Re: [asterisk-users] Security - What inbound variables can attackers populate or use when calling?

2010-08-06 Thread jwexler 
> Use fail2ban.  Also, read some of the security advisories from earlier
this year about being sure to always use a FILTER statement whenever you're
dialing using > a variable (most notably ${EXTEN}).
http://downloads.asterisk.org/pub/security/AST-2010-002.html



Thanks Warren!!

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Saturday, August 07, 2010 1:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Security - What inbound variables can
attackers populate or use when calling?

 

On Fri, Aug 6, 2010 at 10:53 PM,  wrote:

Someone from Amsterdam was trying to register yesterday using an automated
program which tried roughly 1,000 or so username password combinations
before I shut asterisk down and added his/her ip to iptables to drop it. I
wonder if I can configure the system to automatically detect such an attack
in progress (e.g., a 1,000+ registration failures from the same ip is an
'attack') and the ip's to iptables, hosts.deny, etc. on the fly. That might
be another topic I guess?





Use fail2ban.  Also, read some of the security advisories from earlier this
year about being sure to always use a FILTER statement whenever you're
dialing using a variable (most notably ${EXTEN}).
http://downloads.asterisk.org/pub/security/AST-2010-002.html

-- 
Thanks,
--Warren Selby
http://www.selbytech.com

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[asterisk-users] Dahdi issue on sangoma A200

2010-08-06 Thread Max Alex 
Hi All,
I have Sangoma A200 Card installed on my system,
I have centos 5.5 with 64 bit,
Here are the description for asterisk and dahdi.
Asterisk 1.6..2.9
Dahdi: 2.3.0.1
I have two issues with dahdi
1) I am not getting full callerid on my phones from sangoma card to asterisk
users. if i am connecting analog phone directly then i am getting callerid
properly.
I am in india and using Airtel Connection, I have set variables in
chan_dahdi.conf as well for callerid but the not getting full digits in
callerid,
it is coming with 8 digits only.
2) Another issue is when I am hanging up the phone from inbound or outbound
from the dahdi channel, it takes 5-6 seconds to dropping the call.

Here are the confguration file for chan_dahdi.conf
-
;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
;autogenrated on 2010-07-30
;Dahdi Channels Configurations
;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak

[trunkgroups]

[channels]
context=default
usecallerid=yes
callerid=asreceived
hanguponpolarityswitch=yes
answeronpolarityswitch=yes
;cidstart=ring
cidstart=polarity_IN
;cidsignalling=dtmf
cidsignalling=dtmf
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
useincomingcalleridondahditransfer=yes
;callerid=asreceived

;Sangoma AFT-A200 [slot:4 bus:2 span:1]  
context=from-internal
group=1
echocancel=yes
callerid=asreceived
signalling = fxo_ks
channel => 1

context=from-internal
group=1
echocancel=yes
callerid=asreceived
signalling = fxo_ks
channel => 2

context=from-zaptel
group=0
echocancel=yes
callerid=asreceived
signalling = fxs_ks
channel => 3

context=from-zaptel
group=0
echocancel=yes
callerid=asreceived
signalling = fxs_ks
channel => 4
---
Please hemp me for this issues.

Thanks,
Max Alex
Voip Developer
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[asterisk-users] shrinkcallerid

2010-08-06 Thread Jayson Baker 
Am I really the only one having problems with this new "shrinkcallerid"?  I
can't find anything on Google about it.
Was happening on 1.6.2.10 and now on 1.8.0-beta2

In sip.conf shrinkcallerid=no, yet a name like "Joe Smith" ends up being
"JoeSmith"

Whoever though this up anyway is stupid.  Why would you want to strip spaces
out of a caller ID?

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

2010-08-06 Thread Chandrakant Solanki 
Hi

Can you tell me which Linux OS are you used  and what is speex / speex-devel
version.

Can you give details for above?

-- 
Regards,

Chandrakant Solanki

On Fri, Aug 6, 2010 at 6:22 PM, Nasir Iqbal wrote:

> Hi,
>
> May you also need to install *speex-tools* . if problem retain then let us
> know about your Linux distribution and Asterisk version.
>
> Regards
>
> On Fri, Aug 6, 2010 at 4:59 PM, Deepika Nijhawan <
> deepika.nijha...@oxygen8.com> wrote:
>
>>  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
>>
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>   http://www.asterisk.org/hello
>>
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>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Nasir Iqbal
>
> ICT Innovations
> http://www.ictinnovations.com/
>
>
> --
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Re: [asterisk-users] Security - What inbound variables can attackers populate or use when calling?

2010-08-06 Thread Warren Selby 
On Fri, Aug 6, 2010 at 10:53 PM,  wrote:

> Someone from Amsterdam was trying to register yesterday using an automated
> program which tried roughly 1,000 or so username password combinations
> before I shut asterisk down and added his/her ip to iptables to drop it. I
> wonder if I can configure the system to automatically detect such an attack
> in progress (e.g., a 1,000+ registration failures from the same ip is an
> ‘attack’) and the ip’s to iptables, hosts.deny, etc. on the fly. That might
> be another topic I guess?
>
>
Use fail2ban.  Also, read some of the security advisories from earlier this
year about being sure to always use a FILTER statement whenever you're
dialing using a variable (most notably ${EXTEN}).
http://downloads.asterisk.org/pub/security/AST-2010-002.html

-- 
Thanks,
--Warren Selby
http://www.selbytech.com
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Re: [asterisk-users] Security - What inbound variables can attackers populate or use when calling?

2010-08-06 Thread jwexler 
Well, I'm not sure actually. I was attacked in June by someone who racked up
between $800 and $900 in international calls to places in the middle of
Africa, Korea, etc. So, I am motivated to secure this. I have made it much
much more secure, definitely, but am looking for as many ways to further
lock this down as possible.

 

I figure that I should filter every field that someone could possible
interact with Asterisk in case they send characters that might breach
security and allow them some kind of access. Symbols like the amperstand
(&), comma (,), forward slash (/), at (@), pipe (|), etc. I would guess
could be bad.

 

Someone from Amsterdam was trying to register yesterday using an automated
program which tried roughly 1,000 or so username password combinations
before I shut asterisk down and added his/her ip to iptables to drop it. I
wonder if I can configure the system to automatically detect such an attack
in progress (e.g., a 1,000+ registration failures from the same ip is an
'attack') and the ip's to iptables, hosts.deny, etc. on the fly. That might
be another topic I guess?

 

This experience has emphasized the importance of securing the system and
security in asterisk in general.

 

Any insight on this would be really appreciated!

 

Thanks!!

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mike mosier
Sent: Saturday, August 07, 2010 11:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Security - What inbound variables can
attackers populate or use when calling?

 

What kind of attack can they reform calling in?

On Aug 6, 2010 1:12 AM,  wrote:
> I am setting filters, etc. on variables that attackers can send asterisk
> when they call (for example when they initially call into asterisk).
> 
> So far, I am filtering:
> 
> exten
> 
> CALLERID(name)
> 
> CALLERID(num)
> 
> 
> 
> What other fields or variables would an attacker be able to use in the
> packets that they send when placing the call to asterisk?
> 
> 
> 
> Further, I am assuming that in the case that an attacker, first, simply
> dials in normally and then after reaching voice prompts or other, starts
> his/her attack, then all I need to filter in that case is exten. Anything
> else here as well?
> 
> 
> 
> Thanks!!
> 

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

2010-08-06 Thread Frank Church 
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


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

2010-08-06 Thread Bruce Ferrell 
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

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Re: [asterisk-users] Security - What inbound variables can attackers populate or use when calling?

2010-08-06 Thread mike mosier 
What kind of attack can they reform calling in?

On Aug 6, 2010 1:12 AM,  wrote:
> I am setting filters, etc. on variables that attackers can send asterisk
> when they call (for example when they initially call into asterisk).
>
> So far, I am filtering:
>
> exten
>
> CALLERID(name)
>
> CALLERID(num)
>
>
>
> What other fields or variables would an attacker be able to use in the
> packets that they send when placing the call to asterisk?
>
>
>
> Further, I am assuming that in the case that an attacker, first, simply
> dials in normally and then after reaching voice prompts or other, starts
> his/her attack, then all I need to filter in that case is exten. Anything
> else here as well?
>
>
>
> Thanks!!
>
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Re: [asterisk-users] Codec Conversion

2010-08-06 Thread Steve Underwood 
  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. 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.


>> 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 block", they could
> have put one TNETV3010 chip on there, even smaller than the Mindspeed and 
> without the heat sink, and had twice the
> channel capacity as they do now.
TI have had DSP chips with a PCI interface for years, so that 
explanation doesn't make a lot of sense. Of course, these days you need 
a PCI-E interface. I'm not so sure about the status of those in DSP chips.
>> so there is still a place for LPC10 [...]

> e>>  I haven't seen an LPC10 implementation with MOS higher than 2.5.  Due to 
> its

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

2010-08-06 Thread Bruce Ferrell 
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.

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[asterisk-users] Set outgoing number in filename of the recordings

2010-08-06 Thread Rushikesh 
Hi,

Im not able to set the outgoing number in filename for asterisk recordings


Following is what I have done in 
/var/lib/asterisk/agi-bin/recordingcheck file


.
.
.
include("phpagi.php");


/**/

$agi = new AGI();
$temp = $agi->get_variable("agi_dnid") ;   // I have also tried with 
get_variable("DIAL_NUMBER")

if($temp['result'] == 1 ) {
$dnid = $temp['data'] ;
}
else
{
$dnid = "NUMBER" ;
}

$timestamp = $argv[1];
$uniqueid = $argv[2];
$type = $agi->get_variable("ARG2");

.
.
.



Please help me


Regards,
Rishi

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

2010-08-06 Thread jwexler 
This works. I have tested with the following settings:
In regards to the specifics of your question:
In sip.conf:
dynamic_exclude_static=yes

In users.conf, for each user (changing the permit statement to the ip of
each user):
hassip=yes
host=dynamic
registersip=yes
deny=0.0.0.0/0.0.0.0
permit=192.168.1.8/255.255.255.255   (using your ip setting)

Hope that helps

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank Church
Sent: Friday, August 06, 2010 11:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How does deny/permit work in sip.conf?

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?

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Re: [asterisk-users] sip.conf register in realtime DB

2010-08-06 Thread unserossi 
>   You cannot use realtime static and the other realtime tables at the



> same time.  You will need to use realtime and then use something like

> the EXEC command in sip.conf to execute a script that then pulls the

> register statement from your database.  Or use the realtime static table

> for everything.



Using the EXEC-command in sip.conf means I will have to issue a "sip 

reload" when I want to load changes in the database ?!



New information that is put into the REGISTER-database is not available 

without a 'sip reload' ?!



If not, do you see another way to have new registrations without a 'sip 

reload' ?!





Kind regards,



Jonas.



-- 

Why don't you use 'real' realtime meaning to have your sip peers in your 
database?
Then you would not have to do a reload after adding new peers to your db.
And you can still have sip peers additionally in sip.conf.

 
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Re: [asterisk-users] sip.conf register in realtime DB

2010-08-06 Thread Jonas Kellens 
On 08/06/2010 06:45 PM, Carlos Chavez wrote:
> Or use the realtime static table for everything.

What do you mean by "everything" ?! What is this "everything" ?!

You mean all the sip options in a database and so no sip.conf file ?!



Kind regards,

Jonas.

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Re: [asterisk-users] sip.conf register in realtime DB

2010-08-06 Thread Jonas Kellens 
On 08/06/2010 06:45 PM, Carlos Chavez wrote:
>   You cannot use realtime static and the other realtime tables at the
> same time.  You will need to use realtime and then use something like
> the EXEC command in sip.conf to execute a script that then pulls the
> register statement from your database.  Or use the realtime static table
> for everything.

Using the EXEC-command in sip.conf means I will have to issue a "sip 
reload" when I want to load changes in the database ?!

New information that is put into the REGISTER-database is not available 
without a 'sip reload' ?!

If not, do you see another way to have new registrations without a 'sip 
reload' ?!


Kind regards,

Jonas.

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Re: [asterisk-users] How to reuse mysql connection between AGI's

2010-08-06 Thread Steve Edwards 

On Thu, 5 Aug 2010, Faheem wrote:


Hey, Is there any way to share MySQL connection between different agi's.


No.

Each AGI is executed as a separate process. While debugging (not 
interacting with a "real" call), you can executed your AGI completely 
independent from Asterisk from the command line if you feed it appropriate 
cruft via stdin.


Actually when call comes to asterisk box it executes various agi scripts 
sequentially. Each script checks various values by making a 
new MySQL connection and then execute query and then disconnects. 


So, Ideally there should be one connection, and it should be reused 
between each agi and when a call is over it should be disconnected. Is 
there any mechanism to reuse single MySQL connection between agi 
scripts? The agi scripts are written in Perl


I suspect the issue is Perl, not MySQL.

If you are doing something often enough to be a performance issue, why did 
you write it in an interpreted script language instead of a compiled 
language?


In previous experiments, I demonstrated you can execute XXX AGIs written 
in c in the time you can load the interpreter, parse your script, and 
execute a single AGI written in Perl or PHP.


In that experiment, the "null-agi" only read the AGI environment and 
exited.


I added mysql_init(), mysql_real_connect(), and mysql_close() and...

On my wimpy (by current standards) 500MHz AMD Geode I can execute about 20 
"null-agi's" per second. On my almost as wimpy 1.1GHz AMD 8650 I can 
execute about 100 per second.


I'd suggest re-implementing your AGIs in c and "merging" a couple of the 
AGIs together if it makes sense for your environment.


As an alternative, you could go the fastagi() route. This means changing 
your code to execute as a daemon which means you can keep the same MySQL 
connection and you eliminate the AGI process creation overhead.


It also introduces complexity in handling simultaneous call execution, you 
become dependent on that process being available, and you lose some 
flexibility in being able to make changes to your AGIs without affecting 
calls in progress.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000-- 
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Re: [asterisk-users] Need USA DIDs

2010-08-06 Thread Nyamul Hassan 
Sorry for asking this after 6 week of staleness on this post, but do you
know which FCC registration is needed?  We already have a 214.  Is that it?

Thanks.

Regards
HASSAN



On Thu, Jun 24, 2010 at 00:43, Tarek Sawah  wrote:

>  i consuleted didforsale.com regarding the wholesale thing and their
> response was that you should buy a bulk of numbers and make your own api..
> one more thing.. if you are in the USA ..be sure to start your FCC
> registration (if you don't have it yet) because it can be a disaster for US
> companies providing DID numbers to US citizens without FCC license.
>
>
> -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1
> 347 562 2308
>
>
>
> --
> Date: Wed, 23 Jun 2010 23:43:14 +0530
>
> From: rscl.mum...@gmail.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Need USA DIDs
>
> On Wed, Jun 23, 2010 at 9:50 PM, Hall, Rick  wrote:
>
>  Agreed!  Didforsale.com is THE way to go.
>
>
>
> --
> Rick Hall
> Senior Vice President
> ReadyWire Multimedia Solutions
>
>
> Anyone having experience with didww.com ?
>
> Sorry, I forgot to mention I am looking for wholesale DID -- reseller
> option with API to that my customers can select country - city -- DID from
> my website.
>
> Thx
>
>
>
>
> --
> The New Busy is not the too busy. Combine all your e-mail accounts with
> Hotmail. Get 
> busy.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] How does deny/permit work in sip.conf?

2010-08-06 Thread Frank Church 
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.

> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

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Re: [asterisk-users] Asterisk and OCS2007 R2

2010-08-06 Thread unserossi 
I did it straight Asterisk to OCS using the OCS Mediation Server. We do have 
Dialogic Diva Server Cards which are able to be used as Media Gateway too using 
an additional software called SipControl (not 100% sure about the name) but as 
this software needs to be licensed separately I prefer a direct setup like a 
actually have.

Oliver


Oliver,
 



Did you happen to use a Dialog Media gateway in the mix, or is 
this straight Asterisk to OCS? We are implementing this in my shop, and running 
a bit of a ground (trying to use the MediaGateway). Any help anyone could 
provide would be wonderful. Thanks all.
 
Lyle J. McKarns
---
Networking/Linux Engineering Team
n|m Nexus Management
4 Industrial Parkway
Suite 101
Brunswick, Maine 04011
 
Tel (USA)   : 1 207 319 1105
Tel (UK)  : 0207 100 4968
Fax: 1 207 725 8552
Nexus Management, Inc.│ Registered Office:  4 Industrial Parkway, Suite 101, 
Brunswick, Maine.  04011│Company No. 19891257D, Registered in Maine│ A member 
of the Nexus Management Plc group of companies
 
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of unsero...@aol.com
Sent: Friday, August 06, 2010 3:50 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk and OCS2007 R2
 



Hi all,


i finally got it working to use Asterisk 1.6.1.20 as my PSTN gateway for 
OCS2007 R2 following the HowTo 
http://blogs.technet.com/b/gclark/archive/2008/10/09/asterisk-1-6-with-office-communications-server-2007.aspx.

I can call the OCS from Asterisk and vice versa.

The only thing that doesn't work is displaying the callerid(name) in OCS 
Communicator 2007.
I can manipulate the callerid(num) in Asterisk to any value I want using Set(), 
but callerid(name) is not displayed.

Has anyone a similar setup with names being displayed in OCS?

Thanks in advance.
Oliver



 


=
 
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Re: [asterisk-users] Asterisk and OCS2007 R2

2010-08-06 Thread Lyle McKarns 
Oliver,
Did you happen to use a Dialog Media gateway in the mix, or is 
this straight Asterisk to OCS? We are implementing this in my shop, and running 
a bit of a ground (trying to use the MediaGateway). Any help anyone could 
provide would be wonderful. Thanks all.

Lyle J. McKarns
---
Networking/Linux Engineering Team
n|m Nexus Management
4 Industrial Parkway
Suite 101
Brunswick, Maine 04011

Tel (USA)   : 1 207 319 1105
Tel (UK)  : 0207 100 4968
Fax: 1 207 725 8552
Nexus Management, Inc.│ Registered Office:  4 Industrial Parkway, Suite 101, 
Brunswick, Maine.  04011│Company No. 19891257D, Registered in Maine│ A member 
of the Nexus Management Plc group of companies

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of unsero...@aol.com
Sent: Friday, August 06, 2010 3:50 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk and OCS2007 R2

Hi all,

i finally got it working to use Asterisk 1.6.1.20 as my PSTN gateway for 
OCS2007 R2 following the HowTo
http://blogs.technet.com/b/gclark/archive/2008/10/09/asterisk-1-6-with-office-communications-server-2007.aspx.

I can call the OCS from Asterisk and vice versa.

The only thing that doesn't work is displaying the callerid(name) in OCS 
Communicator 2007.
I can manipulate the callerid(num) in Asterisk to any value I want using Set(), 
but callerid(name) is not displayed.

Has anyone a similar setup with names being displayed in OCS?

Thanks in advance.
Oliver


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[asterisk-users] Asterisk and OCS2007 R2

2010-08-06 Thread unserossi 



Hi all,


i finally got it working to use Asterisk 1.6.1.20 as my PSTN gateway for 
OCS2007 R2 following the HowTo 
http://blogs.technet.com/b/gclark/archive/2008/10/09/asterisk-1-6-with-office-communications-server-2007.aspx.

I can call the OCS from Asterisk and vice versa.

The only thing that doesn't work is displaying the callerid(name) in OCS 
Communicator 2007.
I can manipulate the callerid(num) in Asterisk to any value I want using Set(), 
but callerid(name) is not displayed.

Has anyone a similar setup with names being displayed in OCS?

Thanks in advance.
Oliver



 


 
 
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Re: [asterisk-users] Codec Conversion

2010-08-06 Thread Jeff Brower 
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.

> 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.

> 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 block", they could
have put one TNETV3010 chip on there, even smaller than the Mindspeed and 
without the heat sink, and had twice the
channel capacity as they do now.

>>> so there is still a place for LPC10 [...]
e>> I haven't seen an LPC10 implementation with MOS higher than 2.5.  Due to 
its age and expiration of patents, LPC10
>> might be a basis for a 2400 bps open source codec.  But enormous improvement 
>> would be needed to come close to MELPe
>> performance.
>>
>>
> MELPe is definitely a compandent thing, and TI cannot waive fees for
> that. MELP and MELPe are derived from LPC10. Any attempt to improve
> LPC10 would take you down a similar road, though you would need to skirt
> around the patents.

Again, not correct.  Suggest to research the many online independent sources, 
or contact NSA (who initiated the
overall MELPe effort in the 1990s, in response to a need to significantly 
improve over LPC10) and who can give you a
complete IP list.

> Do you really consider MELPe to be an enormous improvement over LPC10?
> Its still pretty lousy compared to a number of options at about 5kbps,
> and RTP overheads mean the gain from going lower than 5k isn't that big.
> The main reason LPC10 and MELPe offer a low bit rate in RTP is the
> minimum packet you can pack 22.5ms frames into sanely is a 90ms one.

In MOS terms, yes.  In VoIP terms, I agree it's not clear cut.  At 2400 bps, a 
90 msec packet would

Re: [asterisk-users] Reinstalling Asterisk due to hardware changes

2010-08-06 Thread Danny Nicholas 
>From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
jeremy.hellst...@synovate.com
>Subject: [asterisk-users] Reinstalling Asterisk due to hardware changes

 

>My purely SIP experiment has failed so I am purchasing a Digium E1/T1 card
to put into my Asterisk box.

 

>I know from the wonderful O'Reilly book that the proper installation is
Zaptel --> libpri --> Asterisk.  Is it possible to simply reinstall in that
order once I have installed the card and have Asterisk successfully work or
is it best to uninstall all three packages and install from scratch?

 

In my experience, all you should need to do is install Zaptel/Dahdi and
libpri, then do a new make && make install on your existing installation.

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[asterisk-users] Reinstalling Asterisk due to hardware changes

2010-08-06 Thread Jeremy.Hellstrom 
My purely SIP experiment has failed so I am purchasing a Digium E1/T1 card to 
put into my Asterisk box.

 

I know from the wonderful O'Reilly book that the proper installation is Zaptel 
à libpri à Asterisk.  Is it possible to simply reinstall in that order once I 
have installed the card and have Asterisk successfully work or is it best to 
uninstall all three packages and install from scratch?

 

Thanks, Jeremy

 

Jeremy Hellstrom   

 

Specialist, IT/Systems/Sampling

Phone (604) 664 2472

Fax (604) 664 2456

Suite 1550

1090 West Georgia St.

Vancouver BC

V6E 3V7

 

www.synovate.com  

 



Vancouver's ultimate focus group facility

State-of-the-art focus group facility, now with FocusVision! Synovate is your 
one-stop shop for all qualitative research needs from recruiting, to 
moderating, to facilitating. To learn more, visit 
www.synovate.com/vancouverfocusgroups 
  

 

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[asterisk-users] Asterisk and OCS2007 R2

2010-08-06 Thread unserossi 
Hi all,

i finally got it working to use Asterisk 1.6.1.20 as my PSTN gateway for 
OCS2007 R2 following the HowTo 
http://blogs.technet.com/b/gclark/archive/2008/10/09/asterisk-1-6-with-office-communications-server-2007.aspx.

I can call the OCS from Asterisk and vice versa.

The only thing that doesn't work is displaying the callerid(name) in OCS 
Communicator 2007.
I can manipulate the callerid(num) in Asterisk to any value I want using Set(), 
but callerid(name) is not displayed.

Has anyone a similar setup with names being displayed in OCS?

Thanks in advance.
Oliver


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Re: [asterisk-users] OT: Grandstream GXV3140

2010-08-06 Thread --[ UxBoD ]-- 
- Original Message -
> On Fri, 6 Aug 2010, --[ UxBoD ]-- wrote:
> 
> > Hi,
> >
> > Do any of you have these phones ? How have you found it ? Are you
> > using them over WiFi or hard wired ? Does it play nicely with
> > Asterisk ?
> >
> > Need to replace my Snom M3s and this phone maybe a contender.
> 
> Full of bugs. Stay away. Vendors have started dropping this phone. I
> use them personally when traveling (to talk to my baby girl), but as a
> commercial product this phone is AWFUL.
> 
> j

Oh, that is very disappointing indeed; especially some of the others bits I 
read on the net :( Desperately trying to find a new phone that will support:

* Centralised management and deployment
* OpenVPN
* Asterisk support

The company I work for have got very despondent with Snom and their lack of 
support.
-- 
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Re: [asterisk-users] Codec Conversion

2010-08-06 Thread Steve Underwood 
  On 08/06/2010 04:43 PM, Jeff Brower wrote:
> Steve-
>
>>On 08/06/2010 05:40 AM, Jeff Brower wrote:
>>> Miguel-
>>>
 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. 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.

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.
>> so there is still a place for LPC10 [...]
> I haven't seen an LPC10 implementation with MOS higher than 2.5.  Due to its 
> age and expiration of patents, LPC10
> might be a basis for a 2400 bps open source codec.  But enormous improvement 
> would be needed to come close to MELPe
> performance.
>
>
MELPe is definitely a compandent thing, and TI cannot waive fees for 
that. MELP and MELPe are derived from LPC10. Any attempt to improve 
LPC10 would take you down a similar road, though you would need to skirt 
around the patents.

Do you really consider MELPe to be an enormous improvement over LPC10? 
Its still pretty lousy compared to a number of options at about 5kbps, 
and RTP overheads mean the gain from going lower than 5k isn't that big. 
The main reason LPC10 and MELPe offer a low bit rate in RTP is the 
minimum packet you can pack 22.5ms frames into sanely is a 90ms one. 
90ms RTP *really* cuts the overheads, compared to the more typical 20ms 
or 30ms packets used for G.729.

As others have mentioned, David Rowe is working on a modern 2400bps 
codec. He did a burst of work some time ago, and then put it aside while 
busy with other things. He recently told me he is restarting the work, 
and he wants to get that codec into good shape before the end of this year.

Steve

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Re: [asterisk-users] Asterisk 1.4 and TE420P

2010-08-06 Thread Warren Selby 
On Fri, Aug 6, 2010 at 11:24 AM, Danny Nicholas  wrote:

>   *>>From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Warren Selby
> >>*Subject:* Re: [asterisk-users] Asterisk 1.4 and TE420P
>
>
>
> >On Fri, Aug 6, 2010 at 10:33 AM, James Texter 
> wrote:
>
> >I have a site running 1.4.17 with Zaptel.  They want to add a TE420P for
> additional T1 capacity.  I'm 99% sure this will work, anyone aware of a
> reason it wont?
>
>
>
>
> >>I've got a client running a TE420P with asterisk 1.4.33.1 and DAHDI 2.2,
> works fine.  They've only got the one card in the box, but it's using all 4
> ports.
>
> To expand on OP’s question, is he going to have to upgrade to 1.4.3X/DAHDI
> to make the TE420P work?
>
>

I've only ever used it with DAHDI.  I've used it with version of asterisk
down to 1.4.22 though, I'm pretty sure.

-- 
Thanks,
--Warren Selby
http://www.selbytech.com
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Re: [asterisk-users] sip.conf register in realtime DB

2010-08-06 Thread Carlos Chavez 
You cannot use realtime static and the other realtime tables at the
same time.  You will need to use realtime and then use something like
the EXEC command in sip.conf to execute a script that then pulls the
register statement from your database.  Or use the realtime static table
for everything.

On Fri, 2010-08-06 at 10:57 +0200, Jonas Kellens wrote:
> Please can anyone help me with this ?!
> 
> I have tried renaming the sip.conf file, or tried including another
> file into sip.conf like sippy.conf and then add sippy.conf =>
> mysql,AsteriskDB,ast_config to extconfig.conf but all this is not
> working.
> 
> The only thing that changes something is my example listed below, but
> then I always have that nasty WARNING which I find odd.
> 
> I need realtime sip registrations (so without having to do a sip
> reload).
> 
> 
> 
> 
> Kind regards,
> 
> Jonas.
> 
> 
> On 08/03/2010 10:13 AM, Jonas Kellens wrote: 
> > Hello list,
> > 
> > scrambling different pieces of info together I've come with the
> > following :
> > 
> > I want to have my "register =>" statements in a MySQL-database, so
> > I've made the following table.
> > 
> > table ast_config :
> > id  1
> > cat_metric  0
> > var_metric  0
> > commented  0
> > filename  sip.conf
> > category  general
> > var_name  register
> > var_val  username:passw...@sip.provider.net
> > 
> > 
> > In ext_config (text file) I have :
> > 
> > sipusers => mysql,AsteriskDB,sip_buddies
> > sippeers => mysql,AsteriskDB,sip_buddies
> > sip.conf => mysql,AsteriskDB,ast_config
> > 
> > In sip.conf (text file) I have also :
> > 
> > sip.conf :
> > rtcachefriends=yes ; Cache realtime friends by adding
> > them to the internal list
> > ; just like friends added from the
> > config file only on a
> > ; as-needed basis? (yes|no)
> > 
> > 
> > After a reload I noticed that the registration came through when I
> > executed "sip show registrations". This realtime works.
> > But I then get a lot of the following messages :
> > 
> > [Aug  2 20:05:05] WARNING[29850]: chan_sip.c:18187 build_peer:
> > Qualify is incompatible with dynamic uncached realtime.  Please
> > either turn rtcachefriends on or turn qualify off on peer 'user2'
> > [Aug  2 20:05:05] WARNING[29850]: chan_sip.c:18187 build_peer:
> > Qualify is incompatible with dynamic uncached realtime.  Please
> > either turn rtcachefriends on or turn qualify off on peer 'user2'
> > 
> > 
> > rtcachefriends is turned on (see above)
> > qualify is on on every peer (and I want it to stay that way)
> > 
> > 
> > 
> > Can anyone tell me what I need to configure to get a 100% working
> > example ?!
> > 
> > 
> > 
> > Kind regards,
> > 
> > Jonas.
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-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] Asterisk 1.4 and TE420P

2010-08-06 Thread Danny Nicholas 
>>From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
>>Subject: Re: [asterisk-users] Asterisk 1.4 and TE420P

 

>On Fri, Aug 6, 2010 at 10:33 AM, James Texter 
wrote:

>I have a site running 1.4.17 with Zaptel.  They want to add a TE420P for
additional T1 capacity.  I'm 99% sure this will work, anyone aware of a
reason it wont?

 


>>I've got a client running a TE420P with asterisk 1.4.33.1 and DAHDI 2.2,
works fine.  They've only got the one card in the box, but it's using all 4
ports.

To expand on OP's question, is he going to have to upgrade to 1.4.3X/DAHDI
to make the TE420P work?

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Re: [asterisk-users] Asterisk 1.4 and TE420P

2010-08-06 Thread Warren Selby 
On Fri, Aug 6, 2010 at 10:33 AM, James Texter wrote:

> I have a site running 1.4.17 with Zaptel.  They want to add a TE420P for
> additional T1 capacity.  I'm 99% sure this will work, anyone aware of a
> reason it wont?
>
> Thanks,
>
> James
>
>
I've got a client running a TE420P with asterisk 1.4.33.1 and DAHDI 2.2,
works fine.  They've only got the one card in the box, but it's using all 4
ports.

-- 
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--Warren Selby
http://www.selbytech.com
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[asterisk-users] Asterisk 1.4 and TE420P

2010-08-06 Thread James Texter 
I have a site running 1.4.17 with Zaptel.  They want to add a TE420P for
additional T1 capacity.  I'm 99% sure this will work, anyone aware of a
reason it wont?

Thanks,

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

2010-08-06 Thread Bruce Ferrell 
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.

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[asterisk-users] OT: Grandstream GXV3140

2010-08-06 Thread --[ UxBoD ]-- 
Hi,

Do any of you have these phones ? How have you found it ? Are you using them 
over WiFi or hard wired ? Does it play nicely with Asterisk ?

Need to replace my Snom M3s and this phone maybe a contender.
-- 
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Re: [asterisk-users] Using a 1.4 config with 1.6

2010-08-06 Thread Paul Belanger 
On Fri, Aug 6, 2010 at 10:36 AM, Roderick A. Anderson
 wrote:
> I have a rather simple setup running under Asterisk 1.4.  I'd like to
> move it to a new install of 1.6.  Before I bring it online are there any
> gotchas I should look for?  A Gotcha README would be better but
> searching with Google and the forums, for me, gets hits that deal with
> hardware issues -- cards etc.  Nothing about depreciated/changed commands.
>
Be sure to read CHANGES and UPGRADE.txt within the source directory of Asterisk.

-- 
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com

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Re: [asterisk-users] Using a 1.4 config with 1.6

2010-08-06 Thread Roderick A. Anderson 
Danny Nicholas wrote:
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roderick A.
> Anderson
> Subject: [asterisk-users] Using a 1.4 config with 1.6
> 
> I have a rather simple setup running under Asterisk 1.4.  I'd like to 
> move it to a new install of 1.6.  Before I bring it online are there any 
> gotchas I should look for?  
> 
> 
> TIA,
> Rod
> 
> I am doing dual testing with 1.4.3X and 1.6X and the only thing that has hit
> me so far is having to change the dialplan delimiters (from "|" to ",")  If
> your dialplan is simple enough, this may not affect you at all.

Excellent!  Mine is very simple.  Open/Closed hours, four menu options.


Thanks,
Rod
-- 


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

2010-08-06 Thread Nasir Iqbal 
Hi,

Currently CentOS yum repository does not provide speex-tools so you have to
install it manaully. follow the steps given below.

1. first remove existing speex pacages yum remove speex*

2. run following to install required rpms
rpm -ivh
http://www.lfarkas.org/linux/packages/centos/5/i386/gstreamer/speex-1.2-0.10.rc1.i386.rpm
rpm -ivh
http://www.lfarkas.org/linux/packages/centos/5/i386/gstreamer/speex-devel-1.2-0.10.rc1.i386.rpm
rpm -ivh
http://www.lfarkas.org/linux/packages/centos/5/i386/gstreamer/speex-tools-1.2-0.10.rc1.i386.rpm

3. reconfigure asterisk

4. verify speex codec via "make menuselect" under "Codec Translators". if
speex is enabled and selected continue with make

Enjoy!

On Fri, Aug 6, 2010 at 7:13 PM, Deepika Nijhawan <
deepika.nijha...@oxygen8.com> wrote:

>  Hi Chandrakant
>
>
>
> I have checked and it shows func_speex module is enabled.
>
> Where can I install speex-tools from ?
>
> Asterisk version 1.6.2.10 and Centos 5.5 are installed.
>
>
>
> ---
>
>
>
> Kind Regards,
>
>
>
> *Deepika Nijhawan*
>
> *VoIP Engineer*
>
> * *
>
> *Oxygen8* Communications
>
>
>
>
>
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-- 
Nasir Iqbal

ICT Innovations
http://www.ictinnovations.com/
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[asterisk-users] How does deny/permit work in sip.conf?

2010-08-06 Thread Frank Church 
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?

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Re: [asterisk-users] Using a 1.4 config with 1.6

2010-08-06 Thread Danny Nicholas 
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roderick A.
Anderson
Subject: [asterisk-users] Using a 1.4 config with 1.6

I have a rather simple setup running under Asterisk 1.4.  I'd like to 
move it to a new install of 1.6.  Before I bring it online are there any 
gotchas I should look for?  


TIA,
Rod

I am doing dual testing with 1.4.3X and 1.6X and the only thing that has hit
me so far is having to change the dialplan delimiters (from "|" to ",")  If
your dialplan is simple enough, this may not affect you at all.


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[asterisk-users] Using a 1.4 config with 1.6

2010-08-06 Thread Roderick A. Anderson 
I have a rather simple setup running under Asterisk 1.4.  I'd like to 
move it to a new install of 1.6.  Before I bring it online are there any 
gotchas I should look for?  A Gotcha README would be better but 
searching with Google and the forums, for me, gets hits that deal with 
hardware issues -- cards etc.  Nothing about depreciated/changed commands.


TIA,
Rod
-- 

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

2010-08-06 Thread Deepika Nijhawan 
Hi Chandrakant

 

I have checked and it shows func_speex module is enabled. 

Where can I install speex-tools from ?

Asterisk version 1.6.2.10 and Centos 5.5 are installed.

 

---

 

Kind Regards,

 

Deepika Nijhawan

VoIP Engineer

 

Oxygen8 Communications 

 

 

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Re: [asterisk-users] CDR report

2010-08-06 Thread Olivier 
2010/8/5 Dario Quiroz 

> Hi all!
> Are someone using a CDR report? I have an Asterisk 1.6 running perfect but
> I need a web based report of CDRs.
> Nothing big, only the basic. Have anybody a how-to or a link?
>

Look for asterisk-stats (or cdr-stats as it has been renamed) : users are
rather pleased with it.


> Thanks in advance!!
>
> --
> Atenciosamente,
>
> ---
>
>  Dario Quiroz
>
> Analista de Suporte
>
>(71) 9275-9080
> darioqui...@gmail.com
>
> ---
>
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[asterisk-users] T.38 fax between ATA's and Asterisk and Cisco PGW 2200

2010-08-06 Thread P Z 

To provide a reliable fax solution for users connected to a Asterisk 1.6.2.6 
server i have tested a few T.38 capable ATA's:
- Patton M-ATA
- Grandstream HandyTone 486
- Fritz!Box 7170

I have tried Asterisk 1.6.2.6 compiled with SpanDSP-0.0.6pre17 and also 
Asterisk 1.6.2.6 with Fax for Asterisk installed.

These Asterisk servers are connected to a Cisco PGW 2200 + AS5400XM.


Sending fax messages from all ATA's to the PSTN (so ATA -> Asterisk -> PGW -> 
PSTN) failed with a variety of error messages so i tested the different steps 
one by one.


ATA's -> Asterisk ReceiveFax:
So far i have only succeeded in sending fax messages from the Fritz!Box 7170 to 
both Asterisk configurations using the ReceiveFax application.
Sending fax messages from the other ATA's to Asterisk using the ReceiveFax 
application failed.


ATA's -> PGW:
To exclude Asterisk i have connected the ATA's directly to the PGW; no success 
either.


Asterisk -> PGW:
To exclude the ATA's i used the Asterisk SendFax application to send a TIFF 
file to a landline each time with a different fax machine connected to it. 
Results:


Asterisk 1.6.2.6 compiled with SpanDSP-0.0.6pre17 :

asterisk[1367]: WARNING[18591]: app_fax.c:223 in phase_e_handler: Error 
transmitting fax. result=19: Received other than DIS while waiting for DIS.
asterisk[1367]: WARNING[18591]: app_fax.c:820 in transmit: Transmission failed

asterisk[1367]: WARNING[18906]: app_fax.c:223 in phase_e_handler: Error 
transmitting fax. result=20: Received no response to DCS or TCF.
asterisk[1367]: WARNING[18906]: app_fax.c:820 in transmit: Transmission failed

asterisk[1367]: WARNING[18986]: app_fax.c:223 in phase_e_handler: Error 
transmitting fax. result=49: The call dropped prematurely.
asterisk[1367]: WARNING[18986]: app_fax.c:817 in transmit: Transmission error


Asterisk 1.6.2.6 with Fax for Asterisk :

asterisk[7092]: WARNING[3167]: res_fax.c:1529 in sendfax_t38_init: Audio FAX 
not allowed on channel 'SIP/out.to.pgw-000b3f49' and T.38 negotiation failed; 
aborting.
asterisk[7092]: ERROR[3167]: res_fax.c:1650 in sendfax_exec: error initializing 
channel 'SIP/out.to.pgw-000b3f49' in T.38 mode

asterisk[7092]: VERBOSE[3226]: -- FAX handle 0: [ 028.000627 ], entering 
CLOSING state
asterisk[7092]: VERBOSE[3225]: -- Channel 'SIP/out.to.pgw-000b3f72' FAX 
session '11' is complete, result: 'FAILED' (FAX_FAILURE_TRAINING), error: 
'3RD_FRM_CHECK_ERROR', pages: 0, resolution: 'unknown', transfer rate: '2400', 
remoteSID: ''

asterisk[7092]: WARNING[3272]: res_fax.c:1529 in sendfax_t38_init: Audio FAX 
not allowed on channel 'SIP/out.to.pgw-000b3f8b' and T.38 negotiation failed; 
aborting.
asterisk[7092]: ERROR[3272]: res_fax.c:1650 in sendfax_exec: error initializing 
channel 'SIP/out.to.pgw-000b3f8b' in T.38 mode


My questions:

- Does anyone have experience with T.38 fax with a setup like this: ATA -> 
Asterisk -> PGW -> PSTN?

- Does anyone have experience in connecting Asterisk to a Cisco PGW 2200 + 
AS5400XM?

- Are there any tools to debug T.38 traffic?

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

2010-08-06 Thread Nasir Iqbal 
Hi,

May you also need to install *speex-tools* . if problem retain then let us
know about your Linux distribution and Asterisk version.

Regards

On Fri, Aug 6, 2010 at 4:59 PM, Deepika Nijhawan <
deepika.nijha...@oxygen8.com> wrote:

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



-- 
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ICT Innovations
http://www.ictinnovations.com/
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Re: [asterisk-users] Codec Conversion

2010-08-06 Thread Michael Graves 
On Fri, 06 Aug 2010 07:40:44 -0500, Michael Graves wrote:

>On Fri, 6 Aug 2010 03:43:33 -0500 (CDT), Jeff Brower wrote:
>
>
>
>>> 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.
>>
>>> so there is still a place for LPC10 [...]
>>
>>I haven't seen an LPC10 implementation with MOS higher than 2.5.  Due to its 
>>age and expiration of patents, LPC10
>>might be a basis for a 2400 bps open source codec.  But enormous improvement 
>>would be needed to come close to MELPe
>>performance.
>>
>>-Jeff
>
>I wonder where David Rowe's newer CODEC2 fits into this discussion?
>(http://codec2.org/)
>
>Clearly it's not implemented anywhere yet, but it may prove yet useful
>in very bandwidth constrained applications. Oh yes. It's completely
>open source and should not be subject to patent issues.
>
>Michael

The more appropriate link should have been
http://www.rowetel.com/blog/?page_id=452

Michael
--
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Twitter mjgraves




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Re: [asterisk-users] Codec Conversion

2010-08-06 Thread Michael Graves 
On Fri, 6 Aug 2010 03:43:33 -0500 (CDT), Jeff Brower wrote:



>> 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.
>
>> so there is still a place for LPC10 [...]
>
>I haven't seen an LPC10 implementation with MOS higher than 2.5.  Due to its 
>age and expiration of patents, LPC10
>might be a basis for a 2400 bps open source codec.  But enormous improvement 
>would be needed to come close to MELPe
>performance.
>
>-Jeff

I wonder where David Rowe's newer CODEC2 fits into this discussion?
(http://codec2.org/)

Clearly it's not implemented anywhere yet, but it may prove yet useful
in very bandwidth constrained applications. Oh yes. It's completely
open source and should not be subject to patent issues.

Michael
--
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mgravesmstvp.com
http://www.mgraves.org
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c713-201-1262
sip:mgra...@mstvp.onsip.com
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Twitter mjgraves




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

2010-08-06 Thread Chandrakant Solanki 
On Fri, Aug 6, 2010 at 5:29 PM, Deepika Nijhawan <
deepika.nijha...@oxygen8.com> wrote:

>  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*
>
> * *
>

Hi

Go For asterisk top directory.

And follow below steps to check whether speex function module is enable or
not.

./configure
make menuselect
  => Go for "Dialplan Function"
  => Then func_speex.

if func_speex shows [XXX] this symbol that means func_speex module is not
enable. And if you select func_speex then it shows dependency below of
module list.

-- 
Regards,

Chandrakant Solanki
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[asterisk-users] [Asterisk-Users] How do I install speex for asterisk?

2010-08-06 Thread Deepika Nijhawan 
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] Asterisk 1.6 without DAHDI

2010-08-06 Thread Kevin P. Fleming 
On 08/05/2010 06:25 PM, Roderick A. Anderson wrote:
> Kevin P. Fleming wrote:
>> On 08/05/2010 03:52 PM, Roderick A. Anderson wrote:
>>> I have a Linux-Vserver guest running CentOS 5.5 with Asterisk 1.6 
>>> installed from the asterisk.org and digium.com repositories.
>>>
>>> I have Asterisk starting (service asterisk start) but see errors about 
>>> dahdi in /var/log/asterisk/messages.
>>>
>>> ... ERROR[25658] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No 
>>> such file or directory
>>>
>>> Linux-Vservers don't allow, under normal circumstances, guests to fiddle 
>>> with /dev.  I could create all the entries in /dev/dahdi but as far as I 
>>> can determine I have no need of dahdi -- Asterisk 1.6.2, SIP only 
>>> connections, and currently no conference call needs.
>>>
>>> Is there a way to stop Asterisk (safe_asterisk) from even trying to load 
>>> dahdi?
>>
>> Yes; don't load codec_dahdi.so in Asterisk. Use 'noload' in your
>> modules.conf file. What packages have you installed from the
>> asterisk.org and digium.com yum repositories?
> 
> Thanks Kevin.
> 
> I made that entry and now there are no more dahdi errors in the log file.
> 
> Here is the command I used to install Asterisk.
> 
> yum install asterisk16 asterisk16-configs asterisk16-voicemail
> 
> And here are some RPM queries
> 
> # rpm -qa | grep asterisk
> asterisk-sounds-core-en-gsm-1.4.19-1_centos5
> asterisk16-core-1.6.2.10-1_centos5
> asterisk16-configs-1.6.2.10-1_centos5
> asterisk16-dahdi-1.6.2.10-1_centos5
> asterisk16-voicemail-1.6.2.10-1_centos5
> asterisk16-doc-1.6.2.10-1_centos5
> asterisk16-1.6.2.10-1_centos5

If you aren't going to use DAHDI, don't install the asterisk16-dahdi
package :-) Without that package, codec_dahdi wouldn't even be on your
system, and you would never have had this problem. With that package
(which provides codec_dahdi, among other things), the assumption is that
you want that module loaded.

> There are also these packages.
> 
> # rpm -qa | grep dahdi
> kmod-dahdi-linux-2.3.0.1-1_centos5.2.6.18_194.8.1.el5
> dahdi-firmware-tc400m-MR6.12-1_centos5
> dahdi-firmware-oct6114-128-1.05.01-1_centos5
> dahdi-firmware-2.0.2-1_centos5
> asterisk16-dahdi-1.6.2.10-1_centos5
> kmod-dahdi-linux-fwload-vpmadt032-2.3.0.1-1_centos5.2.6.18_194.8.1.el5
> dahdi-firmware-oct6114-064-1.05.01-1_centos5
> dahdi-firmware-hx8-2.06-1_centos5
> dahdi-linux-2.3.0.1-1_centos5

You don't need to have any of these DAHDI packages installed at all.

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

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Re: [asterisk-users] sip.conf register in realtime DB

2010-08-06 Thread Jonas Kellens 

Please can anyone help me with this ?!

I have tried renaming the sip.conf file, or tried including another file 
into sip.conf like sippy.conf and then add sippy.conf => 
mysql,AsteriskDB,ast_config to extconfig.conf but all this is not working.


The only thing that changes something is my example listed below, but 
then I always have that nasty WARNING which I find odd.


I need realtime sip registrations (so without having to do a sip reload).




Kind regards,

Jonas.


On 08/03/2010 10:13 AM, Jonas Kellens wrote:

Hello list,

scrambling different pieces of info together I've come with the 
following :


I want to have my "register =>" statements in a MySQL-database, so 
I've made the following table.


table ast_config :
id  1
cat_metric  0
var_metric  0
commented  0
filename  sip.conf
category  general
var_name  register
var_val username:passw...@sip.provider.net


In ext_config (text file) I have :

sipusers => mysql,AsteriskDB,sip_buddies
sippeers => mysql,AsteriskDB,sip_buddies
sip.conf => mysql,AsteriskDB,ast_config

In sip.conf (text file) I have also :

sip.conf :
rtcachefriends=yes ; Cache realtime friends by adding them 
to the internal list
; just like friends added from the 
config file only on a

; as-needed basis? (yes|no)


After a reload I noticed that the registration came through when I 
executed "sip show registrations". This realtime works.

But I then get a lot of the following messages :
/
[Aug  2 20:05:05] WARNING[29850]: chan_sip.c:18187 build_peer: Qualify 
is incompatible with dynamic uncached realtime.  Please either turn 
rtcachefriends on or turn qualify off on peer 'user2'
[Aug  2 20:05:05] WARNING[29850]: chan_sip.c:18187 build_peer: Qualify 
is incompatible with dynamic uncached realtime.  Please either turn 
rtcachefriends on or turn qualify off on peer 'user2'/



rtcachefriends is turned on (see above)
qualify is on on every peer (and I want it to stay that way)



Can anyone tell me what I need to configure to get a 100% working 
example ?!




Kind regards,

Jonas.
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Re: [asterisk-users] Codec Conversion

2010-08-06 Thread Jeff Brower 
Steve-

>   On 08/06/2010 05:40 AM, Jeff Brower wrote:
>> Miguel-
>>
>>> 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.

> so there is still a place for LPC10 [...]

I haven't seen an LPC10 implementation with MOS higher than 2.5.  Due to its 
age and expiration of patents, LPC10
might be a basis for a 2400 bps open source codec.  But enormous improvement 
would be needed to come close to MELPe
performance.

-Jeff


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Re: [asterisk-users] rolling over Master.csv CDR File

2010-08-06 Thread Ishfaq Malik 

Have a look into the command logrotate

http://www.cyberciti.biz/faq/how-do-i-rotate-log-files/

On 05/08/10 21:26, Ujjval Karihaloo wrote:


Is there a setting to roll over the Master.csv CDR File in 
/var/log/asterisk/cdr-csv,  from and ZIP the older file once its gets 
a certain size?




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