[asterisk-users] Asterisk 12.4.0 not able to install pjsip

[Sameer Rathod]

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 ?

-- 
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Sameer Rathod
8109413462
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Re: [asterisk-users] Asterisk 12.4.0 not able to install pjsip

[Scott Griepentrog]

​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
>
>
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Re: [asterisk-users] Certified Asterisk 11.6 Menuselect

[Rusty Newton]

On Mon, Jul 21, 2014 at 9:36 AM, Ryan Wagoner  wrote:
> Has there been a change in the way certified Asterisk is being packaged?
> Starting with certified Asterisk 11.6 has all the extended options are
> checked by default in menuslect? Certified Asterisk 11.2 does not have them
> checked and neither does certified Asterisk 1.8.15?

Thanks for taking note. I've filed an issue here
https://issues.asterisk.org/jira/browse/ASTERISK-24104


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Re: [asterisk-users] Native architecture never available in menuselect

[Rusty Newton]

On Mon, Jul 21, 2014 at 12:36 AM, CDR  wrote:
> I want to compile Asterisk always for the native architecture of the
> machine, and I find that it is never available. It says
> Depends on: native_arch(E)
>Can use: N/A
>  Conflicts with: N/A
>  Support Level: core
>
> This is Fedora 20
> gcc (GCC) 4.8.3 20140624 (Red Hat 4.8.3-1)
> many thanks
> Philip

I've rarely seen a machine that it isn't available on. The exception
for me was a virtualbox machine in one particular case.

You may get more help if you describe more about the CPU/architecture
that the machine uses.


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Re: [asterisk-users] Asterisk 12.4 IMAP VM Issue - Can't move messages between folders

[Dave Fullerton]

On 07/17/2014 09:46 AM, Dave Fullerton wrote:

Hello all,
   I'm running into an issue with Asterisk 12.4 and IMAP voicemail. I
have asterisk set up to connect to my Dovecot IMAP server and I can
leave and retrieve messages from my inbox and old messages. However, I
am unable to move messages between folders. I get a message from
asterisk stating "Sorry the users mailbox can't accept more messages".
Here is my setup:

In Voicemail.conf I have the following set:
imapgreetings=no
imapparentfolder=Voicemail
imapfolder=Voicemail.Inbox

On my imap server, I have the following folder structure:

INBOX
Sent
Junk
Drafts
Voicemail
|-Family
|-Inbox
|-Work

I did a packet capture on my imap server and found that when I go to
move a message from Old messages to Family the following happens:
Asterisk issues a "CREATE Voicemail.Family" which succeeds with "OK
Create completed" (The folder is successfully created if it does not
exist, I can see it in thunderbird).
Then Asterisk issues a "COPY 1 Family" which fails with "NO [TRYCREATE]
Mailbox Doesn't exist: Family" I don't think Asterisk is using the value
of imapparentfolder when copying the message. The COPY command should be
"COPY 1 Voicemail.Family".

Is there something I am missing in my configuration or is this a bug?

Thank you
-Dave



I think I have tracked the issue down to save_to_folder in 
app_voicemail.c. The third argument to mail_move/mail_copy needs to be 
different, but my C is not strong enough to know what I need to change 
it to. Any suggestions?


-Dave

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[asterisk-users] Return to transferor if no answer after REFER

[Jānis Rukšāns]

Hi all,

Consider the following scenario. Extension A is in a call with B. At
some point, B transfers A to C by sending REFER:

   A  Asterisk   B C
   |  |  | |
   |<-- INVITE/200 -->|<-- INVITE/200 -->| |
   :  :  : :
   |  |<--- REFER A r: C | |
   |  | 200 OK ->| |
   |  | INVITE C ->|
   |  | NOTIFY ->| |
   :  :  : :

If the INVITE to C is answered, B is connected to C, and A is hung up
(receives BYE), which is the expected behaviour.

However, if C does not answer (is busy, etc), Asterisk hangs up on A and
B anyway. What I'd like to have is to connect B back to A instead.

Is this possible, and if yes, what am I missing? Any pointers are
welcome.

Thanks,
Jānis


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[asterisk-users] Limit Asterisk

[Eduardo Leones]

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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Re: [asterisk-users] Limit Asterisk

[Scott Griepentrog]

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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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



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[asterisk-users] Any way to get rid of X-Asterisk?

[Nick Cameo]

Long story... Would be nice if we can remove this
on BYEs

X-Asterisk-HangupCause: Normal Clearing.
X-Asterisk-HangupCauseCode: 16.

Kind Regards,

Nick.
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Re: [asterisk-users] Any way to get rid of X-Asterisk?

[David Lam]

This is defined in chan_sip.c. Simply edit the source file and recompile.


On Wed, Jul 23, 2014 at 1:39 PM, Nick Cameo  wrote:

> Long story... Would be nice if we can remove this
> on BYEs
>
> X-Asterisk-HangupCause: Normal Clearing.
> X-Asterisk-HangupCauseCode: 16.
>
> Kind Regards,
>
> Nick.
>
>
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[asterisk-users] SIP configuration in realtime static and realtime dynamic

[Robin Kipp]

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


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Re: [asterisk-users] Any way to get rid of X-Asterisk?

[Nick Cameo]

Yeah I can do that Anything in sip.conf that we can set?

N.


On Wed, Jul 23, 2014 at 4:39 PM, David Lam  wrote:

> This is defined in chan_sip.c. Simply edit the source file and recompile.
>
>
> On Wed, Jul 23, 2014 at 1:39 PM, Nick Cameo  wrote:
>
>> Long story... Would be nice if we can remove this
>> on BYEs
>>
>> X-Asterisk-HangupCause: Normal Clearing.
>> X-Asterisk-HangupCauseCode: 16.
>>
>> Kind Regards,
>>
>> Nick.
>>
>>
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>
>
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Re: [asterisk-users] Limit Asterisk

[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 
mailto: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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Check us out at: http://digium.com · http://asterisk.org





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Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

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Re: [asterisk-users] Limit Asterisk

[Eduardo Leones]

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
>
>
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Re: [asterisk-users] Limit Asterisk

[Eduardo Leones]

Thanks for the feedback.

In this case SSD disks you think it solves?




2014-07-23 17:29 GMT-03:00 Scott Griepentrog :

> 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
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Re: [asterisk-users] Any way to get rid of X-Asterisk?

[David Lam]

As far as I know, the code that is set to sent these parameters are static
and not affected by the sip.conf settings.. If someone finds otherwise, let
me know.


On Wed, Jul 23, 2014 at 1:53 PM, Nick Cameo  wrote:

> Yeah I can do that Anything in sip.conf that we can set?
>
> N.
>
>
> On Wed, Jul 23, 2014 at 4:39 PM, David Lam  wrote:
>
>> This is defined in chan_sip.c. Simply edit the source file and recompile.
>>
>>
>> On Wed, Jul 23, 2014 at 1:39 PM, Nick Cameo  wrote:
>>
>>> Long story... Would be nice if we can remove this
>>> on BYEs
>>>
>>> X-Asterisk-HangupCause: Normal Clearing.
>>> X-Asterisk-HangupCauseCode: 16.
>>>
>>> Kind Regards,
>>>
>>> Nick.
>>>
>>>
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>>
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Re: [asterisk-users] Limit Asterisk

[Ron Wheeler]

Do the calculations for both and see what the answer is.
The nice thing about having a model is that you can test configurations 
without actually having to build one until you are confident that it 
should work.


Ron


On 23/07/2014 5:04 PM, Eduardo Leones wrote:

Thanks for the feedback.

In this case SSD disks you think it solves?




2014-07-23 17:29 GMT-03:00 Scott Griepentrog >:


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
mailto: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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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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Re: [asterisk-users] Limit Asterisk

[Steve Edwards]

Please don't top-post.

On Wed, 23 Jul 2014, Eduardo Leones wrote:


In this case SSD disks you think it solves?


Don't buy hardware until you've identified (either empirical or 
calculated) the bottleneck.


But...

SSDs do rock. I recently observed (via vmstat 5) a Samsung 840 topping out 
at 460,000 blocks per second. I can remember when 10,000 was big :)


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Re: [asterisk-users] Limit Asterisk

[Chris Bagnall]

On 23/7/14 10:29 pm, Steve Edwards wrote:

Don't buy hardware until you've identified (either empirical or
calculated) the bottleneck.


If you've plenty of spare RAM (and at 16GB I'd suggest you probably do), 
I'd throw in the possibility of recording to RAM disk, then moving the 
calls to hard disk during your quiet (or closed) hours.



SSDs do rock. I recently observed (via vmstat 5) a Samsung 840 topping
out at 460,000 blocks per second. I can remember when 10,000 was big :)


This. The 840 is a great bit of kit - we've replaced nearly all our 
spinning disks with a mix of Samsung 830 and 840 drives.


Kind regards,

Chris
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Re: [asterisk-users] Any way to get rid of X-Asterisk?

[Eric Wieling]

From sip.conf.sample in 11.10.0

;use_q850_reason = no ; Default "no"
  ; Set to yes add Reason header and use Reason header if 
it is available.



From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Lam
Sent: Wednesday, July 23, 2014 5:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Any way to get rid of X-Asterisk?

As far as I know, the code that is set to sent these parameters are static and 
not affected by the sip.conf settings.. If someone finds otherwise, let me know.

On Wed, Jul 23, 2014 at 1:53 PM, Nick Cameo 
mailto:sym...@gmail.com>> wrote:
Yeah I can do that Anything in sip.conf that we can set?

N.

On Wed, Jul 23, 2014 at 4:39 PM, David Lam 
mailto:software...@gmail.com>> wrote:
This is defined in chan_sip.c. Simply edit the source file and recompile.

On Wed, Jul 23, 2014 at 1:39 PM, Nick Cameo 
mailto:sym...@gmail.com>> wrote:
Long story... Would be nice if we can remove this
on BYEs

X-Asterisk-HangupCause: Normal Clearing.
X-Asterisk-HangupCauseCode: 16.

Kind Regards,

Nick.


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Re: [asterisk-users] Any way to get rid of X-Asterisk?

[Nick Cameo]

> From sip.conf.sample in 11.10.0
>
>
>
> ;use_q850_reason = no ; Default "no"
>
>   ; Set to yes add Reason header and use Reason header
> if it is available.
>
>
>
>
>
Using 1.8.7. Shiza

Thanks as always guys.

N.
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Re: [asterisk-users] Any way to get rid of X-Asterisk?

[David Lam]

This adds the Q850 reason header but doesn't get rid of the other
Asterisk-Hangup headers, at least in v11.


On Wed, Jul 23, 2014 at 2:53 PM, Eric Wieling  wrote:

> From sip.conf.sample in 11.10.0
>
>
>
> ;use_q850_reason = no ; Default "no"
>
>   ; Set to yes add Reason header and use Reason header
> if it is available.
>
>
>
>
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Lam
> *Sent:* Wednesday, July 23, 2014 5:07 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Any way to get rid of X-Asterisk?
>
>
>
> As far as I know, the code that is set to sent these parameters are static
> and not affected by the sip.conf settings.. If someone finds otherwise, let
> me know.
>
>
>
> On Wed, Jul 23, 2014 at 1:53 PM, Nick Cameo  wrote:
>
> Yeah I can do that Anything in sip.conf that we can set?
>
>
>
> N.
>
>
>
> On Wed, Jul 23, 2014 at 4:39 PM, David Lam  wrote:
>
> This is defined in chan_sip.c. Simply edit the source file and recompile.
>
>
>
> On Wed, Jul 23, 2014 at 1:39 PM, Nick Cameo  wrote:
>
> Long story... Would be nice if we can remove this
>
> on BYEs
>
>
>
> X-Asterisk-HangupCause: Normal Clearing.
>
> X-Asterisk-HangupCauseCode: 16.
>
>
>
> Kind Regards,
>
>
>
> Nick.
>
>
>
>
>
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Re: [asterisk-users] Limit Asterisk

[Steve Edwards]

On Wed, 23 Jul 2014, Chris Bagnall wrote:


The 840 is a great bit of kit...


The 850 is supposed to be shipping next week. It's got 3d VNAND so the 
chip geometry can be bigger -- higher speeds and greater reliability.


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