Re: [asterisk-users] SIP configuration in realtime static and realtime dynamic

[Ishfaq Malik]

On 23 July 2014 21:53, Robin Kipp  wrote:

> Hi all,
> I’m currently in the process of familiarizing myself with Asterisk, and am
> trying to move certain configuration objects (such as SIP peers) into a
> MySQL database, accessed by Asterisk using the ODBC connector.
> Now, I’ve imported the sippeers MySQL table from the contrib directory of
> the Asterisk source, and I could add SIP users in here. However, I
> currently don’t understand whether this realtime dynamic configuration
> table is meant to replace or just supplement sip.conf. This is because the
> sippeers table does not offer certain fields for entries in the [general]
> section of my sip.conf file, such as the ‚udpbindaddr‘ variable.
> So, am I supposed to put all that in the database by adding appropriate
> table columns, or can I leave this in the sip.conf file and chan_sip.so
> will read both the file and MySQL table once loaded? Also, is there anyway
> that I could use templates, so that I don’t have to redefine everything for
> each SIP peer?
> Thanks a lot for help!
> Robin
>
>
>
Hi

It supplements it.

In fact, you can define some peers in the sip.conf and some in the MySQL
table. However, if you do add any in the sip.conf directly, you'll have to
do a sip reload which will clear your realtime cache.

Regards

Ish

-- 

Ishfaq Malik
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Company: Packnet Limited
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[asterisk-users] audio gain in SIP channel

[lore]

hello all,
i'm trying to do what in object with an asterisk box 11.11 on centos6.5,
using functions
AGC and VOLUME, but seems that does not work at all.
There is a way to check this values during setup/call?
Maybe is it not possible realize what i'd like to do?

Could anyone can help me on this?

thanks a lot in advance

regards

Lorenzo
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Re: [asterisk-users] audio gain in SIP channel

[Rafael dos Santos Saraiva]

Hi

To using VOLUME function the syntax is:
Set(VOLUME(rx)=+n)
Set(VOLUME(rx)=-n)
Set(VOLUME(tx)=+n)
Set(VOLUME(tx)=-n)

I think is not possible retrieve the value of the channel.




Att,
*Rafael dos Santos Saraiva*



2014-07-24 7:52 GMT-03:00 lore :

> hello all,
> i'm trying to do what in object with an asterisk box 11.11 on centos6.5,
> using functions
> AGC and VOLUME, but seems that does not work at all.
> There is a way to check this values during setup/call?
> Maybe is it not possible realize what i'd like to do?
>
> Could anyone can help me on this?
>
> thanks a lot in advance
>
> regards
>
> Lorenzo
>
> --
> _
> -- 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
>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Asterisk 12.4.0 not able to install pjsip

[Sameer Rathod]

I am using centos 6.4 and 6.5

steps to install pjprojects is

a) git clone https://github.com/asterisk/pjproject pjproject

b) cd pjproject

c) ./configure --prefix=/usr --enable-shared --disable-sound
--disable-resample --disable-video --disable-opencore-amr

d) make dep

e) make install

f) ldconfig

g) export PKG_CONFIG_PATH=/usr/lib/pkgconfig

h) cd ../..asterisk

i) ./configure

j) make menuselect

In this it is showing XXX for pjlib

Please let me know if any more information is needed





On Wed, Jul 23, 2014 at 6:23 PM, Scott Griepentrog 
wrote:

> ​1) What platform are you on (i.e. Ubuntu/Centos/etc)
>
> 2) What steps did you take to install the PJSIP libraries?​
>
>
> On Wed, Jul 23, 2014 at 7:30 AM, Sameer Rathod 
> wrote:
>
>> Hi,
>>
>> I had tried all the steps which I used to inatall  Asterisk 12.3.2
>>
>> Pjsip in Asterisk 12.3.2 is working but in new release Asterisk 12.4.0
>> it is not working I am getting XXX in make menuselect resource_module. I
>> tried all trouble shooting steps along with ldconfig etc.
>>
>> I think its a bug can any one help me on this ?
>>
>> --
>> Regards
>> Sameer Rathod
>> 8109413462
>>
>>
>> --
>> _
>> -- 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
>>
>
>
>
> --
> [image: Digium logo]
> Scott Griepentrog
> Digium, Inc · Software Developer
> 445 Jan Davis Drive NW · Huntsville, AL 35806 · US
> direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029
> Check us out at: http://digium.com · http://asterisk.org
>
> --
> _
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>



-- 
Regards
Sameer Rathod
8109413462
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Re: [asterisk-users] Limit Asterisk

[Scott Griepentrog]

Whether SSD drives allow you to add any additional calls depends entirely
on whether or not they can be written to faster than the SAS drives you
have.  My experience shows SSD's can be twice as fast as run-of-the-mill
SATA, but the performance difference compared to SAS is likely not as
great, and could even be worse.  You'll need to test two drives to find
out.  I recommend mounting both to test them and copying a very large ISO
file using dd which will give you the transfer rate when finished.  Then
you should have your answer.


On Wed, Jul 23, 2014 at 4:03 PM, Eduardo Leones <
edua...@ypytecnologia.com.br> wrote:

> Thanks for the feedback.
>
> In this case SSD disks you think it solves?
>
>
> Eduardo
>
>
> 2014-07-23 18:01 GMT-03:00 Ron Wheeler :
>
>  I would also do some math on the bandwidth requirement.
>>
>> If you divide your disk bandwidth by your recording bit rate what is the
>> theoretical maximum number of calls that you can record at once? Assumes
>> that you have infinite CPU and memory and that you can actually drive the
>> disks at their maximum.
>> If this comes out to 300, you are already there. If it comes out to 3000,
>> you have something wrong in your setup or your assumptions and a target to
>> work towards.
>>
>> What quality are you using in the recording? 44k per second(CD quality
>> sound)  uses a lot more bandwidth than 3K (telephone quality)
>> What encoding are you using?
>> How low a bit rate can you use and still have usable recordings? If they
>> are for legal or audit use, you can go pretty low. If you are recording
>> soundtracks for reuse in training or publication, you may require higher
>> bit rates.
>>
>> If you disable recording, how many simultaneous calls can you support?
>> Just to be sure that recording is the issue.
>>
>> Ron
>>
>>
>> On 23/07/2014 4:29 PM, Scott Griepentrog wrote:
>>
>>  Your bottleneck is most likely your drive bandwidth.  Even with SAS
>> drives, you'll need to move to a raid 5+ solution with 6+ drives to
>> continue to increase the concurrent calls, or use a storage appliance.
>>
>>  To confirm this, install the tool nmon and use the v and d options to
>> bring up the resource usage indicators and drive busy/throughput statistics.
>>
>>
>>
>> On Wed, Jul 23, 2014 at 2:48 PM, Eduardo Leones <
>> edua...@ypytecnologia.com.br> wrote:
>>
>>>  people
>>>
>>>  I have a running Asterisk 1.8.28 in great Dell server with two xeon
>>> processors and 16gb of ram and HD SAS 15k (Raid 1). This server is
>>> recording all calls (placed to record the audio in a ram disk), the entire
>>> CDR goes straight to MySQL by cdr_mysql.so. Each call runs some validation
>>> and AGI's have an auto dialer system that generates calls over the manager.
>>> Calls originate and terminate via SIP (no transcode).
>>>
>>>  With this structure, even being a great server, we can not spend 150
>>> simultaneous calls. When it reaches 140, the load average goes up a lot and
>>> the calls start to get very bad audio, tear, etc.. Using the top we see
>>> that all the processing is for asterisk. In this scenario, I think there is
>>> some limitation in Asterisk, or even the manager due to the auto dialer.
>>>
>>>  Can anyone give me any tips where I can look where is the bottleneck?
>>> I need to get at least 250 calls that server quality.
>>>
>>>  tks
>>>
>>>
>>> --
>>> _
>>> -- 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
>>>
>>
>>
>>
>>  --
>>  [image: Digium logo]
>> Scott Griepentrog
>> Digium, Inc · Software Developer
>> 445 Jan Davis Drive NW · Huntsville, AL 35806 · US
>> direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029
>> Check us out at: http://digium.com · http://asterisk.org
>>
>>
>>
>>
>> --
>> Ron Wheeler
>> President
>> Artifact Software Inc
>> email: rwhee...@artifact-software.com
>> skype: ronaldmwheeler
>> phone: 866-970-2435, ext 102
>>
>>
>> --
>> _
>> -- 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
>>
>
>
> --
> _
> -- 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
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>http://lists.di

Re: [asterisk-users] Limit Asterisk

[Eduardo Leones]

Thank you all for the answers. I will do tests to find the problem.

One other question I have, in the scenario that I sent, how bad would be to
transcode G711 to G729 in 70% of calls? There is a study that shows a
statistically loss of performance (concurrent calls) with active transcode?

tks




2014-07-24 8:54 GMT-03:00 Scott Griepentrog :

> Whether SSD drives allow you to add any additional calls depends entirely
> on whether or not they can be written to faster than the SAS drives you
> have.  My experience shows SSD's can be twice as fast as run-of-the-mill
> SATA, but the performance difference compared to SAS is likely not as
> great, and could even be worse.  You'll need to test two drives to find
> out.  I recommend mounting both to test them and copying a very large ISO
> file using dd which will give you the transfer rate when finished.  Then
> you should have your answer.
>
>
> On Wed, Jul 23, 2014 at 4:03 PM, Eduardo Leones <
> edua...@ypytecnologia.com.br> wrote:
>
>> Thanks for the feedback.
>>
>> In this case SSD disks you think it solves?
>>
>>
>> Eduardo
>>
>>
>> 2014-07-23 18:01 GMT-03:00 Ron Wheeler :
>>
>>  I would also do some math on the bandwidth requirement.
>>>
>>> If you divide your disk bandwidth by your recording bit rate what is the
>>> theoretical maximum number of calls that you can record at once? Assumes
>>> that you have infinite CPU and memory and that you can actually drive the
>>> disks at their maximum.
>>> If this comes out to 300, you are already there. If it comes out to
>>> 3000, you have something wrong in your setup or your assumptions and a
>>> target to work towards.
>>>
>>> What quality are you using in the recording? 44k per second(CD quality
>>> sound)  uses a lot more bandwidth than 3K (telephone quality)
>>> What encoding are you using?
>>> How low a bit rate can you use and still have usable recordings? If they
>>> are for legal or audit use, you can go pretty low. If you are recording
>>> soundtracks for reuse in training or publication, you may require higher
>>> bit rates.
>>>
>>> If you disable recording, how many simultaneous calls can you support?
>>> Just to be sure that recording is the issue.
>>>
>>> Ron
>>>
>>>
>>> On 23/07/2014 4:29 PM, Scott Griepentrog wrote:
>>>
>>>  Your bottleneck is most likely your drive bandwidth.  Even with SAS
>>> drives, you'll need to move to a raid 5+ solution with 6+ drives to
>>> continue to increase the concurrent calls, or use a storage appliance.
>>>
>>>  To confirm this, install the tool nmon and use the v and d options to
>>> bring up the resource usage indicators and drive busy/throughput statistics.
>>>
>>>
>>>
>>> On Wed, Jul 23, 2014 at 2:48 PM, Eduardo Leones <
>>> edua...@ypytecnologia.com.br> wrote:
>>>
  people

  I have a running Asterisk 1.8.28 in great Dell server with two xeon
 processors and 16gb of ram and HD SAS 15k (Raid 1). This server is
 recording all calls (placed to record the audio in a ram disk), the entire
 CDR goes straight to MySQL by cdr_mysql.so. Each call runs some validation
 and AGI's have an auto dialer system that generates calls over the manager.
 Calls originate and terminate via SIP (no transcode).

  With this structure, even being a great server, we can not spend 150
 simultaneous calls. When it reaches 140, the load average goes up a lot and
 the calls start to get very bad audio, tear, etc.. Using the top we see
 that all the processing is for asterisk. In this scenario, I think there is
 some limitation in Asterisk, or even the manager due to the auto dialer.

  Can anyone give me any tips where I can look where is the bottleneck?
 I need to get at least 250 calls that server quality.

  tks


 --
 _
 -- 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

>>>
>>>
>>>
>>>  --
>>>  [image: Digium logo]
>>> Scott Griepentrog
>>> Digium, Inc · Software Developer
>>> 445 Jan Davis Drive NW · Huntsville, AL 35806 · US
>>> direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029
>>> Check us out at: http://digium.com · http://asterisk.org
>>>
>>>
>>>
>>>
>>> --
>>> Ron Wheeler
>>> President
>>> Artifact Software Inc
>>> email: rwhee...@artifact-software.com
>>> skype: ronaldmwheeler
>>> phone: 866-970-2435, ext 102
>>>
>>>
>>> --
>>> _
>>> -- 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 

Re: [asterisk-users] audio gain in SIP channel

[lore]

thanks a lot Rafael.
could you tell me also something about AGC(rx)=?
I mean, i've tryed

Set(AGC(rx)=)
Set(AGC(rx)=)
Set(DENOISE(tx)=on)
Set(DENOISE(rx)=on)

using =8000, 16000 and 32000 but all calls looked like to have se same
audio gain.

thanks for your rapid reply.


2014-07-24 13:12 GMT+02:00 Rafael dos Santos Saraiva :

> Hi
>
> To using VOLUME function the syntax is:
> Set(VOLUME(rx)=+n)
> Set(VOLUME(rx)=-n)
> Set(VOLUME(tx)=+n)
> Set(VOLUME(tx)=-n)
>
> I think is not possible retrieve the value of the channel.
>
>
>
>
> Att,
> *Rafael dos Santos Saraiva*
>  
>
>
> 2014-07-24 7:52 GMT-03:00 lore :
>
>> hello all,
>> i'm trying to do what in object with an asterisk box 11.11 on centos6.5,
>> using functions
>> AGC and VOLUME, but seems that does not work at all.
>> There is a way to check this values during setup/call?
>> Maybe is it not possible realize what i'd like to do?
>>
>> Could anyone can help me on this?
>>
>> thanks a lot in advance
>>
>> regards
>>
>> Lorenzo
>>
>> --
>> _
>> -- 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
>>
>
>
> --
> _
> -- 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
>



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conta un cazzo, 1941 ... sono anche un autore"
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Re: [asterisk-users] audio gain in SIP channel

[Rafael dos Santos Saraiva]

I dont using these functions (AGC/ DENOISE). My suggestion... try invert
the priorities:
Set(DENOISE(tx)=on)
Set(DENOISE(rx)=on)
Set(AGC(rx)=)
Set(AGC(rx)=)

And try higher values.. is more easy the perception if the values are
larger than default.


Att,
*Rafael dos Santos Saraiva*



2014-07-24 9:23 GMT-03:00 lore :

> thanks a lot Rafael.
> could you tell me also something about AGC(rx)=?
> I mean, i've tryed
>
> Set(AGC(rx)=)
> Set(AGC(rx)=)
> Set(DENOISE(tx)=on)
> Set(DENOISE(rx)=on)
>
> using =8000, 16000 and 32000 but all calls looked like to have se same
> audio gain.
>
> thanks for your rapid reply.
>
>
> 2014-07-24 13:12 GMT+02:00 Rafael dos Santos Saraiva  >:
>
>> Hi
>>
>> To using VOLUME function the syntax is:
>> Set(VOLUME(rx)=+n)
>> Set(VOLUME(rx)=-n)
>> Set(VOLUME(tx)=+n)
>> Set(VOLUME(tx)=-n)
>>
>> I think is not possible retrieve the value of the channel.
>>
>>
>>
>>
>> Att,
>> *Rafael dos Santos Saraiva*
>>  
>>
>>
>> 2014-07-24 7:52 GMT-03:00 lore :
>>
>>>  hello all,
>>> i'm trying to do what in object with an asterisk box 11.11 on centos6.5,
>>> using functions
>>> AGC and VOLUME, but seems that does not work at all.
>>> There is a way to check this values during setup/call?
>>> Maybe is it not possible realize what i'd like to do?
>>>
>>> Could anyone can help me on this?
>>>
>>> thanks a lot in advance
>>>
>>> regards
>>>
>>> Lorenzo
>>>
>>> --
>>> _
>>> -- 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
>>>
>>
>>
>> --
>> _
>> -- 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
>>
>
>
>
> --
> "Chi vive sperando muore cagando ... Lo Russo isoletta dell'Egeo che non
> conta un cazzo, 1941 ... sono anche un autore"
>
> --
> _
> -- 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
>
-- 
_
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Re: [asterisk-users] Limit Asterisk

[Eduardo Leones]

Another question, what audio format I use in MixMonitor to maintain a
connection with reasonable quality and reduce the use of I / O disk? Today
I use wav.


tks


2014-07-24 9:05 GMT-03:00 Eduardo Leones :

> Thank you all for the answers. I will do tests to find the problem.
>
> One other question I have, in the scenario that I sent, how bad would be
> to transcode G711 to G729 in 70% of calls? There is a study that shows a
> statistically loss of performance (concurrent calls) with active transcode?
>
> tks
>
>
>
>
> 2014-07-24 8:54 GMT-03:00 Scott Griepentrog :
>
> Whether SSD drives allow you to add any additional calls depends entirely
>> on whether or not they can be written to faster than the SAS drives you
>> have.  My experience shows SSD's can be twice as fast as run-of-the-mill
>> SATA, but the performance difference compared to SAS is likely not as
>> great, and could even be worse.  You'll need to test two drives to find
>> out.  I recommend mounting both to test them and copying a very large ISO
>> file using dd which will give you the transfer rate when finished.  Then
>> you should have your answer.
>>
>>
>> On Wed, Jul 23, 2014 at 4:03 PM, Eduardo Leones <
>> edua...@ypytecnologia.com.br> wrote:
>>
>>> Thanks for the feedback.
>>>
>>> In this case SSD disks you think it solves?
>>>
>>>
>>> Eduardo
>>>
>>>
>>> 2014-07-23 18:01 GMT-03:00 Ron Wheeler :
>>>
>>>  I would also do some math on the bandwidth requirement.

 If you divide your disk bandwidth by your recording bit rate what is
 the theoretical maximum number of calls that you can record at once?
 Assumes that you have infinite CPU and memory and that you can actually
 drive the disks at their maximum.
 If this comes out to 300, you are already there. If it comes out to
 3000, you have something wrong in your setup or your assumptions and a
 target to work towards.

 What quality are you using in the recording? 44k per second(CD quality
 sound)  uses a lot more bandwidth than 3K (telephone quality)
 What encoding are you using?
 How low a bit rate can you use and still have usable recordings? If
 they are for legal or audit use, you can go pretty low. If you are
 recording soundtracks for reuse in training or publication, you may require
 higher bit rates.

 If you disable recording, how many simultaneous calls can you support?
 Just to be sure that recording is the issue.

 Ron


 On 23/07/2014 4:29 PM, Scott Griepentrog wrote:

  Your bottleneck is most likely your drive bandwidth.  Even with SAS
 drives, you'll need to move to a raid 5+ solution with 6+ drives to
 continue to increase the concurrent calls, or use a storage appliance.

  To confirm this, install the tool nmon and use the v and d options to
 bring up the resource usage indicators and drive busy/throughput 
 statistics.



 On Wed, Jul 23, 2014 at 2:48 PM, Eduardo Leones <
 edua...@ypytecnologia.com.br> wrote:

>  people
>
>  I have a running Asterisk 1.8.28 in great Dell server with two xeon
> processors and 16gb of ram and HD SAS 15k (Raid 1). This server is
> recording all calls (placed to record the audio in a ram disk), the entire
> CDR goes straight to MySQL by cdr_mysql.so. Each call runs some validation
> and AGI's have an auto dialer system that generates calls over the 
> manager.
> Calls originate and terminate via SIP (no transcode).
>
>  With this structure, even being a great server, we can not spend 150
> simultaneous calls. When it reaches 140, the load average goes up a lot 
> and
> the calls start to get very bad audio, tear, etc.. Using the top we see
> that all the processing is for asterisk. In this scenario, I think there 
> is
> some limitation in Asterisk, or even the manager due to the auto dialer.
>
>  Can anyone give me any tips where I can look where is the
> bottleneck? I need to get at least 250 calls that server quality.
>
>  tks
>
>
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 Ron Wheeler
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 Artifact Software Inc
 email: rwhee...@artifact-software.com
 skype: 

[asterisk-users] Bria softphone registration problems on DNS SRV cluster

[Noah Engelberth]

I have a pair of Asterisk 11.5.1 servers operating as a load balanced cluster, 
with DNS SRV records set up to weight them 60/40 relative to each other (both 
at priority 0).  The back-end is MySQL Realtime, and everything works pretty 
well with the Cisco SPA phones & ATAs that represent the majority of my 
endpoints.

I recently tried to add an iPhone with the Bria softphone application, to 
provide a wireless handset option for my customers (since the Cisco SPA302D 
wireless handsets are decidedly not durable).  The Bria application will 
register, but within 1 registration timeout cycle, it will stop receiving 
incoming calls, and show as unregistered.  Sometimes it can send an outbound 
call and re-register, but most of the time it doesn't.

Bria support has provided me some information here: 
https://support.counterpath.com/responses/bria-for-iphone-does-not-maintain-registration#comment-21492

Based on post #4, it looks like they're basically tagging both servers in the 
process of trying to re-register, and eventually send a REGISTER packet with 
authorization based on both servers' nonces, but Asterisk still rejects it with 
a 401.

Is there something I should be looking at on the Asterisk side that would fix 
this problem?  Does anyone know if there have been changes in authorization 
handling since 11.5.1 that would fix the issue?


Thank you,

Noah Engelberth
MetaLINK Technologies

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Re: [asterisk-users] Limit Asterisk

[Steve Edwards]

Please don't top-post.

Please trim irrelevant posts.

On Thu, 24 Jul 2014, Eduardo Leones wrote:

Another question, what audio format I use in MixMonitor to maintain a 
connection with reasonable quality and reduce the use of I / O disk?


I think the question is premature.

You have a resource limitation. Until you know what that limitation is, 
you can't really make intelligent changes.


Is it I/O activity or I/O bandwidth?

Are you swapping? (Swapping is 'death' to performance.)

Are you running out of CPU?

If you're planning on transcoding to something as computationally 
intensive as 729, do you have gobs of excess CPU capacity? If not, you'll 
just be trading 1 resource limitation for another.


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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] Asterisk 12.4.0 not able to install pjsip

[Matthew Jordan]

On Thu, Jul 24, 2014 at 6:15 AM, Sameer Rathod  wrote:

> I am using centos 6.4 and 6.5
>
> steps to install pjprojects is
>
> a) git clone https://github.com/asterisk/pjproject pjproject
>
> b) cd pjproject
>
> c) ./configure --prefix=/usr --enable-shared --disable-sound 
> --disable-resample --disable-video --disable-opencore-amr
>
> d) make dep
>
> e) make install
>
> f) ldconfig
>
> g) export PKG_CONFIG_PATH=/usr/lib/pkgconfig
>
> h) cd ../..asterisk
>
> i) ./configure
>
> j) make menuselect
>
> In this it is showing XXX for pjlib
>
> Please let me know if any more information is needed
>
>
What is the output of "pkg-config --print-provides libpjproject"? For that
matter, does "pkg-config --list-all" show libpjproject as a package?

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[asterisk-users] How to diagnose early media on a PRI

[Justin Killen]

I have a dialplan (freepbx) that plays a busy signal in-band when an extension 
is busy (before an Answer).  Stripped down, it looks like this:
exten => 1005,n,PlayTones(busy)
exten => 1005,n,Busy(20)
Note that there is no Answer() prior to this.  Our trunk is a PRI.

When I call into this extension from outside, I get about 25 seconds of 
ringing, followed by a hangup.  Looking at the asterisk logs, 20 seconds of 
that delay is AFTER the PlayTones() function is invoked.  I talked with our 
Telco about this, and they want to refer to in-band tones prior to answer as a 
media cut-through.  The tech said that it is enabled on their end, and he did 
some test calls and got some ISDN trap logs.  He is saying that the PBX is 
playing the ring-back tone instead of the busy tone, but I don't think that's 
the case (If I add an Answer() to the dialplan, I do in fact hear the busy 
tone).
Is there anybody out there who has experience with reading/analyzing IDSN trap 
logs (Q931) that can help me narrow down where the issue is and how to fix it?

Thanks,
-Justin

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[asterisk-users] TLS/TCP behind NAT; Signaling issues with offnet phones

[D.H. Williams]

Issue is what subject says.  Here is the background.

Version:  11.11.0
Topology:  Asterisk Box at our Data Center behind Cisco Firewall.
 Everything works fine from remote offices over a VPN.  Issue is sales team
would like to connect up to our Asterisk box remotely (offnet).  Common
enough solution, I'm guessing.

So, I've opened all the correct holes on the firewall and hammered out
inspection with Cisco.  UDP transport works like a champ, but obviously we
are sending SIP across as clear text when they are on wireless outside the
office.  I know TLS/SRTP isn't completely secure, but we can file it as
"good enough" for now.

I've tested this out by using my softphone (Bria 4) on non company wireless
network and captured packets via Wireshark and have pinpointed the issue,
but not sure how to circumvent it.

I started with TLS, but set transport to TCP as the issue is similar on
each and TCP shows what I am going to bet is also the issue with TLS.  Here
is a breakdown:

1.  Softphone registers fine.
2.  Can place a call fine.  Media works fine (used
media_address= command to resolve this, btw).
3.  When I go to disconnect/transfer/place the call on hold from softphone,
pretty much anything that requires signaling, my packet captures reveals
that I'm trying to do this using the private IP of my Asterisk box (Nat,
again, is on the firewall at data center), and I get TCP retransmissions.
 so the fact it isn't working makes sense, because my local box doesn't
know how to get to a private IP address.

I've tried using externaddr in sip.conf to no avail.  Is there some command
I'm missing?  Obviously if I put an interface with a public IP on the
outside I'd bet that would resolve this problem, but sort of like having
that guy behind a hardware firewall :)

I'm to the point of telling them to fire up a VPN on be done with it, but
all the same I am curious if there is a way with tcp/tls transport to fix
this because, well, I'm curious.

Thanks in advanced for looking at this!

DH
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Re: [asterisk-users] TLS/TCP behind NAT; Signaling issues with offnet phones

[D.H. Williams]

Just found the solution in case someone down the line stumbles across this.
 externaddr only works with localnet defined in sip.conf.

Again, was simply misled due to UDP working but TCP not working.

This also resolved the issue with TLS which makes sense.


On Thu, Jul 24, 2014 at 5:12 PM, D.H. Williams  wrote:

> Issue is what subject says.  Here is the background.
>
> Version:  11.11.0
> Topology:  Asterisk Box at our Data Center behind Cisco Firewall.
>  Everything works fine from remote offices over a VPN.  Issue is sales team
> would like to connect up to our Asterisk box remotely (offnet).  Common
> enough solution, I'm guessing.
>
> So, I've opened all the correct holes on the firewall and hammered out
> inspection with Cisco.  UDP transport works like a champ, but obviously we
> are sending SIP across as clear text when they are on wireless outside the
> office.  I know TLS/SRTP isn't completely secure, but we can file it as
> "good enough" for now.
>
> I've tested this out by using my softphone (Bria 4) on non company
> wireless network and captured packets via Wireshark and have pinpointed the
> issue, but not sure how to circumvent it.
>
> I started with TLS, but set transport to TCP as the issue is similar on
> each and TCP shows what I am going to bet is also the issue with TLS.  Here
> is a breakdown:
>
> 1.  Softphone registers fine.
> 2.  Can place a call fine.  Media works fine (used
> media_address= command to resolve this, btw).
> 3.  When I go to disconnect/transfer/place the call on hold from
> softphone, pretty much anything that requires signaling, my packet captures
> reveals that I'm trying to do this using the private IP of my Asterisk box
> (Nat, again, is on the firewall at data center), and I get TCP
> retransmissions.  so the fact it isn't working makes sense, because my
> local box doesn't know how to get to a private IP address.
>
> I've tried using externaddr in sip.conf to no avail.  Is there some
> command I'm missing?  Obviously if I put an interface with a public IP on
> the outside I'd bet that would resolve this problem, but sort of like
> having that guy behind a hardware firewall :)
>
> I'm to the point of telling them to fire up a VPN on be done with it, but
> all the same I am curious if there is a way with tcp/tls transport to fix
> this because, well, I'm curious.
>
> Thanks in advanced for looking at this!
>
> DH
>
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