Re: [asterisk-users] Load testing SIP registration attempts

[Ben Ford]

>
> Would you use SIPp ?


Definitely.

 Any example scenario ?


The testsuite is a great place to look for example SIPp scenarios. Check
out tests/channels/pjsip/registration - there are quite a few different
scenarios in there to pick from, and you can tailor them to whatever you
need.

On Tue, Nov 3, 2020 at 9:20 AM Olivier  wrote:

> Hello,
>
> How would you test how a PJSIP-powered Asterisk 13 instance resist to
> hostile REGISTRATION attempts ?
>
> Would you use SIPp ? Any example scenario ?
> Would you go with an alternative tool ? Which one would you pick ?
>
> Best regards
> --
> _
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>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
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>
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-- 
Benjamin Ford
Software Engineer
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Check us out at www.sangoma.com and www.asterisk.org
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Re: [asterisk-users] Asterisk versions?

[Ben Ford]

Hey Dave,

In the case of 13 and 16, these are LTS versions which means that they get
long term service. 17 is a standard release. The benefit of an LTS is that
you can expect it to get bug fixes and improvements for an extended period
of time without anything major being changed. If you find an LTS version
that has everything you need, it's probably the safest version to choose.
Any noteworthy changes between releases in a version (16.8 to 16.9, for
example) will be documented in the CHANGES and UPGRADE.txt files. If you're
worried about configuration changes or syntax, this is the place to look.

 Of course, as a developer I love to see people on the most up to date
version, but it's all user preference :)

I would recommend checking out the wiki for more details on the version you
are looking at.

On Mon, May 11, 2020, 5:04 PM Dave Woodfall  wrote:

> Hi all,
>
> I'm a fairly long time user of Asterisk, but I'm new to this list.  I
> used to use the old forums some few years ago.
>
> I wanted to ask why there are different Asterisk versions, as shown
> by the announcements in the past week or 2:
>
> Asterisk 13.33.0
> Asterisk 16.10.0
> Asterisk 17.4.0
>
> I'm currently using 16.8.0 and wondering if I should upgrade to
> 16.10.0, or perhaps give 17.4.0 a try.
>
> Are there any differences I should be aware of, like config file
> syntax or similar things?
>
> Thanks,
>
> --
> Dave
>
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>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
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>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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Re: [asterisk-users] What does "all 3 app_voicemail variants can now be built" implies exactly ?

[Ben Ford]

>
> From Astricon 2019 notes [1], you can read "[a]ll 3 app_voicemail variants
> can now be built".
> What does it mean ?


At compilation, you can specify which voicemail modules you would like to
build. You can select all 3 modules if you want.

Is this change tied with a specific Asterisk version ?


 This should be in 17.

Is possible to change from ODBC to IMAP without re-compilation ?
> Is it also possible to mix mailbox types on a single system ?


You can find more information in this blog post[1]. Only one module can be
loaded at a time.

[1]:
https://blogs.asterisk.org/2019/11/20/announcing-a-new-compile-option-for-app_voicemail-storage/

On Tue, Jan 21, 2020 at 10:20 AM Olivier  wrote:

> Hello,
>
> From Astricon 2019 notes [1], you can read "[a]ll 3 app_voicemail
> variants can now be built".
> What does it mean ?
>
> Is this change tied with a specific Asterisk version ?
> Is possible to change from ODBC to IMAP without re-compilation ?
> Is it also possible to mix mailbox types on a single system ?
>
> [1] https://wiki.asterisk.org/wiki/display/AST/Agenda+2019
>
> Best regards
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
Benjamin Ford
Software Engineer
256-428-6147
Check us out at www.sangoma.com and www.asterisk.org
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Re: [asterisk-users] multiple softphone clients and same/different account credentials

[Ben Ford]

I'm no expert on the user side of things, but I would prefer option A. Of
course, this is completely your preference. Asterisk will allow either
option, so you have some flexibility there. One of the advantages of option
A is that you can have multiple devices (like you mentioned) that can all
be rung at once simply if the user has a desk phone, mobile work phone, etc.

On Mon, Nov 25, 2019 at 6:44 PM Greg Troxel  wrote:

> (I'm new to Asterisk, after having started VOIP with vat on the mbone in
> the 90s.)
>
> I am setting up my first Asterisk system, and trying to read
> docs/guidance and follow best practices.  I have read the 5th Edition of
> "Asterisk: The Definitive Guide" and like the 3rd Edition on the web it
> recommends that hardphones and softphones both have a unique name
> distinct from any concept of extension.
>
>
> http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeviceConfig_id283201.html
>
> Then the 5th edition goes on to give an example with a hardphone and a
> softphone associated with one individual, where the hardphone is named
> by MAC address and the softphone by JIM_VANM_SOFT (p. 61).
>
> Despite talking about separating extensions, phone names, and people, it
> seems clear that a softphone is usually personal to a person (unless
> it's a desk phone via a computer, but I'm talking about the personal
> type).
>
> THe book does not address the notion that a user might be given
> credentials and then configure them on a number of softphone-type
> devices simultaneously, e.g. a smartphone, a tablet, and two laptops.
> When getting service from an ITSP, it seems there are credentials and
> they don't want to know the details of how many softphones you are
> using.
>
> So which option is preferred?
>
>   A) Have a softphone aor/auth_user/password for a particular human, and
>   expect them to configure it on multiple devices.  Do not worry that 1)
>   multiple are registered at once (because that's normal in SIP) and 2)
>   asterisk has no idea which is which (because the intent is to place a
>   call to that person)
>
>   B) issue credentials per device and keep them all separate.  Use
>   extensions.conf to ring them all
>
> Having written the question out carefully, it seems obvious that A is
> the way to do this, but it's sort of contrary to the advice in the book
> so I thought I would ask.
>
> Thanks,
> Greg
>
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> _
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>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
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>
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-- 
Benjamin Ford
Software Engineer
256-428-6147
Check us out at www.sangoma.com
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[asterisk-users] Queues, Penalties, RINGALL, and Use Cases

[Ben Ford]

Hey everyone,

I posted a topic on the community forum which can be found here:
https://community.asterisk.org/t/request-for-feedback-queues-penalties-ringall-and-use-cases/80960

It covers a scenario encountered while debugging an issue, and I would love
some feedback on it to help make a decision on how to fix it moving forward.

Thanks!

-- 
*Benjamin Ford*
Digium - A Sangoma Company | Software Engineer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US

Check us out at: https://digium.com · https://sangoma.com
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Re: [asterisk-users] How does verbosity work?

[Ben Ford]

I just saw I forgot to link the wiki page:
https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace

Hope this helps.

On Mon, Sep 16, 2019 at 10:13 AM Dan Cropp  wrote:

> Thank you Ben
>
>
>
> *From:* asterisk-users  *On
> Behalf Of *Ben Ford
> *Sent:* Monday, September 16, 2019 9:58 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> *Subject:* Re: [asterisk-users] How does verbosity work?
>
>
>
> Is there a good way to diagnose what asterisk is doing for each thread id?
>
>
>
> If you are able to get a backtrace during the time the issue is occurring,
> that could provide some insight. You can get a backtrace while Asterisk is
> running by following the steps on [1].
>
>
>
> Is there a verbose level that would be recommended to track the threading?
> Also, does the verbose help if it may be confbridge related?
>
>
>
> Verbose is a very chatty debug that spits out information about what's
> going on. You could try with verbose 3, and if that doesn't provide the
> information you're looking for, you can dynamically bump it up at runtime.
> If you are looking for something specific, you can always do a search in
> the confbridge files for the verbose message and set the verbose level to
> whatever number that message specifies.
>
>
>
> On Mon, Sep 16, 2019 at 9:44 AM Dan Cropp  wrote:
>
> I’m trying to track down a CPU spike we are seeing in a system.
>
>
>
> We have tracked down the spike to a single CPU and TID using that CPU.
> Indications are that it’s asterisk running this TID.
>
>
>
> I’m trying to figure out what asterisk is doing on this thread around that
> time, but haven’t been able to match the tid to anything I’m seeing in
> asterisk debugging.
>
> Is there a good way to diagnose what asterisk is doing for each thread id?
>
>
>
> We enabled the debugging in the logger.conf.  Is there a verbose level
> that would be recommended to track the threading?
>
> Also, does the verbose help if it may be confbridge related?
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> --
>
> *Benjamin Ford*
>
> Digium - A Sangoma Company | Software Engineer
>
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> <https://maps.google.com/?q=445+Jan+Davis+Drive+NW+-+Huntsville,+AL+35806+-+US=gmail=g>
>
> Check us out at: https://digium.com · https://sangoma.com
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
*Benjamin Ford*
Digium - A Sangoma Company | Software Engineer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
<https://maps.google.com/?q=445+Jan+Davis+Drive+NW+-+Huntsville,+AL+35806+-+US=gmail=g>
Check us out at: https://digium.com · https://sangoma.com
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Re: [asterisk-users] How does verbosity work?

[Ben Ford]

>
> Is there a good way to diagnose what asterisk is doing for each thread id?


If you are able to get a backtrace during the time the issue is occurring,
that could provide some insight. You can get a backtrace while Asterisk is
running by following the steps on [1].

Is there a verbose level that would be recommended to track the threading?
> Also, does the verbose help if it may be confbridge related?


Verbose is a very chatty debug that spits out information about what's
going on. You could try with verbose 3, and if that doesn't provide the
information you're looking for, you can dynamically bump it up at runtime.
If you are looking for something specific, you can always do a search in
the confbridge files for the verbose message and set the verbose level to
whatever number that message specifies.


On Mon, Sep 16, 2019 at 9:44 AM Dan Cropp  wrote:

> I’m trying to track down a CPU spike we are seeing in a system.
>
>
>
> We have tracked down the spike to a single CPU and TID using that CPU.
> Indications are that it’s asterisk running this TID.
>
>
>
> I’m trying to figure out what asterisk is doing on this thread around that
> time, but haven’t been able to match the tid to anything I’m seeing in
> asterisk debugging.
>
> Is there a good way to diagnose what asterisk is doing for each thread id?
>
>
>
> We enabled the debugging in the logger.conf.  Is there a verbose level
> that would be recommended to track the threading?
>
> Also, does the verbose help if it may be confbridge related?
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
*Benjamin Ford*
Digium - A Sangoma Company | Software Engineer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US

Check us out at: https://digium.com · https://sangoma.com
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Re: [asterisk-users] CyberMegaPhone WebRTC Video Conference demo

[Ben Ford]

Forgot to link the wiki page. Sorry about that!

[1]:
https://wiki.asterisk.org/wiki/display/AST/Installing+and+Configuring+CyberMegaPhone

On Fri, Jan 4, 2019 at 3:20 PM Ben Ford  wrote:

> Hey there,
>
> I'm guessing you're following the tutorial [1] on how to set up Cyber Mega
> Phone (correct me if I'm wrong). I would double check to make sure all
> configuration matches up with what the wiki provides. Feel free to share
> configuration here too.
>
> You could try substituting the domain with "localhost" to see if that
> produces a different result. You may also need to navigate to "
> https://localhost:8089/ws;, or substitute 'localhost' with your IP.
> Whatever it is you decide to use, make sure it's consistent between all
> instances.
>
> On Fri, Jan 4, 2019 at 11:24 AM Dan Cropp  wrote:
>
>> I am trying to run the CyberMegaPhone demo to see the WebRTC Video
>> Conference demonstration from AstriDevCon 2017
>>
>>
>>
>> I have been able to make WebRTC work on this same box with SIPML5 demo
>> but not the CMP2K.
>>
>>
>>
>> When I attempt to access the https://myip:8089/cmp2k I am prompted for
>> the unsecure web.  I enable unsecure web.  (Using the asterisk local
>> certificate generation from the SIPML5 demo).
>>
>>
>>
>> After that, I’m only seeing “Access Denied” web page.
>>
>> “You do not have permission to access the requested URL.
>>
>> ….
>>
>> Asterisk/16.1.1
>>
>>
>>
>>
>>
>> I do not get to the Welcome to Cyber Mega Phone 2K Ultimate Dynamic
>> Edition page with the Account, Connect, Call buttons.
>>
>>
>>
>>
>>
>> When I look at the CLI
>>
>> CLI> http show status
>>
>> HTTP Server Status:
>>
>> Prefix:
>>
>> Server: Asterisk/16.1.1
>>
>> Server Enabled and Bound to 0.0.0.0:8088
>>
>>
>>
>> HTTPS Server Enabled and Bound to 0.0.0.0:8089
>>
>>
>>
>> Enabled URI's:
>>
>> /httpstatus => Asterisk HTTP General Status
>>
>> /static/... => Asterisk HTTP Static Delivery
>>
>> /ws => Asterisk HTTP WebSocket
>>
>>
>>
>> Enabled Redirects:
>>
>>   /cmp2k => /static/cyber_mega_phone_2k/index.html
>>
>>
>>
>> Any suggestions on what I am doing wrong?
>>
>>
>>
>> Dan
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> *Benjamin Ford*
> Digium - A Sangoma Company | Software Engineer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> <https://maps.google.com/?q=445+Jan+Davis+Drive+NW+-+Huntsville,+AL+35806+-+US=gmail=g>
> Check us out at: https://digium.com · https://sangoma.com
>
>
>

-- 
*Benjamin Ford*
Digium - A Sangoma Company | Software Engineer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
<https://maps.google.com/?q=445+Jan+Davis+Drive+NW+-+Huntsville,+AL+35806+-+US=gmail=g>
Check us out at: https://digium.com · https://sangoma.com
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  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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Re: [asterisk-users] CyberMegaPhone WebRTC Video Conference demo

[Ben Ford]

Hey there,

I'm guessing you're following the tutorial [1] on how to set up Cyber Mega
Phone (correct me if I'm wrong). I would double check to make sure all
configuration matches up with what the wiki provides. Feel free to share
configuration here too.

You could try substituting the domain with "localhost" to see if that
produces a different result. You may also need to navigate to "
https://localhost:8089/ws;, or substitute 'localhost' with your IP.
Whatever it is you decide to use, make sure it's consistent between all
instances.

On Fri, Jan 4, 2019 at 11:24 AM Dan Cropp  wrote:

> I am trying to run the CyberMegaPhone demo to see the WebRTC Video
> Conference demonstration from AstriDevCon 2017
>
>
>
> I have been able to make WebRTC work on this same box with SIPML5 demo but
> not the CMP2K.
>
>
>
> When I attempt to access the https://myip:8089/cmp2k I am prompted for
> the unsecure web.  I enable unsecure web.  (Using the asterisk local
> certificate generation from the SIPML5 demo).
>
>
>
> After that, I’m only seeing “Access Denied” web page.
>
> “You do not have permission to access the requested URL.
>
> ….
>
> Asterisk/16.1.1
>
>
>
>
>
> I do not get to the Welcome to Cyber Mega Phone 2K Ultimate Dynamic
> Edition page with the Account, Connect, Call buttons.
>
>
>
>
>
> When I look at the CLI
>
> CLI> http show status
>
> HTTP Server Status:
>
> Prefix:
>
> Server: Asterisk/16.1.1
>
> Server Enabled and Bound to 0.0.0.0:8088
>
>
>
> HTTPS Server Enabled and Bound to 0.0.0.0:8089
>
>
>
> Enabled URI's:
>
> /httpstatus => Asterisk HTTP General Status
>
> /static/... => Asterisk HTTP Static Delivery
>
> /ws => Asterisk HTTP WebSocket
>
>
>
> Enabled Redirects:
>
>   /cmp2k => /static/cyber_mega_phone_2k/index.html
>
>
>
> Any suggestions on what I am doing wrong?
>
>
>
> Dan
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
*Benjamin Ford*
Digium - A Sangoma Company | Software Engineer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US

Check us out at: https://digium.com · https://sangoma.com
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Re: [asterisk-users] Database re-connect issue

[Ben Ford]

Looks like there were some changes that went into 1.8 that were supposed to
fix the problem, and the issue was closed because the version the reporter
was on hit EOL. If you think this is a bug within Asterisk, please file an
issue with the appropriate information so we can triage it!

As a side note, make sure you are on version 13 or 15. Any other version
will be autoclosed since bug fixes are not done due to EOL.

Hope this helps!

On Wed, Mar 7, 2018 at 7:29 AM, D'Arcy Cain  wrote:

> I had a problem inserting CDR records into my PostgreSQL database.
> According to the log it failed to open the database at startup.  I
> searched and found the following report.
>
> https://issues.asterisk.org/jira/browse/ASTERISK-15820
>
> However, it looks like it was closed after 14 months and nothing was
> done about the problem.  Is there any point to opening another report?
> How do we make sure that the issue doesn't get forgotten?
>
> --
> D'Arcy J.M. Cain
> Vybe Networks Inc.
> http://www.VybeNetworks.com/
> IM:da...@vex.net VoIP: sip:da...@vybenetworks.com
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
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[asterisk-users] AMI "Queues" Action

[Ben Ford]

Currently the AMI "Queues" action outputs the same text that the CLI
outputs when running a "queue show" command, which does not conform with
the AMI spec. It should follow the same format as other AMI actions,
structured in a key value list. The "QueueStatus" action outputs the
information that "Queues" should output, so the current plan is to remove
the "Queues" action entirely. Is anyone opposed to this?
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Re: [asterisk-users] [asterisk-dev] AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben Klang)

[Ben Klang]

Il giorno Oct 30, 2014, alle ore 4:57 PM, Paul Albrecht palbre...@glccom.com 
ha scritto:
 
 On Oct 29, 2014, at 2:45 PM, Ben Klang bkl...@mojolingo.com 
 mailto:bkl...@mojolingo.com wrote:
 
 
 On 10/28/2014 06:03 PM, Ben Langfeld wrote:
 On 28 October 2014 19:47, Derek Andrew derek.and...@usask.ca 
 mailto:derek.and...@usask.ca wrote:
 What is the alternative to the dial plan? Is everyone talking about 
 getting rid of the statements like:
 exten = s,1,
 
 what is the alternative? 
 
 Remote applications based on APIs like ARI. This is the start of the 
 discussion, and please remember that nothing has been decided or even 
 presented as a robust plan yet. This is brain-storming.
 
 Additionally, note that the original proposal was to deprecate AMI/AGI in 
 favour of ARI once it is feature complete with those protocols; an 
 entirely lesser change than the removal of the dialplan in its entirety.
  
 
 Since this thread has my name on it, I guess it’s past time that I explain 
 my motivation for making the suggestion, and try to restore some of the 
 context that was present in the discussion at AstriDevCon.
 
 Before I jump into the details of my proposal, I’d like to clarify terms...
 
 
 It’s intellectually dishonest to redefine the terms of an argument to 
 presuppose your own conclusion. If you don’t intend to use the term 
 “deprecate” as it is commonly understood by software developers and users 
 than you should avoid the use of the term “deprecate” so that others clearly 
 understand your argument. If you really mean “deprecate” as commonly 
 understood by software developers and users then you should be prepared to 
 defend that proposition.

I had thought that the term “deprecate” was already understood to be the 
definition I gave, but earlier posts on the mailing list seemed to indicate 
confusion. My definition mirrors the Wikipedia definition: 
https://en.wikipedia.org/wiki/Deprecation 
https://en.wikipedia.org/wiki/Deprecation.  Perhaps I just should have linked 
to that originally, as their explanation is even better than my own.

In any event, what we are talking about is the deprecation as I defined it. If 
you prefer another word for it, I’m fine with that too.  I just want to be 
clear that my proposal is to discourage use of AMI/AGI in new projects, but not 
to immediately remove it.

  
 Now, on to what I originally proposed...
 
 
 It’s clear from the title of the agenda item what was proposed. You proposed 
 deprecating AMI/AGI and that entails deprecating the dial plan. The fact that 
 deprecating the dial plan is now on the table is a direct consequence of your 
 proposal. This is reflected in both comments made at AstiCon and Matt’s 
 summary of  Astricon on the development list. You can’t have it both ways. 
 You want to deprecate dial plan or not. Which is it? 

Actually, AMI/AGI and Dialplan are separate.  You can disable AMI and you can 
unload res_agi.so. Dialplan/extensions.conf continue to work just fine.  
Certainly AMI/AGI make use of Dialplan, but deprecating AMI/AGI doesn’t mean 
you have to deprecate Dialplan.

 
 It is my opinion that while AGI and AMI are probably individually fixable, 
 doing so would cause backward-incompatible changes…
 
 Deprecating the dial plan and AGI/AMI is incompatible going forward. What is 
 supposed to happen? Are users supposed to throw away there applications 
 whenever ARI/Stasis is feature complete? Is ARI/Stasis really any easier to 
 use than the dial plan? Are we all supposed to use Adhearsion? 
 

You’re certainly welcome to use Adhearsion :) For what it’s worth, Adhearsion 
will continue to support AMI/AGI because we have to until ARI is 
feature-complete.  For Adhearsion users, the transition to ARI should be 
seamless because that’s one of the things that the framework promises: to paper 
over the idiosyncrasies of the underlying protocols.

If you don’t want to use Adhearsion, I’d recommend you look at ARI for 
developing new projects.  There are libraries in many languages that make it 
easy to use. It’s got a great start and will only improve as people continue to 
use it and develop additional features.  Today, it is not yet a replacement for 
AMI/AGI, but I’m very optimistic that it will be in the near future.

I suspect that I’m not convincing to you, and that you want to continue using 
AMI/AGI. That’s fine, I’m not telling you to throw out any code.  I think 
Asterisk’s historical policy toward backward compatibility and removing 
features speaks for itself.  Rather than continue to debate the semantics of my 
proposal, I’d like to continue the discussion on how we can improve ARI and 
improve the state of the world for all Asterisk developers in the years to come.

/BAK/
-- 
Ben Klang
Principal/Technology Strategist, Mojo Lingo
bkl...@mojolingo.com mailto:bkl...@mojolingo.com
+1.404.475.4841

Mojo Lingo -- Voice applications that work like magic
http://mojolingo.com http://mojolingo.com/
Twitter: @MojoLingo

[asterisk-users] Moderated News Aggregation for Asterisk

[Ben Merrills]

Hi all. 

Just wanted to let people know about a small project I started over the weekend 
to help me keep up with news about Asterisk. http://asterisktimes.xdev.net/

Some of the other new sites are either not there anymore or slow to update, so 
I've come up with a different idea for keeping Asterisk news up-to date and in 
one place. For the moment, I call it Asterisk Times.

OK, so maybe not the best name, but it's a work in progress.

So what is it? Well, this is an attempt to create a moderated, aggregated news 
platform for Asterisk. We want developers, 3rd party companies, open source 
tools, in fact anyone who does anything noteworthy with Asterisk to tell us. 
And the best way to do it, is by letting us have an RSS feed into your own 
announcements or news.

With that, we can then review and submit news to the aggregator on your behalf, 
which then shows up on the homepage of this website.

I hope people see value in this, as I know I do. This isn't run for profit or 
commercial reasons, it's just because I think as a community we deserve a 
better, more frequently updated news site.

The URL will change (or at least get its own dedicated URL) once the project is 
off the ground and I can see people getting value from it.

Any suggestions or feedback welcome.

Thanks,

Ben (aka skrusty on irc)

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[asterisk-users] Index of AGI Scripts

[Ben Merrills]

Hi, 

I wasn't sure quite where to inform people about this, but often I find it hard 
to find good AGI scripts written by the community, and voip-info is so often 
out of date. So I create a simple website for people to list their own, or 
freely available AGI scripts all in one place. 

http://www.theagigallery.co.uk

Any feedback would be welcomed at this time, improvements, issues, general 
comments etc as I am sure there will be plenty.

I've added some that I use and that I've found online already, but I would ask 
that others add any they know of too, and help build up a nice big free 
database of AGI scripts for the general Asterisk community.

Thanks, Ben


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[asterisk-users] Introducing Sippy Cup: SIPp Load Testing Made Easy

[Ben Klang]

Hello everyone,

Recently we've been focusing quite heavily on making Adhearsion[0] faster.  To 
do that, we needed a convenient way to test our Asterisk voice apps.  The 
obvious tool in the Open Source world is SIPp[1]. SIPp is great! Though it's a 
little clumsy to use sometimes, especially if you're trying to use it to drive 
interactive calls like an IVR.

So to make our own lives easier, we created Sippy Cup.  I wanted to announce it 
here in the hopes that it makes your lives easier as well.

Sippy Cup is an Open Source (MIT license) piece of software that allows you to 
define an entire SIPp load test profile in a single, simple YAML format.  This 
includes not only the test steps, but also the load generation parameters such 
as calls per second, maximum concurrent calls, and the total number of calls to 
place.

But what's REALLY useful is Sippy Cup's ability to dynamically generate PCAP 
audio.  If you've ever needed to drive an IVR from SIPp you're probably 
familiar with the pains - it usually requires capturing an actual call, 
isolating the RTP, and then giving it to SIPp to play back.  Sippy Cup makes 
that easier by actually generating uLaw silence interspersed with appropriately 
timed RFC4733 DTMF.  That alone has saved us tremendous time when tweaking our 
load test scenarios.

Blog announcement of the project:
https://mojolingo.com/blog/2013/introducing-sippy-cup-sipp-load-testing-made-easy/

Github sources:
https://github.com/bklang/sippy_cup

Enjoy!

/BAK/

[0]: http://adhearsion.com
[1]: http://sipp.sourceforge.net
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Ben Klang
Principal/Technology Strategist, Mojo Lingo
bkl...@mojolingo.com
+1.404.475.4841

Mojo Lingo -- Voice applications that work like magic
http://mojolingo.com
Twitter: @MojoLingo



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Re: [asterisk-users] Can't make Asterisk send authentication to remote peer on INVITE [SOLVED]

[Ben WIlliams]

Problem solved - It was missing username setting, I had assumed
fromuser would be used for authentication.

On Sat, Apr 14, 2012 at 9:30 PM, Ben WIlliams
bwilliams+aster...@jadeworld.com wrote:
 This is a really simple problem that I just can't get to work. There
 are two Asterisk servers with the following sip user and peer. When a
 call is attempted, Asterisk is not sending authentication details in
 response to the 401. Note, if the secret is blank on 172.16.0.2 test,
 the INVITE succeeds.

 on 172.16.0.2:

 [test]
 type=friend
 secret=abcde
 host=dynamic
 context=demo

 on 172.16.0.1 :

 [natty]
 type=peer
 host=172.16.0.2
 fromuser=test
 secret=abcde

 originate SIP/natty/1234568 extension 200
  == Using SIP RTP CoS mark 5
 Audio is at 172.16.0.1 port 19486
 Adding codec 0x2 (gsm) to SDP
 Adding codec 0x4 (ulaw) to SDP
 Adding codec 0x8 (alaw) to SDP
 Adding non-codec 0x1 (telephone-event) to SDP
 Reliably Transmitting (no NAT) to 172.16.0.2:5060:
 INVITE sip:1234568@172.16.0.2 SIP/2.0
 Via: SIP/2.0/UDP 172.16.0.1:5066;branch=z9hG4bK59f50e30;rport
 Max-Forwards: 70
 From: asterisk sip:test@172.16.0.1:5066;tag=as1689b2b6
 To: sip:1234568@172.16.0.2
 Contact: sip:test@172.16.0.1:5066
 Call-ID: 2353cf0e59596e285c684b44220f8915@172.16.0.1
 CSeq: 102 INVITE
 User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2.1
 Date: Sat, 14 Apr 2012 09:10:38 GMT
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
 Supported: replaces, timer
 Content-Type: application/sdp
 Content-Length: 290

 v=0
 o=root 1594270426 1594270426 IN IP4 172.16.0.1
 s=Asterisk PBX 1.6.2.9-2ubuntu2.1
 c=IN IP4 172.16.0.1
 t=0 0
 m=audio 19486 RTP/AVP 3 0 8 101
 a=rtpmap:3 GSM/8000
 a=rtpmap:0 PCMU/8000
 a=rtpmap:8 PCMA/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=ptime:20
 a=sendrecv

 ---

 --- SIP read from UDP:172.16.0.2:5060 ---
 SIP/2.0 401 Unauthorized
 Via: SIP/2.0/UDP
 172.16.0.1:5066;branch=z9hG4bK59f50e30;received=172.16.0.1;rport=5066
 From: asterisk sip:test@172.16.0.1:5066;tag=as1689b2b6
 To: sip:1234568@172.16.0.2;tag=as1a6c2364
 Call-ID: 2353cf0e59596e285c684b44220f8915@172.16.0.1
 CSeq: 102 INVITE
 Server: Asterisk PBX 1.6.2.9-2ubuntu2.1
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
 Supported: replaces, timer
 WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=7a03a1d3
 Content-Length: 0


 -
 --- (11 headers 0 lines) ---
 Transmitting (no NAT) to 172.16.0.2:5060:
 ACK sip:1234568@172.16.0.2 SIP/2.0
 Via: SIP/2.0/UDP 172.16.0.1:5066;branch=z9hG4bK59f50e30;rport
 Max-Forwards: 70
 From: asterisk sip:test@172.16.0.1:5066;tag=as1689b2b6
 To: sip:1234568@172.16.0.2;tag=as1a6c2364
 Contact: sip:test@172.16.0.1:5066
 Call-ID: 2353cf0e59596e285c684b44220f8915@172.16.0.1
 CSeq: 102 ACK
 User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2.1
 Content-Length: 0


 ---
 [Apr 14 21:10:38] NOTICE[31158]: chan_sip.c:17975
 handle_response_invite: Failed to authenticate on INVITE to
 'asterisk sip:test@172.16.0.1:5066;tag=as1689b2b6'

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Re: [asterisk-users] Can't make Asterisk send authentication to remote peer on INVITE

[Ben WIlliams]

172.16.0.1 is not sending the authentication details to 172.16.0.2
when 172.16.0.2 responds with 401.

On Sat, Apr 14, 2012 at 9:30 PM, Ben WIlliams
bwilliams+aster...@jadeworld.com wrote:
 This is a really simple problem that I just can't get to work. There
 are two Asterisk servers with the following sip user and peer. When a
 call is attempted, Asterisk is not sending authentication details in
 response to the 401. Note, if the secret is blank on 172.16.0.2 test,
 the INVITE succeeds.

 on 172.16.0.2:

 [test]
 type=friend
 secret=abcde
 host=dynamic
 context=demo

 on 172.16.0.1 :

 [natty]
 type=peer
 host=172.16.0.2
 fromuser=test
 secret=abcde

 originate SIP/natty/1234568 extension 200
  == Using SIP RTP CoS mark 5
 Audio is at 172.16.0.1 port 19486
 Adding codec 0x2 (gsm) to SDP
 Adding codec 0x4 (ulaw) to SDP
 Adding codec 0x8 (alaw) to SDP
 Adding non-codec 0x1 (telephone-event) to SDP
 Reliably Transmitting (no NAT) to 172.16.0.2:5060:
 INVITE sip:1234568@172.16.0.2 SIP/2.0
 Via: SIP/2.0/UDP 172.16.0.1:5066;branch=z9hG4bK59f50e30;rport
 Max-Forwards: 70
 From: asterisk sip:test@172.16.0.1:5066;tag=as1689b2b6
 To: sip:1234568@172.16.0.2
 Contact: sip:test@172.16.0.1:5066
 Call-ID: 2353cf0e59596e285c684b44220f8915@172.16.0.1
 CSeq: 102 INVITE
 User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2.1
 Date: Sat, 14 Apr 2012 09:10:38 GMT
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
 Supported: replaces, timer
 Content-Type: application/sdp
 Content-Length: 290

 v=0
 o=root 1594270426 1594270426 IN IP4 172.16.0.1
 s=Asterisk PBX 1.6.2.9-2ubuntu2.1
 c=IN IP4 172.16.0.1
 t=0 0
 m=audio 19486 RTP/AVP 3 0 8 101
 a=rtpmap:3 GSM/8000
 a=rtpmap:0 PCMU/8000
 a=rtpmap:8 PCMA/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=ptime:20
 a=sendrecv

 ---

 --- SIP read from UDP:172.16.0.2:5060 ---
 SIP/2.0 401 Unauthorized
 Via: SIP/2.0/UDP
 172.16.0.1:5066;branch=z9hG4bK59f50e30;received=172.16.0.1;rport=5066
 From: asterisk sip:test@172.16.0.1:5066;tag=as1689b2b6
 To: sip:1234568@172.16.0.2;tag=as1a6c2364
 Call-ID: 2353cf0e59596e285c684b44220f8915@172.16.0.1
 CSeq: 102 INVITE
 Server: Asterisk PBX 1.6.2.9-2ubuntu2.1
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
 Supported: replaces, timer
 WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=7a03a1d3
 Content-Length: 0


 -
 --- (11 headers 0 lines) ---
 Transmitting (no NAT) to 172.16.0.2:5060:
 ACK sip:1234568@172.16.0.2 SIP/2.0
 Via: SIP/2.0/UDP 172.16.0.1:5066;branch=z9hG4bK59f50e30;rport
 Max-Forwards: 70
 From: asterisk sip:test@172.16.0.1:5066;tag=as1689b2b6
 To: sip:1234568@172.16.0.2;tag=as1a6c2364
 Contact: sip:test@172.16.0.1:5066
 Call-ID: 2353cf0e59596e285c684b44220f8915@172.16.0.1
 CSeq: 102 ACK
 User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2.1
 Content-Length: 0


 ---
 [Apr 14 21:10:38] NOTICE[31158]: chan_sip.c:17975
 handle_response_invite: Failed to authenticate on INVITE to
 'asterisk sip:test@172.16.0.1:5066;tag=as1689b2b6'

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[asterisk-users] Release Announcement: Adhearsion 2.0 for Asterisk 1.8+

[Ben Klang]

Today marks another milestone in the Adhearsion project: the release of 
Adhearsion 2.0.  There has been a fury of activity in the last few days as we 
have worked hard to update documentation and release a brand new look-and-feel 
for the Adhearsion website.  We hope you like it.

So, with a small flourish and no small amount of relief, I'm pleased to 
announce the immediate availability of Adhearsion 2.0, the open source 
framework for the creation of voice applications.

Here are some highlights of the changes relative to the latest Adhearsion 1.x:

* Adhearsion now supports multiple telephony engines! In particular we support 
Asterisk (as always) as well newly added support for PRISM via the 
open-standard Rayo protocol

* CallControllers make telephone functionality more Ruby-esque, more testable 
and are scientifically shown to make you happier

* A self-documenting configuration engine (rake config:show)

* A completely revamped plugin system makes adding and sharing Adhearsion 
functionality better than ever

* Did I mention the new website design and documentation?

* Way more stuff than I can reasonably list here.  You should check out the 
CHANGELOG and the Upgrade documentation.


I would like to take a moment and recognize the team that made this happen.  
The Adhearsion project has exploded in the last year, and many of the people 
who worked so hard to bring you Adhearsion 2 are actually new to the community 
within the last year!  A special thanks to Ben Langfeld who has driven much of 
this development effort and contributed fixes to many bugs and added new 
functionality in some of our dependency packages in the process of making this 
happen.  I also want to thank our sponsors, especially Tropo, for not only 
funding direct development, but helping to evangelize and organize.  Tropo has 
been a fantastic collaborator throughout Adhearsion's lifetime.

Now, you might be thinking all of the above sounds great, but how stable can 
it really be? Is it webscale?  The answer is very stable and yes, 
respectively.  But I don't want you to just take my word for it.  A few weeks 
back, I bet Ben Langfeld a double sawbuck (that is, an Andrew Jackson, a USD 
$20) that Adhearsion 2 wasn't ready to take a fully loaded server's worth of 
traffic.  And he muttered something about me not keeping the faith, and then 
took me up on that bet.  So now we're going to do it live.  In the next couple 
of weeks we are going to do a live broadcast of a load test, pushing Adhearsion 
to scale on both Asterisk and PRISM.  We are going to see just how webscale 
it is, and we're going to be streaming the event live on Ustream so you all can 
join in the fun.  The loser (hopefully me) will be well and truly prepared to 
take your jeers and fork over the cash.  Look for an announcement soon for 
where and when.  It's about as geeky fun as telephony gets.  I hope you'll come 
join us.

In the meantime, go check out Adhearsion 2!

On behalf of the Adhearsion 2 development team, thanks for being you.
-- 
Ben Klang
bkl...@mojolingo.com
404.475.4841

Mojo Lingo -- Voice applications that work like magic
http://mojolingo.com
Twitter: @MojoLingo



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[asterisk-users] Adhearsion 1.2.0 Released at Lone Star Ruby Conference V

[Ben Klang]

Yesterday I announced the release of Adhearsion 1.2.0 during my presentation 
State of the Art Telephony with Ruby at Lone Star Ruby Conference V in 
Austin, Texas. LSRC has long been a supporter of the Adhearsion and Asterisk 
projects and I felt it fitting to use the presentation there as the opportunity 
to make this exciting new release.

In case you missed it, an overview of some of the new features was discussed in 
a previous blog post entitled Upcoming Features in Adhearsion 1.2 on the Mojo 
Lingo blog. A quick recap here:

Native support for text-to-speech on both Asterisk and Tropo
Several useful new methods, including #play_or_speak, #record_to_file, #input!, 
#interruptible_play!, #speak and more.
Enhancements to existing methods such as #input and #play
Fixes to the way we load Bundler gems with the ahn executable
Improvments to the logging system to enable tracing the activities of a 
specific call through the logs
Also interesting to the Adhearsion community is the greatly improved support 
for Tropo via AGItate and the ability to easily create sophisticated 
text-to-speech renderings with the new RubySpeech library. These projects were
As always, the latest and greatest Adhearsion is available through RubyGems and 
sources are on Github.

Thanks once again to past core contributor Ben Langfeld and welcome to new 
contributor Lance Gleason. Thanks also to IfByPhone for sponsoring several of 
the improvements (especially Text-to-Speech).

/BAK/
-- 
Ben Klang
bkl...@mojolingo.com
404.475.4841

Mojo Lingo -- Voice applications that work like magic
http://mojolingo.com
Twitter: @MojoLingo
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Re: [asterisk-users] missing argument on AGI

[Ben Klang]

On Feb 24, 2011, at 5:27 PM, Ron wrote:

 Hi All,
 
 I'm using the asterisk 1.4.39.2 with phpagi 2.20 I have setup a dial plan:
 
 [callback-outbound]
 exten = _00.,1,Macro(callout|${EXTEN})
 
 [macro-callout]
 exten = s,1,AGI(getchannel.php|${ARG1})
 exten = s,2,Dial(Local/${OUTBOUND}@from-internal/nj||tr)
 exten = s,3,Hangup()
 
 but for some reason i am not receiving the argument:
 Executing [s@macro-callout:2] Dial(SIP/201-0004, 
 Local/@from-internal/nj||tr) in new stack
 [Feb 24 21:47:11] NOTICE[1901] chan_local.c: No such extension/context 
 @from-internal while calling Local channel
 
 the number is missing, i get the number from the agi, below is the debug:
 
 21:47:10]-- Executing [006583232393-1-201@callback-outbound:1] 
 Macro(SIP/201-0004, callout|006583232393-1-201) in new stack
 21:47:10]-- Executing [s@macro-callout:1] AGI(SIP/201-0004, 
 getchannel.php|006583232393-1-201) in new stack
 21:47:10]-- Launched AGI Script /var/lib/asterisk/agi-bin/getchannel.php
 21:47:10]AGI Tx  agi_request: getchannel.php
 21:47:10]AGI Tx  agi_channel: SIP/201-0004
 21:47:10]AGI Tx  agi_language: en
 21:47:10]AGI Tx  agi_type: SIP
 21:47:10]AGI Tx  agi_uniqueid: 1298555228.12
 21:47:10]AGI Tx  agi_callerid: unknown
 21:47:10]AGI Tx  agi_calleridname: unknown
 21:47:10]AGI Tx  agi_callingpres: 0
 21:47:10]AGI Tx  agi_callingani2: 0
 21:47:10]AGI Tx  agi_callington: 0
 21:47:10]AGI Tx  agi_callingtns: 0
 21:47:10]AGI Tx  agi_dnid: unknown
 21:47:10]AGI Tx  agi_rdnis: unknown
 21:47:10]AGI Tx  agi_context: macro-callout
 21:47:10]AGI Tx  agi_extension: s
 21:47:10]AGI Tx  agi_priority: 1
 21:47:10]AGI Tx  agi_enhanced: 0.0
 21:47:10]AGI Tx  agi_accountcode:
 21:47:10]AGI Tx 
 21:47:10]AGI Rx  EXEC Noop
 21:47:10]-- AGI Script Executing Application: (Noop) Options: ((null))   
 == THIS SHOULD DISPLAY THE ARGUMENT
 21:47:10]AGI Tx  200 result=0
 21:47:11]AGI Rx  EXEC Set CALLERID(num)=
 21:47:11]-- AGI Script Executing Application: (Set) Options: 
 (CALLERID(num)=)
 21:47:11]AGI Tx  200 result=0
 21:47:11]AGI Rx  EXEC Set OUTBOUND=
 21:47:11]-- AGI Script Executing Application: (Set) Options: (OUTBOUND=)

^ --- This

You are relying on the channel variable ${OUTBOUND} to be set when you invoke 
the Dial() string, but the AGI script is not setting OUTBOUND correctly.  In 
both cases you illustrate (the Noop and the Set) the variable isn't making it 
from PHP to AGI.  Check your PHP script to make sure the data is what you think 
it is.

/BAK/
-- 
Ben Klang
bkl...@mojolingo.com
404.475.4841

Mojo Lingo -- Voice applications that work like magic
http://mojolingo.com
Twitter: @MojoLingo


 21:47:11]AGI Tx  200 result=0
 
 
 my php code include something:
 
 #!/usr/bin/php-cgi -q
 ?php
 include('phpagi/phpagi.php');
 $agi=new AGI();
 
 $param = $argv[1];
 
 $agi - exec(Noop,$param);
 
 ..
 ..
 ..
 ..
 ?
 
 not sure where to check next i'm stumped, hope somebody can help. thanks in 
 advance.
 
 Regards
 Ron
 
 
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[asterisk-users] Adhearsion 1.0.1 Released

[Ben Klang]

The Adhearsion team announces the release of Adhearsion version 1.0.1. 
Adhearsion is an open source Ruby-language framework for creating telephony 
applications.  This update primarily addresses compatibility with newer 
versions of other software but also adds native support for Bundler to newly 
created Adhearsion applications.

Here are some highlights from the changelog:

Handling of new Asterisk 1.6/1.8 events
Improved control of Asterisk Queues
Two new dialplan methods have been added: say_chars and say_phonetic
Ruby 1.9 is now an officially supported platform
Fix compatibility with Rails 3
Bundler now included by default for new Adhearsion applications
Not bad for a dot release!  You can read the full CHANGELOG here.

As always I'd like to thank the Adhearsion community for their contributions to 
this release.  Special thanks to contributors Ben Langfeld, Robert Jackson and 
Matthew Clark.

To install Adhearsion just type gem install adhearsion at your nearest command 
prompt.  For help getting started, checkout our Wiki and Getting Started pages. 
 As always, you can find us on irc.freenode.net #adhearsion or our Google 
Groups mailing list.  Contributors welcome!  Check out the sources on 
Adhearsion's Github.


/BAK/
-- 
Ben Klang
bkl...@mojolingo.com
404.475.4841

Mojo Lingo -- Voice applications that work like magic
http://mojolingo.com
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Re: [asterisk-users] Usage Reports

[Ben Schorr]

I think we installed it with all of the defaults - but maybe MYSQL isn't
there.  I'll check on that, thanks!

 

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
www.rolandschorr.com http://www.rolandschorr.com/ 
b...@rolandschorr.com mailto:b...@rolandschorr.com 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Duncan
Turnbull
Sent: Thursday, December 30, 2010 2:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Usage Reports

 

Freepbx really needs its own list but it doesn't seem to have one

 

But - if you have mysql setup and records being logged then the reports
should show you usage on a daily, weekly, monthly level. Make sure you
built asterisk with cdrs logged into mysql - its in the addons 

 

Cheers Duncan

 

On 30/12/2010, at 8:36 PM, Ben Schorr wrote:





We're using FreePBX 2.8 and there is a Reports tab but it doesn't seem
to actually do anything.  Is there some secret/trick to getting a report
out of it that will tell us which extensions are placing calls?  I've
tried every query on the form that I can think of.  Is the reporting
disabled by default or ???

 

Any tips/pointers appreciated.

 

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
www.rolandschorr.com http://www.rolandschorr.com/ 
b...@rolandschorr.com mailto:b...@rolandschorr.com 

 

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[asterisk-users] Usage Reports

[Ben Schorr]

We're using FreePBX 2.8 and there is a Reports tab but it doesn't seem
to actually do anything.  Is there some secret/trick to getting a report
out of it that will tell us which extensions are placing calls?  I've
tried every query on the form that I can think of.  Is the reporting
disabled by default or ???

 

Any tips/pointers appreciated.

 

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
www.rolandschorr.com http://www.rolandschorr.com/ 
b...@rolandschorr.com mailto:b...@rolandschorr.com 

 

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[asterisk-users] Adhearsion 1.0 - Now Showing

[Ben Klang]

Thanks to the hard work of many people in the Adhearsion community, I am 
pleased to be able to announce the immediate availability of Adhearsion version 
1.0.  Since Jay Phillips first began work on the project in 2006 Adhearsion has 
changed the way developers think about telephony applications.  Now with 
several years of operating experience and multitudes of applications deployed 
to production, it is time to acknowledge this important milestone.

What does Adhearsion 1.0 mean?

• A battle-tested API.  Adhearsion 1.0 has defined a well-tested API 
and it has been proven with over two years of real-world deployment experience.
• A “stable” branch.  Adhearsion’s exposed API will remain stable 
throughout this major version number series.  No backward-incompatible changes 
will be made, making it safer for developers to trust future upgrades.
• Updated documentation.  Thanks to Justin Dupree of Tropo, 
Adhearsion’s docs received a lot of TLC in the form of content updates and a 
migration to Github Wiki.  Check them out at docs.adhearsion.com
• Gem-based Components.  The final feature added to Adhearsion prior to 
1.0 is the ability to install and use components via RubyGems.  Learn more 
about that in the Gem-based Components wiki page here 
(http://github.com/adhearsion/adhearsion/wiki/Gem-based-components).

To get your hands on Adhearsion 1.0 run, don’t walk, to your nearest command 
line and issue a

sudo gem install adhearsion.

As always, don’t be a stranger.  We can be found on the Adhearsion mailing list 
(http://groups.google.com/group/adhearsion), in IRC (irc.freenode.net 
#adhearsion) and on the Adhearsion website (http://adhearsion.com).

PS: Come to the formal release announcement at AstriCon, room 6, at 1:45PM 
today!

/BAK/
-- 
Ben Klang
Adhearsion Project, Lead Maintainer
b...@alkaloid.net
http://adhearsion.com



-- 
Ben Klang
Alkaloid Networks LLC
b...@alkaloid.net
404.475.4850
http://projects.alkaloid.net


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[asterisk-users] Announcing Adhearsion 0.8.5

[Ben Klang]

We in the Adhearsion community are happy to announce the release of version 
0.8.5 of our framework.  Adhearsion is a featureful framework for developing 
Asterisk-based applications using the Ruby programming language.  This latest 
release adds exciting new support for XMPP within Adhearsion applications.  
Additionally we have focused on fixing outstanding issues, improving 
documentation and including contributions from many new community members.

There are several easy ways to get Adhearsion:

• “gem install adhearsion” in your favorite terminal window
• Download the tarball 
(http://github.com/adhearsion/adhearsion/tarball/0.8.5)
• Clone our Git repository (http://github.com/adhearsion/adhearsion.git)

New in this version of Adhearsion:

• Support for connecting to XMPP, querying roster status, sending and 
receiving messages and other XMPP features.  A sample component is included to 
get you started quickly.
• Allow routing calls into specific Adhearsion dialplan contexts using 
the AGI URI.  Example: agi://localhost/my_stuff will send the call into the 
“my_stuff” context.
• Allow using static MeetMe conference room definitions
• Extend logging objects into ActiveRecord and Blather
• Dozens of smaller bugfixes and enhancements; see the CHANGELOG

Now that the Adhearsion API has been stable for nearly two years, we will be 
looking for a 1.0 release to be made in the next couple of months based on the 
current software.  The 0.8.5 version may be considered a 1.0 Release Candidate. 
 Our community is making a concerted effort to improve documentation, enhance 
examples, create shareable components and fix any remaining bugs in time for 
the big 1.0.

To learn more about the Adhearsion project, visit our website at 
http://adhearsion.com.  Our active community may be found on our mailing list 
or in IRC (irc.freenode.net #adhearsion).

A special shout out to the following people who helped bring you Adhearsion 
0.8.5:

• Ben Langfeld (XMPP support, documentation, code review)
• Michel Vaillancourt (Testing, feedback, ideas)
• Jason Goecke and Tropo (Continued support of Adhearsion and 
AdhearsionConf, project mentoring)
• Chris Matthieu (Creation of AHNHub.com)
• Jay Phillips (Work on AHNHub.com, ideas and discussion at 
AdhearsionConf)

And a thanks to all of the people who reported bugs on our bugtracker, sent us 
pull requests on Github, answered questions on the mailing list or in IRC, and 
generally made our community the awesome group that it is.  Thanks!

/BAK/
-- 
Ben Klang
Alkaloid Networks LLC
b...@alkaloid.net
404.475.4850
http://projects.alkaloid.net


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Re: [asterisk-users] Polycom 331 freezes connecting to FreePBX

[Ben Schorr]

Sorry, I should clarify - we have had a similar setup (IPSEC VPN, Polycom 331) 
working at a different location with a different handset for this same firm.  
We've never gotten the phone/VPN to work at this particular site.  I was just 
trying to explain that it DOES appear to be successfully connecting to the TFTP 
server that provisions the phones.  The TFTP server is sitting right next to 
the Asterisk server so if it can connect to one it should be able to connect to 
the other - basic connectivity, it appears, is working between the sites.

We currently have 57 other Polycom phones (most of them 331s)  working in this 
system, with the current application, just fine, including maybe 10 that 
connect over an IPSEC VPN from a 3rd location.

That does give me an idea though...we could take the handset from the failing 
location to the 3rd location (barely a mile away) and plug it in and see if it 
works there.  If it does then the problem must be somewhere in the connection 
and not with the handset itself.  If it doesn't work in the other location 
either then the problem is probably with the phone and/or it's configuration. 

Make sense?

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
www.rolandschorr.com
b...@rolandschorr.com


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
 boun...@lists.digium.com] On Behalf Of David Backeberg
 Sent: Monday, August 16, 2010 11:04
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Polycom 331 freezes connecting to FreePBX
 
 On Mon, Aug 16, 2010 at 4:21 PM, Ben Schorr b...@rolandschorr.com wrote:
  We gave the phone a static IP address and pointed it to the
  configuration server on the remote end that has the CFG files for it.
  The phone starts up, downloads SIP and the new application and
  otherwise seems to be booting normally.  Then it gets to the LAN
  Properties screen that shows the phone's IP address, MAC address and
 firmware version and then...nothing.
  It just sits there frozen.
 
 I have a suggestion...
 
 Put back the 'old application', and determine whether the 'new application' 
 broke
 your phone boot. Since you don't mention changing anything else, survey says 
 it's
 probably the last thing you changed that broke things.
 
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[asterisk-users] Polycom 331 freezes connecting to FreePBX

[Ben Schorr]

We deployed a single phone handset (Polycom 331) at a remote site.  We
have a IPSEC VPN running between the firewall at the remote site and the
firewall at the site where our Asterisk/FreePBX box lives.  We have used
a similar configuration for this site before and it worked fine.

 

We gave the phone a static IP address and pointed it to the
configuration server on the remote end that has the CFG files for it.
The phone starts up, downloads SIP and the new application and
otherwise seems to be booting normally.  Then it gets to the LAN
Properties screen that shows the phone's IP address, MAC address and
firmware version and then...nothing.  It just sits there frozen.  

 

I assume it's trying to register with the Asterisk server but for some
reason that seems to be failing.

 

I've swapped in a different, brand new, Polycom 331 on that spot and it
does the exact same thing.  From my laptop I can ping the Asterisk
server across the VPN just fine.  All of the network connectivity looks
good, as far as I can tell.

 

Anybody have a hint for what we should be looking at?  I don't see any
obviously blocked ports and the VPN should take care of that anyhow.
I've looked in Polycom's KB but it didn't seem to offer any explanation
for what it means when the phone freezes on the LAN properties screen.

 

Any suggestions welcomed.

 

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
1155 Fort Street Mall
Honolulu, Hawaii 96813
Mobile:  808-782-6306
Fax: 808-533-3677
www.rolandschorr.com http://www.rolandschorr.com/ 
b...@rolandschorr.com mailto:b...@rolandschorr.com 

 

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[asterisk-users] T.38 on a MAX/Lucent/Ascend TNT

[Ben Winslow]

Hello folks,

I've been trying to get T.38 over SIP working with calls terminated by a 
MAX/Lucent/Ascent TNT.  As far as I can tell, SIP and T.38 are actually 
working perfectly; however, I can't get the TNT to properly terminate a 
FAX call.  Does anyone have a working configuration for SIP and T.38 for 
calls from a TNT or APX?

Here's a brief description/diagram of my test setup:

Laptop --RS232-- Modem --POTS-- Channel bank (ADIT 600) --CAS T1-- 
TNT --Ethernet/SIP-- Asterisk --SIP-- t38modem.

The TNT is running TAOS 11.0.4 with both the SIP and realtime fax 
features licensed.  I've tried using several modems, and with each I can 
establish a data call without any problems (although at a maximum of 
31200bps/3429 baud using a 33.6 modem, for reasons I haven't dug into 
yet.)  Whenever I try to send a fax from the laptop, however, the call 
always seems to fail in the first HDLC phase (phase B) with either a 
timeout or error 23 (COMREC invalid command received.)  The modem is 
connected directly to the channel bank and the channel bank is connected 
directly to the TNT in an attempt to reduce the number of variables.

With my current configuration, the call to Asterisk will come up as a 
voice call, then be dumped into t38modem when Asterisk receives the 
1100Hz CNG tone from the sending modem.  At that point, the call is 
dumped into t38modem which re-invites the TNT with the T.38 options, and 
the TNT usually sends a T.38 t30-ind of 0x00 (no-signal) or 0x3a (???), 
although I do occasionally see more promising messages like 
v29-7200-training or t30-data/hdlc-fcs-ok.  Shortly thereafter, the 
dialing modem will give up and terminate the call.

If I try dumping the same call into iaxmodem instead of t38modem, the 
call actually progresses further -- the real modem receives and decodes 
the HDLC CSI/DIS from iaxmodem, but the high-speed trainup always fails 
for calls coming from the TNT.

Does anyone have any advice or suggestions?  Has anyone actually made 
T.38 work with one of these devices running ANY TAOS version?

Thanks,
-- 
Ben Winslow winsl...@pa.net

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

[Ben Dinnerville]

 process_sdp_a_image: 
Transcoding MMR: 0
[May 12 22:47:15] DEBUG[22587]: chan_sip.c:8387 process_sdp: Processing 
media-level (image) SDP a=T38FaxTranscodingMMR:0... OK.
[May 12 22:47:15] DEBUG[22587]: chan_sip.c:9018 process_sdp_a_image: 
Transcoding JBIG: 0
[May 12 22:47:15] DEBUG[22587]: chan_sip.c:8387 process_sdp: Processing 
media-level (image) SDP a=T38FaxTranscodingJBIG:0... OK.
[May 12 22:47:15] DEBUG[22587]: chan_sip.c:5034 change_t38_state: T38 
state changed to 2 on channel SIP/vltb-sbc01-
[May 12 22:47:15] DEBUG[22587]: chan_sip.c:8586 process_sdp: Have T.38 
but no audio, accepting offer anyway
[May 12 22:47:15] DEBUG[22587]: chan_sip.c:20433 handle_request_invite: 
SIP/vltb-sbc01-: This call is UP
[May 12 22:47:15] DEBUG[22587]: chan_sip.c:3561 __sip_xmit: Trying to 
put 'SIP/2.0 100' onto UDP socket destined for 202.167.246.165:5060
[May 12 22:47:15] DEBUG[22725]: chan_sip.c:5034 change_t38_state: T38 
state changed to 3 on channel SIP/vltb-sbc01-
[May 12 22:47:15] DEBUG[22725]: chan_sip.c:10410 add_sdp: Done building 
SDP. Settling with this capability: 0x0 (nothing)
[May 12 22:47:15] DEBUG[22725]: chan_sip.c:3561 __sip_xmit: Trying to 
put 'SIP/2.0 200' onto UDP socket destined for 202.167.246.165:5060
[May 12 22:47:15] DEBUG[22725]: channel.c:2434 ast_settimeout: 
Scheduling timer at (0 requested / 0 actual) timer ticks per second
[May 12 22:47:15] NOTICE[22725]: res_fax.c:1373 sendfax_t38_init: 
Negotiated T.38 for send on SIP/vltb-sbc01-
[May 12 22:47:15] ERROR[22725]: res_fax_digium.c:2114 dgm_fax_start: FAX 
handle 0: failed to queue document '/var/spool/asterisk/fax/campaign_70.tif'
[May 12 22:47:15] ERROR[22725]: res_fax.c:834 generic_fax_exec: channel 
'SIP/vltb-sbc01-' FAX session '0' failure, reason: 'failed to 
start FAX session'
[May 12 22:47:15] DEBUG[22587]: chan_sip.c:4097 __sip_ack: Stopping 
retransmission on '197593892ad59589494f1e56403b7...@202.177.217.102' of 
Response 100: Match Found
sp02*CLI
Disconnected from Asterisk server
[r...@sp02 tmp]# /usr/sbin/safe_asterisk: line 138: 22547 Segmentation 
fault  (core dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f 
${CLIARGS} ${ASTARGS} /dev/${TTY}  /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
cat: /var/run/asterisk.pid: No such file or directory
Automatically restarting Asterisk.



The system is running on Centos 5.2 i386 but I can also re-produce on 
Centos 5.4 x86_64

The issue seems to be with the error - failed to queue document 
'/var/spool/asterisk/fax/campaign_70.tif' However I can confirm that a) 
the file exists, b) it is worl readable and c) is created from a pdf 
with ghostscipt in the recommended fashion - gs -q -dNOPAUSE -dBATCH 
-sDEVICE=tiffg4 -sPAPERSIZE=a4 -sColorMode=mono 
-sOutputFile=campaign_70.tif combo2010.pdf

and can open the file in preview on mac without any issues.

There is no further information in the logs / console I cannot think of 
any other reason why it would not be able to queue the file.

Digium for some reason do not leave the older versions of FFA on their 
site so I am unable to test this on older versions of FFA, only older 
versions of asterisk all of which reproduce the same results (have tried 
asterisk-1.6.1.19.tar.gz
asterisk-1.6.2.2.tar.gz
asterisk-1.6.2.5.tar.gz
asterisk-1.6.2.7.tar.gz)

Anyone else having core dump issues or fax failure issues with 1.2.0? 
This one has kept me up for 2 days now - if I had any hair i would be 
pulling it out now.

Cheers,

Ben


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

[Ben Dinnerville]

Kevin P. Fleming wrote:

 Like I said, it's a known problem, and the fix should be out within a
 day or two. It was reported to us about a week ago, so if you had
 contacted the support department, it's likely they would have been able
 to shortcut your hair-pulling experience :-)
 

Hi Kevin,

Thanks for the update. Unfortunately I contacted the support team early 
on in the process (about 35 hours ago) and to date the only response has 
been Please run the debug process and send us the logs - so there has 
been much hair pulling in the meanwhile. I have ticket WAJ-201081 logged 
and am awaiting a response - however it would be appreciated if I could 
get my hands on the 1.2.1 version and would be more than happy to test 
it and see if the issue is fixed.

Cheers,

Ben


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

[Ben Dinnerville]

Well, I have managed to get my hands on a copy of 1.2.1 rc1 FFA which 
seems to have fixed the core dumping issue but does not appear to have 
fixed the issue that was causing the core dump.

We are still getting an issue with a particular file which I have tried 
multiple different ways to create to no avail. The tiff file is created 
with ghostscript from a pdf as per the guidlines but every time we try 
and fax it we get the following:

[May 13 11:28:09] DEBUG[26959] res_fax_digium.c: FAX handle 0: created 
document queue
[May 13 11:28:09] ERROR[26959] res_fax_digium.c: FAX handle 0: failed to 
queue document '/var/spool/asterisk/fax/campaign_70.tif'
[May 13 11:28:09] DEBUG[26959] res_fax_digium.c: FAX handle 0: freeing 
document queue.
[May 13 11:28:09] DEBUG[26959] res_fax_digium.c: FAX handle 0: closing
[May 13 11:28:09] ERROR[26959] res_fax.c: channel 
'SIP/teleblast-sbc01-0001' FAX session '1' failure, reason: 'failed 
to start F
AX session'

and the call terminates.

tiffinfo for the file shows:

TIFF Directory at offset 0x2aae0 (174816)
   Subfile Type: multi-page document (2 = 0x2)
   Image Width: 1767 Image Length: 2369
   Resolution: 204, 196 pixels/inch
   Bits/Sample: 1
   Compression Scheme: CCITT Group 4
   Photometric Interpretation: min-is-white
   FillOrder: msb-to-lsb
   Orientation: row 0 top, col 0 lhs
   Samples/Pixel: 1
   Rows/Strip: 2369
   Planar Configuration: single image plane
   Page Number: 0-0
   Software: GPL Ghostscript 8.70
   DateTime: 2010:05:12 23:20:00
   Group 4 Options: (0 = 0x0)
TIFF Directory at offset 0x57172 (356722)
   Subfile Type: multi-page document (2 = 0x2)
   Image Width: 1767 Image Length: 2369
   Resolution: 204, 196 pixels/inch
   Bits/Sample: 1
   Compression Scheme: CCITT Group 4
   Photometric Interpretation: min-is-white
   FillOrder: msb-to-lsb
   Orientation: row 0 top, col 0 lhs
   Samples/Pixel: 1
   Rows/Strip: 2369
   Planar Configuration: single image plane
   Page Number: 1-0
   Software: GPL Ghostscript 8.70
   DateTime: 2010:05:12 23:20:11
   Group 4 Options: (0 = 0x0)


And the file is 357026 bytes in size. Can anyone see anything wrong with 
the tiff info or does anyone know of any issues with multiple pages or 
file size with fax for asterisk? Unfortunately I cannot seem to find any 
more information as to why the document couldn't be queued.

Cheers,

Ben


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Re: [asterisk-users] extension not found

[Ben Schorr]

Is there some reason why I keep getting this same message from cool
dude over and over and over?  And under different subject lines?

 

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
www.rolandschorr.com http://www.rolandschorr.com/ 
b...@rolandschorr.com

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cool dude
Sent: Friday, February 12, 2010 21:24
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] extension not found

 

hi friend need ur help in dial plan, i want to allow exten 2000 to 2005
can make call outside and exten 2006 to 2010 can not make call outside.
heres my dial plan.
 
sip.conf
 
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=outside
secret=1234
host=dynamic

[2001]
type=friend
context=outside
secret=1234
host=dynamic

[2002]
type=friend
context=outside
secret=1234
host=dynamic

[2003]
type=friend
contex=outside
secret=1234
host=dynamic
 
[2004]
type=friend
contex=outside
secret=1234
host=dynamic

[2005]
type=friend
contex=outside
secret=1234
host=dynamic
 
[2006]
type=friend
contex=internal
secret=1234
host=dynamic

[2007]
type=friend
contex=internal
secret=1234
host=dynamic
[2008]
type=friend
contex=internal
secret=1234
host=dynamic

[2009]
type=friend
contex=internal
secret=1234
host=dynamic

[2010]
type=friend
contex=internal
secret=1234
host=dynamic


## ##
vi /etc/asterisk/extensions.conf
[from-zaptel]
exten = s,1,wait(2)
exten = s,n,dial(sip/2000)
exten = s,n,dial(sip/2001)
exten = s,n,Playback(tt-weasels)
[others]
include = internal
include = outside
[internal]
exten = _20XX,1,Dial(SIP/${EXTEN})
exten = _20XX,n,VoiceMail(${ext...@others,u)
exten = _20XX,n,Hangup()
[outside]
exten = 2001,1,Dial(Zap/1-1/${EXTEN})
exten = 2001,n,Hangup
exten = 2002,1,Dial(Zap/1-1/${EXTEN})
exten = 2002,n,Hangup
exten = 2003,1,Dial(Zap/1-1/${EXTEN})
exten = 2003,n,Hangup
exten = 2004,1,Dial(Zap/1-1/${EXTEN})
exten = 2004,n,Hangup
exten = 2005,1,Dial(Zap/1-1/${EXTEN})
exten = 2005,n,Hangup

this is the log when i am calling from exten 2000 to outside

Connected to Asterisk 1.4.29 currently running on localhost (pid = 2243)
Verbosity is at least 3
[Feb 13 12:05:47] NOTICE[2482]: chan_sip.c:15124 handle_request_invite:
Call from '2002' to extension '9193696136' rejected because extension
not found.

 



Your Mail works best with the New Yahoo Optimized IE8. Get it NOW!
http://in.rd.yahoo.com/tagline_ie8_new/*http:/downloads.yahoo.com/in/in
ternetexplorer/ .

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Re: [asterisk-users] CDR / billsec / originate / local chan

[Ben Dinnerville]

Sean Brady wrote:

 Have you tried removing the /n option from the local channel?  Just a 
 thought, but it's probably worth a try.  You could also try calculating the 
 billsec in the dialplan and write it to the CDR with the adaptive CDR feature 
 in 1.6.2.
 
 Not sure if this is helpful but it was a thought.
 
 -Sean
Hi Sean,

Yeah, tried with and without the /n option - it turns out that there was 
a bug in any release post 1.6.2.0 including the current 1.6.2.2 that 
meant if you established a call using originate and a local channel the 
second leg of the channel would not get bridged in - this bug seems to 
be fixed in the current SVN release and I just need to wait now for a 
actual release.

The issue with cdr is that billsec does not seem to be available on the 
asterisk side of the channel, only the sip/ iax / other tech channel so 
when using originate, you dont get to see the tech side of the 
channel, only the asterisk side if it. There was a mention somewhere on 
dev forum and in the bug tracker that this was because someone deemed 
the asterisk side of the channel as non billable any only the tech side 
as billable so billsec was disabled ( I say disabled because it used to 
work and doesnt now - why capabilities are removed is beyond me)

Cheers,

Ben


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[asterisk-users] originate, local channel and failure extension

[Ben Dinnerville]

Hi All,

I am in the process of migrating from 1.4.20 to 1.6.2.x and have 
stumbled across a number of differences between the 2 versions that 
are forcing me to use local channels in my dialplan (mainly centered 
around the different behavior of CDR fields in the 2 versions) .

Previously, I would place a call via an AMI Originate action similar to:

action:.Originate..
actionid:.1306903_89#AJ_ORIGINATE_25
timeout:.4
exten:.s
async:.true
callerid:..612
context:.campaignType_5
priority:.1
channel:.SIP/trunk1/61212142321

And the campaignType_5 context would handle the call processing, 
capturing a failed dial attempt in the failed exten :

[campaignType_5_]
exten = s,1,Answer()
...


exten = h,1,Set(timefinished=${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)})
exten = h,2,GotoIf($[${TIMETOPRESS}foo = foo]?h,20)
...

exten = failed,1,Set(DIGITPRESSED=98)
exten = failed,2,Set(TIMETOPRESS=${timestarted})
exten = failed,3,Set(CALLSTATUS=6)
exten = failed,4,Set(timestarted=${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)})
exten = failed,5,Set(timefinished=${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)})

It looks like I am now being forced to use a local exten so I can access 
all the CDR variables I need, with the way the call is established now 
looking like:

action:.Originate..
actionid:.1306903_89#AJ_ORIGINATE_25
timeout:.4
exten:.s
async:.true
callerid:..612
context:.campaignType_5
priority:.1
channel:.Local/61212142...@outboundsip/n

And both a campaignType_5 context similar to that above and a 
outboundsip context similar to (an over simplified version):

[outboundsip]
exten = _XX.,1,Dial(SIP/trunk1/${EXTEN})
exten = _XX.,n,Hangup

exten = h,1,NoOp(Billsec is: ${CDR(billsec)})


My question - previously I would handle the failure of the call in the 
campaignType_5 context, now I have 2 contexts where I could possibly get 
a failure - should I be handling the failure in the campaignType_5 
context (the context that is defined in the originate action), 
outboundsip context (the context used for the local channel part of the 
call) or both? and why?

I have to say, the way that Orignate and cdr's used to work in 1.4.20 
was a hell of a lot easier to manage than the current versions.

Cheers,

Ben


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Re: [asterisk-users] How to run a remote PHP script and still have access to audio stream?

[Ben Dinnerville]

Leo Burd wrote:
 Hello there,
 
 I'm trying to figure out how to run a PHP script on a remote machine and 
 still have access to the audio stream associated with the call. 
 
 Ideally, I'd love to play/record audio files directly from/to the remote 
 server without having to copy them back and forth to the Asterisk 
 server.  What is the best way to do this?
 
 Is it possible to combine EAGI with FastAGI in PHP?
 
 Thanks in advance,
 
 Leo
 
 

There is the EAGI protocol that will allow this but the easiest way I 
find to do this sort of thing is to have a shared file system between 
the app / web server and the asterisk server(s). We run a clustered 
setup with 12 asterisk systems and a clustered jboss environment with a 
NFS mount shared between all the systems. For applications such as call 
recording asterisk does the monitor into the NSF mounted share / 
directory which is also visible on the jboss servers (mainly for 
permission checks / security etc) and the web server (for download etc). 
As long as you have a scheme that ensures you do not have duplicate file 
names (which can be controlled by a central database and via your php 
script) then you will not have any issues. There are a number of other 
file system alternatives out there that will achieve the same thing but 
NFS seems to be proven and stable and we have not any issues with it to 
date. Your php script can then be a simple AGI / FastAGI that simply 
executes a Playback(path/to/nfs/directory/file) - you can also 
incorporate things like checking if the file exists in your PHP script 
and implementing access restrictions etc.

We also share our sounds directory between systems this way so that all 
our sounds only have to reside in one place but are visible across all 
the systems.

Cheers,

Ben


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Re: [asterisk-users] Not able to compile asterisk, zaptel, libpri in /usr/src

[Ben Schorr]

Please see http://www.officeforlawyers.com/howask.htm 

 

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
www.rolandschorr.com http://www.rolandschorr.com/ 
b...@rolandschorr.com mailto:b...@rolandschorr.com 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of 
aster...@opensourcesolution.in
Sent: Sunday, February 07, 2010 19:10
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Not able to compile asterisk, zaptel,libpri in 
/usr/src

 

Not able to compile asterisk,zaptel,libpri in /usr/src

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Re: [asterisk-users] Running a script after Dial() ?

[Ben Dinnerville]

Per Jessen wrote:
 I have the following dialplan:
 
 ; calls prefix by '8' are recorded
 exten = _8[01]./_251,1,Set(something=shortened)
 exten = _8[01]./_251,n,Set(WAV=filename)
 exten = _8[01]./_251,n,Monitor(wav,${WAV},mb)
 exten = _8[01]./_251,n,Dial(mISDN/2/${EXTEN:1},,g)
 exten = _8[01]./_251,n,System(send-recorded-conversation ${WAV}.wav
 ${EXTEN:1} emailaddr)
 exten = _8[01]./_251,n,Hangup()
 
 The idea is that the caller may opt to record a conversation by
 prefixing the dialled number with '8'. The wav file would then be
 emailed to him when the call finishes. 
 The recording works fine, but the emailing doesn't - only when the
 called party hangs up first, but if the caller hangs up, the
 System(script) isn't called.  What am I missing here?
 
 
 /Per Jessen, Zürich
 
Sorry, Monitor also has the flag param which allows you to execute a 
command post recording if you want to stick with Monitor and not MixMonitor.


Cheers,

Ben


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Re: [asterisk-users] Running a script after Dial() ?

[Ben Dinnerville]

Why dont you use the MixMonitor application which allows for a system 
command to be passed in as an argument that is executed once the 
recording is finished??? -

MixMonitor(file.ext[|options[|command]])

command will be executed when the recording is over. Any strings 
matching ^{X} will be unescaped to ${X} and all variables will be 
evaluated at the time the application is called. Where command is a 
system (Linux shell) command, see Asterisk cmd System for example values.
The variable MIXMONITOR_FILENAME will contain the name of the file used 
for recordings.
Note do NOT include the dialplan command System(blah), just blah.
If you don't specify a full path of the sound file, the file will be 
stored in the monitor subdir of the path specified with astspooldir in 
asterisk.conf (so default will be /var/spool/asterisk/monitor).
Note that no environment variables are given to command — you must 
pass these on via command-line arguments.

The audio file is closed and processing of command is started *after* 
the 'h' extension priorities have been run.


Cheers,

Ben


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Re: [asterisk-users] Running a script after Dial() ?

[Ben Dinnerville]

 I get a somewhat minimal set of standard shell environment variables 
 (BASH*, HOSTNAME, PWD, TERM, etc) including the same PATH environment 
 variable I passed to Asterisk when it was started.
 
That just means that you cant rely on environment variables in the 
script that you execute and you must make sure your script is fully 
aware of anything it needs to know about itself. IE, dont rely on a path 
being setup and enter full paths to all commands / executables or any 
other variables and explicitly define them in the script - no different 
to running jobs from Cron etc that dont have a full shell environment.


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Re: [asterisk-users] Asterisk 1.6.2 ?

[Ben Dinnerville]

This is usually due to an error with the SIP stack not being loaded due 
to an error - make sure that full logging is on and check your log file 
and search for ERROR and see if there is any mention to SIP (chan_sip.o 
etc), alternatively, start asterisk from the command like with asterisk 
-vdc and watch the output to screen for any errors at 
startup. Fix the error and SIP will start up.


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[asterisk-users] CDR / billsec / originate / local chan

[Ben Dinnerville]

 to how it was working in 
1.4.20? I have been through a number of bug reports, mailing lists 
posts, web sites etc and there seems to be a number of issues reported 
over the years relating to billsec / h exten / originate but it seems to 
change from version to version.


Thanks in advance,

Ben


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Re: [asterisk-users] Smallest possible Asterisk VM

[Ben Schorr]

I think Astlinux comes in under 100MB.

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
www.rolandschorr.com
b...@rolandschorr.com
Twitter: http://www.twitter.com/bschorr
Facebook: http://www.facebook.com/rolandschorr 

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
 boun...@lists.digium.com] On Behalf Of Frank Church
 Sent: Monday, February 01, 2010 19:41
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Smallest possible Asterisk VM
 
 How small can an Asterisk system be, in terms of disk space utilized?
 
 I am looking for just asterisk, with mysql, postgresql, or sqlite,
with PHP and
 Python.
 
 After finishing the build and removing the tools, how small can the
whole
 system be?
 
 100Mb, 200Mb?
 
 Can packages be used to build the whole system, like using debs and
rpms
 alone?
 
 /vfclists
 
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Re: [asterisk-users] Can't get G.729 to work...

[Ben Schorr]

Really?  I didn't see them in the web interface; which is why I turned
to editing the files.  I'll check the web interface again, perhaps I
simply missed them.

Best wishes and aloha, 

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
www.rolandschorr.com
b...@rolandschorr.com


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
 boun...@lists.digium.com] On Behalf Of Andres
 Sent: Wednesday, December 16, 2009 5:43 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Can't get G.729 to work...
 
 Ben Schorr wrote:
 
 O.K., I restored the Allow=ulaw in the sip_general_additional.conf
 file, then I found the individual extension settings in the
 sip_additional.conf file and I added
 
 
 I would not go editing the individual files if you are using FreePBX.
 As soon as you make a change in the web interface it will override any
 manual changes you made.
 
 Simply do it in the web interface for each extension.  You do have a
 parameter called allow and another called disallow in the web
interface when
 editing the extension (its under device options).  Use them.  For
multiple
 entries just separate them with a comma.
 
 Andres
 http://www.neuroredes.com
 
 disallow=all
 allow=g729
 
 to each of the extensions at the remote site.  Then I did a SIP
RELOAD.
 So we'll see how that goes.
 
 Thanks again for the assist - this has been quite an education.
 
 Ben M. Schorr
 Chief Executive Officer
 __
 Roland Schorr  Tower
 www.rolandschorr.com
 b...@rolandschorr.com
 
 
 
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
 boun...@lists.digium.com] On Behalf Of Dave Fullerton
 Sent: Tuesday, December 15, 2009 11:47 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Can't get G.729 to work...
 
 I don't know how FreePBX works, but with vanilla Asterisk you would
do
 something like this with your sip.conf:
 
 [general]
 disallow=all
 allow=ulaw
 allow=g729
 
 [localA]
 callerid=Local phone A 100
 username=localA
 secret=blahblah1
 
 [localB]
 callerid=Local phone B 101
 username=localB
 secret=blah1blah
 
 [remoteA]
 callerid=Remote phone A 102
 disallow=all
 allow=g729
 username=remoteA
 secret=123456
 
 [remoteB]
 callerid=Remote phone B 103
 disallow=all
 allow=g729
 username=remoteB
 secret=654321
 
 You can do this using templates as well, but this will make it
easier
 
 
 to
 
 
 understand. See the disallow/allow lines on the remote peers? Those
 override the settings in the general portion of your sip.conf. With
 
 
 these
 
 
 settings the local phones will use ulaw by default and allow g729
when
 needed.
 
 This will do what you want for the most part. Local phones will use
 
 
 ulaw for all
 
 
 calls between themselves and calls in and out of the PRI. Calls from
a
 
 
 remote
 
 
 phone to a local phone will use g.729 end to end. Calls from a local
 
 
 phone to a
 
 
 remote phone will use ulaw between the local phone and asterisk and
 
 
 g.729
 
 
 between asterisk and the remote phone (this is a limitation of
 
 
 asterisk's
 
 
 codec negotiation). Calls from remote phones will use g.729 all the
 
 
 time.
 
 
 I'm sure there is a way to do this through the freepbx gui, but like
I
 
 
 said, I
 
 
 have no experience with freepbx.
 
 -Dave
 
 
 
 Ben Schorr wrote:
 
 
 O.K., I think I'm catching on.  I only have a single SIP.CONF file
 that ALL of the extensions are using so I'm gathering that I need
to
 set up a separate SIP.CONF file (or perhaps just an included file)
 
 
 for
 
 
 the 8 users at the remote office which ONLY Allows the G.729.
 
 So now I'm figuring out how to do that.
 
 Ben M. Schorr
 Chief Executive Officer
 __
 Roland Schorr  Tower
 www.rolandschorr.com
 b...@rolandschorr.com
 
 
 
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 
 
 [mailto:asterisk-users-
 
 
 boun...@lists.digium.com] On Behalf Of Dave Fullerton
 Sent: Tuesday, December 15, 2009 11:05 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Can't get G.729 to work...
 
 That's a bit misleading. Yes calls that travel over a PRI will be
 
 
 using ulaw, but
 
 
 only over the PRI leg of the call. The SIP leg can still be using
 
 
 G.729 with
 
 
 asterisk transcoding between the two legs.
 
 Ben, You haven't shown us the contents of your sip.conf file for
 
 
 the
 
 
 peers
 
 
 you are working on but I have a guess as to what is going on. In
 
 
 one
 
 
 of your
 
 
 previous messages you state: I moved G.729 to the top of that
list
 
 
 (just
 
 
 below disallow) I'm guessing your list looks something like this:
 
 disallow=all
 allow=g729
 allow=ulaw
 allow={maybe something else}
 
 This will be fine for all the phones

[asterisk-users] Can't get G.729 to work...

[Ben Schorr]

Asterisk 1.4 - FreePBX - Polycom 330 and 501 phones.

 

I've got G.729 loaded in the modules on the Asterisk server and on the
Polycom phones I've set G.729 to be the first preference of codec, but
still when I go SIP SHOW CHANNELS during active calls it still shows
(ULAW) (G.711) as the codec in use.

 

I'm a newbie at Asterisk, can anybody suggest what I might be
overlooking?

 

Best wishes and aloha, 

 

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
1155 Fort Street Mall
Honolulu, Hawaii 96813
Mobile:  808-782-6306
Fax: 808-533-3677
www.rolandschorr.com http://www.rolandschorr.com/ 
b...@rolandschorr.com mailto:b...@rolandschorr.com 

 

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Re: [asterisk-users] Can't get G.729 to work...

[Ben Schorr]

Ahhh...yes, I think that may have been it.  I moved G.729 to the top of
that list (just below disallow) and now I have a restart when
convenient pending.  Is that sufficient or do I have to actually reboot
the server for the change to take effect?

Best wishes and aloha, 

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
www.rolandschorr.com
b...@rolandschorr.com

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
 boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere
 Sent: Tuesday, December 15, 2009 8:30 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Can't get G.729 to work...
 
 
 On Tue, 15 Dec 2009, Ben Schorr wrote:
 
  Asterisk 1.4 - FreePBX - Polycom 330 and 501 phones.
 
 
 
  I've got G.729 loaded in the modules on the Asterisk server and on
the
  Polycom phones I've set G.729 to be the first preference of codec,
but
  still when I go SIP SHOW CHANNELS during active calls it still shows
  (ULAW) (G.711) as the codec in use.
 
 
 
  I'm a newbie at Asterisk, can anybody suggest what I might be
  overlooking?
 
 
 In the sip.conf entry for your peer you need to specify the codec
negotiation
 order.  Though you put g.729 first on the phone, asterisk probably has
ulaw
 first, and is taking precedence.  In the sip.conf entry put this:
 
 disallow=all
 allow=g729
 
 j
 
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Re: [asterisk-users] Can't get G.729 to work...

[Ben Schorr]

O.K., thanks, I'm catching on (slowly).  Waiting for the next call to
see if the SIP.CONF change did the trick.

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
www.rolandschorr.com
b...@rolandschorr.com


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
 boun...@lists.digium.com] On Behalf Of Danny Nicholas
 Sent: Tuesday, December 15, 2009 9:15 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Can't get G.729 to work...
 
 You should only need a reboot for DAHDI changes (not always then...)
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben
Schorr
 Sent: Tuesday, December 15, 2009 1:08 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Can't get G.729 to work...
 
 Ahhh...yes, I think that may have been it.  I moved G.729 to the top
of that
 list (just below disallow) and now I have a restart when convenient
 pending.  Is that sufficient or do I have to actually reboot the
server for the
 change to take effect?
 
 Best wishes and aloha,
 
 Ben M. Schorr
 Chief Executive Officer
 __
 Roland Schorr  Tower
 www.rolandschorr.com
 b...@rolandschorr.com
 
  -Original Message-
  From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
  boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere
  Sent: Tuesday, December 15, 2009 8:30 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] Can't get G.729 to work...
 
 
  On Tue, 15 Dec 2009, Ben Schorr wrote:
 
   Asterisk 1.4 - FreePBX - Polycom 330 and 501 phones.
  
  
  
   I've got G.729 loaded in the modules on the Asterisk server and on
 the
   Polycom phones I've set G.729 to be the first preference of codec,
 but
   still when I go SIP SHOW CHANNELS during active calls it still
shows
   (ULAW) (G.711) as the codec in use.
  
  
  
   I'm a newbie at Asterisk, can anybody suggest what I might be
   overlooking?
  
 
  In the sip.conf entry for your peer you need to specify the codec
 negotiation
  order.  Though you put g.729 first on the phone, asterisk probably
has
 ulaw
  first, and is taking precedence.  In the sip.conf entry put this:
 
  disallow=all
  allow=g729
 
  j
 
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http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
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Re: [asterisk-users] Can't get G.729 to work...

[Ben Schorr]

O.K., interestingly enough when I call our extensions from my mobile
phone it still seems to be using ULAW, but when they dial out it seems
to be using G.729 now.

Is there something in Dahdi that I need to configure so that inbound
calls (from the PRI on a Digium TE205) use G.729 to get to the phones
too?

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
www.rolandschorr.com
b...@rolandschorr.com


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
 boun...@lists.digium.com] On Behalf Of j...@jeff.net
 Sent: Tuesday, December 15, 2009 9:13 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Can't get G.729 to work...
 
 
 
 On Tue, 15 Dec 2009, Ben Schorr wrote:
 
  Ahhh...yes, I think that may have been it.  I moved G.729 to the top
  of that list (just below disallow) and now I have a restart when
  convenient pending.  Is that sufficient or do I have to actually
  reboot the server for the change to take effect?
 
 Just do a sip reload at the asterisk CLI prompt and you will be good
to go.  It
 won't cutoff any calls in progress.  Then reboot your phone.
 
 Cheers,
 
 j
 
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Re: [asterisk-users] Can't get G.729 to work...

[Ben Schorr]

Sorry, I think I may have misspoke...

What I'm hoping for is that all of the connections between my phones (or
at least a particular group of them) and my Asterisk server will use
G.729.  Currently it seems like it usually is, but not always, and I
haven't figured out the pattern.

All of our calls fall into two categories:

Internal calls - one extension to another within our single Asterisk
server org.
External calls - To/From one of our extensions out thru the PRI line to
our carrier (Hawaiian Tel) to phone numbers out in the world.

For some reason it appears that inbound calls from out in the world are
going to our phones using ULAW, but outbound calls to the world are
using G.729.

That's progress but...how can I get my Asterisk server to use G.729 to
pass those incoming calls to my phones?

Best wishes and aloha, 

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
www.rolandschorr.com
b...@rolandschorr.com


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
 boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere
 Sent: Tuesday, December 15, 2009 9:54 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Can't get G.729 to work...
 
 
 On Tue, 15 Dec 2009, Ben Schorr wrote:
 
  O.K., interestingly enough when I call our extensions from my mobile
  phone it still seems to be using ULAW, but when they dial out it
seems
  to be using G.729 now.
 
  Is there something in Dahdi that I need to configure so that inbound
  calls (from the PRI on a Digium TE205) use G.729 to get to the
phones
  too?
 
 A Dahdi channel over a PRI will always be ulaw - that is the encoding
on the
 PRI (at least in the US).  If your phones are using G.729 then
transcoding will
 be taking place within asterisk for the bridge between the channels.
 
 My guess is you are looking at the PRI channel.  There should be
another
 channel for the phone.  That should always be G.729 now.
 
 Cheers,
 
 j
 
 
  Ben M. Schorr
  Chief Executive Officer
  __
  Roland Schorr  Tower
  www.rolandschorr.com
  b...@rolandschorr.com
 
 
  -Original Message-
  From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
  boun...@lists.digium.com] On Behalf Of j...@jeff.net
  Sent: Tuesday, December 15, 2009 9:13 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] Can't get G.729 to work...
 
 
 
  On Tue, 15 Dec 2009, Ben Schorr wrote:
 
  Ahhh...yes, I think that may have been it.  I moved G.729 to the
top
  of that list (just below disallow) and now I have a restart when
  convenient pending.  Is that sufficient or do I have to actually
  reboot the server for the change to take effect?
 
  Just do a sip reload at the asterisk CLI prompt and you will be
  good
  to go.  It
  won't cutoff any calls in progress.  Then reboot your phone.
 
  Cheers,
 
  j
 
  ___
  -- Bandwidth and Colocation Provided by http://www.api-digital.com
--
 
  asterisk-users mailing list
  To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users
 
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  To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
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Re: [asterisk-users] Can't get G.729 to work...

[Ben Schorr]

Oh, dear.  So my users with less-than-ideal bandwidth are stuck with
drop-outs and poor sound quality because they can't use the reduced
bandwidth codec for those calls?  :-(

They've been complaining that they often end up on a call where one or
both parties are cutting in and out.  Unfortunately it's only this one
remote site, with about 8 users, who connect across a VPN to the site
where the server is.  We've tried increasing their bandwidth and
tweaking the QOS settings on their firewalls but so far we haven't been
able to solve it.  I was hoping that switching to a lower bandwidth
CODEC would give them the call reliability they need.

If not then I guess I'm back to the drawing board, with increasingly
impatient users, trying to troubleshoot their call quality issues.

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
www.rolandschorr.com
b...@rolandschorr.com


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
 boun...@lists.digium.com] On Behalf Of Danny Nicholas
 Sent: Tuesday, December 15, 2009 10:19 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Can't get G.729 to work...
 
 IMO you can only use the G.729 on a SIP call.  If the call falls onto
the PRI
 framework, ulaw will be forced.
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben
Schorr
 Sent: Tuesday, December 15, 2009 2:11 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Can't get G.729 to work...
 
 Sorry, I think I may have misspoke...
 
 What I'm hoping for is that all of the connections between my phones
(or at
 least a particular group of them) and my Asterisk server will use
G.729.
 Currently it seems like it usually is, but not always, and I haven't
figured out
 the pattern.
 
 All of our calls fall into two categories:
 
 Internal calls - one extension to another within our single Asterisk
server org.
 External calls - To/From one of our extensions out thru the PRI line
to our
 carrier (Hawaiian Tel) to phone numbers out in the world.
 
 For some reason it appears that inbound calls from out in the world
are going
 to our phones using ULAW, but outbound calls to the world are using
G.729.
 
 That's progress but...how can I get my Asterisk server to use G.729 to
pass
 those incoming calls to my phones?
 
 Best wishes and aloha,
 
 Ben M. Schorr
 Chief Executive Officer
 __
 Roland Schorr  Tower
 www.rolandschorr.com
 b...@rolandschorr.com
 
 
  -Original Message-
  From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
  boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere
  Sent: Tuesday, December 15, 2009 9:54 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] Can't get G.729 to work...
 
 
  On Tue, 15 Dec 2009, Ben Schorr wrote:
 
   O.K., interestingly enough when I call our extensions from my
mobile
   phone it still seems to be using ULAW, but when they dial out it
 seems
   to be using G.729 now.
  
   Is there something in Dahdi that I need to configure so that
inbound
   calls (from the PRI on a Digium TE205) use G.729 to get to the
 phones
   too?
 
  A Dahdi channel over a PRI will always be ulaw - that is the
encoding
 on the
  PRI (at least in the US).  If your phones are using G.729 then
 transcoding will
  be taking place within asterisk for the bridge between the channels.
 
  My guess is you are looking at the PRI channel.  There should be
 another
  channel for the phone.  That should always be G.729 now.
 
  Cheers,
 
  j
 
  
   Ben M. Schorr
   Chief Executive Officer
   __
   Roland Schorr  Tower
   www.rolandschorr.com
   b...@rolandschorr.com
  
  
   -Original Message-
   From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-
   boun...@lists.digium.com] On Behalf Of j...@jeff.net
   Sent: Tuesday, December 15, 2009 9:13 AM
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   Subject: Re: [asterisk-users] Can't get G.729 to work...
  
  
  
   On Tue, 15 Dec 2009, Ben Schorr wrote:
  
   Ahhh...yes, I think that may have been it.  I moved G.729 to the
 top
   of that list (just below disallow) and now I have a restart
when
   convenient pending.  Is that sufficient or do I have to
actually
   reboot the server for the change to take effect?
  
   Just do a sip reload at the asterisk CLI prompt and you will be
   good
   to go.  It
   won't cutoff any calls in progress.  Then reboot your phone.
  
   Cheers,
  
   j
  
   ___
   -- Bandwidth and Colocation Provided by
http://www.api-digital.com
 --
  
   asterisk-users mailing list
   To UNSUBSCRIBE or update options visit

Re: [asterisk-users] Can't get G.729 to work...

[Ben Schorr]

Well, I know I still have a LOT to learn about Asterisk but...how will
they get their incoming phone calls from their DIDs (which the TelCo
sends to their PRI) if I move the remote office onto a SIP provider?

The PRI doesn't seem to cause any problem for the majority of the users
(at the home site) it's just the 8 users at the remote site who are
complaining of quality issues.

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
www.rolandschorr.com
b...@rolandschorr.com


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
 boun...@lists.digium.com] On Behalf Of Danny Nicholas
 Sent: Tuesday, December 15, 2009 10:31 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Can't get G.729 to work...
 
 Why not restrict these 8 users to a SIP provider like (but not)
 bandwidth.com?  By eliminating the PRI element, you should completely
 resolve the problem.
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben
Schorr
 Sent: Tuesday, December 15, 2009 2:26 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Can't get G.729 to work...
 
 Oh, dear.  So my users with less-than-ideal bandwidth are stuck with
drop-
 outs and poor sound quality because they can't use the reduced
bandwidth
 codec for those calls?  :-(
 
 They've been complaining that they often end up on a call where one or
both
 parties are cutting in and out.  Unfortunately it's only this one
remote site,
 with about 8 users, who connect across a VPN to the site where the
server is.
 We've tried increasing their bandwidth and tweaking the QOS settings
on
 their firewalls but so far we haven't been able to solve it.  I was
hoping that
 switching to a lower bandwidth CODEC would give them the call
reliability
 they need.
 
 If not then I guess I'm back to the drawing board, with increasingly
impatient
 users, trying to troubleshoot their call quality issues.
 
 Ben M. Schorr
 Chief Executive Officer
 __
 Roland Schorr  Tower
 www.rolandschorr.com
 b...@rolandschorr.com
 
 
  -Original Message-
  From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
  boun...@lists.digium.com] On Behalf Of Danny Nicholas
  Sent: Tuesday, December 15, 2009 10:19 AM
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
  Subject: Re: [asterisk-users] Can't get G.729 to work...
 
  IMO you can only use the G.729 on a SIP call.  If the call falls
onto
 the PRI
  framework, ulaw will be forced.
 
  -Original Message-
  From: asterisk-users-boun...@lists.digium.com
  [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben
 Schorr
  Sent: Tuesday, December 15, 2009 2:11 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] Can't get G.729 to work...
 
  Sorry, I think I may have misspoke...
 
  What I'm hoping for is that all of the connections between my phones
 (or at
  least a particular group of them) and my Asterisk server will use
 G.729.
  Currently it seems like it usually is, but not always, and I haven't
 figured out
  the pattern.
 
  All of our calls fall into two categories:
 
  Internal calls - one extension to another within our single Asterisk
 server org.
  External calls - To/From one of our extensions out thru the PRI line
 to our
  carrier (Hawaiian Tel) to phone numbers out in the world.
 
  For some reason it appears that inbound calls from out in the world
 are going
  to our phones using ULAW, but outbound calls to the world are using
 G.729.
 
  That's progress but...how can I get my Asterisk server to use G.729
to
 pass
  those incoming calls to my phones?
 
  Best wishes and aloha,
 
  Ben M. Schorr
  Chief Executive Officer
  __
  Roland Schorr  Tower
  www.rolandschorr.com
  b...@rolandschorr.com
 
 
   -Original Message-
   From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-
   boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere
   Sent: Tuesday, December 15, 2009 9:54 AM
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   Subject: Re: [asterisk-users] Can't get G.729 to work...
  
  
   On Tue, 15 Dec 2009, Ben Schorr wrote:
  
O.K., interestingly enough when I call our extensions from my
 mobile
phone it still seems to be using ULAW, but when they dial out it
  seems
to be using G.729 now.
   
Is there something in Dahdi that I need to configure so that
 inbound
calls (from the PRI on a Digium TE205) use G.729 to get to the
  phones
too?
  
   A Dahdi channel over a PRI will always be ulaw - that is the
 encoding
  on the
   PRI (at least in the US).  If your phones are using G.729 then
  transcoding will
   be taking place

Re: [asterisk-users] Can't get G.729 to work...

[Ben Schorr]

I thought I already did that - which is how they now get some (but not
yet all) of their calls on G.729.  scratching head

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
www.rolandschorr.com
b...@rolandschorr.com


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
 boun...@lists.digium.com] On Behalf Of Tim Nelson
 Sent: Tuesday, December 15, 2009 10:29 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Can't get G.729 to work...
 
 - Ben Schorr b...@rolandschorr.com wrote:
  Oh, dear.  So my users with less-than-ideal bandwidth are stuck
with
  drop-outs and poor sound quality because they can't use the reduced
  bandwidth codec for those calls?  :-(
 
  They've been complaining that they often end up on a call where one
or
  both parties are cutting in and out.  Unfortunately it's only this
  one remote site, with about 8 users, who connect across a VPN to the
  site where the server is.  We've tried increasing their bandwidth
and
  tweaking the QOS settings on their firewalls but so far we haven't
  been able to solve it.  I was hoping that switching to a lower
  bandwidth CODEC would give them the call reliability they need.
 
  If not then I guess I'm back to the drawing board, with increasingly
  impatient users, trying to troubleshoot their call quality issues.
 
 
 You need to install the G.729a codec on your system so that it will
transcode
 your calls from ulaw (on your PRI side) to g729 (on your SIP side).
Keep in
 mind that G.729 is a patented codec which requires licensing. The two
 companies offering G.729 for Asterisk(that I know of, please correct
me if
 there are others :-) ) are here:
 
 http://store.digium.com/productview.php?category_id=5product_code=8
 G729CODEC
 http://www.howlertech.com/products/howlets/
 
 I've always used the Digium G.729 and it has worked flawlessly. I've
also
 heard good things about Howler G.729 but never used it personally.
 
 Tim Nelson
 Systems/Network Support
 Rockbochs Inc.
 (218)727-4332 x105
 
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Re: [asterisk-users] Can't get G.729 to work...

[Ben Schorr]

Yes, the routers are another issue we're dealing with.  We've configured
them to prioritize traffic to/from our Asterisk server but I'm not
convinced that setting is really working as expected.  So we're working
with the vendor on that.

The effective bandwidth is 6Mb/sec down x 1.4Mb/sec up (so basically
1.4x1.4 on the VPN).  For 8 users, where rarely more than 2-3 of them
are on the phone at any given time, that should be sufficient I think.

They DO have to share the connection with their web browsing and e-mail
and such but as best we've been able to tell they aren't saturating
their connections - usually not more than 4-5 of the 8 are using their
computers at any given time and most of them just do e-mail and local
apps that shouldn't touch the Internet connection.

Frankly I'm puzzled that they have these issues and the problems rarely
seem to happen when I call them.  I'll go to their site and make a few
calls from one of their phones and it sounds perfect to me.  But three
days later all I hear is how frustrated they are because these new VOIP
phones suck and they can never hear anybody and...  sigh

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
www.rolandschorr.com
b...@rolandschorr.com


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
 boun...@lists.digium.com] On Behalf Of Danny Nicholas
 Sent: Tuesday, December 15, 2009 10:54 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Can't get G.729 to work...
 
 Do your routers allow giving these users maximum priority?  What is
the
 effective bandwidth on the VPN connection?
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben
Schorr
 Sent: Tuesday, December 15, 2009 2:47 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Can't get G.729 to work...
 
 I thought I already did that - which is how they now get some (but not
yet all)
 of their calls on G.729.  scratching head
 
 Ben M. Schorr
 Chief Executive Officer
 __
 Roland Schorr  Tower
 www.rolandschorr.com
 b...@rolandschorr.com
 
 
  -Original Message-
  From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
  boun...@lists.digium.com] On Behalf Of Tim Nelson
  Sent: Tuesday, December 15, 2009 10:29 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] Can't get G.729 to work...
 
  - Ben Schorr b...@rolandschorr.com wrote:
   Oh, dear.  So my users with less-than-ideal bandwidth are stuck
 with
   drop-outs and poor sound quality because they can't use the
reduced
   bandwidth codec for those calls?  :-(
  
   They've been complaining that they often end up on a call where
one
 or
   both parties are cutting in and out.  Unfortunately it's only
this
   one remote site, with about 8 users, who connect across a VPN to
the
   site where the server is.  We've tried increasing their bandwidth
 and
   tweaking the QOS settings on their firewalls but so far we haven't
   been able to solve it.  I was hoping that switching to a lower
   bandwidth CODEC would give them the call reliability they need.
  
   If not then I guess I'm back to the drawing board, with
increasingly
   impatient users, trying to troubleshoot their call quality issues.
  
 
  You need to install the G.729a codec on your system so that it will
 transcode
  your calls from ulaw (on your PRI side) to g729 (on your SIP side).
 Keep in
  mind that G.729 is a patented codec which requires licensing. The
two
  companies offering G.729 for Asterisk(that I know of, please correct
 me if
  there are others :-) ) are here:
 
 
 http://store.digium.com/productview.php?category_id=5product_code=8
  G729CODEC
  http://www.howlertech.com/products/howlets/
 
  I've always used the Digium G.729 and it has worked flawlessly. I've
 also
  heard good things about Howler G.729 but never used it personally.
 
  Tim Nelson
  Systems/Network Support
  Rockbochs Inc.
  (218)727-4332 x105
 
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 http://lists.digium.com/mailman/listinfo/asterisk-users
 
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Re: [asterisk-users] Can't get G.729 to work...

[Ben Schorr]

O.K., I think I'm catching on.  I only have a single SIP.CONF file that
ALL of the extensions are using so I'm gathering that I need to set up a
separate SIP.CONF file (or perhaps just an included file) for the 8
users at the remote office which ONLY Allows the G.729.

So now I'm figuring out how to do that.

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
www.rolandschorr.com
b...@rolandschorr.com


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
 boun...@lists.digium.com] On Behalf Of Dave Fullerton
 Sent: Tuesday, December 15, 2009 11:05 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Can't get G.729 to work...
 
 That's a bit misleading. Yes calls that travel over a PRI will be
using ulaw, but
 only over the PRI leg of the call. The SIP leg can still be using
G.729 with
 asterisk transcoding between the two legs.
 
 Ben, You haven't shown us the contents of your sip.conf file for the
peers
 you are working on but I have a guess as to what is going on. In one
of your
 previous messages you state: I moved G.729 to the top of that list
(just
 below disallow) I'm guessing your list looks something like this:
 
 disallow=all
 allow=g729
 allow=ulaw
 allow={maybe something else}
 
 This will be fine for all the phones in the office but the remote
phones need
 to ONLY have disallow=all and allow=g729 in their entries in sip.conf
as Jeff's
 reply stated. By having the allow=ulaw entry in there you are giving
asterisk
 permission to allow any call that is already in the ulaw format (calls
from the
 PRI) to remain in that format when contacting your remote phones. If
you're
 still stick post your sip.conf (with the passwords removed) and we can
help
 you out.
 
 -Dave
 
 
 Danny Nicholas wrote:
  IMO you can only use the G.729 on a SIP call.  If the call falls
onto the
  PRI framework, ulaw will be forced.
 
  -Original Message-
  From: asterisk-users-boun...@lists.digium.com
  [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben
Schorr
  Sent: Tuesday, December 15, 2009 2:11 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] Can't get G.729 to work...
 
  Sorry, I think I may have misspoke...
 
  What I'm hoping for is that all of the connections between my phones
(or
  at least a particular group of them) and my Asterisk server will use
  G.729.  Currently it seems like it usually is, but not always, and I
  haven't figured out the pattern.
 
  All of our calls fall into two categories:
 
  Internal calls - one extension to another within our single Asterisk
  server org.
  External calls - To/From one of our extensions out thru the PRI line
to
  our carrier (Hawaiian Tel) to phone numbers out in the world.
 
  For some reason it appears that inbound calls from out in the world
are
  going to our phones using ULAW, but outbound calls to the world are
  using G.729.
 
  That's progress but...how can I get my Asterisk server to use G.729
to
  pass those incoming calls to my phones?
 
  Best wishes and aloha,
 
  Ben M. Schorr
  Chief Executive Officer
  __
  Roland Schorr  Tower
  www.rolandschorr.com
  b...@rolandschorr.com
 
 
  -Original Message-
  From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
  boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere
  Sent: Tuesday, December 15, 2009 9:54 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] Can't get G.729 to work...
 
 
  On Tue, 15 Dec 2009, Ben Schorr wrote:
 
  O.K., interestingly enough when I call our extensions from my
mobile
  phone it still seems to be using ULAW, but when they dial out it
  seems
  to be using G.729 now.
 
  Is there something in Dahdi that I need to configure so that
inbound
  calls (from the PRI on a Digium TE205) use G.729 to get to the
  phones
  too?
  A Dahdi channel over a PRI will always be ulaw - that is the
encoding
  on the
  PRI (at least in the US).  If your phones are using G.729 then
  transcoding will
  be taking place within asterisk for the bridge between the
channels.
 
  My guess is you are looking at the PRI channel.  There should be
  another
  channel for the phone.  That should always be G.729 now.
 
  Cheers,
 
  j
 
  Ben M. Schorr
  Chief Executive Officer
  __
  Roland Schorr  Tower
  www.rolandschorr.com
  b...@rolandschorr.com
 
 
  -Original Message-
  From: asterisk-users-boun...@lists.digium.com
  [mailto:asterisk-users-
  boun...@lists.digium.com] On Behalf Of j...@jeff.net
  Sent: Tuesday, December 15, 2009 9:13 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] Can't get G.729 to work...
 
 
 
  On Tue, 15 Dec 2009, Ben Schorr wrote:
 
  Ahhh...yes, I think

Re: [asterisk-users] Can't get G.729 to work...

[Ben Schorr]

O.K., so for now (as a test) I just commented out the allow=ULAW line
in the SIP.conf (actually it's sip_general_additional.conf on this
FreePBX box) and that does seem to be forcing all traffic to G.729.

I think ultimately I'd like to let the local users use ULAW because it
seems to sound better and just force the 8 remote users to use G.729,
but for now I can live with this while I figure out how to do that.

Thanks for all of your help - and I welcome any additional pointers
you'd like to offer.

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
www.rolandschorr.com
b...@rolandschorr.com


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
 boun...@lists.digium.com] On Behalf Of Ben Schorr
 Sent: Tuesday, December 15, 2009 11:16 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Can't get G.729 to work...
 
 O.K., I think I'm catching on.  I only have a single SIP.CONF file
that ALL of the
 extensions are using so I'm gathering that I need to set up a separate
 SIP.CONF file (or perhaps just an included file) for the 8 users at
the remote
 office which ONLY Allows the G.729.
 
 So now I'm figuring out how to do that.
 
 Ben M. Schorr
 Chief Executive Officer
 __
 Roland Schorr  Tower
 www.rolandschorr.com
 b...@rolandschorr.com
 
 
  -Original Message-
  From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
  boun...@lists.digium.com] On Behalf Of Dave Fullerton
  Sent: Tuesday, December 15, 2009 11:05 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] Can't get G.729 to work...
 
  That's a bit misleading. Yes calls that travel over a PRI will be
 using ulaw, but
  only over the PRI leg of the call. The SIP leg can still be using
 G.729 with
  asterisk transcoding between the two legs.
 
  Ben, You haven't shown us the contents of your sip.conf file for the
 peers
  you are working on but I have a guess as to what is going on. In one
 of your
  previous messages you state: I moved G.729 to the top of that list
 (just
  below disallow) I'm guessing your list looks something like this:
 
  disallow=all
  allow=g729
  allow=ulaw
  allow={maybe something else}
 
  This will be fine for all the phones in the office but the remote
 phones need
  to ONLY have disallow=all and allow=g729 in their entries in
sip.conf
 as Jeff's
  reply stated. By having the allow=ulaw entry in there you are giving
 asterisk
  permission to allow any call that is already in the ulaw format
(calls
 from the
  PRI) to remain in that format when contacting your remote phones. If
 you're
  still stick post your sip.conf (with the passwords removed) and we
can
 help
  you out.
 
  -Dave
 
 
  Danny Nicholas wrote:
   IMO you can only use the G.729 on a SIP call.  If the call falls
 onto the
   PRI framework, ulaw will be forced.
  
   -Original Message-
   From: asterisk-users-boun...@lists.digium.com
   [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben
 Schorr
   Sent: Tuesday, December 15, 2009 2:11 PM
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   Subject: Re: [asterisk-users] Can't get G.729 to work...
  
   Sorry, I think I may have misspoke...
  
   What I'm hoping for is that all of the connections between my
phones
 (or
   at least a particular group of them) and my Asterisk server will
use
   G.729.  Currently it seems like it usually is, but not always, and
I
   haven't figured out the pattern.
  
   All of our calls fall into two categories:
  
   Internal calls - one extension to another within our single
Asterisk
   server org.
   External calls - To/From one of our extensions out thru the PRI
line
 to
   our carrier (Hawaiian Tel) to phone numbers out in the world.
  
   For some reason it appears that inbound calls from out in the
world
 are
   going to our phones using ULAW, but outbound calls to the world
are
   using G.729.
  
   That's progress but...how can I get my Asterisk server to use
G.729
 to
   pass those incoming calls to my phones?
  
   Best wishes and aloha,
  
   Ben M. Schorr
   Chief Executive Officer
   __
   Roland Schorr  Tower
   www.rolandschorr.com
   b...@rolandschorr.com
  
  
   -Original Message-
   From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-
   boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere
   Sent: Tuesday, December 15, 2009 9:54 AM
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   Subject: Re: [asterisk-users] Can't get G.729 to work...
  
  
   On Tue, 15 Dec 2009, Ben Schorr wrote:
  
   O.K., interestingly enough when I call our extensions from my
 mobile
   phone it still seems to be using ULAW, but when they dial out it
   seems
   to be using G.729 now.
  
   Is there something

Re: [asterisk-users] Can't get G.729 to work...

[Ben Schorr]

O.K., I restored the Allow=ulaw in the sip_general_additional.conf file,
then I found the individual extension settings in the
sip_additional.conf file and I added 

disallow=all
allow=g729

to each of the extensions at the remote site.  Then I did a SIP RELOAD.
So we'll see how that goes.

Thanks again for the assist - this has been quite an education.

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
www.rolandschorr.com
b...@rolandschorr.com


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
 boun...@lists.digium.com] On Behalf Of Dave Fullerton
 Sent: Tuesday, December 15, 2009 11:47 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Can't get G.729 to work...
 
 I don't know how FreePBX works, but with vanilla Asterisk you would do
 something like this with your sip.conf:
 
 [general]
 disallow=all
 allow=ulaw
 allow=g729
 
 [localA]
 callerid=Local phone A 100
 username=localA
 secret=blahblah1
 
 [localB]
 callerid=Local phone B 101
 username=localB
 secret=blah1blah
 
 [remoteA]
 callerid=Remote phone A 102
 disallow=all
 allow=g729
 username=remoteA
 secret=123456
 
 [remoteB]
 callerid=Remote phone B 103
 disallow=all
 allow=g729
 username=remoteB
 secret=654321
 
 You can do this using templates as well, but this will make it easier
to
 understand. See the disallow/allow lines on the remote peers? Those
 override the settings in the general portion of your sip.conf. With
these
 settings the local phones will use ulaw by default and allow g729 when
 needed.
 
 This will do what you want for the most part. Local phones will use
ulaw for all
 calls between themselves and calls in and out of the PRI. Calls from a
remote
 phone to a local phone will use g.729 end to end. Calls from a local
phone to a
 remote phone will use ulaw between the local phone and asterisk and
g.729
 between asterisk and the remote phone (this is a limitation of
asterisk's
 codec negotiation). Calls from remote phones will use g.729 all the
time.
 
 I'm sure there is a way to do this through the freepbx gui, but like I
said, I
 have no experience with freepbx.
 
 -Dave
 
 
 
 Ben Schorr wrote:
  O.K., I think I'm catching on.  I only have a single SIP.CONF file
  that ALL of the extensions are using so I'm gathering that I need to
  set up a separate SIP.CONF file (or perhaps just an included file)
for
  the 8 users at the remote office which ONLY Allows the G.729.
 
  So now I'm figuring out how to do that.
 
  Ben M. Schorr
  Chief Executive Officer
  __
  Roland Schorr  Tower
  www.rolandschorr.com
  b...@rolandschorr.com
 
 
  -Original Message-
  From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
  boun...@lists.digium.com] On Behalf Of Dave Fullerton
  Sent: Tuesday, December 15, 2009 11:05 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] Can't get G.729 to work...
 
  That's a bit misleading. Yes calls that travel over a PRI will be
  using ulaw, but
  only over the PRI leg of the call. The SIP leg can still be using
  G.729 with
  asterisk transcoding between the two legs.
 
  Ben, You haven't shown us the contents of your sip.conf file for
the
  peers
  you are working on but I have a guess as to what is going on. In
one
  of your
  previous messages you state: I moved G.729 to the top of that list
  (just
  below disallow) I'm guessing your list looks something like this:
 
  disallow=all
  allow=g729
  allow=ulaw
  allow={maybe something else}
 
  This will be fine for all the phones in the office but the remote
  phones need
  to ONLY have disallow=all and allow=g729 in their entries in
sip.conf
  as Jeff's
  reply stated. By having the allow=ulaw entry in there you are
giving
  asterisk
  permission to allow any call that is already in the ulaw format
  (calls
  from the
  PRI) to remain in that format when contacting your remote phones.
If
  you're
  still stick post your sip.conf (with the passwords removed) and we
  can
  help
  you out.
 
  -Dave
 
 
  Danny Nicholas wrote:
  IMO you can only use the G.729 on a SIP call.  If the call falls
  onto the
  PRI framework, ulaw will be forced.
 
  -Original Message-
  From: asterisk-users-boun...@lists.digium.com
  [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben
  Schorr
  Sent: Tuesday, December 15, 2009 2:11 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] Can't get G.729 to work...
 
  Sorry, I think I may have misspoke...
 
  What I'm hoping for is that all of the connections between my
phones
  (or
  at least a particular group of them) and my Asterisk server will
use
  G.729.  Currently it seems like it usually is, but not always, and
I
  haven't figured out the pattern.
 
  All of our calls fall into two categories

[asterisk-users] Today's problem: Inbound call routing

[Ben Schorr]

O.K., so AsteriskNow 1.4.26.2, FreePBX 2.6.0RC2.1  We have a Digium
TE205P connected to a single span if ISDN PRI.  The Telco has assigned
us two local numbers to test incoming calls.  I created an inbound route
for one of those DID's and assigned it to one of our extensions.  Sounds
simple enough.

 

Too simple, apparently, when I dial the number the caller gets a
recording that it's a non-working number and this is what I see in the
CLI:

 

Extension '8085255935' in context 'default' from '808xxx' does not
exist.  Rejecting call on channel 0/1, span 1

 

So...other than creating the inbound route and assigning it to an
extension I apparently have to do something else.  Any suggestions as to
what that might be?

 

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
1155 Fort Street Mall
Honolulu, Hawaii 96813
Mobile:  808-782-6306
Fax: 808-533-3677
www.rolandschorr.com http://www.rolandschorr.com/ 
b...@rolandschorr.com

 

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Re: [asterisk-users] Today's problem: Inbound call routing

[Ben Schorr]

 -Original Message-
  Too simple, apparently, when I dial the number the caller gets a
  recording that it's a non-working number and this is what I see in
the
  CLI:
 
  Extension '8085255935' in context 'default' from '808xxx' does
not
  exist.  Rejecting call on channel 0/1, span 1
 
 That is a pretty clear error message.

Yes, I thought so.  But how do I fix it?

  So...other than creating the inbound route and assigning it to an
  extension I apparently have to do something else.  Any suggestions
as
  to what that might be?
 
 You manage your dialplan with FreePBX. This mailing list supports
Asterisk. I
 have no problem with questions about FreePBX systems. But they should
 also be phrased as Asterisk questions. This is a FreePBX question.

I see, so this isn't an Asterisk problem it's a FreePBX problem?


Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
www.rolandschorr.com
b...@rolandschorr.com




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Re: [asterisk-users] Today's problem: Inbound call routing

[Ben Schorr]

Sorry, I'm brand new at Asterisk (and/or FreePBX).  I'm going to have to
figure out what all those things are before I can show them.

I'll have to get back to you.

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
www.rolandschorr.com
b...@rolandschorr.com


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
 boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
 Sent: Friday, October 09, 2009 9:54 AM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Today's problem: Inbound call routing
 
 On Fri, Oct 09, 2009 at 06:15:43AM -1000, Ben Schorr wrote:
   -Original Message-
Too simple, apparently, when I dial the number the caller gets a
recording that it's a non-working number and this is what I see
in
  the
CLI:
   
Extension '8085255935' in context 'default' from '808xxx'
does
  not
exist.  Rejecting call on channel 0/1, span 1
   
   That is a pretty clear error message.
 
  Yes, I thought so.  But how do I fix it?
 
So...other than creating the inbound route and assigning it to
an
extension I apparently have to do something else.  Any
suggestions
  as
to what that might be?
  
   You manage your dialplan with FreePBX. This mailing list supports
  Asterisk. I
   have no problem with questions about FreePBX systems. But they
   should also be phrased as Asterisk questions. This is a FreePBX
question.
 
  I see, so this isn't an Asterisk problem it's a FreePBX problem?
 
 Creating an inbound route is FreePBX speak. This is a FreePBX
question.
 Please ask an Asterisk question.
 
 For instance, show a dialplan trace, show the respective dialplan,
show the
 respective channel configuration.
 
 --
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
 
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[asterisk-users] Can dial long distance but not local?

[Ben Schorr]

AsteriskNOW 1.4.26.2 with a  Digium TE205P connected to an ISDN PRI
(single span).  I'm sure I just have something goofed up in the
dialplans?  I have a bunch of Polycom 331 IP phones connecting to the
server.  I can dial the other extensions in the system fine and I can
dial long distance outgoing but cannot seem to get it to dial local (7
digit) calls.

 

I see this in the CLI:

 

-- Executing [...@macro-dialout-trunk:19] Dial(SIP/801-09b6e498,
DAHDI/g1/5551212|300|) in new stack 
-- Requested transfer capability: 0x00 - SPEECH 
-- Called g1/5551212 
-- Channel 0/1, span 1 got hangup, cause 28 
-- Hungup 'DAHDI/1-1' 
== Everyone is busy/congested at this time (1:0/0/1) 
-- Executing [...@macro-dialout-trunk:20] Goto(SIP/801-09b6e498,
s-CHANUNAVAIL|1) in new stack 
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) 
-- Executing [s-chanunav...@macro-dialout-trunk:1]
GotoIf(SIP/801-09b6e498, 1?noreport) in new stack 
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3) 
-- Executing [s-chanunav...@macro-dialout-trunk:3]
NoOp(SIP/801-09b6e498, TRUNK Dial failed due to CHANUNAVAIL
(hangupcause: 2  - failing through to other trunks) in new stack 



Also when I check the PRI DEBUG I see an Error 28 which indicates an
invalid number format.  But I'm just sending 5551212, which should be
o.k.

 

I'm a newbie at this...any suggestions welcomed.

 

-Ben-

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Re: [asterisk-users] Can dial long distance but not local?

[Ben Schorr]

Thanks Cary,

 

If I send as 10 digits (dialing the local area code) I get an error from
the phone co that I've dialed an incorrect number.  Unfortunately I live
in Hawaii where the 808 area code covers the whole state but is treated
differently on neighbor-islands.

 

So if I dial 8085551212 I get information for the neighbor islands
rather than for our local island.

 

I think I tried the 1+ trick with the same results, but I'll give it
another try just in case I overlooked it.

 

 

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
www.rolandschorr.com http://www.rolandschorr.com/ 
b...@rolandschorr.com mailto:b...@rolandschorr.com 

Twitter: http://www.twitter.com/bschorr

 

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch
Sent: Wednesday, October 07, 2009 11:04 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Can dial long distance but not local?

 

Perhaps send it as 10 digits or 1+? 

 

Cary Fitch

 



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben Schorr
Sent: Wednesday, October 07, 2009 3:58 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Can dial long distance but not local?

 

AsteriskNOW 1.4.26.2 with a  Digium TE205P connected to an ISDN PRI
(single span).  I'm sure I just have something goofed up in the
dialplans?  I have a bunch of Polycom 331 IP phones connecting to the
server.  I can dial the other extensions in the system fine and I can
dial long distance outgoing but cannot seem to get it to dial local (7
digit) calls.

 

I see this in the CLI:

 

-- Executing [...@macro-dialout-trunk:19] Dial(SIP/801-09b6e498,
DAHDI/g1/5551212|300|) in new stack 
-- Requested transfer capability: 0x00 - SPEECH 
-- Called g1/5551212 
-- Channel 0/1, span 1 got hangup, cause 28 
-- Hungup 'DAHDI/1-1' 
== Everyone is busy/congested at this time (1:0/0/1) 
-- Executing [...@macro-dialout-trunk:20] Goto(SIP/801-09b6e498,
s-CHANUNAVAIL|1) in new stack 
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) 
-- Executing [s-chanunav...@macro-dialout-trunk:1]
GotoIf(SIP/801-09b6e498, 1?noreport) in new stack 
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3) 
-- Executing [s-chanunav...@macro-dialout-trunk:3]
NoOp(SIP/801-09b6e498, TRUNK Dial failed due to CHANUNAVAIL
(hangupcause: 2  - failing through to other trunks) in new stack 

Also when I check the PRI DEBUG I see an Error 28 which indicates an
invalid number format.  But I'm just sending 5551212, which should be
o.k.

 

I'm a newbie at this...any suggestions welcomed.

 

-Ben-

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Re: [asterisk-users] Can dial long distance but not local?

[Ben Schorr]

No, unfortunately not, our local area code is split among multiple
islands. You have to dial it to reach a neighbor island.  If you dial it
with a local number it tries to find that number on a neighbor island
(then usually fails).

I'm trying to find out from them if they want us to dial something else
first though - like a 1 or a 9.

I tried pridialplan=local as well as pridialplan=unknown.  No
improvement either way.


Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
www.rolandschorr.com
b...@rolandschorr.com
Twitter: http://www.twitter.com/bschorr



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Wednesday, October 07, 2009 11:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can dial long distance but not local?

Ben Schorr wrote:

 AsteriskNOW 1.4.26.2 with a Digium TE205P connected to an ISDN PRI 
 (single span). I'm sure I just have something goofed up in the 
 dialplans? I have a bunch of Polycom 331 IP phones connecting to the 
 server. I can dial the other extensions in the system fine and I can 
 dial long distance outgoing but cannot seem to get it to dial local (7
 digit) calls.


My guess is that your provider requires the full 10 digits even for 
local calls.

Doug


-- 
 
Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little
Temporary Safety, deserve neither Liberty nor Safety.


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[asterisk-users] Paging with Pickup

[Asaf Ben Aroch]

Hi,

I'm trying to achieve the following feature that's common in Avaya
systems:

 

A user page all extensions in a full duplex mode- he can hear all, and
all can hear him via their phones' speaker. When one of the extensions
picks up the handset, the call is bridged between the pager and the
person who picked up. All the rest are disconnected.

 

Does anyone have an idea or ever implemented such behavior? I was
thinking about adding an extra step for the picking up user to dial **
and do a dynamic feature that will run a script to kick all the rest
from the Meetme of the Page().

 

Thoughts?

 

A

 

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Re: [asterisk-users] Country numbering plan resources

[Ben Thompson]

On Sat, Dec 13, 2008 at 04:10:47PM +1300, Michael wrote:
 Is there any good free / accurate online resources with detailed country 
 numbering plans? Failing that let's get something running ourselves.

Here is some info for the UK :-

http://www.ofcom.org.uk/telecoms/ioi/numbers/numbers_administered/

I've spotted a small number of errors in the data, and I can give you
the details if you want.

Cheers

Ben

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Re: [asterisk-users] Server for 25-30 phones, sip trunks over the net

[Ben Hauger]

Well, we're running an asterisk 1.4.x system with a te220 span adapter 
(T1 PRI). 83 internal SIP peers, mostly Polycom Soundpoint IP series 
phones. It's a single-CPU dual-core Pentium E6420. The OS is CentOS 4.5 
(x86_64). Seems to work well, though it's only busy with call switching, 
voicemail, and call recording.

Cheers,
Ben

nb wrote:
 What cpu/memmory configurations have people had good luck with for a
 small office asterisk server... (polycom's/poe linksys)

 Dell's got their $200 SC440 going again until the 12th... and I'm
 thinking it might be just the ticket...

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Re: [asterisk-users] Overlap dialing via SIP

[Ben Thompson]

On Mon, Jul 21, 2008 at 05:10:15PM +0100, Ben Thompson wrote:

 [outbound-international]
 exten = _900XX,1,Set(oldexten=${EXTEN})
 exten = _900XX,2,Goto(international-number-length-check,s,1)
 
 [international-number-length-check]
 exten = s,1,Answer
 exten = s,2,WaitExten(8)
 
 exten = _X,1,Set(enddigits=${EXTEN})
 exten = _X,2,NoOp(${TIMESTAMP} ok 13 digits - we dial 
 ${oldexten}${enddigits})
 exten = _X,3,Dial(${OUTBOUNDTRUNK}/${oldexten}${enddigits})
 exten = _X,4,Congestion()
 exten = _X,104,Busy()
 
 exten = _XX,1,Set(enddigits=${EXTEN})
 exten = _XX,2,NoOp(${TIMESTAMP} ok 14 digits - we dial 
 ${oldexten}${enddigits})
 exten = _XX,3,Dial(${OUTBOUNDTRUNK}/${oldexten}${enddigits})
 exten = _XX,4,Congestion()
 exten = _XX,104,Busy()
 
 exten = _XXX,1,Set(enddigits=${EXTEN})
 exten = _XXX,2,NoOp(${TIMESTAMP} ok 15 digits - we dial 
 ${oldexten}${enddigits})
 exten = _XXX,3,Dial(${OUTBOUNDTRUNK}/${oldexten}${enddigits})
 exten = _XXX,4,Congestion()
 exten = _XXX,104,Busy()
 
 exten = t,1,Dial(${OUTBOUNDTRUNK}/${oldexten})
 exten = t,2,Congestion()
 exten = t,102,Busy()
 
 
 This works fairly well but I have noticed that occasionally the WaitExten 
 feature does
 not seem to catch the first digits if they are dialed too quickly. It is 
 almost as if
 there is a some sort of delay and the thirteenth digit is sometimes missed.


In answer to my own email I have found that the Background() function
works slightly better :-

[outbound-international]
exten = _900XX,1,Set(oldexten=${EXTEN})
exten = _900XX,2,Goto(international-number-length-check,s,1)

[international-number-length-check]
exten = s,1,Background()

exten = _X,1,Set(enddigits=${EXTEN})
exten = _X,2,NoOp(${TIMESTAMP} ok 13 digits - we dial ${oldexten}${enddigits})
exten = _X,3,Goto(international-dialout,${oldexten}${enddigits},1)

exten = _XX,1,Set(enddigits=${EXTEN})
exten = _XX,2,NoOp(${TIMESTAMP} ok 14 digits - we dial ${oldexten}${enddigits})
exten = _XX,3,Goto(international-dialout,${oldexten}${enddigits},1)

exten = _XXX,1,Set(enddigits=${EXTEN})
exten = _XXX,2,NoOp(${TIMESTAMP} ok 15 digits - we dial 
${oldexten}${enddigits})
exten = _XXX,3,Goto(international-dialout,${oldexten}${enddigits},1)

exten = t,1,NoOp(timeout so dial just 12 digits ${oldexten})
exten = t,2,Goto(international-dialout,${oldexten}${enddigits},1)

[international-dialout]
exten = _900XX,1,Macro(dialout-pstn)
exten = _900XXX,1,Macro(dialout-pstn)
exten = _900,1,Macro(dialout-pstn)
exten = _900X,1,Macro(dialout-pstn)


In general I have found that Overlap Dialing works very well and it is
a worthwhile feature to have. If there are any others in the UK
who would like to collaborate with me on maintaining an up to date list
of UK mappings please let me know. I would be happy to maintain a
webpage or somthing like that where people could access the info in an
asterisk friendly format.

Ben Thompson

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Re: [asterisk-users] Overlap dialing via SIP

[Ben Thompson]

On Tue, Jul 22, 2008 at 11:41:45AM +0100, Gordon Henderson wrote:

  [0906]
  exten = _90906XXX,1,Macro(dialout-pstn)
 
 york.ac.uk and you're allowing 0906 numbers? Where do I sign up ;-)

Err no, this is not my actual dialplan - just an example.


 Personally I think you're making life hard for yourself, although 
 potentially nice for the users, I guess.
 
 Or maybe you want to look at the ! match pattern, or just give-up on 
 overlap dialling. I like to be able to 'edit' numbers on my phone before 

OK, the ! match pattern sounds interesting. Does this allow
overlap dialing though?

Thanks

Ben Thompson

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[asterisk-users] Overlap dialing via SIP

[Ben Thompson]

Hi

I have set up an asterisk system which allows the use of Overlap Dialing from
SIP handsets. In order to do this I had to list the various patterns of numbers
which can be dialed in the UK. We also dial with a prefix of '9' for and outside
line so much of my dialplan looks like this :-

[084x]
exten = _9084,1,Macro(dialout-pstn)

[outbound-national]
exten = _90[1-2]X,1,Macro(dialout-pstn)

[087x]
exten = _9087,1,Macro(dialout-pstn)

[0906]
exten = _90906XXX,1,Macro(dialout-pstn)

...


I was able to download the mappings for 0800 numbers and other special ranges
from the ofcom website and I have incorporated these. For international dialing
I have not been able to find an easy way of doing this so I created the folling
contexts whcih make use of the WaitExten feature :-

[outbound-international]
exten = _900XX,1,Set(oldexten=${EXTEN})
exten = _900XX,2,Goto(international-number-length-check,s,1)

[international-number-length-check]
exten = s,1,Answer
exten = s,2,WaitExten(8)

exten = _X,1,Set(enddigits=${EXTEN})
exten = _X,2,NoOp(${TIMESTAMP} ok 13 digits - we dial ${oldexten}${enddigits})
exten = _X,3,Dial(${OUTBOUNDTRUNK}/${oldexten}${enddigits})
exten = _X,4,Congestion()
exten = _X,104,Busy()

exten = _XX,1,Set(enddigits=${EXTEN})
exten = _XX,2,NoOp(${TIMESTAMP} ok 14 digits - we dial ${oldexten}${enddigits})
exten = _XX,3,Dial(${OUTBOUNDTRUNK}/${oldexten}${enddigits})
exten = _XX,4,Congestion()
exten = _XX,104,Busy()

exten = _XXX,1,Set(enddigits=${EXTEN})
exten = _XXX,2,NoOp(${TIMESTAMP} ok 15 digits - we dial 
${oldexten}${enddigits})
exten = _XXX,3,Dial(${OUTBOUNDTRUNK}/${oldexten}${enddigits})
exten = _XXX,4,Congestion()
exten = _XXX,104,Busy()

exten = t,1,Dial(${OUTBOUNDTRUNK}/${oldexten})
exten = t,2,Congestion()
exten = t,102,Busy()


This works fairly well but I have noticed that occasionally the WaitExten 
feature does
not seem to catch the first digits if they are dialed too quickly. It is almost 
as if
there is a some sort of delay and the thirteenth digit is sometimes missed.

Can anyone suggest why WaitExten might be ocasionally missing a digit or can 
anyone think
of a better way of doing this?

Thanks

Ben Thompson



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[asterisk-users] One-way audio after music on hold

[Ben Wellborn]

I'm running Connected to Asterisk 1.4.18-1.
We have a T1 PRI and a SIP trunk coming in from our IVR.
When people call in on the SIP trunk, I experience the following
problem:
I put the caller on hold, MOH starts.  MOH quits after approx ten
seconds.  When I pick the call back up off of hold, they can't hear me
for about 20 seconds (sometimes longer or not at all) but I can hear
them.

This doesn't happen with calls on the T1.

Has anyone seen this before?  Any solutions out there?

Thanks,
Ben W


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[asterisk-users] Asterisk 1.4 reliability problems

[Ben Willcox]

Hello All,

We have been experiencing some ongoing reliability problems with
Asterisk for quite some time, and I am trying to find out if anyone else
has experienced the same problems.

We are running asterisk 1.4.17~dfsg-2+b1 on Debian Lenny, with a Digium
PRI card, and have approximately 120 sip peers, mostly Snom 360s, with a
few Grandstream GXP2000 and a handful of Handytone 486 units. 

The symptoms, when they occur, are as follows:

-The inability to receive incoming calls to our ISDN PRI (callers get a
busy tone), this starts off becoming intermittent but becomes permanent.

-Asterisk cli commands work once, but then no longer return any data
until disconnecting and reconnecting to the cli, i.e. sip show peers,
show channels etc.

-Internal SIP calls stop working

-Calls remain stuck in queues, the queue members do not ring, and show
as Busy when issuing a 'queue show' command.


We've actually had these sort of problems for many months now, which
originally started when we were running Asterisk 1.2 on Gentoo. We have
done a large amount of fault finding and testing, which has involved a
replacement ISDN card, reinstall on complete different server hardware,
and changing to Asterisk 1.4 on Debian Lenny.

I believe there may be two separate issues here - we did track down one
problem to our cacti and nagios monitoring scripts, which were
connecting and disconnecting to the manager interface several times per
minute, which eventually caused asterisk to give the above symptoms,
although in addition to the above, asterisk would consume 100% cpu on
the box, and eventually need a hard-reboot of the server. I posted about
this to the list a few weeks ago, and it was confirmed that this could
cause such a problem. After stopping these services the problems were
much reduced.

However, we have now completely disabled the manager interface
(enabled=no in manager.conf), and yesterday the problem occurred again -
a restart of asterisk got everything going again.
So really I'm at a loss as to where to go from here. A colleague of mine
also has the same problem at his site running Asterisk 1.4 on Debian
Lenny, he has never used the manager interface, and has completely
different server hardware and ISDN card, so I wonder if it's a Debian
specific problem?

One option is to try reverting back to Asterisk 1.2, but that isn't
really a long-term solution. We also had major problems with 1.2 with
our Snom 360 phones, as with any Snom firmware  6.2.2 there was a
serious problem whereby on hangup the channels were not cleared down,
meaning we had many outgoing ISDN calls held open for many hours until
we realised the problem. This problem does not occur in Asterisk 1.4,
although we have many log messages such as:

chan_sip.c: Remote host can't match request BYE to call callid

so I don't know if this is anything to worry about?

Any help would be gratefully received!

Thanks,
Ben



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Re: [asterisk-users] Asterisk 1.4 reliability problems

[Ben Willcox]

Hi All,

Thanks for all the replies. Here are my responses to the responses:

On Tue, 2008-03-18 at 06:13 -0400, Al Baker wrote:
 Curious, you mention a number of problems that have gone on for months
 Question:  Have you reported ANY or ALL of them to DIGIUM and if so
   what has been their response on each of these problems ?

We have been working very closely with the reseller that supplied us
with the system, and although we have made progress over this time and
they have given us a lot of technical support, I now feel that it will
be quicker to progress the current issues independently. I don't know if
the issues were escalated as far as Digium though.

Tzafrir Cohen wrote:
 The symptoms you mention suggest some sort of deadlock. Please enable
 debug and the full log. Maybe this will provide some hints. But please
 check that the full log is rotated in /etc/logrotate.d/asterisk .
 
 Can you reproduce this situation? e.g.: by extensive usage of the
 manager interface? If so, it might help for testing.

I will enable full debug logging. I suspect that we could reproduce the
original problem with the manager interface by stress testing it with
multiple connections, but I'm not sure if this is the same problem that
we are currently experiencing.
I also want to avoid causing problems on our production system at the
moment, as it is rather 'delicate' as far as the users are concerned at
the moment.

Steve Totaro wrote:
 Why not try a different OS such as CentOS for now?  That would be my
 next step.

I have considered this, to at least to establish whether it is a Debian
specific problem, either with the asterisk packages themselves, or some
other configuration or package issue. I am umming and ahhing between
this and Gordon's suggestion below:

Gordon Henderson wrote:
 Personally, I'd go back to Debian, but stick to stable (Etch) and
 then 
 compile and install a custom kernel tailored exactly to your
 hardware, 
 then compile and install your own asterisk from source.

I'm thinking that this may be the way I should go, then I will have the
freedom to install any version of asterisk that I need, whilst also
keeping my favourite distro.

Doug Lytle wrote:
 Two things,
 
 1.)  On your queue setup, avoid using AgenCallbackLogin, it's known
 to 
 cause deadlocked channels.
 2.)  Restart the Asterisk service once a week.  I do this via a CRON
 job 
 at 3am on Sundays.

We're actually not using Agents on our queues, just SIP channels, so
hopefully this is not the problem. We simulate 'agents' logging in and
out by pausing and unpausing queue members.
I am now going to add a cron job to restart asterisk daily, in the hope
that until the problem is resolved properly, at least it will help
relieve some of the pain by making it stable for a full 24hrs at a time.

Matt Florell wrote:
 I would suggest upgrading to at least 1.4.18. I was able to run it for
 about 2 weeks and almost one million calls before I could get it to
 crash, and the 1.4.19RC2 seems to fix even more of the locking issues
 as well. I know a lot of these problems still existed under 1.4.17.

A million calls sounds good, but 2 weeks, not so good. It's a bit
disappointing to me that crashing /ever/ is acceptable, I had always had
the understanding that asterisk was supposed to be rock-solid. I suppose
it's some consolation that its not just me that has problems!

Thanks for all the input. I think short term I will restart asterisk
daily, then the action plan is to revert back to Debian Etch, and then
install asterisk 1.4.18 from source, and hopefully this will improve
things.

Thanks,
Ben

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[asterisk-users] WARNING[4294]: chan_zap.c:8851 zt_pri_error: 1 copying 5 bytes LLC

[Ben Thompson]

Hi

I have recently upgraded my Asterisk system (from 1.2 to 1.4) and I have 
started to
notice the following messages when I recieve a call on my Zap channel
:-

[Mar 11 09:20:17] WARNING[4294]: chan_zap.c:8851 zt_pri_error: 1 copying 5 
bytes LLC

I have a single PRI ISDN 30 link to a Siemens Realitis DX. Here is my
zapata.conf :-

[channels]
echocancel=no
echocancelwhenbridged=no
rxgain=-5.0
txgain=0
musiconhold=default
language=en
context=from-realitis
switchtype=euroisdn
pridialplan=unknown
prilocaldialplan=unknown
priindication=outofband
signalling=pri_net
usecallerid=yes
cidsignalling=v23
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
callgroup=1
pickupgroup=1
immediate=no
overlapdial=yes
progzone=uk
resetinterval=43200
group=1
channel = 1-15
channel = 17-31

Here is my /etc/zaptel.conf :-

loadzone=us
defaultzone=us
span=1,0,0,ccs,hdb3,crc4
bchan=1-15,17-31
hardhdlc=16
span=2,0,0,ccs,hdb3,crc4
bchan=32-46,48-62
hardhdlc=47

The second span is not used and just has a loopback plug
connected. Could anyone advise me what the error message means?

Thanks

Ben Thompson

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Re: [asterisk-users] WARNING[4294]: chan_zap.c:8851 zt_pri_error: 1 copying 5 bytes LLC

[Ben Thompson]

On Tue, Mar 11, 2008 at 09:25:19AM +, Ben Thompson wrote:
 Hi
 
 I have recently upgraded my Asterisk system (from 1.2 to 1.4) and I have 
 started to
 notice the following messages when I recieve a call on my Zap channel
 :-
 
 [Mar 11 09:20:17] WARNING[4294]: chan_zap.c:8851 zt_pri_error: 1 copying 5 
 bytes LLC
 
 I have a single PRI ISDN 30 link to a Siemens Realitis DX. Here is my
 zapata.conf :-
 
 [channels]
 echocancel=no
 echocancelwhenbridged=no
 rxgain=-5.0
 txgain=0
 musiconhold=default
 language=en
 context=from-realitis
 switchtype=euroisdn
 pridialplan=unknown
 prilocaldialplan=unknown
 priindication=outofband
 signalling=pri_net
 usecallerid=yes
 cidsignalling=v23
 hidecallerid=no
 callwaiting=yes
 usecallingpres=yes
 callwaitingcallerid=yes
 threewaycalling=yes
 transfer=yes
 canpark=yes
 cancallforward=yes
 callreturn=yes
 callgroup=1
 pickupgroup=1
 immediate=no
 overlapdial=yes
 progzone=uk
 resetinterval=43200
 group=1
 channel = 1-15
 channel = 17-31
 
 Here is my /etc/zaptel.conf :-
 
 loadzone=us
 defaultzone=us
 span=1,0,0,ccs,hdb3,crc4
 bchan=1-15,17-31
 hardhdlc=16
 span=2,0,0,ccs,hdb3,crc4
 bchan=32-46,48-62
 hardhdlc=47
 
 The second span is not used and just has a loopback plug
 connected. Could anyone advise me what the error message means?

Also, here is a section of debug :-

1 -- Restarting T203 counter
1  Protocol Discriminator: Q.931 (8)  len=34
1  Call Ref: len= 2 (reference 1371/0x55B) (Originator)
1  Message type: SETUP (5)
1  [04 03 80 90 a3]
1  Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer 
capability: Speech (0)
1   Ext: 1  Trans mode/rate: 64kbps, circuit-mode 
(16)
1   Ext: 1  User information layer 1: A-Law (35)
1  [18 03 a1 83 93]
1  Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Preferred  
Dchan: 0
1 ChanSel: Reserved
1Ext: 1  Coding: 0  Number Specified  Channel Type: 3
1Ext: 1  Channel: 19 ]
1  [6c 06 00 83 32 34 39 39]
1  Calling Number (len= 8) [ Ext: 0  TON: Unknown Number Type (0)  NPI: 
Unknown Number Plan (0)
1Presentation: Presentation allowed of network 
provided number (3)  '2499' ]
1  [70 04 80 38 34 31]
1  Called Number (len= 6) [ Ext: 1  TON: Unknown Number Type (0)  NPI: Unknown 
Number Plan (0)  '841' ]
1  [7c 03 80 90 a3]
1  Low-layer compatibilty (len= 5) [ 1 0x80 1 0x90 1 0xA3 1  ]
1 -- Making new call for cr 1371
1 -- Processing Q.931 Call Setup
1 -- Processing IE 4 (cs0, Bearer Capability)
1 -- Processing IE 24 (cs0, Channel Identification)
1 -- Processing IE 108 (cs0, Calling Party Number)
1 -- Processing IE 112 (cs0, Called Party Number)
1 -- Processing IE 124 (cs0, Low-layer Compatibility)
[Mar 11 11:32:31] WARNING[4294]: chan_zap.c:8851 zt_pri_error: 1 copying 5 
bytes LLC 
1 q931.c:4226 q931_receive: call 1371 on channel 19 enters state 6 (Call 
Present)
1 -- Restarting T203 counter
1 q931.c:3302 q931_setup_ack: call 1371 on channel 19 enters state 25 (Overlap 
Receiving)
1 -- Restarting T203 counter
1  Protocol Discriminator: Q.931 (8)  len=14
1  Call Ref: len= 2 (reference 1371/0x55B) (Terminator)
1  Message type: SETUP ACKNOWLEDGE (13)
1  [18 03 a9 83 93]
1  Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Exclusive  
Dchan: 0
1 ChanSel: Reserved
1Ext: 1  Coding: 0  Number Specified  Channel Type: 3
1Ext: 1  Channel: 19 ]
1  [1e 02 81 88]
1  Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 
0  Location: Private network serving the local user (1)
1Ext: 1  Progress Description: Inband 
information or appropriate pattern now available. (8) ]
-- Accepting overlap voice call from '2499' to '841' on channel 0/19, span 1
-- Starting simple switch on 'Zap/19-1'
1 -- Restarting T203 counter


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Re: [asterisk-users] Attatch monitor recording to a voicemail

[Ben Willcox]

Jared Smith wrote:

 No, unfortunately this was done under NDA, but the general gist goes like 
 this:

As it happens, I deployed my solution to this on our live PBX today, which I
wrote with some help from another asterisk-users user. Here is what I
came up with:

Firstly, in features.conf I don't use the normal automon function in the
featuremap, but an applicationmap:

[applicationmap]
recordtovm =*1,self,Macro,recordtovm


Then in extensions.conf we have the following additions:

[globals]
DYNAMIC_FEATURES=recordtovm

[macro-recordtovm]
exten = 
s,1,Set(MONITOR_FILENAME=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${CALLERID(num)})
exten = s,n,Set(DYNAMIC_FEATURES=recordtovm)
exten = s,n,MixMonitor(${MONITOR_FILENAME}.wav,b,/etc/asterisk/recordtovm.pl 
${CALLERID(num)} ${MONITOR_FILENAME}.wav)

And finally the perl script recordtovm.pl in /etc/asterisk/ is as follows:

#!/usr/bin/perl -w
#
use strict;

my $monitordir=/var/spool/asterisk/monitor/;
my $vmdir=/var/spool/asterisk/voicemail/default/;
my $vmfolder=INBOX;
my $vmbox=$ARGV[0];
my $vmpath=$vmdir.$vmbox/.$vmfolder;
my $monitorfilename=$ARGV[1];

opendir (DIR, $vmpath);
my @files = grep(/\.txt$/,readdir(DIR));
closedir(DIR);
my @sortedfiles = sort {$b cmp $a} @files;
my $vmid;
if ($sortedfiles[0] =~ /^(msg)(\d\d\d\d)(.txt)/)
{
$vmid=$2;
$vmid++;
}
else
{
$vmid=;
};

open VMFILE, $vmpath/msg$vmid.txt;
print VMFILE ;\n;
print VMFILE ; Message Information file\n;
print VMFILE ;\n;
print VMFILE [message]\n;
print VMFILE origmailbox=$vmbox\n;
print VMFILE context=\n;
print VMFILE macrocontext=\n;
print VMFILE exten=s\n;
print VMFILE priority=\n;
print VMFILE callerchan=\n;
print VMFILE callerid=\n;
print VMFILE origdate=\n;
print VMFILE origtime=\n;
print VMFILE category=\n;
print VMFILE duration=\n;
close VMFILE;

if ($ARGV[1])
{
system(mv $monitordir.$monitorfilename $vmpath/msg$vmid.wav);
};


Seems to work pretty well, we have the Record button on our SNOM phones mapped 
to DTMF *1,
so its a single press to start recording. The perl script doesn't populate the 
origdate and
origtime fields at the moment so you'll need to add this if you want the time 
and date saving
with the message.

Hope this helps,

Ben





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Re: [asterisk-users] ata device but for a soundcard

[Ben Willcox]

Adam Moffett wrote:
 So you want a device that will answer a SIP call, and play the audio out 
 to a speaker? 
 
 You're looking to build a PA system then?

We achieved this using a Grandstream Budgetone configured to 
auto-answer, and just soldered a pair of wires across its speaker 
terminals connected to the audio in of our PA amplifier. It works very well!

Cheers,
Ben

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Re: [asterisk-users] SIP GSM

[Ben Willcox]

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Kev S
 Sent: Tuesday, January 29, 2008 9:53 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] SIP  GSM
 
 With that sort of set up, If for example i get a 8 channel GSM gateway 
 and the X100P can i make more than 1 concurrent call though the gateway 
 with the X100P or does it only support 1 call at a time?
 
 What im looking to do is get a multi channel GSM gateway, and have the 
 ability to make more than 1 call at once through it.

The PorTech MV-372 works nicely with asterisk and is multichannel (2, if 
that counts!)

Cheers,
Ben

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Re: [asterisk-users] Attatch monitor recording to a voicemail

[Ben Willcox]

Jared Smith wrote:

 I've always done this by setting the MONITOR_EXEC channel variable to
 point to an external program that takes care of moving the recording to
 the proper location so that it can be accessed by the user who made the
 recording.  I'll bet if you search for MONITOR_EXEC and
 MONITOR_EXEC_ARGS you'll find exactly what you need.

Thanks Jared,

That gets me halfway there, but what I'm wondering about is the process 
of moving of the recording to the correct place - i.e. should my 
external program do the following:

1) Check the users voicemail directory for existing message filenames
2) Copy the recording into the voicemail directory named msg.WAV 
(incremented depending on number of existing messages)
3) Create the msg.txt file in the correct format

or is there another way that will sort this all out automatically? What 
would happen if a real voicemail drops into that directory while my 
external script is halfway through copying/creating for example...

Cheers,
Ben



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[asterisk-users] Attatch monitor recording to a voicemail

[Ben Willcox]

Hello All,

Our old Lucent Argent system had a feature whereby when you initiate
recording during a call, it would afterwards send the recording as a
voicemail message to the user who initiated the recording.

We use the automon *1 recording function in asterisk, which allows users
to record a call if necessary on the fly. Unfortunately there doesn't
appear to be an easy way for the user to actually access that
recording.
What I'd like to do is replicate the functionality described above, so
that when the call finishes, asterisk sends a voicemail to the user
containing the monitor recording.

I'm not sure of the best way to achieve this - is there a method I can
use within the asterisk dialplan, or do I need to manipulate the automon
wav file and voicemail directory contents outside of asterisk to put the
recording in the users mailbox?

Thanks,
Ben

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Re: [asterisk-users] urgent-channels

[Ben Willcox]

Khaled Chehab wrote:
 Hi All
 
 
 
 I am using asterisk 1.2.4
 
 
 
 Please see the results when I execute Sip show channels
 
 *569 *active SIP channels

What phones are you using? We had a similar problem with Snom 360 phones
with firmware version  6.2.2 and asterisk 1.2, whereby channels would
not hangup correctly.

Cheers,
Ben


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Re: [asterisk-users] Disappearing B-Channels

[Ben Willcox]

Mark Greene wrote:
 I could do that. The only issue is that I don't understand why others 
 with my setup have not had to do the same. What's unique about my TDMoE 
 setup that makes it intolerant to channel restarts? I did everything by 
 the book.

We had a similar problem, and although your symptoms are not identical, 
we had a massive ongoing problem with the reliability of our ISDN 
channels, where they would randomly stop working until eventually we 
could no longer make any outgoing or incoming calls via our PRI line. A 
restart of asterisk was needed to get it working again.

We finally discovered that the problem was our use of cacti, monitoring 
channel usage via the manager interface. We were hitting the manager 
interface with 4 connects, queries and disonnects per minute, and over 
the course of a day, asterisk would start dropping calls and other 
strange behaviour. I guess there is a memory leak or similar with the 
asterisk manager interface.

Perhaps not your particular problem, but thought I'd throw it in there 
anyway as you never know...

Cheers,
Ben

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Re: [asterisk-users] Asterisk as XMPP component. How to use it ?

[Ben Willcox]

Olivier wrote:
 At the opposite, I think it could be useful for an Asterisk server to 
 act as XMPP User Activity provider (ie update XEP-0108 field with 
 on-the-phone value).
 Do you agree ?
 Is there any XMPP client supporting User Activity ?
 Is Asterisk capable of getting or sending such User Activity messages ?

The Openfire XMPP server (http://www.igniterealtime.org/) has an 
asterisk plugin which uses the manager interface to send 'On the phone' 
status to XMPP clients. It works very well.
It also has the capability to Pause and Unpause queue members depending 
on idle status, but that is very annoying and we turn that off!

Cheers,
Ben

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Re: [asterisk-users] Grandstream GXP2020 / 2000

[Ben Schorr]

I have a client using the Grandstream phones (not sure which model but it looks 
fairly low-end) and they're lukewarm on them.  The display doesn't tilt up for 
easy viewing and the sound quality on the speaker phone leaves something to be 
desired apparently.

But as basic, inexpensive, Asterisk handsets they do the job.

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr  Tower
www.rolandschorr.com
[EMAIL PROTECTED]


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Chavez
Sent: Tuesday, September 25, 2007 7:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Grandstream GXP2020 / 2000

On Tue, 2007-09-25 at 10:59 +0200, Erik Wartusch wrote:
 Hi,
 
 Has somebody experiences with the Grandstream GXP2020 / 2000 phones in 
 a business graded installation (with really traffic on  not 3 
 calls a day ;-) ) Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4 
 in generall)
 
I have not tried the 2020 yet but the GXP-2000 works fairly well.  The 
only complaint I had from a very busy installation (a travel agency) is that 
the handset gets hot after prolonged use.  This may have been because the 
office itself was hot during summer and after they installed an AC the problem 
is no longer there.

--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
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[asterisk-users] Hangup detection and trombining

[Ben Dinnerville]

Hi All,

I hate to post yet another bloody hangup detection issue on the list, but
I have been pulling my hair out no end of late with a hangup detection issue
on 1 system (have a few others out there with TDM400's and no issue but this
one is causing a real headache)

The scenario is - system with TDM04B, a call comes in on a analogue line,
rings internally and then diverts to a mobile on a second analogue line, so
we in effect have a trombone happening where a call comes in on 1 analogue
and back out on another analogue.

Hangup detection seems to be working most of the time, but on a regular
basis does not (about once every 2 days or so). We cannot get hangup
supervision / polarity reversal or any other smart way of detecting a
hangup, so are using busydetect. What seems to be happening is that on
trombone'd calls when both parties hangup, there is a busy tone being played
on each leg of the call back down each line. Some times we seem to get lucky
and the tones are played in sync and a hangup occurs, but other times the
tones are out of sync with each other and are overlapping, causing a
non-normal tone on the line(s) or a continuous tone rather than a 'beep beep
beep' which means the card / system cannot detect a hangup via busy detect.

Can anyone out there confirm if my assumptions are correct re the dual'ing
or the tones and the effect it will have on hangup detection?

And if correct, can anyone recommend a work around to get hangup detection
working in such a scenario?

Cheers,

Ben
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Re: [asterisk-users] TC400B and show transcoder

[Ben Dinnerville]

Kevin P. Fleming wrote:
 Ben Dinnerville wrote:
 
 The problem occurs when we have external (pstn) calls coming into / out 
 of the system (via an iax trunk), in which case we have no control over 
 frame size, as well as occurring with handsets directly connected to the 
 system.
 
 Please contact Digium Support to work through these problems, as you
 have unlimited installation support with the purchase of the product.
 
 No, the TC400B does not provide any Zaptel 'spans', so it does not
 provide a timing source.
 
 What documentation is referring to the 'show transcoder' command? That
 command is not in either Asterisk 1.2 or 1.4, so we need to get that
 documentation fixed...
 
Hi Kevin,

Am working through the rma process at the moment :)

The document that describes the show transcoder command is the pdf on 
the digium website -

http://www.digium.com/elqNow/elqRedir.htm?ref=http://www.digium.com/docs/TC400B/TC400B-user-manual.pdf

It is the only documentation I have been able to find on the card.

As mentioned, the command does exist (only available when a card is 
present) on 1.2 but not 1.4 - if it is not meant to be in 1.4 i would 
recommend getting it in there, it is nice to be able to know what is 
happening on the system


Cheers,

Ben


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[asterisk-users] TC400B and show transcoder

[Ben Dinnerville]

Hi All,

I have recently installed a TC400B card into a system and am trying to 
get it to work. As far as I ca tell from the docco on Digiums website, 
there is no config as such unless you want to enable  / disable only 1 
codec, otherwise by default it runs as 92 channels of either.

I have tried asterisk 1.4.9, 1.4.10 and 1.4.10.1 along with zaptel 1.4.4 
and addons 1.4.2. The zaptel modules all apear to be loaded correctly 
(loading wctc4xxp loads up zttranscode and zaptel). Dmesg shows that the 
card has been found:

Registered codec translator 'DTE Encoder' with 92 transcoders 
(srcs=000c, dsts=0101)
Registered codec translator 'DTE Decoder' with 92 transcoders 
(srcs=0101, dsts=000c)
Zaptel DTE (g.729a / g.723.1 5.3kbps) Transcoder support LOADED (firm 
ver = 56)
Found and successfully installed a Wildcard TC: Wildcard TC400P+TC400M


and the card has its own interrupt -
193:18715321896779   IO-APIC-level  tc400b

But when ever we need to do a transcode, ie playing back a wav file on a 
g729 channel, the audio is complete rubbish, with a lot of stutters in 
it (sounds like a recording does when you upload a file in the wrong 
sample rate etc) - the file that we are playing back is a wav file that 
has existed on the system and has been successfully played back with the 
soft g729 transcoding and also plays back fine when the channel is alaw, 
just not when the channel is g729. The same issue occurs when a 
transcode has to happen from a handset to a IP trunk, eg alaw on the 
handset and g729 on the trunk channel, the audio stream is non 
comprehensible.

The other issue is that whilst all the modules apear to be loaded 
ocrrectly, and a show translation shows that the codes are supported 
without the presence of a g729 key:

pbxla*CLI core show translation
  Translation times between formats (in milliseconds) for one 
second of data
   Source Format (Rows) Destination Format (Columns)

   g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc 
g726 g722
  g723-   3113 32 32 -   12 
3-
   gsm3   -222 21 23 -   11 
2-
  ulaw1   2-12 21 21 -   11 
2-
  alaw1   21-2 21 21 -   11 
2-
  g726aal23   222- 21 23 -   11 
1-
 adpcm3   2222 -1 23 -   11 
2-
  slin2   1111 1- 12 -   10 
1-
 lpc103   2222 21 -3 -   11 
2-
  g7292   3113 32 3- -   12 
3-
 speex-   ---- -- -- -- 
--
  ilbc4   3333 32 34 -- 
3-
  g7263   2221 21 23 -   11 
--
  g722-   ---- -- -- -- 
--

The show transcoder command listed in the documentation does not exist. 
There is no show transcoder or core show transcoder command 
available on the system. I have checked the menu options for the build 
and cannot see any specific item that needs to be enabled for this 
command to be available but have a feeling that the lack of this command 
and the horrible transcoded audio quality are related. Or is it just 
that the show transcoder command is only available in 1.2 and not in 1.4?


Another quick (hopefully) question - does the TC400 card provide a 
zaptel timing source, or do you still need to load ztdummy in the case 
of not having another card in the system?


Any info or experiences would be great.

Thanks in advance.

Ben


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Re: [asterisk-users] TC400B and show transcoder

[Ben Dinnerville]

Andres wrote:


 Try to compare the frame size you are receiving from asterisk and set 
 your phone to transmit the same frame size.  I would guess the card 
 appears to have problems when the frame size is different.  Please try 
 and report back.  I am curious about this.

The problem occurs when we have external (pstn) calls coming into / out 
of the system (via an iax trunk), in which case we have no control over 
frame size, as well as occurring with handsets directly connected to the 
system.


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[asterisk-users] Outbound SIP authentication with dynamic credentials

[Yossi Ben Hagai]

Hi All,

I'm working on the following scenario:
VoIP Gateway -- Asterisk server -- Proxy server -- PSTN
   |
  XMLRPC
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[asterisk-users] Outbound SIP authentication with dynamic credentials

[Yossi Ben Hagai]

Hi All,

I'm working on the following scenario:
VoIP Gateway -- Asterisk server -- Proxy server -- PSTN
   | |
  XMLRPC  Radius
In this call flow a prepaid caller places a call over the VoIP gateway to
the Asterisk server acting as an IVR,
the server collects the user ID, PIN, and B-number and authenticate the
credentials using an XML-RPC interface and plays the balance to the
caller. so far so good.
the next step is to send an Invite message for the B-number over to the
proxy server. the issue is that the proxy server does not trust
the asterisk server and the actual call duration limit and accounting is
handled by the proxy server which requires the asterisk server to proxy-auth
itself sending the user ID and PIN details as the MD5 auth credentials.

Now the asterisk can act as a client and authenticate itself, but this
required the credentials to be hard coded on sip.conf.
currently my last resort would be to put the sip.conf on MySQL RT and create
peers dynamically for each call, but it doesn't seem like an elegant
solution.
Is there a way to specify the auth credentials on the Dial() command?

Thanks.
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[asterisk-users] ivr testing script

[Ben Wellborn]

Hello All,
I want to run a bash script on my asterisk-now machine that will test
our IVR at another site.  I need it to dial the number and, if it
answers, listen for a few seconds and then hang up.  If there are return
codes from the commands that can help me determine the status of the
line, I can then send an email to alert if there was busy tone or no
answer.
I've fiddled with a bash script to send sip show peers and parse it,
but that's about the most of my experience interfacing to asterisk.
Is there a way to do what I want?

Thanks,
Ben W


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Re: [asterisk-users] SIP accounts from MYSQL.

[Yossi Ben Hagai]

Asterisk realtime is what you are looking for. the subject is explained very
clearly including configuration examples and DB schema on the following
links:
http://www.voip-info.org/wiki-Asterisk+RealTime
http://www.asteriskdocs.org/modules/news/article.php?storyid=28

I won't go over the process as it is detailed in the links above, but
basically you should compile the asterisk-addons, configure the res_mysql
with the proper DB details, create a table to hold sip.conf and optionally
extensions.conf then configure extconfig to map the newly created tables.

Joss.


On 5/27/07, Jonson Player [EMAIL PROTECTED] wrote:


Hello,
I just want to put all my sip accounts in mysql and asterisk use it from
mysql. How can I do that, could you be more specific because I readed alot
on wiki and i'm lost... I don't know what to modify in Makefile from channel
directory. I use asterisk 1.4.4, that is already compiled and i also have
CDR in mysql. I must create manny accounts and I want to realize that from
mysql. Thank you for your support guys.


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Re: [asterisk-users] Asterisk and iBasis

[Yossi Ben Hagai]

Asterisk supports it and the good news is that you don't have to do anything
for it to work.

On 5/19/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:


Hi,

We are currently trying to setup Asterisk with iBasis. One
question/problem we have is that Ibasis has told us to send the INVITEs to
one IP address and all media to a different IP address. How can we do that
in Asterisk?

Thanks

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Re: [asterisk-users] Asterisk and iBasis

[Yossi Ben Hagai]

iBasis, like many providers uses a softswitch in which separate elements
handle the signaling (SIP/H.323) and media gateways handle the media (RTP).

when you send a call with the Dial command you state iBasis signaling
address and the Asterisk sets it's own media IP/Port in the SDP. when iBasis
send back a response it states it's own media IP/Port in the SDP (which can
be different from the signaling IP) so the asterisk will know where to send
the RTP packets. so in terms of asterisk configuration you don't need to do
anything different from what you would usually do..


On 5/19/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:


Thanks for the prompt response, but would you care to explain this a bit
further?

It could be due to my ignorance, and if so, I apologize. But, how can we
send the INVITE to one IP and then the media to a different one? Do we just
simply send the call to the INVITE IP using the Dial command and that's it?

Thanks

On Sat, May 19, 2007 12:29 pm, Yossi Ben Hagai [EMAIL PROTECTED] said:

 Asterisk supports it and the good news is that you don't have to do
anything
 for it to work.

 On 5/19/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:

 Hi,

 We are currently trying to setup Asterisk with iBasis. One
 question/problem we have is that Ibasis has told us to send the INVITEs
to
 one IP address and all media to a different IP address. How can we do
that
 in Asterisk?

 Thanks

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Re: [asterisk-users] Need a RTP/SIP Proxy to be used as SBC (Session Border Controller)

[Yossi Ben Hagai]

Check rtpproxy from portone for media proxy and nat traversal.
http://www.voip-info.org/wiki/view/Portaone+rtpproxy

another option is the MediaProxy from AG projects:
http://www.voip-info.org/wiki-MediaProxy

Joss.
On 5/11/07, Jean-Marc Salsa [EMAIL PROTECTED] wrote:


Hi all,

I have been using asterisk to do such kind of thing,
But I must admitt, this is not 100 % conveniant (Mainly because Asterisk
isn't a SIP Proxy).

I just wanted to know if you knew/used some kind of SBC or packages which
would deal both with SIP AND RTP !
SER/OpenSER woulc be a good SIP Proxy ... but then how to deal with RTP ?

Any tip, info greatly welcome !

Thanks,

JM

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[asterisk-users] Redirect Message Waiting Indication

[Ben Brown]

I have several users that occassionally uses a single Wifi Phone in place
of his/her desk phone. I have everything setup to allow him/her to
redirect his/hers desk phone traffic to the wifi phone and to have the
wifi phone masqurade the caller ID. However, I would also like to redirect
the message waiting indication. Can this be done from the dialplan?

THanks

BEN BROWN

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Re: [asterisk-users] List of telemarketers??

[Yossi Ben Hagai]

Regarding (2) - you can either provide a realtime query service supporting
web service interface which can be consumed using virtually any programming
language and it would be very easy to build an AGI script around it.
the second option would be to periodically update a flat file (csv) and
provide ftp access - this way you won't have to sustain the load of the
realtime queries as the demand grows and the numbers can be provisioned into
PBX which doesn't have public Internet access.

personally I don't have a use for such a DB, but I'm willing to help on
setting it up for the community if needed.

Joss.


On 5/9/07, Ritesh Agrawal [EMAIL PROTECTED] wrote:


Does anyone know if there is a known list of telemarketers?
Something like http://whocalled.us/ with an easier access?

We could all benefit if there was such a thing :-)
If there is enough interest, I could put up a database that everyone can
benefit from.

I just need some suggestions on:
(1) Adding new numbers based on community responses (some rule to sanity
check)
(2) Method that everyone would prefer to access the dbase.

Ritesh


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Re: [asterisk-users] Two Connected Servers Sound Quailty

[Yossi Ben Hagai]

Hi Matt,

you didn't mention what type/bw of each site Internet connection, i suggest
that you try to split the scenario into smaller pieces:
- run long term pings between the server while you make a call and check for
packet loss.
- make internal calls between extensions on the same branch and verify that
both servers work okay (eliminating Internet connectivity)
- register a UA from one site to a server on the other site, make a call and
viceversa (eliminating a problem on one of the servers).
- check for speed/duplex setting on NIC and switch port.
- check if the sound quality issues are symmetric (does both sides
experience the sound cut or it only happens on a specific site).
- make sure you don't use G.711 as it consumes bw and from the codec
list you've mentioned has the lowest tolerance to packet loss.

Since the problems are intermittent my bet is that someone in the office is
have the p2p client work overtime or sending lots emails with funny
attachments


On 4/28/07, Matt Gardner [EMAIL PROTECTED] wrote:


Ok this is my first post and I will try to keep it short.

I have searched everywhere and haven't found an answer to my question

I have two Trixbox servers that are connected over the Internet via an
IAX2 connection.  We are experiencing very poor sound quality.  I have tried
many different codecs gsm, ilbc, g729, g711 and all seem to have the same
problem. (All though g729 seems to work the best but still isn't reliable)
The problems are intermittent sometimes the sound will cut out for 3-4
seconds and other times the sound will just be loosing every other word, and
other times it sounds just fine.

Also, we have been using Skype over this same Internet connection and have
very good sound quality with very few lost words.

So here are my questions.

First, is it a correct assumption to say that because Skype works well
over this connection then I should be able to get asterisk to work over this
connect.  I am hoping that Skype isn't better then asterisk in this area.

If I should be able to get the same sound quality could you point me in
the right direction on how to achieve this.  (I have tried messing with the
jitterbuffer but haven't been able to find very good docs on how to utilize
this functionality so about all I have done is set jitterbuffer=yes)

Thanks in advance.

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Re: [asterisk-users] Re: call dispatching - legacy application

[Yossi Ben Hagai]

Hi Adriano,

I agree with Time Bandit - AGI is what you are looking for. I recently had a
similar scenario where I had the check the cid of every customer calling to
a support qeue and check the payment status against a windows CRM app. if
the customer has an unsettled debt the call is redirected to sales rep.
The call flow in your case should be something like this:
- Answer call.
- depending on the API available to the legacy VB app (or AS/400 directly)
you can either write a local script on the * server to perform the query or
host a FastAGI script on the VB app server (or on any other server that can
perform the DB query).
- The script input is the cid and the output should be a channel variable
assigned to operator exten.
- Dial the right operator based on the channel variable assigned by the
script (you can also dial the operator within the AGI - but it better to
keep AGI as simple as possible and run as little time as possible.

one last tip - keep in mind that cid is not always available (presentation
restricted) or customer may be calling from different location
(cellphone/office) which is not on your DB - so you might want to present an
IVR menu that allows the customer to enter the cid for the requested
service.

Joss

On 4/27/07, adriano ghezzi [EMAIL PROTECTED] wrote:


well more indeep the actual process is

myparser a php script connected telnet to aah manager get and parse
events
it grab cid from manager event (incoming call)
it passes cid to a legacy visual basic app that query a db on ibm as/400
the query return info about customer's status and opened tickets
now it should instruct asterisk to send the call to the right operator,
because at the moment I'm not able to do it i do:
the call get dispatched Normal way
i wait the call ends, the php parser inform another (master vb app)
that open a pop-up on the pc of the operator that processed the call
and update the customer'ts ticket

what i would like is to dispatch the call to a specific operator eg
the preferred customer's operator.

thanks to Brad.
ciao!




2007/4/26, adriano ghezzi [EMAIL PROTECTED]:
 Hy all

 need to preprocess
 1) incoming call get caller id lookup some info in my db,
 2) based on the result dispatch the call to the right operator

 step 1 is ok I developped a small .php script that connect manager and
 parse events, now I have to tell AAH do dispatch call to the right
 operator

 questions
 1) is this the right practice ?
 2) where to find a complete manager api reference, (to buy too)

 note that
 there is a legacy application that query the db actually php script
 send the request to this app and wait for response

 I'm a programmer at very first installation of AAH , just testing
capabilities

 thanks in advance for any help and suggestion.

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Re: [asterisk-users] tone generation

[Yossi Ben Hagai]

Check the Milliwatt() cmd here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Milliwatt
It sends 1000Hz, but you can derive from it.

Joss.


On 4/24/07, Jerry Geis [EMAIL PROTECTED] wrote:


Does asterisk have a way in the dialplan to generate tones?
Say I want to play a tone 300Hz for 3 seconds.
Can I do that?

If not, can I use some system command to generate the wav file
then just have asterisk play it?

Jerry

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Re: [asterisk-users] Asterisk Pix firewalls

[Yossi Ben Hagai]

I second that. the PIX has SIP fixup which allows RTP traffic to pass
dynamically based on SDP information, so you don't need to create a rule for
the RTP range - just allow SIP UDP 5060.

On 4/25/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:


On Tue, Apr 24, 2007 at 11:04:53PM -0400, Lee Jenkins wrote:
 Noah Miller wrote:

 SIP:
 TCP and UDP port 5060 (signalling) - can be changed in sip.conf
 UDP ports 1-2 (RTP stream) - can be changed in rtp.conf
 

Yes. See rtp.conf (at least on your side).

Also, if the firewall understands SIP, it may be smart enough to open
the ports for the relevant RTP ports upon the beginning of a SIP
session. So consider trying not to open any port for RTP.

--
  Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] duration sec and billing sec in cdr

[Yossi Ben Hagai]

Looks okay to me. either the number you are testing with your VoIP provider
has an automated response which answers the call at the same sec you sent
the Invite request or the provider is sending False Answer Supervision...do
a sip debug and check while you make the call.

On 4/16/07, Adam KOSA [EMAIL PROTECTED] wrote:


Hi guys,

i've installed asterisk to handle multiple voip accounts.  I've looked
at CDR configs, and managed to have cdr-csv files growing after each
call.  It would be easier to check my locak asterisk cdr's than logging
into each account and check them at the provider website.

i found that if i ring my sip softphone from my ata, bill seconds are
counted correctly.  however, if i call via a voip provider, bill seconds
are counted incorrectly.  Example:

this call went to a pstn number

New call from 551 --- 94361abcdefg (context: internal)
Dialed: SIP/[EMAIL PROTECTED]
Call start: 2007-04-14 20:10:55
Answered  : 2007-04-14 20:10:55
Call end  : 2007-04-14 20:11:10
Duration  : 15 sec
Bill  : 15 sec


this call went to my ata from the sip softphone:

New call from 551 --- 505 (context: internal)
Dialed: SIP/505|45
Call start: 2007-04-15 07:58:11
Answered  : 2007-04-15 07:58:15
Call end  : 2007-04-15 07:58:43
Duration  : 32 sec
Bill  : 28 sec


i've searched and google'd the wiki, but found only accounting software
and cdr extensions for providers, but that's not what i need.

thanks for any help
Adam
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Re: [asterisk-users] duration sec and billing sec in cdr

[Yossi Ben Hagai]

The Playback command is auto-answering the call. you can use
Playback(please_wait,noanswer) to fix it.

Joss.


On 4/16/07, Adam KOSA [EMAIL PROTECTED] wrote:


Hi, and thanks for the suggestions!

Matt wrote:
 Sounds like your VoIP provider is incorrectly sending you an Answer
before
 the call actually completes.  I would contact your VoIP provider.


 I suppose it could also be possible that YOU have an Answer() statement
 that
 is only on your VoIP trunk.  I would double check that, and then contact
 your VoIP provider to see if they have any suggestions.

this is what's most likely as i have no experience in asterisk configs.
I've checked the extension.conf settins, they are:

exten = _94./_5[05][15],1,Playback(please_wait)
exten = _94./_5[05][15],n,Set(CALLERID(name)=my_voip_username)
exten = _94./_5[05][15],n,Dial(SIP/00${EXTEN:[EMAIL PROTECTED])

and for the internal numbers:

exten = _NXZ,1,Set(TIMEOUT(digit)=2)
exten = _NXZ,2,Dial(SIP/${EXTEN},45)
exten = _NXZ,3,VoiceMail(b${EXTEN})
exten = _NXZ,103,VoiceMail(u${EXTEN})

 Basically, SOMEONE (your or voipstunt) is answering the call before it
 should be answered.


i will check this with more voip providers to see if they or i have
messed up something (but it's probably going to be me, i just don't know
where to start looking).

thanks again
Adam
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