with some switches.  There is also a
scheduled check-config that you can set in the phone provisioning.

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Re: [asterisk-users] Polycoms rebooting themselves

[Andrew Latham]

On Wed, Aug 17, 2011 at 6:35 PM, Mike Diehl mdi...@diehlnet.com wrote:
 On Wednesday 17 August 2011 4:12:21 pm Doug Lytle wrote:
 Mike Diehl wrote:
  Any other ideas?

 They should be writing out logs to your ftp server (If your provisioning
 them that way).

 At the moment, my web server isn't capable of receiving the phones POST
 request.  Sounds like that's going to change... soon!

 --

 Take care and have fun,
 Mike Diehl.

On large installs SYSLOG is a better option.

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Re: [asterisk-users] Where to proceed next

[Andrew Latham]

On Thu, Aug 11, 2011 at 3:04 PM, Danny Nicholas da...@debsinc.com wrote:
 Hello list,

       I presently use the 1.4 releases because I enjoy sleeping
 at night.  I understand that 1.4 reaches end-of-life in a little over 8
 months (https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions).  I
 also know (as best as I can) that no genie is going to make Asterisk 1.4 go
 “poof” on this date.  My clients would probably sleep better thinking they
 were running a PBX that didn’t have this “drop dead” date however.   Since
 1.6.X has the same time constraints as 1.4, it seems it would be a waste of
 time going that direction.  Should I go down the 1.8 .X path to have 4 years
 of time, but the headaches that have been documented here, or pursue the
 10.X which is presently considered Beta? (is it really beta,  or just
 relabeled 1.8?).

 Thanks

 Danny Nicholas

Regardless of what release you choose to use.  The best thing you can
do is to check that all the features and configurations you use are in
the test-suite. Look at bamboo and see how the tests are going.  If
you have a feature in your dial-plan that concerns you, share it and
think of a way to test this feature.

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Re: [asterisk-users] burned module X400M

[Andrew Latham]

On Thu, Aug 4, 2011 at 10:36 AM, Agustina Berretta
agustina.berre...@gmail.com wrote:
 Hello folks!

 How can I be sure a module was burned by high tension?

 I installed the module, configured it using dahdi_genconf -vv

 but when I type: asterisk -rx dahdi show channels I don´t see the module.

 Thanks a lot

cat /proc/dahdi/*

If there is no /proc/dahdi then maybe you don't have the modules loaded.


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Re: [asterisk-users] Lightning and thunder

[Andrew Latham]

On Wed, Jul 27, 2011 at 9:44 AM, Claude Hayn chayn...@gmail.com wrote:
 We are frequently losing power during lightning storms.  (Yes we have UPS,
 but often by the time power comes back up the UPS has run out of juice)

 We are using Asterisk with a T1/PRI card as a front end connected to our
 PBX.  Whenever there is a power outage both the Asterisk box and the PBX
 automatically reboot when power returns.

 The Asterisk box automatically reconnects to the ITSP SIP Peer, and the PBX
 to the T1/PRI Card Asterisk box.

 Incoming calls connect, but outbound calls will not complete until the
 Asterisk box is manually rebooted again.

 Does anyone know of a solution for this issue?  Having to get up in the late
 night to manually reboot the Asterisk box is getting old!

 Thank you,

 Claude

You may want to look at http://www.networkupstools.org/ to control you
power down in a graceful manner.

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Re: [asterisk-users] [1.4] Minimal installation?

[Andrew Latham]

On Mon, Jul 18, 2011 at 9:20 AM, Gilles codecompl...@free.fr wrote:
 Hello,

 I'd like to run Asterisk on an embedded device, where space is scarce.
 It should be able to handle calls from a VoIP provider in SIP, calls
 from the PSTN through Dahdi, and voicemail.

 If someone's already done this, I'd like to know which
 directories/files are required for a basic install?

 Does this look right?
 =
 /bin/asterisk

 /etc/asterisk/
        asterisk.conf
        logger.conf
        modules.conf
        sip.conf
        extensions.conf
        voicemail.conf

 /etc/init.d/asterisk

 /usr/lib/asterisk/modules/

 /var/lib/asterisk/agi-bin/moh - /var/lib/asterisk/sounds/moh
 /var/lib/asterisk/sounds/
 /var/lib/asterisk/agi-bin/static-http/

 /var/spool/asterisk/
 =

1. Sound files are likely the biggest issue.

2. DAHDI installs all firmwares by default, find what you need and
remove the rest.

3. Config files are mostly white space use this.

#Removes beginning and ending white space
sed -i 's/^[ \t]*//;s/[ \t]*$//' /etc/asterisk/*.conf

#Deletes empty lines
sed -i '/^$/d' /etc/asterisk/*.conf

#Adds a line return above a [
sed -i '/^\[/{x;p;x;}' /etc/asterisk/*.conf

# Deletes comments that starts with ; at the beginning of a line
sed -i '/^\;/d' /etc/asterisk/*.conf

# Deletes comments after the ; at any place
sed -i 's/;.*//' /etc/asterisk/*.conf

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Re: [asterisk-users] Using Firewall to protect Asterisk

[Andrew Latham]

On Fri, Jul 15, 2011 at 12:47 PM, CDR vene...@gmail.com wrote:
 I need to keep out all connection from 5 countries, which originate
 most of the Denial of Service attacks. The entries are
 around 9000 if used as xx.xx.0.0/16. I heard that there is a smarter
 way to do this by using User Tables in iptables, that will keep the
 speed equal to LOG(x). I already tried using  a straight list and it
 kills the box. Unless a smarter way us found, there is no way to use
 iptables.

 Federico

DROP will remove the vast majority of bad networks.  Fail2ban[2] for
the rest or recent[3] with triggers at port 139 will get the rest.

[1] http://www.spamhaus.org/drop/
[2] http://www.fail2ban.org/wiki/index.php/Main_Page
[3] http://snowman.net/projects/ipt_recent/

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Re: [asterisk-users] TDM400p susceptible to EMI?

[Andrew Latham]

On Wed, Jul 13, 2011 at 2:51 PM, Steve Edwards
asterisk@sedwards.com wrote:
 I have a TDM400p with 3 fxs and 1 fxo daughter cards.

 It's in a mini-itx case with a 'right-angle' PCI riser card so the TDM400p
 is 'sandwiched' between the Atom D525 CPU and the 2.5 hard drive.

 I'm getting a bunch of clicks and pops on all ports.

 Has anybody had a similar experience? Did you find a solution?

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

How is it grounded?  Silly I know but its possible.

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Re: [asterisk-users] TDM400p susceptible to EMI?

[Andrew Latham]

On Wed, Jul 13, 2011 at 5:16 PM, Tim Nelson tnel...@rockbochs.com wrote:
 - Original Message -
  On Wed, Jul 13, 2011 at 2:51 PM, Steve Edwards
  asterisk@sedwards.com wrote:
  I have a TDM400p with 3 fxs and 1 fxo daughter cards.
 
  It's in a mini-itx case with a 'right-angle' PCI riser card so the
  TDM400p
  is 'sandwiched' between the Atom D525 CPU and the 2.5 hard drive.
 
  I'm getting a bunch of clicks and pops on all ports.
 
  Has anybody had a similar experience? Did you find a solution?

 On Wed, 13 Jul 2011, Andrew Latham wrote:

  How is it grounded? Silly I know but its possible.

 This box is using a picoPSU-80 80w DC-DC 'power supply' fed from an
 inline 'laptop brick.'

 I ran a separate lead from the chassis to the grounding plug on the
 same
 'duplex' wall outlet. No joy.


 picoPSU's are typically pretty good. I wouldn't suspect it in this case then 
 unless your power supply is underpowered for the hardware's current draw.

 --Tim

I also use the pico-PSUs and have not had any issues.

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Re: [asterisk-users] References customers

[Andrew Latham]

On Sun, Jul 10, 2011 at 5:22 PM, bilal ghayyad bilmar...@yahoo.com wrote:
 I mean:

 What are the customers (big customers I mean) that they installed Asterisk in 
 their company to be as a reference?

 Example: Toyota, GM, Hilton, Shiraton hotel, ... etc

 An example of such companies, whom?

 Is there a link that mention them?

 Regards
 Bilal

 ---
 What do you mean by customers? Are you looking for
 testimonials from
 satisfied users?

http://www.digium.com/  scroll down to the bottom  Google, Yahoo,
US Army, IBM  just a few small little businesses.


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Re: [asterisk-users] How to without GUI

[Andrew Latham]

On Fri, Jul 1, 2011 at 1:55 PM, Danny Nicholas da...@debsinc.com wrote:
 Hey gang,

     I’ve got a CISCO SPA3102 that I want to set up.  My
 environment is not favorable for using the Asterisk GUI interface – does
 anybody have step by step how to set up a SIP trunk just by editing
 shudder sip.conf?



 Thanks in Advance

 Danny Nicholas

from http://svn.asterisk.org/svn/asterisk/trunk/configs/sip.conf.sample

;--- sample definition for a provider
;[provider1]
;type=peer
;host=sip.provider1.com
;fromuser=4015552299  ; how your provider knows you
;remotesecret=youwillneverguessit ; The password we use to authenticate to them
;secret=gissadetdu; The password they use to contact us
;callbackextension=123; Register with this server and
require calls coming back to this extension
;transport=udp,tcp; This sets the transport type to
udp for outgoing, and will
; ;   accept both tcp and udp. Default
is udp. The first transport
; ;   listed will always be used for
outgoing connections.


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Re: [asterisk-users] How to without GUI

[Andrew Latham]

On Fri, Jul 1, 2011 at 2:23 PM, Doug Lytle supp...@drdos.info wrote:
 Danny Nicholas wrote:

 step by step how to set up a SIP trunk just by editing shudder sip.conf

 You'll find that most here don't use a GUI.

 Doug


Doug

Many people get addicted to the users.conf and res_phoneprov for
automatic phone provisioning.  The GUI works quite well and is very
light.  It works with only the manager and java/emca-scripting.  We
need to extend the res_phoneprov to work with other configuration
files.

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Re: [asterisk-users] Clarification of the terms shown on CLI

[Andrew Latham]

On Tue, Jun 28, 2011 at 4:53 PM, Bruce B bruceb...@gmail.com wrote:
 Hi everyone,
 When doing a sip show settings on Asterisk 1.6.2.18, I see the following:
   Match Auth Username:    No
   Allow unknown access:   Yes
   Allow subscriptions:    Yes
   Allow overlap dialing:  Yes
   Allow promsic. redir:   No
   Enable call counters:   No
 What do each of above signify?
 Thanks

from http://svn.asterisk.org/svn/asterisk/trunk/configs/sip.conf.sample

;match_auth_username=yes; if available, match user entry using the
; 'username' field from the authentication line
; instead of the From: field.
;allowguest=no  ; Allow or reject guest calls (default is yes)
; If your Asterisk is connected to the Internet
; and you have allowguest=yes
; you want to check which services you offer 
everyone
; out there, by enabling them in the default 
context (see below).
;allowsubscribe=no  ; Disable support for subscriptions.
(Default is yes)
allowoverlap=no ; Disable overlap dialing support.
(Default is yes) --- btw this one is funny
;promiscredir = no  ; If yes, allows 302 or REDIR to
non-local SIP address
; Note that promiscredir when
redirects are made to the
; local system will cause loops since
Asterisk is incapable
; of performing a hairpin call.
;callcounter = yes  ; Enable call counters on devices.
This can be set per
; device too.

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Re: [asterisk-users] gmkioliyuiuuuioihlllkppuipppppookkkgkkkhklfgko

[Andrew Latham]

On Mon, Jun 20, 2011 at 11:47 PM, Alex Balashov
abalas...@evaristesys.com wrote:
 I nominate this for most imaginative use of Asterisk-users of 2011.

 --
 Alex Balashov - Principal
 Evariste Systems LLC
 260 Peachtree Street NW
 Suite 2200
 Atlanta, GA 30303
 Tel: +1-678-954-0670
 Fax: +1-404-961-1892
 Web: http://www.evaristesys.com/

 On Jun 20, 2011, at 8:43 PM, Marcelo marcelol...@gmail.com wrote:



 Sent from my iPhone

 --


butt dial FTW

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Re: [asterisk-users] Dahdi 2.4.0 and Squeeze

[Andrew Latham]

On Tue, Jun 14, 2011 at 9:44 AM, Olivier oza_4...@yahoo.fr wrote:
 Hi,

 I'm using a two-years old installation script for the first time on a
 Squeeze (linux 2.6.32) platform.
 For an unknown reason (might be an obvious one), Dahdi can't be loaded
 anymore.

 1. First of all, it seems /dev/dahdi content was previously (ie in Lenny)
 owned by asterisk:asterisk (asterisk is run as asterisk).
 Now it is owned by root.
 Any clue about this ?

 2. Secondly, I changed /dev/dahdi content ownership by hand. Then when I'm
 trying to load chan_dahdi, I can read :

 module load chan_dahdi
 Unable to load module chan_dahdi
 Command 'module load chan_dahdi' failed.
 [Jun 14 15:41:53] WARNING[8150]: chan_dahdi.c:1469 dahdi_open: Unable to
 specify channel 1: No such device or address
 [Jun 14 15:41:53] ERROR[8150]: chan_dahdi.c:8816 mkintf: Unable to open
 channel 1: No such device or address
 here = 0, tmp-channel = 1, channel = 1
 [Jun 14 15:41:53] ERROR[8150]: chan_dahdi.c:14229 build_channels: Unable to
 register channel '1-2'

 Suggestions ?

 Regards

Look at the init.d file to see who it is started as. You may want to
test by su as the Asterisk user.

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Re: [asterisk-users] TFTP to be installed in Linux same asterisk machine to be used with Cisco

[Andrew Latham]

On Sat, Jun 11, 2011 at 10:29 AM, bilal ghayyad bilmar...@yahoo.com wrote:
 Hi All;

 Any one can suggest a TFTP server to be installed in Fedora (same machine 
 that Asterisk is installed) to be used for Cisco IP Phones to download the 
 required firmware and configuration files.

 Thanks for the help in advance.
 Regards
 Bilal


Try this 
http://etel.wiki.oreilly.com/wiki/index.php/Dynamic_Phone_Provisioning_with_res_phoneprov_and_TFTP

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Re: [asterisk-users] issues.asterisk.org/jira not working

[Andrew Latham]

On Wed, Jun 8, 2011 at 3:20 PM, Russell Bryant russ...@digium.com wrote:
 A number of people are reporting that Safari is not working properly with 
 JIRA.  Use Firefox or Chrome for now.

 --
 Russell Bryant
 Digium, Inc.   |   Engineering Manager, Open Source Software
 445 Jan Davis Drive NW    -     Huntsville, AL 35806  -  USA
 www.digium.com  -=-  www.asterisk.org -=- blogs.asterisk.org


This could be an issue with the CA keys used in Safari.  I remember
having to chain load a root key for a server just for iphone support a
while back.  looking

Apache option is SSLCertificateChainFile /full/path/to/your.ca-bundle

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Re: [asterisk-users] issues.asterisk.org/jira not working

[Andrew Latham]

On Wed, Jun 8, 2011 at 5:56 PM, Satish Patel satish...@hotmail.com wrote:

  It not working on iPhone. It's saying not able to make secure connection

 --
 Sent from my iPhone

Satish, Can you share what the SSL/TLS Cert says?  Safari and mobile
platforms have a smaller list of CAs, just to make life hard for us
sysadmin types...

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Re: [asterisk-users] OT, free software for SIP ladder diagrams?

[Andrew Latham]

On Tue, May 17, 2011 at 1:12 PM, Steve Edwards
asterisk@sedwards.com wrote:
 I was debugging a turnup with Global Crossing the other day and they
 presented me with a web page that displayed a 'ladder diagram' of a call
 including a ton of detail all neatly organized in tabs and links so you
 could drill down to any level of detail needed.

 The copyright notice says 'Copyright© 2008 Empirix.'

 Is there any free software available to analyze a pcap or similar packet
 dump with similar features?

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

I think there are some, I saw one mentioned on the BACNet mailing
list...  ahh yes http://cloudshark.org/ takes tshark and wireshark
uploads... close to what you are looking for...

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Re: [asterisk-users] OT, free software for SIP ladder diagrams?

[Andrew Latham]

On Tue, May 17, 2011 at 2:32 PM, Steve Edwards
asterisk@sedwards.com wrote:
 On Tue, May 17, 2011 at 1:12 PM, Steve Edwards
 asterisk@sedwards.com wrote:

 I was debugging a turnup with Global Crossing the other day and they
 presented me with a web page that displayed a 'ladder diagram' of a call
 including a ton of detail all neatly organized in tabs and links so you
 could drill down to any level of detail needed.

 The copyright notice says 'Copyright© 2008 Empirix.'

 Is there any free software available to analyze a pcap or similar packet
 dump with similar features?

 On Tue, 17 May 2011, Andrew Latham wrote:

 I think there are some, I saw one mentioned on the BACNet mailing list...
  ahh yes http://cloudshark.org/ takes tshark and wireshark uploads... close
 to what you are looking for...

 Thanks, but not unless I've missed a lot on first glance.

 Cloudshark looks like an online simplified version of wireshark.

 The Empirix product provides a much higher level overview as well as
 allowing you to drill down if needed. The Empirix product appears to be much
 more oriented towards analyzing the life of a call rather than every packet
 on the wire.

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


Yes, I should have put more emphasis on close.  If you find anything
please pass it along to the list.  Did you get a screenshot by any
chance?

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Re: [asterisk-users] Light indicator managed by Asterisk

[Andrew Latham]

On Thu, May 12, 2011 at 12:50 PM, Jonas Kellens
jonas.kell...@telenet.be wrote:
 Hello,

 is there some way to make Asterisk light up a certain light on an IP-phone ?

 Like MWI, the message waiting indicator can light up if there is voicemail.

 Could this light, or even other lights (like BLF-buttons) be used to give a
 visual notification to the user ?

 For example : if a certain value is set in the Mysql-DB and Asterisk reads
 out this value, can Asterisk react upon it inside the dialplan to make a
 light lit up ?

 2nd example : if a certain extension is called, can we perform inside the
 dialplan an action that makes a light lit up on a Snom or Yealink IP-phone ?

 I don't know if all this is at all possible, but it doesn't harm asking I
 guess...

 If BLF works, then maybe more things are possible in the same way. Just
 thinking outside the box here.


 Kind regards,
 Jonas.

On snom and other phones it is easy...
http://wiki.snom.com/Interoperability/PBX/Asterisk#Extension_Monitoring_.28BLF.29_.26_Call_Pick-Up

Also look at SLA
http://svn.asterisk.org/svn/asterisk/trunk/configs/sla.conf.sample

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Re: [asterisk-users] Really, really loud ringers

[Andrew Latham]

On Mon, May 9, 2011 at 4:40 PM, Justin Sherrill
justin.sherr...@americanrocksalt.com wrote:
 Anyone have some recommended equipment for alerting people to calls in a 
 noisy environment?

 I have Polycom IP550 phones set up in some really noisy environments - our 
 mine hoists - and they tend to drown out the ringers.  I'm using Clarity 
 WR100s now.  They're analog devices, attached to Linksys PAP2T ATAs as part 
 of a call group to get a loud (advertised as 95dB) ring out there, but it 
 still could be louder.  Maybe a light-up option would be better.

 The old phone system here had some huge loudspeakers that someone had wired 
 right into the speakers of the old digital phones.  I haven't figured out yet 
 if they need a different voltage, or even if they still work; they were not 
 responding when I replaced the attached phones.

 Justin C. Sherrill - American Rock Salt
 p: 585-991-6825 f: 585-991-6926 c: 585-298-6826

Look for ADA devices.  The Disabilities Act has encouraged some nice
products. And it allows for you to get ISDN service anywhere in the
country...

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Re: [asterisk-users] Supermicro X7SPE (Atom) as Asterisk server?

[Andrew Latham]

On Sat, May 7, 2011 at 3:05 AM, Vahan Yerkanian va...@arminco.com wrote:
 On 5/6/11 11:52 PM, Andrew Latham wrote:

 On Fri, May 6, 2011 at 2:48 PM, Vahan Yerkanianva...@arminco.com  wrote:

 Has anyone used this board as an Asterisk server?

 http://www.supermicro.com/products/motherboard/ATOM/ICH9/X7SPE.cfm?typ=HIPMI=Y

 I'm mostly interested about the possible compatibility issues this board
 may
 have with the AEX800 card.

 Yes that is a great system and the built-in IPMI is a livesaver...  if
 you are using a full size harddrive you need to apply some protection
 to the card in the case (the superserver 1U).  They are close but not
 touching...

 Thanks for the info, are you using the AEX800 with it?
 How's the load, and what actual performance do you have?

I put my AEX800 in one to test and it works fine.  Normally we use
them with E1 cards.  It has 4 cores (2cores + hyperthreading) and
works very well.  Load is low.  Priced almost the same if not cheaper
than a Soekris this is a great solution for small installs. Use a
picoPSU for a fanless setup..

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Re: [asterisk-users] Supermicro X7SPE (Atom) as Asterisk server?

[Andrew Latham]

On Sat, May 7, 2011 at 10:12 AM, Ryan Wagoner rswago...@gmail.com wrote:
 On Fri, May 6, 2011 at 2:52 PM, Andrew Latham lath...@gmail.com wrote:
 On Fri, May 6, 2011 at 2:48 PM, Vahan Yerkanian va...@arminco.com wrote:
 Has anyone used this board as an Asterisk server?
 http://www.supermicro.com/products/motherboard/ATOM/ICH9/X7SPE.cfm?typ=HIPMI=Y

 I'm mostly interested about the possible compatibility issues this board may
 have with the AEX800 card.

 Yes that is a great system and the built-in IPMI is a livesaver...  if
 you are using a full size harddrive you need to apply some protection
 to the card in the case (the superserver 1U).  They are close but not
 touching...

 --
 ~~~ Andrew lathama Latham lath...@gmail.com ~~~

 I have the X7SPE-HF-D525, 1U SC503-200B case and SSD for my firewall.
 Just keep in mind the 1U case with no fans is like an oven. In a 75F
 room the system temp was 132F and the CPU was 163F. This is within
 operating limits of the Atom platform. However I'm not sure I would
 want a hard drive and telco card in there as well. I ended up putting
 a 40mm rated for 7cfm of airflow fan in the case. The temps dropped
 dramatically to 120F system and 131F.

 Ryan

We are using the standard power supply but both of our data centers
are kept at 14C to 16C.  Most of the PBXs get virtualized but a few
customers pay for their own box.

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Re: [asterisk-users] Supermicro X7SPE (Atom) as Asterisk server?

[Andrew Latham]

On Fri, May 6, 2011 at 2:48 PM, Vahan Yerkanian va...@arminco.com wrote:
 Has anyone used this board as an Asterisk server?
 http://www.supermicro.com/products/motherboard/ATOM/ICH9/X7SPE.cfm?typ=HIPMI=Y

 I'm mostly interested about the possible compatibility issues this board may
 have with the AEX800 card.

Yes that is a great system and the built-in IPMI is a livesaver...  if
you are using a full size harddrive you need to apply some protection
to the card in the case (the superserver 1U).  They are close but not
touching...

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Re: [asterisk-users] receive faxes

[Andrew Latham]

On Wed, May 4, 2011 at 10:12 AM, Eric Wieling ewiel...@nyigc.com wrote:

 Does Hylafax support T.38?

 The free fax works just fine with DAHDI.  I've never tried to do T.38 with 
 that since it seems like it would be complicated and not give me much over 
 using DAHDI.


There is the t38modem[1] project.  But as others will mention, there a
many faxing patents and the t38 is not the same on all vendors. Watch
your back.

1. http://t38modem.sourceforge.net/


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Re: [asterisk-users] receive faxes

[Andrew Latham]

On Wed, May 4, 2011 at 12:00 PM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
...
 (For my part, I'm actually surprised that nobody came up with a proper
 protocol for encapsulating the stream of zeros and ones that make up a fax
 transmission but rely on the precise timing inherent with a circuit-switched
 network, into something more suitable for sending over a packet-switched
 network.  That would have fixed it good and proper.)

 --
 AJS

 Answers come *after* questions.

AJS, thanks, love the humor.  Faxing is considered a legal method of
doing business in many areas.  Maybe lobbing for more effective
digital signatures would help get faxing removed from our everyday
lives.

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Re: [asterisk-users] Having redundancy, so if first IP failed then send for the other

[Andrew Latham]

On Tue, May 3, 2011 at 5:31 PM, bilal ghayyad bilmar...@yahoo.com wrote:
 Hi All;

 I need to configure the SIP account so if first IP address failed then to 
 send for the second IP address. How to do this?

 While configuring the sip account, at the host parameter, can I give two IP 
 addresses separated by comma? Or what should I do to have such redundancy?

 Regards
 Bilal

Try DNS SRV http://en.wikipedia.org/wiki/SRV_record


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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

[Andrew Latham]

On Thu, Apr 28, 2011 at 1:34 PM, satish patel satish...@hotmail.com wrote:
 Where did you download asterisk 1.10 or trunk ? I search and found nothing.
 could your point me there?

 -S

svn co http://svn.asterisk.org/svn/asterisk/trunk /usr/src/asterisk_trunk


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Re: [asterisk-users] DHCP / DNS

[Andrew Latham]

No [1]

1. AsteriskNOW does have some of these services as do many
distributions like Zentyal.

On Wed, Apr 27, 2011 at 2:04 PM, Thomas Perron thomas.per...@gmail.com wrote:
 Are there any internal DHCP or DNS services built-in to the Asterisk code?

 --



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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

[Andrew Latham]

On Wed, Apr 27, 2011 at 3:34 PM, Olle E. Johansson o...@edvina.net wrote:
 Friends,

 We have a discussion on asterisk-dev about the maintenance of the 1.4 branch. 
 According to the release plans, support for 1.4 was scheduled to close in 
 April 2011 - basically now. After that, only security patches would be 
 committed. This is already a delay from the original plan published by 
 Russell Bryant.

 Unfortunately, I think this is way too early. My feeling and experience is 
 that 1.8 is not ready for production in the environments I work in - large 
 scale installations. Customers are not planning migration and all new 
 installs are still 1.4. Tests we've been doing with 1.8 has failed within 
 just a short time and so badly that customers has not paid me to spend any 
 further time with 1.8.

 Last time we went through this process with a LTS release (which we did not 
 know then) it took over one year before we had a stable product to migrate 
 away from 1.2 and jump on the 1.4 track. Hopefully, with the help of 
 community, we can move up to 1.8 late this year or early next year. For me 
 1.8 is the focus, it's the LTS release.

 Not having a supported 1.4 version from the Digium-hosted repositories will 
 mean that we will have to move to separate repositories or branch off from 
 the main track. I already maintain a ton of subversion branches with various 
 patches to 1.4 It takes a lot of time to manage this version that is a fork 
 from the main 1.4 branch. I will soon have to start working with subversion 
 branches for 1.8 to create a compatible version for my customers to test, 
 since most of the patches is not part of 1.8. After a few years of doing 
 this, I know the work involved with managing code myself.

 The Digium team wants to go ahead and not support 1.4 any more, I want to 
 keep 1.4 open for normal bug fixes. What do you think?

 Kevin proposed that the community maintains the 1.4 branch without support 
 from the Digium team. I don't think that's a good solution, but it may be the 
 only solution.  I haven't got the resources to manage the 1.4 code myself, so 
 I won't step forward as a maintainer if I can't get proper funding. Anyone 
 else out there that has the time and resources to manage the code?

 Feel free to send me mail off list if you have ideas or suggestions on how to 
 solve this - or continue the discussion here.

 Regards,
 /Olle

 PS. Please don't start a discussion about 1.8 quality in this thread, that's 
 a separate issue. I just want to know what you think about closing 1.4 
 support now. If you want to discuss 1.8 quality, start a new thread. Thanks.

Olle

I(me, my opinion, my feelings, my commercial view) am on the side of
dropping support for 1.4 and 1.6.  1.8 had some major issues which are
resolved/being worked on with more energy as older platforms are shut
down. If a large enough security issue showed up, I hope we would all
try to do the right thing and push it back to 1.6 and 1.4. Support
must end sometime. Merging changes across many versions is very
difficult and time consuming.  Asterisk GUI is very limited do to its
1.4 support code.  There are users that still use 1.2 and are very
happy.  They are not looking for new features. I hope the 1.4 / 1.6
users can survive while they test the 1.8 branch and share why or why
not it will fit their needs.

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Re: [asterisk-users] The new ConfBridge application is now in Asterisk Trunk!

[Andrew Latham]

On Mon, Apr 25, 2011 at 11:10 AM, Paul Belanger pabelan...@digium.com wrote:
 On 11-04-25 10:49 AM, David Backeberg wrote:

 On Mon, Apr 25, 2011 at 10:40 AM, C. Savinovich
 c.savinov...@itntelecom.com  wrote:

 Does this ConfBridge requires a hardware timing source?

 No, and neither does MeetMe with modern DAHDI.

 This is the issue the OP was referencing.  MeetMe depends on DAHDI, which is
 not easily installable in at VM or cloud environment. ConfBridge() does not.

 --
 Paul Belanger
 Digium, Inc. | Software Developer
 twitter: pabelanger | IRC: pabelanger (Freenode)
 Check us out at: http://digium.com  http://asterisk.org


Meetme works well under Linux KVM and there is an open task about the
minimum install here: https://issues.asterisk.org/view.php?id=18467

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Re: [asterisk-users] core show channels consise in asterisk 1.8.3

[Andrew Latham]

On Mon, Apr 18, 2011 at 3:06 PM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
 On 11-04-18 02:47 PM, Jerry Geis wrote:
 When I do core show channels concise over the AMI interface
 how do I specify that I want to see the actual channel number like
 DAHDI/4/xxx
 where 4 is the actual channel.

 RIght now I am seeing   DAHDI/i1/x where i1 is the span.

 I could have sworn I saw this issue already reported, but I can't seem to find
 it. Can you test with the latest 1.8 branch to see if it has already been 
 resolved?

 I tried for a few minutes to find the issue on Mantis as I'm almost positive
 that I've seen it filed, but I can't find it.

 Leif.

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Leif, you are correct.  As a lurker I read that ticket, I think it was
about CDR reporting the channel without the g or i.


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Re: [asterisk-users] Asterisk thread limit

[Andrew Latham]

On Wed, Apr 13, 2011 at 10:50 AM, satish patel satish...@hotmail.com wrote:
 Hi Guys!

 I'm middle of testing my asterisk-1.8.3.2 just make sure how much call it
 could handle in production so following is my senario.

 [sipp_client]---[Asterisk][sipp_server]

 sipp_client
 ./sipp -sf uac_pcap.xml -d 10 -i 172.30.254.211 -s 2000 172.30.1.47 -l
 1000 -r 250 -rp 5000 -m 1000

 sipp_server
 ./sipp -sn uas -i 172.30.245.208


 In above if i set -r 250 -rp 5000 calls per sec. in this case my asterisk
 stopped accepting calls at  382 active calls and sipp client through error
 1302704824.872674: Can create thread to send RTP packets. (But asterisk is
 still live to accept calls)
 

 I have ulimit is set to unlimited so just wondering is there any asterisk
 number of thread limitation which we can set to go beyond this boundary?

 -S

Memory limit or load limit might cause this also.

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Re: [asterisk-users] asterisk and fail2ban

[Andrew Latham]

On Wed, Mar 30, 2011 at 9:38 AM, vip killa vipki...@gmail.com wrote:
 so does anyone use fail2ban w/ asterisk or most people use sshguard?

Vip, the overall message is that it takes layers of
settings/configurations to secure an installation.

Simple Guide
1. alwaysauthreject = yes in
http://svn.asterisk.org/svn/asterisk/trunk/configs/sip.conf.sample
2. Static firewall rules
2.1 Drop invalid traffic
2.2 Slow ICMP and TCP Reset attacks
2.3 Disable unneeded services
3. Dynamic firewall rules
3.1 Fail2ban (works ok, but you should test it)
3.2 Portscanning Block
(http://www.newartisans.com/2007/09/neat-tricks-with-iptables.html)
3.3 Other solutions
3.4 Bad Network Lists (http://www.spamhaus.org/drop/)
4. Auditing.   None of the above will work if not audited or reviewed
on a regular basis.
5. Reporting.  With Monthly reporting you can see trends and make good choices.


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Re: [asterisk-users] Debugging not going to log file

[Andrew Latham]

On Tue, Mar 29, 2011 at 11:05 AM, Dean Hoover kb7...@gmail.com wrote:
 I have an Asterisk server running 1.6.2.13, where I can't seem to get
 the increased logging to save to the /var/log/asterisk/messages file.
 I have tried using the standard core set debug 10 and core set
 verbose 10, as well as specifically pointing it to the filename with
 core set debug 10 /var/log/asterisk/messages.  Still, only the most
 serious errors are being reported to the messages log file.

 It seems to work fine with my other Asterisk running 1.4.23.1.  Is
 there something else that I'm missing?

 Dean Hoover
 Waukesha, Wisconsin

I had this happen a month ago, don't feel bad...

In http://svn.asterisk.org/svn/asterisk/branches/1.4/configs/logger.conf.sample
check for debug on the end of the logging method.

;debug = debug
console = notice,warning,error
;console = notice,warning,error,debug  --- Look here
messages = notice,warning,error
;full = notice,warning,error,debug,verbose

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Re: [asterisk-users] asterisk and fail2ban

[Andrew Latham]

On Tue, Mar 29, 2011 at 2:34 PM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
 On 3/29/2011 12:25 PM, Steve Edwards wrote:
 On Tue, 29 Mar 2011 12:10:59 -0500, Sherwood McGowan

 First thing I'd do is restrict the ip blocks your sip endpoints can
 register/call from in sip.conf (or your database's table for sip
 endpoints)

 On Tue, 29 Mar 2011, Gilles wrote:

 Thanks for the idea, but it's not possible, as the Asterisk must be
 accessible for road warriors and receive SIP calls from anyone.

 Really? How many callers are you expecting from North Korea, Libya,
 China, Iran, etc?


 Thanks Steve, you just emailed exactly what I was going to say...

 Remember guys, there's a LOT of IP blocks out there that are almost
 definitely not going to be somewhere you expect to receive SIP traffic
 from.

 Where are you located? Where do your road warriors usually travel? Start
 by blocking countries that are not going to be expected to send traffic
 98% of the time. When I first started out as a consultant, I helped get
 a certain U.S. ITSP up and running, and we reduced fraud and hack
 attempts DRASTICALLY simply by blocking most of the countries that are
 pretty much known for the prolific numbers of hackers. Sure, we had
 like, 2 customers call in to say they had traveled abroad (or sent their
 device to a family/friend abroad) and couldn't get their device to
 register. But seriously, it was rare.

 Either way, just a suggestion

 --
 Sherwood McGowan sherwood.mcgo...@gmail.com
 Carrier, ITSP, Call Center, and PBX Solutions Consultant

First step should be on the AS level.  If you do not have access to
advertised networks then use http://www.spamhaus.org/drop/ The
Spamhaus Don't Route Or Peer List and the script in the FAQ.

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Re: [asterisk-users] asterisk and fail2ban

[Andrew Latham]

On Mon, Mar 28, 2011 at 9:20 AM, vip killa vipki...@gmail.com wrote:
 Is anyone using asterisk with fail2ban? I have it working except it takes
 way more break-in attempts than what is set in maxretry in jail.conf
 For example, I get an email saying:
 The IP 199.204.45.19 has just been banned by Fail2Ban after 181 attempts
 against ASTERISK.
 when maxretry = 5 in jail.conf
 Perhaps someone else is experiencing this or has resolved it, thank you in
 advance for your time.

If you fixed the logging issue discussed here
http://www.fail2ban.org/wiki/index.php/Asterisk then I would assume
your logging has problems.

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Re: [asterisk-users] White papers or success cases to convince a customer?

[Andrew Latham]

On Fri, Mar 25, 2011 at 7:05 PM, Carlos Chavez cur...@telecomabmex.com wrote:
     Can anyone recommend some White Papers or Success Cases that we can use
 to ease the mind of a customer that has not heard much about Asterisk?  All
 they know is Avaya at this point.

 --
 Carlos Chavez
 Director de Tecnología
 Telecomunicaciones Abiertas de México S.A. de C.V.
 Tel: +52-55-91169161 Ext 2001

There is a small list here http://www.digium.com/en/company/casestudies/

I would suggest you watch the Keynote speech by Kevin at the last
Astricon...  I think he mentions some numbers...


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Re: [asterisk-users] One PRI card with 2 (or more) Telcos

[Andrew Latham]

On Fri, Mar 18, 2011 at 3:15 PM, Tiago Geada tiago.ge...@gmail.com wrote:
 Just a follow up with a bit more information

 asterisk*CLI module show like timing
 Module                         Description                              Use
 Count
 res_timing_pthread.so          pthread Timing Interface                 0

 res_timing_dahdi.so            DAHDI Timing Interface                   40

 2 modules loaded
 asterisk*CLI

 --

  [root@asterisk ~]# dahdi_test -c 100

 Opened pseudo dahdi interface, measuring accuracy...

 99.999% 99.999% 99.992% 99.997% 99.998% 99.995% 99.998% 99.996%

 99.997% 99.998% 99.997% 99.994% 99.991% 99.999% 99.998% 99.998%

 99.995% 99.993% 99.998% 99.999% 99.998% 99.995% 99.992% 99.998%

 100.000% 99.998% 99.995% 99.992% 99.999% 99.998% 99.998% 99.999%

 99.995% 99.999% 99.999% 99.998% 99.999% 99.997% 99.999% 99.998%

 99.998% 99.996% 99.992% 99.998% 99.998% 99.999% 99.996% 99.992%

 99.999% 99.998% 99.997% 99.997% 99.997% 99.998% 99.995% 99.994%

 99.995% 99.992% 99.999% 99.993% 99.990% 99.995% 99.993% 99.999%

 99.997% 99.993% 99.999% 99.996% 99.998% 99.996% 99.993% 99.995%

 99.992% 99.998% 99.993% 99.993% 99.999% 99.993% 99.998% 99.996%

 99.993% 99.996% 99.996% 99.994% 99.999% 99.996% 99.996% 99.992%

 99.999% 99.996% 99.991% 99.996% 99.992% 99.998% 99.997% 99.994%

 99.998% 99.995%

 --- Results after 98 passes ---

 Best: 100.000 -- Worst: 99.990 -- Average: 99.996163, Difference: 99.998235

 --

  [root@asterisk ~]# cat
 /sys/devices/system/clocksource/clocksource0/current_clocksource
 tsc
  [root@asterisk ~]# cat
 /sys/devices/system/clocksource/clocksource0/available_clocksource
 tsc hpet acpi_pm jiffies

 On 18 March 2011 17:52, Tiago Geada tiago.ge...@gmail.com wrote:

 Hi list!
 We currently have a PRI gateway composed by a box with two Digium
 quad-span PRI cards (a TE420 and a ).
 One of the cards is filled with TELCO1, while the other has first two
 slots filled with TELCO2, and 3rd slot with TELCO3.
 I am currently having (timer ?) issues on TELCO3 (span 7)
 D-Chan (202 as determined by dahdi_genconf ) is constantly failing causing
 on-going calls to terminate.
 Problem clears immediately tho. I send a copy of the log with pri debug at
 a time of problems...
 Is there a problem having 2 telcos on the same PRI card?
 Would somebody help?

 asterisk*CLI pri show span 7
 Primary D-channel: 202
 Status: Provisioned, Up, Active
 Switchtype: EuroISDN
 Type: CPE
 Overlap Dial: 0
 Logical Channel Mapping: 0
 Timer and counter settings:
   N200: 3
   N202: 3
   K: 7
   T200: 1000
   T202: 1
   T203: 1
   T303: 4000
   T305: 3
   T308: 4000
   T309: 6000
   T313: 4000
   T-HOLD: 4000
   T-RETRIEVE: 4000
   T-RESPONSE: 4000
 Overlap Recv: No

 and

 [Mar 18 17:04:05] VERBOSE[19844] chan_dahdi.c: TEI=0 MDL-ERROR (I): T200
 expired N200 times sending RR/RNR in state 8(Timer recovery)
 [Mar 18 17:04:05] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME
 [Mar 18 17:04:05] VERBOSE[19844] chan_dahdi.c: Changing from state 8(Timer
 recovery) to 5(Awaiting establishment)
 [Mar 18 17:04:06] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME
 [Mar 18 17:04:07] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME
 [Mar 18 17:04:08] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME
 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: TEI=0 MDL-ERROR (G): T200
 expired N200 times sending SABME in state 5(Awaiting establishment)
 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Changing from state
 5(Awaiting establishment) to 4(TEI assigned)
 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: TEI=0 DL event:
 Q931_DL_EVENT_DL_RELEASE_IND(3)
 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Start T309 for call cref=56
 on channel 2
 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Start T309 for call cref=64
 on channel 3
 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Start T309 for call cref=58
 on channel 4
 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Start T309 for call cref=66
 on channel 6
 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME
 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Changing from state 4(TEI
 assigned) to 5(Awaiting establishment)
 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c:   == Primary D-Channel on
 span 7 down
 [Mar 18 17:04:09] WARNING[19844] chan_dahdi.c: No D-channels available!
  Using Primary channel 202 as D-channel anyway!
 [Mar 18 17:04:10] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME
 [Mar 18 17:04:11] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME
 [Mar 18 17:04:12] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME
 [Mar 18 17:04:13] VERBOSE[19844] chan_dahdi.c: TEI=0 MDL-ERROR (G): T200
 expired N200 times sending SABME in state 5(Awaiting establishment)
 [Mar 18 17:04:13] VERBOSE[19844] chan_dahdi.c: Changing from state
 5(Awaiting establishment) to 4(TEI assigned)
 [Mar 18 17:04:13] VERBOSE[19844] chan_dahdi.c: TEI=0 DL event:
 Q931_DL_EVENT_DL_RELEASE_IND(3)
 [Mar 18 17:04:13] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending 

Re: [asterisk-users] One PRI card with 2 (or more) Telcos

[Andrew Latham]

On Fri, Mar 18, 2011 at 3:17 PM, Adrian Serafini
adrian-li...@wombit.com wrote:

 Is there a problem having 2 telcos on the same PRI card?

 I think you go with one master timer as the Telco.  Then the other spans are
 secondary, tertiary, quaternary timers.

 Adrian


Adrian

This only works when all the providers are using a common clock like
some areas in the USA.  This is not the case all around the world.

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Re: [asterisk-users] Extract Remote-Party-ID from incoming INVITE indialplan

[Andrew Latham]

On Wed, Mar 16, 2011 at 10:50 AM, Jonas Kellens
jonas.kell...@telenet.be wrote:
 is it possible to extract the Remote-Party-ID from an incoming call in the
 dialplan ? Is there some kind of function for this ?

 Kind regards,
 Jonas.

1.8 Documentation on Connected Line update.  Works like magic.
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information

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Re: [asterisk-users] Is asterisk 1.8 stable version to upgrade from asterisk 1.6 on live server?

[Andrew Latham]

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

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Re: [asterisk-users] doorphone?

[Andrew Latham]

On Wed, Mar 9, 2011 at 12:53 PM, Darrick Hartman (lists)
dhart...@djhsolutions.com wrote:
 On 03/09/2011 02:57 AM, Dan Journo wrote:

   could anybody suggest a usable doorphone and magnetic door opener

   hardphone system for me, please? Of course should be connectable to

   asterisk. I am in the EU, should be available here.

 I would recommend using a normal doorphone, and connecting it to a SIP
 gateway like the PAP2T.

 Otherwise, you need a network connection directly into the doorphone
 unit, and some people don't like that because it can give a
 hacker/burglar, direct access to your internal network.

 Hope that helps.

 Dan Journo

 That's not always true.  Some door phones have a remote unit that connects
 to the network and a local device at the door, giving some better security.

 I've used the Valcom VIP-172 phones.  They are simple and work well. Very
 good support if you need to call them.

 http://www.valcom.com/Home_links/sipdoorintercom.htm

 Darrick
 --
 Darrick Hartman
 DJH Solutions, LLC
 http://www.djhsolutions.com

To repeat and support Darrick's point.  Using a doorphone that is
analog and or coax for the last 3+ meters will save some headaches
down the road.  I have used Valcom, Viking and others.  With a Xorcom
appliance you can also have the contact closure I/O to open doors or
ring phones.

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Re: [asterisk-users] server performance....

[Andrew Latham]

On Fri, Mar 4, 2011 at 2:13 PM, viswavardhanreddy karna
viswavardhanre...@gmail.com wrote:
 Hi every one,
  I am doing some experiments on asterisk server
 performance.. How can we know server performance? can any one explain me
 plz
  I have 2 doubts regarding the asterisk server performance...

 1. When can we know asterisk server performance?
     1. when server is in idle state ?
     2. when the server is in busy state?

 can any one please tell me when can the server performance is known i mean
 when server is busy or in idle state?

 Best Regards,
 viswavardhanreddy


Many people test their servers with call-setups and call tear-downs.
Using another tool like sipp you can send 100-1000s of call-setups and
then do call tear-downs.  You can also use transcoding loops to test
the load.  If you have 1 call that is sent to a context where it dials
exten+1 and continues the loop until a target number, you can then set
the codec for each dialed number.  I know that there are many methods
of testing and this is just a common one.

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Re: [asterisk-users] Recieve_Fax caused crash 1.8.2.3

[Andrew Latham]

On Fri, Feb 25, 2011 at 9:49 AM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
 On 11-02-24 08:56 PM, Andrew Latham wrote:

 And I go back to triple check and compare revision numbers...  You are
 100% correct, the revision numbers in our local repository are wrong,
 someone pushed the 1.8.3 RC3 into our local 1.8.2 branch.  I apologize
 and will work to better control my trust of other engineers as this is
 twice in one week I have looked like an ass.  International bandwidth
 limits change how you work and as a business force the mirroring of as
 many sources as possible.

 No worries, you had my heart going there pretty good for a moment!

 Leif!

Me, too, when our repository did not match up, my heart skipped a
beat. Time for fresh checkouts.

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Re: [asterisk-users] Recieve_Fax caused crash 1.8.2.3

[Andrew Latham]

On Thu, Feb 24, 2011 at 6:05 PM, Bryant Zimmerman brya...@zktech.com wrote:
 I had an issue today where receive_fax caused an asterisk switch to crash.
 The switch is 1.8.2.3 version. The call was coming from a fax machine. The
 call started receive_fax answered and then asterisk stopped responding. I
 was able to log into asterisk but it would not do a core restart now nor
 would it take any calls or show an peer registrations.
 I had to kill the asterisk process and restart it.  As best we can tell
 there was no attempt by the sender to intentionally send any malformed
 packets that should have caused this. I see there is a security patch
 1.8.2.4 that lists some RTP security issues. is it possible that this fix
 may address what I ran into as well?

 Thanks

 zktech

There are many updates in 1.8.2.4 that may fix your issue.  If you are
running any version of 1.8 it should be a quick update.

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Re: [asterisk-users] Recieve_Fax caused crash 1.8.2.3

[Andrew Latham]

On Thu, Feb 24, 2011 at 7:43 PM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
 On 11-02-24 04:08 PM, Andrew Latham wrote:

 There are many updates in 1.8.2.4 that may fix your issue.  If you are
 running any version of 1.8 it should be a quick update.

 I wouldn't say many. There is one fix in 1.8.2.4 over 1.8.2.3.

 From the ChangeLog:

        * Asterisk 1.8.2.4 Released.

        * AST-2011-002: Multiple array overflow and crash vulnerabilities in
          UDPTL code


 The release announcement for AST-2011-002 is here:
 http://downloads.asterisk.org/pub/security/AST-2011-002.pdf

 Leif.

And I go back to triple check and compare revision numbers...  You are
100% correct, the revision numbers in our local repository are wrong,
someone pushed the 1.8.3 RC3 into our local 1.8.2 branch.  I apologize
and will work to better control my trust of other engineers as this is
twice in one week I have looked like an ass.  International bandwidth
limits change how you work and as a business force the mirroring of as
many sources as possible.

~~~ Andrew lathama Latham lath...@gmail.com ~~~

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Re: [asterisk-users] Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available

[Andrew Latham]

On Tue, Feb 22, 2011 at 12:16 PM, Ishfaq Malik i...@pack-net.co.uk wrote:
 Has this issue been fixed in this release of 1.8 (or even in the
 previous 1.8.2.3)?

 https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403

 Thanks

 Ish


Ishfaq, I spoke to soon and was looking at the wrong checkout.  The
1.8.2.4 does NOT have the patch from issue 18403.  Asterisk Branch
1.8.3 does have the patch which happened just 1 day after the 1.8.2.4
release. I must have lost the release email because I can only find
the tag in SVN.  I was confused and hope I did not cause you any
confusion.

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Re: [asterisk-users] Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available

[Andrew Latham]

On Tue, Feb 22, 2011 at 12:16 PM, Ishfaq Malik i...@pack-net.co.uk wrote:
 Has this issue been fixed in this release of 1.8 (or even in the
 previous 1.8.2.3)?

 https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403

 Thanks

 Ish

 snip 

 --
 Ishfaq Malik
 Software Developer
 PackNet Ltd

 Office:   0161 660 3062

Yes, you can take the two minutes to search for Must release lock in
 http://svn.asterisk.org/svn/asterisk/tags/1.8.2.4/channels/chan_sip.c

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Re: [asterisk-users] Cisco 7945G phone with asterisk

[Andrew Latham]

On Wed, Feb 16, 2011 at 7:32 AM, ast guy ast...@gmail.com wrote:
 Hi,

  Anyone who has deployed Cisco 7945G phone with asterisk, kindly share your
 experience.

 /ag


I did some 7941's a few months ago with SIP.  They work pretty well.
Make a console cable for the AUX port and you can see them load.  I
had to add Spanish menus to them so I ended up hacking the load
process (Cisco does not support language files with SIP firmware). The
7945 should just have more features.

~~~ Andrew lathama Latham lath...@gmail.com ~~~

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Re: [asterisk-users] Connect Asterisk to a cell phone

[Andrew Latham]

On Wed, Feb 16, 2011 at 2:49 AM, logan logan...@gmail.com wrote:
 Hello,

 Are there any gateways which allow me to hook a cellphone to Asterisk and
 use that line for routing my calls? Basically, I'm looking to play around a
 bit and if I can get to connect a cellphone with Asterisk then that would be
 great.

 Thanks,
 Hitesh
 PS: I have tried to search on the web, but didn't find any pointers on how
 to do so.


There are several pages of information here:
https://wiki.asterisk.org/wiki/display/AST/Mobile+Channel

~~~ Andrew lathama Latham lath...@gmail.com ~~~

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