Re: [asterisk-users] [Asterisk-Users] How do I install speex for asterisk?

2010-08-08 Thread Faheem 
Try "make menu" and select the speex module.
make sure to do a  "make clean" also.

Faheem, Muhammad  VoIP Developer @ Vopium 



--- On Fri, 8/6/10, Deepika Nijhawan  wrote:

From: Deepika Nijhawan 
Subject: [asterisk-users] [Asterisk-Users] How do I install speex for asterisk?
To: asterisk-users@lists.digium.com
Date: Friday, August 6, 2010, 4:59 PM




 
 






 



Hi,
 

   

I
have followed steps which were mentioned on forum and given below. Still
couldn’t get speex working. On test calls getting error
“chan_sip.c: sip_call: No audio format found to offer.” 

   

# yum install speex 

# yum install speex-devel 

# cd /usr/src/asterisk 

# make clean 

# make 

# service asterisk stop 

# make install 

#
service asterisk start 

   

Also,
it is not showing speex translation on “core show translation recalc 10”.
 

   

Can
anybody please tell if missing some step in this.  

   

   

   

--- 

   

Kind
Regards, 

   

Deepika Nijhawan 

VoIP Engineer 

   

Oxygen8 Communications  

   



 



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Re: [asterisk-users] Detecting Called party Ring indication (and act on it)

2010-08-08 Thread Juan Miguel 
Hello Ketema:
I found this, i hope you serve

http://les.net/asterisk/pddpatch/



2009/8/15 Ketema Harris 

> is there a way to have asterisk short circuit the dial timeout
> parameter based on called party sending ring progress ?
>
> History:  I have multiple routes that a call can take.  Some routes
> are not so good and take a long time.
>
> Currently I use the Dial time out parameter, but it times out whether
> or not the called party is ringing or not (basically has to answer)
> I think the term is PDD (Post Dial Delay)? I wish to move on to the
> next route if the progress is longer than X sec, but if I get
> indication of ringing then stay on that route.
>
> Thanks
>
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Re: [asterisk-users] Monitor asterisk

2010-08-08 Thread Nasir Iqbal 
I agree with you and suggest you to use CLI command via AMI, for example
"Command core show channels"

I prefer CLI commands when they are available, as they return an aggregate
response as compared to AMI you do not need to filter, identity, and group
multiple responses / events to get result of a single command!

Regards

On Sun, Aug 8, 2010 at 11:28 PM, Richard Zulu wrote:

> Thanks Nasri,
>
> I don't want to only be able to use the CLI because I need the Helpdesk and
> application support Unit to be able to monitor, and they are not all the
> techy with CLI and stuff..
>
>
>
>
> On Sun, Aug 8, 2010 at 5:00 AM, Nasir Iqbal wrote:
>
>> Hi
>>
>> following asterisk cli commands can help
>>
>> show channels, show uptime and show sysinfo
>>
>> here is an example
>>
>> asterisk -x "core show sysinfo"
>>
>> On Sun, Aug 8, 2010 at 12:25 AM, Richard Zulu wrote:
>>
>>>
>>> Hey guys,
>>>
>>> I have my asterisk box running without a gui. I now need to monitor
>>> usage, calls, traffic of voice calls on this asterisk server. I cannot now
>>> install a gui because the configs will be wiped out, how can i go about
>>> monitoring all the above?
>>>
>>> --
>>> Richard Zulu
>>> Managing Director
>>> Time Information Company
>>> P.O Box 31842
>>> Clock Tower
>>> Kampala, Uganda
>>> www.time.co.ug
>>>
>>> Mobile :+256752624006
>>> Skype: zulu.richard
>>>
>>>
>>> --
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>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>>
>> --
>> Nasir Iqbal
>>
>> ICT Innovations
>> http://www.ictinnovations.com/
>>
>>
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>
>
>
> --
> Richard Zulu
> Managing Director
> Time Information Company
> P.O Box 31842
> Clock Tower
> Kampala, Uganda
> www.time.co.ug
>
> Mobile :+256752624006
> Skype: zulu.richard
>
>
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Re: [asterisk-users] Monitor asterisk

2010-08-08 Thread Kevin Keane 
Do you have a Nagios server? Then you could use that to monitor various aspects 
of Asterisk.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Zulu
Sent: Sunday, August 08, 2010 11:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Monitor asterisk

Thanks Nasri,

I don't want to only be able to use the CLI because I need the Helpdesk and 
application support Unit to be able to monitor, and they are not all the techy 
with CLI and stuff..



On Sun, Aug 8, 2010 at 5:00 AM, Nasir Iqbal 
mailto:na...@ictinnovations.com>> wrote:
Hi

following asterisk cli commands can help

show channels, show uptime and show sysinfo

here is an example

asterisk -x "core show sysinfo"
On Sun, Aug 8, 2010 at 12:25 AM, Richard Zulu 
mailto:richard.z...@time.co.ug>> wrote:

Hey guys,

I have my asterisk box running without a gui. I now need to monitor usage, 
calls, traffic of voice calls on this asterisk server. I cannot now install a 
gui because the configs will be wiped out, how can i go about monitoring all 
the above?

--
Richard Zulu
Managing Director
Time Information Company
P.O Box 31842
Clock Tower
Kampala, Uganda
www.time.co.ug

Mobile :+256752624006
Skype: zulu.richard


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P.O Box 31842
Clock Tower
Kampala, Uganda
www.time.co.ug

Mobile :+256752624006
Skype: zulu.richard

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Re: [asterisk-users] How does deny/permit work in sip.conf?

2010-08-08 Thread Matt Riddell 
On 7/08/10 3:47 PM, Frank Church wrote:
> On 7 August 2010 03:54, Bruce Ferrell  wrote:
>> On 08/06/2010 07:30 PM, Bruce Ferrell wrote:
>>> On 08/06/2010 02:16 PM, Frank Church wrote:
>>>
 On 6 August 2010 16:21, Bruce Ferrell  wrote:


> On 08/06/2010 07:45 AM, Frank Church wrote:
>
>
>> I have been seeing some attempts to register devices on my Asterisk
>> and I want to reconfigure it so that devices will be registered only
>> if they are from the correct address, ie 192.168.1.8/255.255.255.255.
>>
>> I thought using a config like
>>
>> deny=0.0.0.0/0.0.0.0
>> permit=192.168.1.8/255.255.255.255
>>
>> but it is not working the way I thought?
>>
>> Does that need a host=static.ip entry to work, rather than the
>> deny/permit option?
>>
>> Does using a host=dynamic setting override any deny/permit and
>> port=5060 options?
>>
>> Does being a peer or a user make a difference here?
>>
>>
>>
>>
> I had this same problem once.  host=or host=dynamic if you
> want to use permit/deny.  Permit/deny and host=dynamic allows a sip peer
> or user to have a range of addresses.
>
> --
>
>
 Does permit/deny  have any influence on registration, or is it related
 to the destinations it can call to or receive call from?

 How do you stop an asterisk server from accepting registrations when
 the IP is outside a subnet even if the username and secret are
 correct?

 When host=dynamic registrations are accepted even if the pemit IP is
 different from the registered device's IP address. Does permit/deny
 work on a  single IP address eg 192.168.4.111/255.255.255.2555


 The same seems to apply in the [general] section, with contactdeny and
 contacnt permit

 When I set

 contactdeny=0.0.0.0/0.0.0.0
 contactpermit=192.168.4.111/255.255.255.255

 Devices whose IP is not 192.168.4.111 are able to register.



>>> When I've used permit/deny, I did it in conjunction with insecure set to
>>> port,invite to allow gateways that didn't register and don't use
>>> username/secret to originate calls but only from the ip range in
>>> permit.  In fact it was for a provider that had gateways on a large
>>> number of IP addresses, all in the same CIDR block and I didn't want to
>>> do an entry for each of  more than 100 gateways.
>>>
>>> contactpermit/contactdeny *should* work as you are suggesting that you
>>> want I've never tried that.  I may attempt it tonight and see on my 1.4
>>> system.
>>>
>>>
>>
>> To follow up on my own reply.  I just tried this with one of my standard
>> peers that I use for a softphone on a 1.6.2.10  and see the registration
>> attempt come in at the console and a warning comes up
>>
>> : Host '192.0.2.40' disallowed by contact ACL (violating IP 192.0.2.40)
>> : Registration denied because of contact ACL
>>
>> The peer does show in sip show peers and the softphone (twinkle) shows a
>> Registration Fails with a 603 denied.
>>
>> So I'd say it's working
>>
>> --
>
> I am using 1.4.27 and it doesn't seem to work.
>
> I should probably try the 1.6 series

Are you using deny before permit?

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___

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[asterisk-users] PBX Status-like module for AsteriskNow?

2010-08-08 Thread Frank Tarczynski 
 I've moved from trixbox to AsteriskNow.  Does anyone know if there's
something like the PBX Status screen for AsteriskNow?

A module the shows the status of SIP and IAX2 registry and peers, etc
for all individual entries?

The FreePBX System Status screen shows when something fails to register
but doesn't show who it is.

Frank

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Re: [asterisk-users] Monitor asterisk

2010-08-08 Thread Richard Zulu 
Thanks Nasri,

I don't want to only be able to use the CLI because I need the Helpdesk and
application support Unit to be able to monitor, and they are not all the
techy with CLI and stuff..




On Sun, Aug 8, 2010 at 5:00 AM, Nasir Iqbal wrote:

> Hi
>
> following asterisk cli commands can help
>
> show channels, show uptime and show sysinfo
>
> here is an example
>
> asterisk -x "core show sysinfo"
>
> On Sun, Aug 8, 2010 at 12:25 AM, Richard Zulu wrote:
>
>>
>> Hey guys,
>>
>> I have my asterisk box running without a gui. I now need to monitor usage,
>> calls, traffic of voice calls on this asterisk server. I cannot now install
>> a gui because the configs will be wiped out, how can i go about monitoring
>> all the above?
>>
>> --
>> Richard Zulu
>> Managing Director
>> Time Information Company
>> P.O Box 31842
>> Clock Tower
>> Kampala, Uganda
>> www.time.co.ug
>>
>> Mobile :+256752624006
>> Skype: zulu.richard
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Nasir Iqbal
>
> ICT Innovations
> http://www.ictinnovations.com/
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>



-- 
Richard Zulu
Managing Director
Time Information Company
P.O Box 31842
Clock Tower
Kampala, Uganda
www.time.co.ug

Mobile :+256752624006
Skype: zulu.richard
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Re: [asterisk-users] Codec Conversion

2010-08-08 Thread Jeff Brower 
Steve-

>   On 08/07/2010 03:15 AM, Jeff Brower wrote:
>> Steve-
>>
>>> El 05/08/10 14:50, Tim Nelson escribió:
 - "michel freiha"wrote:
> Dear Sir,
>
> I tried to convert ilbc to ulaw and get the same result...Bad Voice
 Quality
> Regards
>
 Again, iLBC is poor quality to begin with. You can't take a poor audio
 sample and make it better by converting it to a codec with better
 'resolution'. An audio sample full of robot voice is going to sound
 like the same robot voice even if you transcode it to a better quality
 codec, whether that is G.729, G.711u, or the latest 'HD Voice' codecs.

 --Tim
>>> This just made me remember some comment on the iax.conf sample file...
>>>
>>> disallow=lpc10; Icky sound quality...  Mr. Roboto.
>> LPC10 is a very old codec, from early 1980s.  LPC10 doesn't do a good 
>> job with pitch detection so it tends to
>> have
>> a
>> 'robotic' sound.  With advent of MELPe, anyone needing bitrates 2400 or 
>> less should not be using LPC10.
>>
>> -Jeff
> MELPe is patent encumbered,
 Not if used for govt/defense purposes.  For commercial-only purposes, TI 
 will waive royalty fees if their chip is
 used
 in the product.  It would have been nice if Digium had considered the many 
 advantages of using a DSP pioneer such
 as
 TI before putting a Mindspeed chip on their TC400B card.
>>> I think all the IP for MELP is now in the hands of Compandent, and TI no
>>> longer has the ability to waive royalties.
>> That is not correct.  Compandent has filed copyrights on certain files 
>> associated with a C549 chip assembly language
>> implementation they did under contract to NSA around 2001.  TI has patent 
>> rights on 2400 bps, TI + Microsoft on 1200
>> bps, and TI + Microsoft + Thales Group on 600 bps.  Microsoft's IP came 
>> about as a result of acquiring a company
>> called SignalCom around 2001.  If the noise pre-processor is used, then 
>> there is some AT&T IP.  To verify this, you
>> can search dsprelated.com (specifically, look for posts discussing this 
>> issue on comp.dsp), and you can also read
>> the
>> "Compandent IPR" section of the MELPe Wikipedia page
>> (http://en.wikipedia.org/wiki/Mixed_Excitation_Linear_Prediction).  That 
>> section was authored by the Compandent's
>> founder, Oded Gottesman.  Oded is a super sharp, very hard working guy.
>>
>> Compandent also claims a copyright on some C code in the file melp_syn.c 
>> (synthesis filter).  I have read
>> discussions
>> by DSP experts indicating the copyrighted section of code can be implemented 
>> in alternative ways, but Oded may say
>> that's not accurate.
> That guy is PITA. He must have driven a lot of people away from MELP by
> the way he acts. He really annoys the regulars in the comp.dsp group by
> posting astroturf questions about MELP, and giving astroturf replies
> about how fantastic it is. That probably shapes a lot of my attitude to
> MELP. :-)
>>> Either way, government use
>>> and use with TI silicon are two niches that might work out well, and
>>> everything else is a problem for several more years. If you are going to
>>> pay royalties for a low bit rate codec, IMBE is probably a better option.
>> I would disagree because IMBE source is not available.  MELPe source is 
>> available and can be downloaded online.
> Depends what you mean by available. IMBE is patented, just like MELP is
> patented. Licence either, and implementations are available.

I meant that MELPe C source code is available for non-commercial purposes 
(academic, R&D, bug fixes and other source
level improvements) without payment and without signing a license agreement 
with a corporation (such as Digital Voice
with IMBE).

> IMBE has
> the great benefit of being widely used for commercial and amateur low
> bit rate channels. For example, amateur radio uses IMBE - an anomaly
> which is one of the drivers for David Rowe's work on an open low bit
> rate codec. Transcoding at low bit rates is a disaster, so using a codec
> you won't need to transcode is a big plus.

Yes all good points.  IMBE and AMBE have surely been successful, testaments to 
the Digital Voice guys and their
pioneering work in the LBR codec area.

>>> TI is a good option, but what do you have against Mindspeed? Choosing a
>>> good option for this kind of card is mostly about managing the patent
>>> licence fees. I assume Mindspeed gave Digium the best option for doing
>>> that, within Digium's volume constraints.
>> My understanding in talking to Digium engineers at Globalcom and other trade 
>> shows back in 2006 is they were worried
>> about interfacing the TI TNET series devices over the PCI bus.  They would 
>> have needed an FPGA with some non-trivial
>> logic programming, so I understand their decision.  But if they had got past 
>> their FPGA "writer's

Re: [asterisk-users] How to track a call result originated from originate AMI command

2010-08-08 Thread Jim Dickenson 
We track status of calls and many other actions using user events in our 
dialplan. The dial and queue commands allow for either agi or macros to be 
executed just before a connection is made. Use option g in dial to allow one to 
execute a user event after the dial command finishes. Use the h extension to 
track hang ups. We set an action token variable to a unique value for each 
originate and all the user events have this token so we can tie them back to 
the original originate. We turn off most AMI read message classes to cut down 
on packet volume.
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Aug 8, 2010, at 7:18 AM, Nasir Iqbal wrote:

> Hi,
> 
> Confusing! you are not alone here. Actually there is no unified development 
> approach exist in Asterisk, every module, application introduce a new way to 
> handle same things!! And the "monitoring" is most difficult part! you have to 
> write different parsing algos to get each bit of information, and 
> unfortunately you have to rewrite most of your code for every new release!
> 
> And regarding your question, I recommend you to use AGI for monitoring here 
> is some tips for you
> in originate command use extension as destination.
> create "failed" extension in same context.
> you can include some variables in originate command which can be used later 
> in dialplan.
> use AGI scripts in "destination" and "failed" extensions to get and save call 
> status in database.
> Regards
> 
> On Sun, Aug 8, 2010 at 6:10 PM, thiyagu venkatesan  
> wrote:
> Hi All,
> 
> 
> I want to track a call that is originated using originate AMI command through 
> AstManProxy server.
> 
> 
> I m using AstManProxy server and I developed an AstManProxy client.
> By using my AstManClient program I can able to login AstManProxy server.
> 
> 
> Now I can able to issue/send originate command to generate a call but I m 
> very confuse that I cannot able to track my
> call.
> 
> 
> The AMI events were very confusing and I m getting various events with 
> different uniqueid value.
> For a single call I m getting events with four or five uniqueid.
> I also filtered using specific channel but also I m getting events with 
> different uniqueid.
> 
> 
> How can I find the below status for the call generated using originate 
> command through AMI events,
> 1. Answer
> 2. No Answer
> 3. Busy
> 
> 
> Can any one help me for this.
> 
> 
> Thanks,
> 
> 
> Thiyagu VOIP
> 
> 
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> 
> 
> -- 
> Nasir Iqbal
> 
> ICT Innovations
> http://www.ictinnovations.com/
> 
> -- 
> _
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Re: [asterisk-users] How to track a call result originated from originate AMI command

2010-08-08 Thread Nasir Iqbal 
Hi,

Confusing! you are not alone here. Actually there is no unified
development approach exist in Asterisk, every module, application introduce
a new way to handle same things!! And the "monitoring" is most difficult
part! you have to write different parsing algos to get each bit of
information, and unfortunately you have to rewrite most of your code for
every new release!

And regarding your question, I recommend you to use AGI for monitoring here
is some tips for you

   - in originate command use extension as destination.
   - create "failed" extension in same context.
   - you can include some variables in originate command which can be used
   later in dialplan.
   - use AGI scripts in "destination" and "failed" extensions to get and
   save call status in database.

Regards

On Sun, Aug 8, 2010 at 6:10 PM, thiyagu venkatesan
wrote:

> Hi All,
>
>
> I want to track a call that is originated using originate AMI command
> through AstManProxy server.
>
>
> I m using AstManProxy server and I developed an AstManProxy client.
> By using my AstManClient program I can able to login AstManProxy server.
>
>
> Now I can able to issue/send originate command to generate a call but I m
> very confuse that I cannot able to track my
> call.
>
>
> The AMI events were very confusing and I m getting various events with
> different uniqueid value.
> For a single call I m getting events with four or five uniqueid.
> I also filtered using specific channel but also I m getting events with
> different uniqueid.
>
>
> How can I find the below status for the call generated using originate
> command through AMI events,
> 1. Answer
> 2. No Answer
> 3. Busy
>
>
> Can any one help me for this.
>
>
> Thanks,
>
>
> Thiyagu VOIP
>
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-- 
Nasir Iqbal

ICT Innovations
http://www.ictinnovations.com/
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[asterisk-users] How to track a call result originated from originate AMI command

2010-08-08 Thread thiyagu venkatesan 
Hi All,


I want to track a call that is originated using originate AMI command
through AstManProxy server.


I m using AstManProxy server and I developed an AstManProxy client.
By using my AstManClient program I can able to login AstManProxy server.


Now I can able to issue/send originate command to generate a call but I m
very confuse that I cannot able to track my
call.


The AMI events were very confusing and I m getting various events with
different uniqueid value.
For a single call I m getting events with four or five uniqueid.
I also filtered using specific channel but also I m getting events with
different uniqueid.


How can I find the below status for the call generated using originate
command through AMI events,
1. Answer
2. No Answer
3. Busy


Can any one help me for this.


Thanks,


Thiyagu VOIP
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Re: [asterisk-users] Asterisk Crashes - Segmentation Fault

2010-08-08 Thread Manmohan Singh Jandu 
Hi Dan,

I was trying to make the Invite working. I am getting following error
when i try to make a call.

[Aug  8 16:55:22] NOTICE[15082]: chan_local.c:534 local_call: No such
extension/context 73...@default while calling Local channel
[Aug  8 16:55:22] NOTICE[15082]: channel.c:4042
__ast_request_and_dial: Unable to call channel Local/73281
[Aug  8 16:55:22] ERROR[12166]: pbx.c:9301 device_state_cb: Received
invalid event that had no device IE
[Aug  8 16:55:22] ERROR[12166]: app_queue.c:1099 device_state_cb:
Received invalid event that had no device IE

Following is my dialplan in /etc/asterisk/extensions.conf

[outgoing]
exten => _73...,1,Dial(SIP/callman02&SIP/callman01/${EXTEN:2})
exten => _73...,n,Congestion

following is in lib/defines.php

//Outcall defaults
define ("CHAN_TYPE", "Local"); //Use Local to let dialplan decide which chan
define ("OUT_CONTEXT", "outgoing"); //Select a context to place the call from
define ("OUT_PEER", ""); // Use this if not using CHAN_TYPE Local
define ("OUT_CALL_CID", "Parlez <1996>"); // Caller ID for Invites

--Manmohan Singh
On Fri, Aug 6, 2010 at 12:46 AM, Dan Austin  wrote:
> Manmohan wrote:
>> I commented locale.php in defines.php and it perfectly worked well.
>
>> Now i am wondering what is this invite participants for, while adding
>> conference. wherein it asks for first name, lastname, emailaddress &
>> telephone number..
> The 'Invite Others' option is mostly for installs that do not have
> a consistent e-mail environment, and are using the SERVER mailer.
> This feature lets the server send invite emails to multiple parties.
> In my environments we have Exchange and Outlook, so I prefer the CLIENT
> mailer, and I can manage the invitations in my mail client
>
>> Let me brief you how i had done this setup. I had created a SIP trunk
>> between Cisco Call manager and Asterisk server. And i am using webmeetme
>> for Audio conferencing.
> Sounds familiar.  I put this package together after wasting too much
> money and time trying to make an expensive Cisco conferencing solution
> work.
>
>> Other than the invite participants, while the conf call is going on we
>> get couple of more options, when we click to the current ongoing conference
>> number.
>
>> End call -- To end the conference call
> Yes
>
>> Extend -- I am sure this is to extend the time of the call for which its
>> scheduled for, but not sure on how much time does it extends by default
>> OR is there any way to define the customized time on whatever required.
> 10 minutes is the default.  I thought I had made it configurable in 
> lib/defines.php,
> but no I have it hard coded in conf_add (to be fixed in the next release now).
> You can search for +600 and change it to any value you like.
>
>> Invite-- When i click this button it asks me telephone number. I assume this
>> is any number which asterisk server can reach as per the dialplan configured
>> in extension.conf in /etc/asterisk.. Though this invite button looks pretty
>> much interesting to use but whenever i enter any phone number it says
>> "System error" not sure if am understand this wrongly.
> You understand it correctly, but the default settings are likely not working.
> Check out the section 'Outcall defaults' in lib/defines.php.  It is likely you
> need to change the OUT_CONTEXT at a minimum.
>
> Dan
>
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-- 
Thanks & Regards
Manmohan Singh Jandu

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[asterisk-users] Asterisk 1.6.2 FastAGI Hangup Problem

2010-08-08 Thread Abeed Saleh 
Hi all,

I'm writing my own FastAGI server. I noticed when I send agi Hangup command
with and without the channel, asterisk does not hangup the channel and waits
for the AGI server to close the connection.

Is this how it's supposed to be?

Thanks

- Abeed
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