[asterisk-users] SIP calls dropping at 15 minutes

2015-11-20 Thread Steve Edwards 
I have a problem where SIP calls from some providers are dropping at 15 
minutes.


The topology is: Client sends calls to a host running OpenSIPS, OpenSIPS 
sends calls to an Asterisk server.


Below,

'Client' is the IP address of the client's host (running 
FPBX-2.8.1(1.8.20.0)


'OpenSIPS' is the IP address of my host running OpenSIPS 1.7.2-tls

'Asterisk' is the IP address of my host running Asterisk 11.17.1.

The relevant snippet of opensips.cfg is:

# 317
if  ($rU =~ '317*')
{
ds_select_dst(
  '02'  # set-id (in dispatcher.list)
, '4'   # algorithm (4 = round-robin)
  );
forward();
return;
}

where set-id 02 is 'sip:Asterisk:5061'

The 'Flow' diagram from Wireshark from a tcpdump on the OpenSIPS host 
follows, hopefully the email clients will not mung it too much.


|Time | Client| Asterisk  |
| |   | OpenSIPS  | 
|7.158764 | INVITE SDP (g711U g7  |   |SIP From: "760xxx"   (5060)   |   |
|7.159003 |   | INVITE SDP (g711U g7  |SIP 
Request
| |   |(5060)   -->  (5061)   |
|7.161857 |   | 100 Trying|   |SIP 
Status
| |   |(5060)   <--  (5061)   |
|7.161958 | 100 Trying|   |   |SIP 
Status
| |(5060)   <--  (5060)   |   |
|7.538268 |   | 200 OK SDP (g711U te  |SIP 
Status
| |   |(5060)   <--  (5061)   |
|7.538411 | 200 OK SDP (g711U te  |   |SIP 
Status
| |(5060)   <--  (5060)   |   |
|7.585703 | ACK   |   |   |SIP 
Request
| |(5060)   -->  (5060)   |   |
|7.585941 |   | ACK   |   |SIP 
Request
| |   |(5060)   -->  (5061)   |
|7.586548 | INVITE SDP (g711U te  |   |SIP From: 
"760xxx"   (5060)   |   |
|7.586726 |   | INVITE SDP (g711U te  |SIP 
Request
| |   |(5060)   -->  (5061)   |
|7.587792 |   | 100 Trying|   |SIP 
Status
| |   |(5060)   <--  (5061)   |
|7.587922 | 100 Trying|   |   |SIP 
Status
| |(5060)   <--  (5060)   |   |
|7.588003 |   | 200 OK SDP (g711U te  |SIP 
Status
| |   |(5060)   <--  (5061)   |
|7.588081 | 200 OK SDP (g711U te  |   |SIP 
Status
| |(5060)   <--  (5060)   |   |
|7.635401 | ACK   |   |   |SIP 
Request
| |(5060)   -->  (5060)   |   |
|7.635674 |   | ACK   |   |SIP 
Request
| |   |(5060)   -->  (5061)   |
|907.588019|   | INVITE SDP (g711U te  |SIP 
Request
| |   |(5060)   <--  (5061)   |
|907.590138|   | 100 Giving a try  |SIP 
Status
| |   |(5060)   -->  (5061)   |
|907.590261|   | INVITE SDP (g711U te  |SIP 
Request
| |   |(5060)   -->  (5061)   |
|907.591294|   | 481 Call/Transaction  |SIP 
Status
| |   |(5060)   <--  (5061)   |
|907.591420|   | ACK   |   |SIP 
Request
| |   |(5060)   -->  (5061)   |
|907.591467|   | 481 Call/Transaction  |SIP 
Status
| |   |(5060)   -->  (5061)   |
|907.592140|   | ACK   |   |SIP 
Request
| |   |(5060)   <--  (5061)   |
|907.867923|   | BYE   |   |SIP 
Request
| |   |(5060)   <--  (5061)   |
|907.868231|   | BYE

[asterisk-users] Which router/firewall would you use for a virtual-PBX Asterisk installation?

2015-11-20 Thread Ernie Dunbar 

Hi everyone.

We've got a fairly large base of customers who use our Asterisk server 
for phone service in a virtual PBX kind of way, where the server is 
security hardened and exposed to the internet for them to connect to 
remotely with SIP and IAX. It's certainly not the sort of affair where 
we're running it as a PBX just within the building. As a result, we see 
network traffic coming through eth0 between 512 Kbps and about 3.0 Mbps, 
depending on the time of day.


We haven't so far been using a hardware firewall/router on our server 
network, but it's becoming increasingly clear that we need to. We have 
enough experience to know that Asterisk is pretty sensitive when it 
comes to network hardware in our situation - we've had to replace one 
otherwise perfectly good 100 Mbps network switch because it simply 
wasn't able to keep up with the amount of streaming audio we put through 
it, and it badly affected voice quality. We have other traffic flowing 
through our server network too, including a significant amount of e-mail 
and web traffic, although that's not quite as sensitive to the quality 
of our network hardware.


If you've got these large requirements for Asterisk, I'd love to hear 
what you use for a router, and whether that router has met your needs. 
It would also be nice to hear about what kinds of routers to avoid that 
you may have tried in the past and found lacking.


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Re: [asterisk-users] Which router/firewall would you use for a virtual-PBX Asterisk installation?

2015-11-20 Thread jg 



Hi everyone.

We've got a fairly large base of customers who use our Asterisk server for phone service in a 
virtual PBX kind of way, where the server is security hardened and exposed to the internet for 
them to connect to remotely with SIP and IAX. It's certainly not the sort of affair where 
we're running it as a PBX just within the building. As a result, we see network traffic coming 
through eth0 between 512 Kbps and about 3.0 Mbps, depending on the time of day.


We haven't so far been using a hardware firewall/router on our server network, but it's 
becoming increasingly clear that we need to. We have enough experience to know that Asterisk 
is pretty sensitive when it comes to network hardware in our situation - we've had to replace 
one otherwise perfectly good 100 Mbps network switch because it simply wasn't able to keep up 
with the amount of streaming audio we put through it, and it badly affected voice quality. We 
have other traffic flowing through our server network too, including a significant amount of 
e-mail and web traffic, although that's not quite as sensitive to the quality of our network 
hardware.


If you've got these large requirements for Asterisk, I'd love to hear what you use for a 
router, and whether that router has met your needs. It would also be nice to hear about what 
kinds of routers to avoid that you may have tried in the past and found lacking.


I am working at a scale of about 10 Mbps and I am using customized pfSense setups. Essentially, 
I am also using Asterisk as a session border controller as part of the router/firewall. I am 
using a multi step procedure to keep unwanted traffic away from the application software, which 
includes geo IP filtering and blocking based on Snort alarms. So far I haven't seen the 
necessity to block anything based on Asterisk logs, but I'll plan to add that feature to 
pfBlockeNG as a custom IPv4 (and IPv6) list.


It's too early for recommendations or public demo software, but I am planning to add my SBC to 
pfSense 2.3 superseding the current Asterisk package. If necessary, pfSense allows for traffic 
shaping and a couple of other neat feature, that are usually not part of small firewalls.


jg

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Re: [asterisk-users] Which router/firewall would you use for a virtual-PBX Asterisk installation?

2015-11-20 Thread Telium Technical Support 
Well router and firewall are very different...it depends on what you are
trying to accomplish.

If you are trying to secure an Asterisk-based call center, get a real
security product.  Look here for details:
http://www.voip-info.org/wiki/view/Asterisk+security

This covers firewall, Asterisk lock-down, and Asterisk specific security.
The average break-in/fraud cost is $25,000 per day.  (watch the Astricon
videos for more details).  So going cheap on security isn't a smart move for
a commercial installation.

If you just want a router/switch, figure out the simultaneous call capacity
x codec demands in bps, and there is your backplane switching speed
requirements.  Even with 100 simultaneous calls at g711, a lower end Cisco
(3xx) router/switch will have no problem.

-M-

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ernie Dunbar
Sent: Friday, November 20, 2015 3:25 PM
To: Asterisk Users
Subject: [asterisk-users] Which router/firewall would you use for a
virtual-PBX Asterisk installation?

Hi everyone.

We've got a fairly large base of customers who use our Asterisk server 
for phone service in a virtual PBX kind of way, where the server is 
security hardened and exposed to the internet for them to connect to 
remotely with SIP and IAX. It's certainly not the sort of affair where 
we're running it as a PBX just within the building. As a result, we see 
network traffic coming through eth0 between 512 Kbps and about 3.0 Mbps, 
depending on the time of day.

We haven't so far been using a hardware firewall/router on our server 
network, but it's becoming increasingly clear that we need to. We have 
enough experience to know that Asterisk is pretty sensitive when it 
comes to network hardware in our situation - we've had to replace one 
otherwise perfectly good 100 Mbps network switch because it simply 
wasn't able to keep up with the amount of streaming audio we put through 
it, and it badly affected voice quality. We have other traffic flowing 
through our server network too, including a significant amount of e-mail 
and web traffic, although that's not quite as sensitive to the quality 
of our network hardware.

If you've got these large requirements for Asterisk, I'd love to hear 
what you use for a router, and whether that router has met your needs. 
It would also be nice to hear about what kinds of routers to avoid that 
you may have tried in the past and found lacking.

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Re: [asterisk-users] How to custom the message on call busy or no answer in asterisk

2015-11-20 Thread Thyda ENG 
I found this but I don't know where the busy tone place, I wanna replace
this file, do you have any idea ?
​
 Screen Shot 2015-11-21 at 1.49.47 PM.png

​

On Fri, Nov 20, 2015 at 1:51 PM, Julien Sansonnens 
wrote:

> Hi,
> Check the DIALSTATUS variable.
> http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
>
> Regards, Julien
>
> --
>
>
>
> 2015-11-20 2:15 GMT+01:00 Thyda ENG :
> > Hi,
> >
> > I was wonder is there any way to custom the message on the call busy or
> no
> > answer I actually get the error code from asterisk server on busy or no
> > answer. Can I custom the text message or custom the message to sound ?
> > Anyone have any idea could u please share me ?
> >
> >
> > Thank,
> >
> > Thyda
> >
> > --
> > _
> > -- 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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>http://www.asterisk.org/hello
>
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>
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[asterisk-users] Error while compiling asterisk asterisk-1.8.32.3

2015-11-20 Thread Jayesh Labade 
Hi,

I encountered following error while compiling asterisk-1.8.32.3. I am
using Debian 8(Jessie) 64 bit version.

make[1]: *** [chan_dahdi.so] Error 1
Makefile:351: recipe for target 'channels' failed
make: *** [channels] Error 2

Detailed error attached in log file.

Best Regards,
Jayesh Labade
   [CC] sig_ss7.c -> sig_ss7.o
   [LD] chan_dahdi.o sig_analog.o sig_pri.o sig_ss7.o -> chan_dahdi.so
sig_analog.o: In function `ast_atomic_fetchadd_int':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/lock.h:600: multiple 
definition of `ast_atomic_fetchadd_int'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/lock.h:600: 
first defined here
sig_analog.o: In function `ast_atomic_dec_and_test':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/lock.h:646: multiple 
definition of `ast_atomic_dec_and_test'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/lock.h:646: 
first defined here
sig_analog.o: In function `ast_tvdiff_sec':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:45: multiple 
definition of `ast_tvdiff_sec'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:45: first 
defined here
sig_analog.o: In function `ast_tvdiff_us':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:64: multiple 
definition of `ast_tvdiff_us'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:64: first 
defined here
sig_analog.o: In function `ast_tvdiff_ms':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:78: multiple 
definition of `ast_tvdiff_ms'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:78: first 
defined here
sig_analog.o: In function `ast_tvzero':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:95: multiple 
definition of `ast_tvzero'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:95: first 
defined here
sig_analog.o: In function `ast_tvcmp':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:106: multiple 
definition of `ast_tvcmp'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:106: 
first defined here
sig_analog.o: In function `ast_tveq':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:125: multiple 
definition of `ast_tveq'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:125: 
first defined here
sig_analog.o: In function `ast_tvnow':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:135: multiple 
definition of `ast_tvnow'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:135: 
first defined here
sig_analog.o: In function `ast_tv':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:171: multiple 
definition of `ast_tv'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:171: 
first defined here
sig_analog.o: In function `ast_samp2tv':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:186: multiple 
definition of `ast_samp2tv'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/time.h:186: 
first defined here
sig_analog.o: In function `_ast_malloc':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/utils.h:477: multiple 
definition of `_ast_malloc'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/utils.h:477: 
first defined here
sig_analog.o: In function `_ast_calloc':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/utils.h:500: multiple 
definition of `_ast_calloc'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/utils.h:500: 
first defined here
sig_analog.o: In function `_ast_realloc':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/utils.h:536: multiple 
definition of `_ast_realloc'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/utils.h:536: 
first defined here
sig_analog.o: In function `_ast_strdup':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/utils.h:563: multiple 
definition of `_ast_strdup'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/utils.h:563: 
first defined here
sig_analog.o: In function `_ast_strndup':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/utils.h:592: multiple 
definition of `_ast_strndup'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/utils.h:592: 
first defined here
sig_analog.o: In function `_ast_vasprintf':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/utils.h:631: multiple 
definition of `_ast_vasprintf'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/utils.h:631: 
first defined here
sig_analog.o: In function `ast_threadstorage_get':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/threadstorage.h:173: multiple 
definition of `ast_threadstorage_get'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/threadstorage.h:173:
 first defined here
sig_analog.o: In function `ast_skip_blanks':
/usr/local/src/asterisk-1.8.32.3/include/asterisk/strings.h:90: multiple 
definition of `ast_skip_blanks'
chan_dahdi.o:/usr/local/src/asterisk-1.8.32.3/include/asterisk/strings.h:90: 
first defined here
sig_analog.o: