Re: [asterisk-users] Removing mailbox and password prompt for voicemail

[Robert Berlin]

A password prompt is avoidable with a ",s" in the VoicemailMain appdata 
  
   Robert Berlin
 Manager of Operations & Systems Development
 Florida High Speed Internet
 (321) 205-1100 x109

  

  
  

  


 From: "D'Arcy J.M. Cain" <da...@vex.net>
Sent: Thursday, August 04, 2016 9:36 AM
To: "Nabeel" <nabeelshik...@gmail.com>
Cc: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Removing mailbox and password prompt for 
voicemail   
On Thu, 4 Aug 2016 14:03:39 +0100
Nabeel <nabeelshik...@gmail.com> wrote:
> I should add, a password is *always* asked if a password has been set.
> There isn't a way to bypass that.

Then something is wrong.

http://darcy.vex.net/star98.mp3

--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net

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Re: [asterisk-users] Removing mailbox and password prompt for voicemail

[Robert Berlin]

Excuse my formatting here, this is my first contribution to the list. I 
believe I understand what you are trying to accomplish. In my dial plans I 
have extension 87 setup to go to a peers voicemail box directly without 
need for a password. The App/Appdata I call is as shown below. Hope this 
helps!
  
 VoicemailMain(${SIPPEER(${CHANNEL(peername)}:mailbox)},s)
  
   Robert Berlin
 Manager of Operations & Systems Development
 Florida High Speed Internet
 (321) 205-1100

  

  
  

  


 From: "D'Arcy J.M. Cain" <da...@vex.net>
Sent: Thursday, August 04, 2016 9:13 AM
To: "Nabeel" <nabeelshik...@gmail.com>
Cc: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Removing mailbox and password prompt for 
voicemail   
On Thu, 4 Aug 2016 14:01:27 +0100
Nabeel <nabeelshik...@gmail.com> wrote:
> You seem to misunderstand even after I have explained. I don't need a
> password when calling my mailbox from my own registered phone (not

And the extension I suggested in answer to your original question will
do that for you. Are you saying that pressing "*98" from your phone
requests a password?

> calling from any other phone). I don't need to call my mailbox from
> other phones on the planet, so I don't need a password. Consider the

You need a password to protect your mailbox from people entering '*'
during your message. The extension I gave you bypasses the password
when you call from your own phone.

> voicemail you get from your mobile network on your mobile phone. You
> don't access it from any phone in the world; you only access it from
> the mobile phone which has your SIM, and you probably don't enter a
> password for it.

Yes but if someone accesses it from another phone it damn well better
ask for a password before serving up my messages.

> The password is only asked if a password has been set. A password is
> also asked if any number is entered after the 'mailbox' prompt.

>From outside phones. What happens when you dial "*98" from your own
phone.

--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net

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Re: [asterisk-users] Audio cutting in and out - asterisk 13.1 cert6 / confbridge

[Robert McGilvray]

"timing test" does similar, it just doesn't do the automatic calculation. 
Confbridge normally operates at a mixing interval of 20ms, which is 50 ticks 
per second. That would be what you would want to test.
If you don't get 50 per second then that means ConfBridge will not provide a 
steady source of media to each participant and it will be up to each remote 
jitterbuffer to handle the delayed traffic. Enough of it and stuff goes wonky. 
You could also see this on a packet capture. That would determine if it's 
timing related or not.

--

Thanks Joshua. We're talking about pretty long gaps in the audio, probably 
around 10-15 seconds which is quite a bit of missed ticks at 20ms sampling. I 
was poking around the timing code trying to get a better understanding of 
things and found that Asterisk uses timerfd_create with CLOCK_MONOTONIC as the 
clock. The man page states CLOCK_MONOTONIC is affected by incremental 
adjustments to the time made by things like NTP.

I may be completely off track here but would something like vmtools that tries 
to correct the clock skew (caused by VMware) be causing some issues here? 
Meaning that if asterisk calls timerfd_create but then the time is adjusted 
could that throw off the timing of the descriptor?

Regards

Bob
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[asterisk-users] Audio cutting in and out - asterisk 13.1 cert6 / confbridge

[Robert McGilvray]

Hello,

We use Asterisk extensively for conferencing - for the last 8 years or so this 
has been the 1.4/1.6/1.8 releases running chan_sip and meetme for up to around 
350 concurrent users. Right around that number DAHDI hit's a hard coded memory 
limit and kicks allocation errors in the log.

[Jun 22 10:04:13] WARNING[9095] app_meetme.c: Unable to open DAHDI pseudo 
channel: Cannot allocate memory

In order to support our growing user count we recently upgraded to 13.1-cert6 
with pjsip and replaced meetme with confbridge. During all of our UAT and load 
testing everything seemed to be fine, there were no perceived audio quality 
issues or any logs that would indicate an issue. Unfortunately now that we're 
in production I'm getting consistent complaints that the audio from 
participants is cutting in and out. It only seems to occur while under load 
with > 350 users but that is anecdotal at best. This is not a simple networking 
issue, we've pretty much ruled that out with various performance testing. That 
was not the case initially and we had incrementing UDP packet receive errors 
which we've eliminated with a bit of tuning.

There are numerous architectural differences between the two installations and 
so far I have not been successful in determining the root cause. I'm reaching 
out to the community and the developers for insight and feedback hoping there 
is prior experience with this issue and how to resolve it. As you can see below 
the most significant difference is probably the use of VMware on the new 
install. I've tuned the ESXi host and guest per VMware's recommendations for 
latency and jitter (full cpu/mem reservations) with no improvement. With all of 
the reading I've done I suspect my issue may come down to a timing source and 
VMware not providing a reliable clock. It seems they allow a backlog of 
interrupts and if it hasn't caught up in 60s they are simply dropped.

Before I rip apart the environment and rebuild on physical I'd like to try and 
confirm that hypothesis. In the past this was a simple matter of running 
dahdi_test which would report the accuracy. I'm not sure how to interpret the 
results of "timing test" in the Asterisk CLI. If I increase the number of ticks 
per second the results are erratic while under load. I'm using the timerfd 
module in Asterisk with a 1000HZ tick kernel and high res timers enabled. I've 
tried both hpet and tsc as system clock sources, both exhibit the same breaks 
in audio. It sounds like someone presses the mute button in the middle of a 
sentence.

Any insight is appreciated!

Here are the specs on the new install:

Physical HWCisco UCS Blade (UCSB-B200-M3)
vMware   ESXI 5.5
VM Guest   4 vCPU w/ 32G of RAM tuned for latency/jitter 
(sensitivity=high) and full cpu/memory reservations.
VM OS  Redhat EL7 kernel 3.10.0-327.13.1.el7.x86_64 with 
tickless disabled e.g nohz=off and 1000HZ.
Asterisk13.1-cert6 using the timerfd module.

Regards
Robert McGilvray
SS GlobeOp
Associate Director, IT Network Security

GlobeOp Financial Services | 1565 Front Street | Yorktown Hts NY 10598

t: +1 (914)-293-3584  |  f: +1 (914)-293-3510
rmcgi...@globeop.com  |  www.ssctech.com<http://www.ssctech.com/>   |   
www.sscglobeop.com<http://www.sscglobeop.com/>
Follow us: Twitter<http://twitter.com/GlobeOp>  |  
Facebook<http://www.facebook.com/pages/SSC-Technologies-Inc/191750415876>  |  
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Re: [asterisk-users] Asterisk 13.1-cert6 Now Available

[Robert McGilvray]

> Are you selectively loading modules? If so you need the new res_pjproject.so 
> loaded.

Yes. That did it, thanks.

Bob

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Re: [asterisk-users] Asterisk 13.1-cert6 Now Available

[Robert McGilvray]

Hello,

This build fails to load res_pjsip.so, it kicks back symbol lookup errors for 
ast_pjproject_get_buildopt. Certified cert4 works fine, pjproject is 2.4.5.

[Apr 21 13:15:34]  Loading res_pjsip.so.
[Apr 21 13:15:34] -- Local IPv4 address determined to be: 10.33.204.12
[Apr 21 13:15:34] -- Local IPv6 address determined to be: 
[fe80::250:56ff:fe95:501a]
[Apr 21 13:15:34]   == Parsing 
'/home/asterisk/asterisk/certified-13.1-cert6/etc/pjsip.conf': Found
[Apr 21 13:15:34]   == Manager registered action PJSIPShowEndpoints
[Apr 21 13:15:34]   == Manager registered action PJSIPShowEndpoint
/home/asterisk/asterisk/certified-13.1-cert6/sbin/asterisk: symbol lookup 
error: /home/asterisk/asterisk/certified-13.1-cert6/lib/modules/res_pjsip.so: 
undefined symbol: ast_pjproject_get_buildopt

ykt1cfbprd1:/home/asterisk/asterisk/certified-13.1-cert6/etc# ldd 
/home/asterisk/asterisk/certified-13.1-cert6/lib/modules/res_pjsip.so
linux-vdso.so.1 =>  (0x7ffc9aa2e000)
libpjsua2.so.2 => /usr/local/lib/libpjsua2.so.2 (0x7fe01653f000)
libstdc++.so.6 => /lib64/libstdc++.so.6 (0x7fe016235000)
libpjsua.so.2 => /usr/local/lib/libpjsua.so.2 (0x7fe015f84000)
libpjsip-ua.so.2 => /usr/local/lib/libpjsip-ua.so.2 (0x7fe015d6e000)
libpjsip-simple.so.2 => /usr/local/lib/libpjsip-simple.so.2 
(0x7fe015b5b000)
libpjsip.so.2 => /usr/local/lib/libpjsip.so.2 (0x7fe015914000)
libpjmedia-codec.so.2 => /usr/local/lib/libpjmedia-codec.so.2 
(0x7fe015709000)
libpjmedia-videodev.so.2 => /usr/local/lib/libpjmedia-videodev.so.2 
(0x7fe015506000)
libpjmedia-audiodev.so.2 => /usr/local/lib/libpjmedia-audiodev.so.2 
(0x7fe015301000)
libpjmedia.so.2 => /usr/local/lib/libpjmedia.so.2 (0x7fe0150be000)
libpjnath.so.2 => /usr/local/lib/libpjnath.so.2 (0x7fe014e9e000)
libpjlib-util.so.2 => /usr/local/lib/libpjlib-util.so.2 
(0x7fe014c7b000)
libsrtp.so.2 => /usr/local/lib/libsrtp.so.2 (0x7fe014a66000)
libgsmcodec.so.2 => /usr/local/lib/libgsmcodec.so.2 (0x7fe01485a000)
libspeex.so.2 => /usr/local/lib/libspeex.so.2 (0x7fe014631000)
libilbccodec.so.2 => /usr/local/lib/libilbccodec.so.2 
(0x7fe014422000)
libg7221codec.so.2 => /usr/local/lib/libg7221codec.so.2 
(0x7fe01421)
libpj.so.2 => /usr/local/lib/libpj.so.2 (0x7fe013ff1000)
libm.so.6 => /lib64/libm.so.6 (0x7fe013cef000)
librt.so.1 => /lib64/librt.so.1 (0x7fe013ae6000)
libpthread.so.0 => /lib64/libpthread.so.0 (0x7fe0138ca000)
libc.so.6 => /lib64/libc.so.6 (0x7fe013509000)
libgcc_s.so.1 => /lib64/libgcc_s.so.1 (0x7fe0132f2000)
    /lib64/ld-linux-x86-64.so.2 (0x7fe0169ef000)

Regards

Robert McGilvray
o: 914 293 3584

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk 
Development Team
Sent: Wednesday, April 20, 2016 12:04 PM
To: Asterisk Users Mailing List
Subject: [asterisk-users] Asterisk 13.1-cert6 Now Available

The Asterisk Development Team has announced the release of Certified Asterisk 
13.1-cert6.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/certified-asterisk

The release of Certified Asterisk 13.1-cert6 resolves several issues reported 
by the community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:
---
 * ASTERISK-25929 - res_pjsip_registrar: AOR_CONTACT_ADDED events
  not raised (Reported by Joshua Colp)
 * ASTERISK-25928 - res_pjsip: URI validation done outside of PJSIP
  thread (Reported by Joshua Colp)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.1-cert6

Thank you for your continued support of Asterisk!
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[asterisk-users] Hold/Resume no audio - Asterisk 13.7.2 / PJSIP 2.4.5 / CUCM 8.6.2

[Robert McGilvray]

Hello,

We currently have a production Asterisk box running 1.8.20.1 which uses MeetMe 
and chan_sip for conferences. I have been testing the new versions of Asterisk 
with PJSIP and ConfBridge but have run into an issue which is preventing us 
from moving forward. Everything works fine until a call is placed on hold, 
after resuming the call the user cannot hear audio from the bridge. The same 
thing works perfectly with 1.8.20.1.

The scenario is: Cisco 8845 SIP (G722) -> CUCM 8.6.2 (also 
tested 10.5, same issue) - > SIP trunk to Asterisk 13.7.2 PJSIP with delayed 
offer ->  ConfBridge.

Tcpdump reveals Asterisk is sending the RTP to the endpoint so I suspect we're 
dealing with a bug / interop issue with the culprit possibly being a=inactive 
lines in the SDP.

I've included a link (on drive) to two separate SIP traces, one using ngrep and 
the other is the output of pjsip logging and the relevant sections of my 
pjsip.conf

https://drive.google.com/folderview?id=0B6XOeEMvID0vX2FxTXNkZWlodWM=sharing

Can anyone offer some insight?

Regards,

BobM
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Re: [asterisk-users] Hold/Resume no audio - Asterisk 13.7.2 / PJSIP 2.4.5 / CUCM 8.6.2

[Robert McGilvray]

I did some more troubleshooting eliminating G722 just in case there was an 
issue with transcoding / MTP which has resulted in a slightly different SDP but 
resume still doesn't work.

Full sip dialog: https://drive.google.com/open?id=0B6XOeEMvID0vZVdOV2NzM3NJZGM

Initial call setup appears to be correct. CUCM sends an early offer to Asterisk 
with SDP, Asterisk responds with a 200 OK with SDP and a=sendrecv.

When I place the call on HOLD the CUCM sends a delayed offer INVITE to Asterisk:

Asterisk responds with a 200 OK containing the SDP with a=sendrecv
CUCM ACKS with an SDP containing a=sendonly

When I resume the call the CUCM sends a delayed offer INVITE to Asterisk:

Asterisk responds with a 200 OK containing the SDP with 
a=recvonly
CUCM ACKS with an SDP containing a=sendonly

I may be missing or interpreting something incorrectly but that does not right 
for a RESUME scenario.

Per RFC32645 the CUCM is responding in one of the two ways permitted:


"If a stream is offered as sendonly, the corresponding stream MUST be
   marked as recvonly or inactive in the answer.  If a media stream is
   listed as recvonly in the offer, the answer MUST be marked as
   sendonly or inactive in the answer.
"

Is this a bug or am I wrong in my interpretation of the dialog?

Thanks!

Robert McGilvray
o: 914 293 3584

From: Robert McGilvray
Sent: Thursday, March 17, 2016 12:55 PM
To: 'asterisk-users@lists.digium.com'
Subject: Hold/Resume no audio - Asterisk 13.7.2 / PJSIP 2.4.5 / CUCM 8.6.2

Hello,

We currently have a production Asterisk box running 1.8.20.1 which uses MeetMe 
and chan_sip for conferences. I have been testing the new versions of Asterisk 
with PJSIP and ConfBridge but have run into an issue which is preventing us 
from moving forward. Everything works fine until a call is placed on hold, 
after resuming the call the user cannot hear audio from the bridge. The same 
thing works perfectly with 1.8.20.1.

The scenario is: Cisco 8845 SIP (G722) -> CUCM 8.6.2 (also 
tested 10.5, same issue) - > SIP trunk to Asterisk 13.7.2 PJSIP with delayed 
offer ->  ConfBridge.

Tcpdump reveals Asterisk is sending the RTP to the endpoint so I suspect we're 
dealing with a bug / interop issue with the culprit possibly being a=inactive 
lines in the SDP.

I've included a link (on drive) to two separate SIP traces, one using ngrep and 
the other is the output of pjsip logging and the relevant sections of my 
pjsip.conf

https://drive.google.com/folderview?id=0B6XOeEMvID0vX2FxTXNkZWlodWM=sharing

Can anyone offer some insight?

Regards,

BobM
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Re: [asterisk-users] DPMA - Asterisk Realtime

[Robert Broyles]

Thanks for the update.

As an Authorized Digium Reseller - it's difficult for me to sell Digium 
phones if the customer can't use the cool features it comes with. They 
can purchase a standard IP Phone (which is what Digium phones are 
without DPMA) from other vendors for a lower price point.


So the ability to use DPMA with Asterisk RT is very important for our 
large deployments.


Anyone willing to contribute towards a bounty for this feature?

--
Robert Broyles

On 5/7/15 7:14 AM, Matthew Jordan wrote:




On Fri, May 1, 2015 at 10:43 AM, Robert Broyles 
rob...@webservicesaz.com mailto:rob...@webservicesaz.com wrote:


We love our Digium phones and DPMA - but we really need it to work
on our Realtime Platform. Otherwise we lose all the cool features
and they are just standard SIP phones.

Anyone working on a solution for this? Or anyone from Digium see
this on the roadmap?


Hey Robert -

We've had a number of requests to have the DPMA work more closely with 
Asterisk Realtime. Right now, that feature isn't planned for an 
upcoming scheduled release, but we do keep track of requests such as 
this. We've made a note of it, and we'll keep evaluating it versus 
other planned and requested features.


Thanks -

Matt

--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com  http://asterisk.org




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[asterisk-users] DPMA - Asterisk Realtime

[Robert Broyles]

We love our Digium phones and DPMA - but we really need it to work on 
our Realtime Platform. Otherwise we lose all the cool features and they 
are just standard SIP phones.


Anyone working on a solution for this? Or anyone from Digium see this on 
the roadmap?



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[asterisk-users] Call Recording doesn't work

[Walter Robert Ditzler]

Hi all,

 

on my atserisk box call recording and cdr doesn't work. In the log files I
have a strange entry - does this have something to do with that?

 

Version: Asterisk 13.1.0

Host: debian wheezy 7.7

 

Thanks a lot for a brief hint .

 

Walter.

 

***

[2015-Jan-26 11:34:04] [PHP-DEPRECATION_WARNING]
(/var/local/asterisk/www/admin/modules/digiumaddoninstaller/functions.inc.ph
p:29) - dl(): dl() is deprecated - use extension=digium_register.so in your
php.ini

[2015-Jan-26 11:34:04] [PHP-WARNING]
(/var/local/asterisk/www/admin/modules/digiumaddoninstaller/functions.inc.ph
p:29) - dl(): Unable to load dynamic library
'/usr/lib/php5/20100525/digium_register.so' -
/usr/lib/php5/20100525/digium_register.so: cannot open shared object file:
No such file or directory

[2015-Jan-26 11:34:06] [PHP-DEPRECATION_WARNING]
(/var/local/asterisk/www/admin/modules/digiumaddoninstaller/functions.inc.ph
p:29) - dl(): dl() is deprecated - use extension=digium_register.so in your
php.ini

[2015-Jan-26 11:34:06] [PHP-WARNING]
(/var/local/asterisk/www/admin/modules/digiumaddoninstaller/functions.inc.ph
p:29) - dl(): Unable to load dynamic library
'/usr/lib/php5/20100525/digium_register.so' -
/usr/lib/php5/20100525/digium_register.so: cannot open shared object file:
No such file or directory

[2015-Jan-26 11:34:07] [PHP-DEPRECATION_WARNING]
(/var/local/asterisk/www/admin/modules/digiumaddoninstaller/functions.inc.ph
p:29) - dl(): dl() is deprecated - use extension=digium_register.so in your
php.ini

[2015-Jan-26 11:34:07] [PHP-WARNING]
(/var/local/asterisk/www/admin/modules/digiumaddoninstaller/functions.inc.ph
p:29) - dl(): Unable to load dynamic library
'/usr/lib/php5/20100525/digium_register.so' -
/usr/lib/php5/20100525/digium_register.so: cannot open shared object file:
No such file or directory

[2015-Jan-26 11:34:20] [PHP-DEPRECATION_WARNING]
(/var/local/asterisk/www/admin/modules/digiumaddoninstaller/functions.inc.ph
p:29) - dl(): dl() is deprecated - use extension=digium_register.so in your
php.ini

[2015-Jan-26 11:34:20] [PHP-WARNING]
(/var/local/asterisk/www/admin/modules/digiumaddoninstaller/functions.inc.ph
p:29) - dl(): Unable to load dynamic library
'/usr/lib/php5/20100525/digium_register.so' -
/usr/lib/php5/20100525/digium_register.so: cannot open shared object file:
No such file or directory

***

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Re: [asterisk-users] OT - Merry Christmas and a Happy and Prosperous 2014

[Robert Krakora]

To you as well.
On Dec 25, 2013 8:56 AM, Nick Cameo sym...@gmail.com wrote:

 God Bless and Merry Christmas to All!

 Nick.

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Re: [asterisk-users] Pulse Audio Motorboating Audio with Asterisk

[Robert Krakora]

https://bbs.archlinux.org/viewtopic.php?pid=920549


On Sat, Jun 8, 2013 at 9:51 AM, Jerry Geis ge...@pagestation.com wrote:

 When I use pulse audio and Asterisk 11.4.0 on the Console/Dsp port
 I get a motorboating sound or warble  - or - just not clear audio.

 When I switch that to ALSA direct it sounds just fine.

 What might be happening with pulse audio that it does not
 sound clear???

 asound.conf below.

 Thanks,

 Jerry

 more /etc/asound.conf
 #
 # Place your global alsa-lib configuration here...
 #

 @hooks [
 {
 func load
 files [
 /etc/alsa/pulse-default.conf
 ]
 errors false
 }
 ]




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MessageNet Systems
101 East Carmel Dr. Suite 105
Carmel, IN 46032
(317)566-1677 Ext 212
(317)663-0808 Fax
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Re: [asterisk-users] Pulse Audio Motorboating Audio with Asterisk

[Robert Krakora]

Pulse Audio 4.0 just came out and has gotten good reviews as it improves
audio quality...I installed it on the devel and support mediaports and will
test tomorrow.

http://www.freedesktop.org/wiki/Software/PulseAudio/Notes/4.0/


On Mon, Jun 10, 2013 at 7:59 PM, Robert Krakora 
rob.krak...@messagenetsystems.com wrote:

 https://bbs.archlinux.org/viewtopic.php?pid=920549


 On Sat, Jun 8, 2013 at 9:51 AM, Jerry Geis ge...@pagestation.com wrote:

 When I use pulse audio and Asterisk 11.4.0 on the Console/Dsp port
 I get a motorboating sound or warble  - or - just not clear audio.

 When I switch that to ALSA direct it sounds just fine.

 What might be happening with pulse audio that it does not
 sound clear???

 asound.conf below.

 Thanks,

 Jerry

 more /etc/asound.conf
 #
 # Place your global alsa-lib configuration here...
 #

 @hooks [
 {
 func load
 files [
 /etc/alsa/pulse-default.conf
 ]
 errors false
 }
 ]




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 Rob Krakora
 MessageNet Systems
 101 East Carmel Dr. Suite 105
 Carmel, IN 46032
 (317)566-1677 Ext 212
 (317)663-0808 Fax




-- 
Rob Krakora
MessageNet Systems
101 East Carmel Dr. Suite 105
Carmel, IN 46032
(317)566-1677 Ext 212
(317)663-0808 Fax
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Re: [asterisk-users] Fw: Stress testing Asterisk

[Robert-GMAIL]

I believe there are options for rtp / audio..

Look at pcap play and rtp echo...

Transcoding would be another beast - if you are allowing it

Sent from my iPhone 5

On May 22, 2013, at 10:02 AM, Tommy Cooper tomcoope...@yahoo.com wrote:

 From the little experience I have I do not think that that is a good way of 
 testing the quality of voice. SIP only initiates and eventually terminates 
 the call, once that the call is connected, SIP and therefore Asterisk are no 
 longer involved. Once the call is connected it is assigned to a trapsport 
 layer protocol such as RTP. RTP is the actual protocol that delivers the 
 voice call between endpoints. I  believe that the setup of your network, QoS, 
 codecs etc... determine the voice quality of your system.
 
  
 - Forwarded Message -
 From: Mitul Limbani mi...@enterux.in
 To: Tommy Cooper tomcoope...@yahoo.com; Asterisk Users Mailing List - 
 Non-Commercial Discussion asterisk-users@lists.digium.com 
 Sent: Wednesday, May 22, 2013 3:23 PM
 Subject: Re: [asterisk-users] Stress testing Asterisk
 
 I have a question here.
 
 How can we test the quality of voice upon increasing the call load?
 
 Can we try passing a voice file using sipp and record the same in dial plan 
 record application ? Is this reliable enough to simulate near real world 
 scenario?
 
 Mitul
 
 On Wednesday, May 22, 2013, Tommy Cooper wrote:
 Thank you for your help I finally solved this issue. Is it possible that my 
 setup can achieve 212 concurrent calls, I am running Asterisk on just 1 core 
 using 3.5 GHz, and 1Gb of RAM?
 
 - Forwarded Message -
 From: Marie Fischer ma...@vtl.ee
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com 
 Sent: Wednesday, May 22, 2013 1:16 PM
 Subject: Re: [asterisk-users] Stress testing Asterisk
 
 
 On 21.05.2013, at 0:05, Tommy Cooper tomcoope...@yahoo.com wrote:
 
  Hi,
  I just installed Sipp 3.3 on CentOS 6.3 and all of the calls Sipp is 
  generating are failing. I am trying to run Sipp on the same machine as 
  Asterisk PBX using the ./sipp -sn uac 192.168.1.115 command.
 
 Do you have a peer and extension configured for SIPP in your Asterisk 
 configuration? You also needat least the -s extension_to_dial option on 
 your sipp command line.
 http://hasnainali.wordpress.com/2009/03/12/using-sipp-for-stress-testing-asterisk/has
  some simple instructions which should get you started.
 If the calls still fail, Asterisk console output would be helpful.
 
 
 
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 -- 
 Regards,
 Mitul Limbani,
 Chief Architech  Founder,
 Enterux Solutions Pvt. Ltd.
 110 Reena Complex, Opp. Nathani Steel, 
 Vidyavihar (W), Mumbai - 400 086. India
 http://www.enterux.com/
 http://www.entvoice.com/
 email: mi...@enterux.in
 DID: +91-22-71967121
 Cell: +91-9820332422
 
 
 
 
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Re: [asterisk-users] Asterisk disconnecting SIP Calls after 15 Minutes

[Robert Krakora]

I am having the same problem with Asterisk 11.2.0 and Linphone and it is
exactly 15 minutes and occurring with SIP running on our LAN.

On Thu, Mar 21, 2013 at 3:31 AM, Florian Wolters flor...@florian-wolters.de
 wrote:

 Hi @ll,

 I just moved my Asterisk Box and changed the Provider and Internet Access
 to a full IP Access by Deutsche Telekom.

 I set up my sip.conf as I found various examples throughout the Net. Calls
 and some other stuff is basically working.

 The problem I ran into is, that the outgoing and incoming calls are
 dropped after exactly 15 Minutes. Solution for this should be setting the
 session-timers to refuse but this doesnt change anything here.

 I am running Asterisk 1.6 on Ubuntu 12.04 LTS and also tried the latest
 Asterisk by Digium without success.

 Has anyone else has the Same problem or is a solution already known? Could
 someone point me in the right direction? I can provide (debug) logs if
 essential.

 Best regards

Flo


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Rob Krakora
MessageNet Systems
101 East Carmel Dr. Suite 105
Carmel, IN 46032
(317)566-1677 Ext 212
(317)663-0808 Fax
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[asterisk-users] app_rtsp.c ported to Asterisk 11.x

[Robert Krakora]

Hi,

If anyone is interested, I have ported app_rtsp.c to Asterisk 11.x.  I have
tested it with GStreamer RTSP server and a C920 webcam streaming H264 SVC
video from one machine to another machine running Linphone.  Contact me at
this e-mail address robkrak...@messagenetsystems.com for source code.

Best Regards,

-- 
Rob Krakora
MessageNet Systems
101 East Carmel Dr. Suite 105
Carmel, IN 46032
(317)566-1677 Ext 212
(317)663-0808 Fax
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Re: [asterisk-users] Google Voice with Asterisk 11/chan_motif

[Robert-GMAIL]

Might also want to check the google hasnt detected an unusual login and is 
asking for the ip to be accepted.

Log in to gmail with that account and check

Sent from my iPhone 5

On Feb 5, 2013, at 4:31 PM, Joshua Colp jc...@digium.com wrote:

 Josue Freitas wrote:
 Thank you!
 
 What about the XMPP traffic? Even when I place calls using GV there's no
 XMPP traffic on 5222.
 
 Do I really need to have the XMPP port (5222) open in the firewall?
 
 Asterisk acts as an XMPP client. It establishes an outgoing connection to 
 port 5222 of the Google Talk XMPP server. No incoming connections occur.
 
 -- 
 Joshua Colp
 Digium, Inc. | Senior Software Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at:  www.digium.com   www.asterisk.org
 
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Re: [asterisk-users] rtptimeout: how to detect it in dialplan?

[Robert Boardman]

On 18 Jan 2013 15:22, Klaus Darilion klaus.mailingli...@pernau.at wrote:

 Hi!

 I want to forward a call to another destination if the outgoing call leg
has an rtptimeout. But as far as I see there is no way to find out if the
hangup was due to a rtp timeout or any other reason. I thought that
HANGUPCAUSE or DIALSTATUS would be set, but they aren't.

 Are there any means to detect an rtp timeout in extensions.conf?

 Thanks
 Klaus

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Re: [asterisk-users] Detect Low Quality Calls - Realtime

[Robert-GMAIL]

Sometimes just the act of collecting performance data degrades the quality

Sent from my iPhone 5

On Jan 6, 2013, at 6:00 AM, XBrian bobo...@yahoo.co.uk wrote:

 Thanks
 
 What would you use to measure jitter / packetloss in real time?
 
 
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Re: [asterisk-users] Verizon SIP trunking Field Trial

[Robert-GMAIL]

Good luck! Finding the right person at VZ has always been a beef of mine


Sent from my iPhone 5

On Jan 5, 2013, at 11:12 AM, Logan Bibby lo...@keobi.com wrote:

 Does anyone have a good contact for their sales? I've attempted calling their 
 Enterprise sales a few times and was just spun around in circles. Having a 
 sales rep I can just call would be awesome.
 
 - Logan
 
 
 On Fri, Jan 4, 2013 at 1:36 PM, Michael L. Young myo...@acsacc.com wrote:
 - Original Message -
  From: Matthew J. Roth mr...@imminc.com
 
  At least Verizon maintains a consistent customer experience.  ; )
 
  Overall, we've found the service to be reliable and stable, but when
  there are problems or changes needed you're dealing with Verizon and
  the
  w...h...e...e...l...s..t...u...r...n..s...l...o...w...l...y.
 
 Haha... that is funny... it is sooo true.
 
 Well, you are right.  Once it is working, it is usually pretty stable.  Just 
 a pain in the butt when things are not working.  Hopefully we can get 
 through the Field Trial and that is all I have to worry about for a while.
 
 Thanks Matthew for all the encouragement as I go down this temporary (I 
 hope) unpleasant path.
 
 Michael
 
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 -- 
 Best regards,
 Logan
 
 Logan Bibby, CEO
 Keobi Communications
 Tuscaloosa, Alabama
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Re: [asterisk-users] Detect Low Quality Calls - Realtime

[Robert-GMAIL]

Asterisk sip show peers lists the qualify value in ms (milliseconds).

Please read up on this and the setting for it in sip.conf config file

Sent from my iPhone 5

On Jan 5, 2013, at 5:30 AM, XBrian bobo...@yahoo.co.uk wrote:

 Joachim, thanks for the reply
 - delay you can somewhat estimate prior to the call (with qualify for example)
 Pls be explicit. How do I use qualify to measure delay
 
 -  The jitter / packetloss you can only figure out when the call is already 
 up for a while. 
 what would you use to measure jitter / packetloss in real time?
 
 
 
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Re: [asterisk-users] Asterisk for Razberry Pi

[Robert Rawlinson]

Wow! Thanks so much for all the information. I now have a lot to look over.
Bob R

On 01/02/2013 10:03 AM, Tzafrir Cohen wrote:
 On Wed, Jan 02, 2013 at 09:55:44AM -0500, Robert Rawlinson wrote:
 Has anyone ported Asterisk to the Razzberry Pi? If so could you point me
 to info on doing so?
 apt-get install asterisk



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[asterisk-users] Asterisk for Razberry Pi

[Robert Rawlinson]

Has anyone ported Asterisk to the Razzberry Pi? If so could you point me
to info on doing so?
Bob R

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Re: [asterisk-users] Motif/XMPP for Google Voice

[Robert]

I again would recommend a more thorough explanation of the configsŠ

I've been using Asterisk for years - but the configs for this need some
explanation in the wikiŠ

The samples contradict what the wiki has.. And as I indicated I could
not get audio working...

On 10/15/12 10:11 AM, Joshua Colp jc...@digium.com wrote:

Joshua Colp wrote:
 asterisk asterisk wrote:
 Dear all,

 Hola,

 I wish to ask a question of the new Motif Channel in asterisk 11.

 I successfully compile the binary and run without error. However, when
 dialing out, no external connection only ringing.

 During testing some issues were uncovered with the Motif channel driver,
 but unfortunately they did not make the last release candidate. My
 suggestion is to get Asterisk 11 from SVN or if you are not comfortable
 with that wait until the official Asterisk 11 release.

The fixes did not make the last release candidate, that is.

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com   www.asterisk.org

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Re: [asterisk-users] Motif XMPP

[Robert]

Well we trashed both 11 installs (11 from tgz on site and 11 from svn) as
the configuration just wouldn't work.

Again reverting to Asterisk 10 with NO network changes / no machine /
firewall changes worked instantly.

We also threw wireshark up and saw no rtp or other such audio path when on
11.

I configured the config files as per Digium wiki.

Maybe there isn't many people who have tried out Motif yet.
Here's hoping that the old school way on Ast10 will be reliable

On 10/10/12 3:48 PM, Joshua Colp jc...@digium.com wrote:

Robert wrote:
 My apologiesŠ I will clarify the situation.

 We set up Motif per Digium's new WIKI on Google Voice for Asterisk 11.
 It completed dialing / ring and answer BUT NO AUDIO.. No errors on the
 console.

I've experienced this once or twice and narrowed it down to the Google
Voice server. It would just refuse to send the information about where
to send media. Call again and it would work fine. I compared the traffic
and saw no difference. The Google Voice platform is very ...
interesting. I've even had this happen using the Google Talk plug-in.

 We upgrade to SVN pull of Asterisk 11 and now Motif gives new errors
(ICE).

The Google Talk and Google Voice support doesn't use ICE like proper
Jingle support. Google has their own and we have basic support for it. I
would be very interested in seeing this error message.

 I gave up as there is little corresponding documentation on MOTIF.

There is a completely documented configuration file which includes every
option (motif.conf.sample), and the wiki page does normally work for
people. As this is a new channel driver and not in widespread use I'm
unsurprised there is little additional documentation.

 We rolled back to Asterisk 10 and got it to work within minutes using
old
 GTALK/Jabber methodology.

Is it working reliably? The signaling between both should be the same so
it would be very weird if it also did not have the same issue. Of course
you would be hitting different servers in the Google network.

 I know the rules about cross-post and before casting stones - I've been
 around Asterisk and other platforms for a long time.

Cheers,

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com   www.asterisk.org

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[asterisk-users] Motif XMPP

[Robert]

Just installed 11 and trying to get MOTIF / XMPP working to E164/PSTN
number.

We can get ring and a connected call ­ but no audioŠ

SIP = ASTERISK = MOTIF

Is there any specific configurations for getting audio to work?


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Re: [asterisk-users] Motif XMPP

[Robert]

My apologiesŠ I will clarify the situation.

We set up Motif per Digium's new WIKI on Google Voice for Asterisk 11.
It completed dialing / ring and answer BUT NO AUDIO.. No errors on the
console.

We upgrade to SVN pull of Asterisk 11 and now Motif gives new errors (ICE).

I gave up as there is little corresponding documentation on MOTIF.

We rolled back to Asterisk 10 and got it to work within minutes using old
GTALK/Jabber methodology.

I know the rules about cross-post and before casting stones - I've been
around Asterisk and other platforms for a long time.

So thanks!

On 10/10/12 2:45 PM, Joshua Colp jc...@digium.com wrote:

Robert wrote:

Hola,

Please in the future don't cross post as you have done to both the
developer list and users list. If it's not related to development of
Asterisk the users list is where it should stay.

 Just installed 11 and trying to get MOTIF / XMPP working to E164/PSTN
 number.

 We can get ring and a connected call ­ but no audioŠ

 SIP = ASTERISK = MOTIF

 Is there any specific configurations for getting audio to work?

You haven't specified what you are calling out through, provided any
console log output, any network information, etc. This is really needed
before anyone can even come close to diagnosing your issue.

To answer your question though there is nothing explicit you configure
to have audio work. It should just work.

Cheers,

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com   www.asterisk.org

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Re: [asterisk-users] Failover router recommendation

[Robert Rosser]

Edgewater 4350 or cheaper vigor 2910 dreytech

On Tue, Oct 9, 2012 at 3:24 PM, Mike Diehl mdiehlena...@gmail.com wrote:

 I hope no one considers this off topic...

 I have a phone customer who wants 2 Internet connections so that if one
 goes down, he can use the other for phone service.

 So, I'd like to get a recommendation for a relatively inexpensive router
 that can perform this function.

 Also, when the failover occurs, the phone's IP address will obviously
 change.  So, how can/should I configure this to minimize my customer's
 down-time?

 TIA,

 Mike Diehl.

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Re: [asterisk-users] asterisk-users Digest, Vol 99, Issue 9

[frangky robert]

  Here is my IP-PBX setupmy setup is : sips softphone - asterisk - xorcom 
  PSTN gateway - pstn line to telcoi'm using xlite for windows
 
  when I make a phone call (sip - outgoing channel),I can hear my own voice 
  so clear. it's very annoying mewhen talking a little loud... any solution? 
 
 Two questions:
 
 (1) Does the problem occur when you make a SIP-to-SIP call, without
 the PSTN being involved?

No, it's happened only when I make a call from sip to pstn line.

 (2) When you hear your own voice in the headset, is it delayed, or
 is just an immediate louder-than-you-want side-tone?

it's immediate voice and very clear, just like talk-to-my-ear with no delay
 If it *does* occur in SIP-to-SIP calls, this would rule out your
 XORCOM and the PSTN as the cause.  If it's only occurring in
 SIP-to-PSTN calls, then the XORCOM and PSTN (or the interaction
 between them) is a likely suspect.
 
 There are several things which can cause this sort of problem.
 
 (A) Direct acoustic feedback within the headset.  In this case, you'd
 probably hear it even if the headset was unplugged entirely.  The
 only cure is to buy a better headset.
 
 (B) Incorrect audio-mixer settings in your PC.  To the PC audio
 infrastructure, a headset usually looks like a microphone
 and a separate speaker.  The audio mixer (hardware and software)
 usually has an ability to mix some of what the microphone hears
 into the speaker output.  If this knob is turned up too high,
 you'll hear your own voice too loudly.  If too low, you won't
 hear your own voice at all when you speak into the headset, and
 many people find this lack of side-tone to be confusing.
 
 The cure here is to adjust the audio side-tone level, either
 in your Windows audio-mixer control panel, or in X-Lite (if
 it has such an adjustment).
 
 (C) Electrical reflection from an analog impedance discontinuity
 in the analog telephone-line system.  This can result from
 a mismatch between the telephone wiring, and the PSTN interface
 device, and can occur at any point in the analog transmission.
 
 If the loud side-tone you hear is *not* delayed noticeably,
 then the impedance mismatch might be at your XORCOM/PSTN
 interface.  The XORCOM may have a software adjustment or
 jumper setting, to match its audio impedance to that of your
 local phone line... try fiddling with these settings to see
 if they reduce the excessive side-tone level.
 
 If the loud side-tone you hear is delayed (it sounds a bit
 like an echo) then it may very well be at the far end of
 the phone line, outside of your own physical control... it
 might be at your local phone office, or anywhere between you
 and the far end of the phone connection.  Not much you can do
 about this.
 
 (D) Audio feedback at the far end of the call, in a cheap phone
 handset.  Sometimes, audio from the back side of the speaker
 in a handset travels through the body of the handset and is
 picked up by the microphone, and results in an audible delayed
 echo of the voice from the far end of the line.  Using a
 better handset, or stuffing the handset full of audio damping
 material (cloth or cotton or fiberglass) is the cure here.

Well, thanks a lot Lee for suggestion and explanation, I'll try this tommorow.

 
 We've often faced this problem with SIP soft phones when the computer's 
 sound system gain was set too high.  You usually have to play around 
 with microphone gain settings to get to the point where the echo 
 disappears with the other party still being able to hear you.

And thanks for your share Raj, I appreciate that..  
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Re: [asterisk-users] I can hear my own voice through the headset

[frangky robert]

Sorry for my last post,




  Here is my IP-PBX setupmy setup is : sips softphone - asterisk - xorcom 
  PSTN gateway - pstn line to telcoi'm using xlite for windows
 
  when I make a phone call (sip - outgoing channel),I can hear my own voice 
  so clear. it's very annoying mewhen talking a little loud... any solution? 
 
 Two questions:
 
 (1) Does the problem occur when you make a SIP-to-SIP call, without
 the PSTN being involved?

No, it's happened only when I make a call from sip to pstn line.

 (2) When you hear your own voice in the headset, is it delayed, or
 is just an immediate louder-than-you-want side-tone?

it's immediate voice and very clear, just like talk-to-my-ear with no delay
 If it *does* occur in SIP-to-SIP calls, this would rule out your
 XORCOM and the PSTN as the cause.  If it's only occurring in
 SIP-to-PSTN calls, then the XORCOM and PSTN (or the interaction
 between them) is a likely suspect.
 
 There are several things which can cause this sort of problem.
 
 (A) Direct acoustic feedback within the headset.  In this case, you'd
 probably hear it even if the headset was unplugged entirely.  The
 only cure is to buy a better headset.
 
 (B) Incorrect audio-mixer settings in your PC.  To the PC audio
 infrastructure, a headset usually looks like a microphone
 and a separate speaker.  The audio mixer (hardware and software)
 usually has an ability to mix some of what the microphone hears
 into the speaker output.  If this knob is turned up too high,
 you'll hear your own voice too loudly.  If too low, you won't
 hear your own voice at all when you speak into the headset, and
 many people find this lack of side-tone to be confusing.
 
 The cure here is to adjust the audio side-tone level, either
 in your Windows audio-mixer control panel, or in X-Lite (if
 it has such an adjustment).
 
 (C) Electrical reflection from an analog impedance discontinuity
 in the analog telephone-line system.  This can result from
 a mismatch between the telephone wiring, and the PSTN interface
 device, and can occur at any point in the analog transmission.
 
 If the loud side-tone you hear is *not* delayed noticeably,
 then the impedance mismatch might be at your XORCOM/PSTN
 interface.  The XORCOM may have a software adjustment or
 jumper setting, to match its audio impedance to that of your
 local phone line... try fiddling with these settings to see
 if they reduce the excessive side-tone level.
 
 If the loud side-tone you hear is delayed (it sounds a bit
 like an echo) then it may very well be at the far end of
 the phone line, outside of your own physical control... it
 might be at your local phone office, or anywhere between you
 and the far end of the phone connection.  Not much you can do
 about this.
 
 (D) Audio feedback at the far end of the call, in a cheap phone
 handset.  Sometimes, audio from the back side of the speaker
 in a handset travels through the body of the handset and is
 picked up by the microphone, and results in an audible delayed
 echo of the voice from the far end of the line.  Using a
 better handset, or stuffing the handset full of audio damping
 material (cloth or cotton or fiberglass) is the cure here.

Well, thanks a lot Lee for suggestion and explanation, I'll try this tommorow.

 
 We've often faced this problem with SIP soft phones when the computer's 
 sound system gain was set too high.  You usually have to play around 
 with microphone gain settings to get to the point where the echo 
 disappears with the other party still being able to hear you.

And thanks for your share Raj, I appreciate that..  
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[asterisk-users] I can hear my own voice through the headset

[frangky robert]

Hi all,
Here is my IP-PBX setupmy setup is : sips softphone - asterisk - xorcom 
PSTN gateway - pstn line to telcoi'm using xlite for windows
when I make a phone call (sip - outgoing channel),I can hear my own voice so 
clear. it's very annoying mewhen talking a little loud... any solution? 
Thanks,
Frangky
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Re: [asterisk-users] Recommended VOIP Monitoring Tools

[Robert-IPhone]

Smokeping with sip probe is quite nice

Sent from BETA iOS6

On Jul 13, 2012, at 4:54 PM, Paul Belanger paul.belan...@polybeacon.com wrote:

 On 12-07-13 08:37 AM, Mike wrote:
 On 12-07-13 06:00 AM, Elliot Murdock wrote:
 Hello,
 
 Which tools are recommendable for monitoring VOIP, bandwidth, server
 alarms, etc.?
 
 Nagios (http://www.nagios.org/) can be configured to monitor pretty much
 anything you want. The (much) harder part is deciding what's relevant to
 monitor, and what your alarm thresholds should be set at.
 
 At $PREVIOUSEMPLOYER, we used Nagios to monitor ~4,000 hosts and about
 8,000 to 10,000 services before we started running into scaling problems
 on a single box.
 Not quite at 4000 hosts, but we are using Nagios with the nsca client. Works 
 great.
 
 -- 
 Paul Belanger | PolyBeacon, Inc.
 Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
 Github: https://github.com/pabelanger | Twitter: 
 https://twitter.com/pabelanger
 
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Re: [asterisk-users] Recommended VOIP Monitoring Tools

[Robert-IPhone]

Thanks whoever is running an auto response ticket system!

Look forward to getting more spam from you!

Sent from BETA iOS6

On Jul 13, 2012, at 4:54 PM, Paul Belanger paul.belan...@polybeacon.com wrote:

 On 12-07-13 08:37 AM, Mike wrote:
 On 12-07-13 06:00 AM, Elliot Murdock wrote:
 Hello,
 
 Which tools are recommendable for monitoring VOIP, bandwidth, server
 alarms, etc.?
 
 Nagios (http://www.nagios.org/) can be configured to monitor pretty much
 anything you want. The (much) harder part is deciding what's relevant to
 monitor, and what your alarm thresholds should be set at.
 
 At $PREVIOUSEMPLOYER, we used Nagios to monitor ~4,000 hosts and about
 8,000 to 10,000 services before we started running into scaling problems
 on a single box.
 Not quite at 4000 hosts, but we are using Nagios with the nsca client. Works 
 great.
 
 -- 
 Paul Belanger | PolyBeacon, Inc.
 Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
 Github: https://github.com/pabelanger | Twitter: 
 https://twitter.com/pabelanger
 
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Re: [asterisk-users] New router, registration problems

[Robert-IPhone]

Linksys firmware?
I've had issues with older firmwares and VoIP


Sent from my iPhone 4S

On Feb 11, 2012, at 1:05 PM, David Woodfall d...@dawoodfall.net wrote:

 On (17:38 11/02/12), David Woodfall d...@dawoodfall.net put forth the 
 proposition:
 On (16:48 11/02/12), David Woodfall d...@dawoodfall.net put forth the 
 proposition:
 I just set up a WRT54GS and now I can't dial out or in.
 
 sip show registry shows:
 
 CODE: SELECT ALL
 Hostdnsmgr Username   Refresh State 
Reg.Time
 draytel.org:5060N  x  120 
 Request Sent
 
 
 I seemed to recall that running in cli always showed a response back, but 
 there's nothing now. Using 1.8.9.2.
 I have my number at draytel set up to dial my mobile if asterisk is down 
 and it keeps doing it as if it's down.
 I setup my server as DMZ in the router, as my old one was. Tried with 
 firewall off.
 
 Any ideas?
 
 I just had a look at debug info and when I dial out I get a
 busy/congested status back. I can see registration packets going out
 but no replies.
 
 Well I'm not sure why but I just stopped asterisk for a few minutes
 and then restarted it and now it registers
 
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Re: [asterisk-users] Virtual Server

[Robert-IPhone]

I run two off virtuozo vps boxes - but capacity will always be the defining 
value

Sent from my iPhone 4S

On Feb 10, 2012, at 9:18 PM, Carlos Rojas crt.ro...@gmail.com wrote:

 Hello everybody
 
 someone in this list, has installed asterisk, in a virtual server like  
 proxmox? I'm thinking  install some asterisk servers in a machine dell xeon 
 64 processor, but I'm not sure, about virtual Server software.
 
 I heard, about proxmox, but I don't know if works fine.
 
 Regards
 
 Carlos
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Re: [asterisk-users] Block Specific Number on Inbound

[Robert Huddleston]

Here's what I do... Changed some variables for obscurity. 911 is the inbound 
#... exten 6000 rings to SIP/TEST

exten = 911,1,GotoIf(${BLACKLIST()}?blacklisted)
exten = 911,n,Macro(stdexten,6000,SIP/test)
exten = 911,n,Playback(transfer,skip)
exten = 911,n(blacklisted),Goto(blacklisted,s,1)


Blacklisted context uses zapateller and plays the intercept message
[blacklisted]
exten = s,1,Answer
exten = s,2,Wait(1)
exten = s,3,Zapateller
exten = s,4,Zapateller
exten = s,5,Playback(ss-noservice)
exten = s,6,Hangup

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stuart Sheldon
Sent: Thursday, December 29, 2011 9:06 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Block Specific Number on Inbound

Check out the X Boy/Girl friend feature.

http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf

Around the middle of the page.

Stu


-Original Message-
From: Kevin Oravits korav...@rcolegal.com
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
Subject: [asterisk-users] Block Specific Number on Inbound
Date: Fri, 30 Dec 2011 01:39:46 +

Greetings,

 

Is there a way to block a specific inbound number? I’ve found code online for 
blocking all nocallerid and all 800, etc. but nothing for a specific number. My 
company is wanting me to block a specific number. Is this possible in Asterisk 
1.4 and 1.6 or do I need to go through my Service Provider?

 

Thanks,

 

Kevin Oravits 

Phone Sys Admin

 


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Re: [asterisk-users] Block Specific Number on Inbound

[Robert Huddleston]

Take a look at Blacklist

 

I love that command and love to send nice intercept messages to the other
side J

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Oravits
Sent: Thursday, December 29, 2011 8:40 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Block Specific Number on Inbound

 

Greetings,

 

Is there a way to block a specific inbound number? I've found code online
for blocking all nocallerid and all 800, etc. but nothing for a specific
number. My company is wanting me to block a specific number. Is this
possible in Asterisk 1.4 and 1.6 or do I need to go through my Service
Provider?

 

Thanks,

 

Kevin Oravits  

Phone Sys Admin

 

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Re: [asterisk-users] India Telecom regulations

[Robert-IPhone]

Right check out Cordia.LT


Sent from my iPhone 4S

On Dec 19, 2011, at 9:58 PM, Raj Mathur (राज माथुर) r...@linux-delhi.org 
wrote:

 On Tuesday 20 Dec 2011, Steve Edwards wrote:
 On Mon, 19 Dec 2011, Nick Khamis wrote:
 SIP in India is illegal.
 
 What about IAX, Skype, VPN, etc?
 
 The only thing that is not permitted is bridging Internet calls with the 
 Indian PSTN.  In fact, that too is allowed if you have a VoIP licence 
 from the government.  Apart from that, as long as you continue using it 
 within your own organisation, any protocol is fine.
 
 IANAL.  TINLA.
 
 Regards,
 
 -- Raj
 -- 
 Raj Mathur  || r...@kandalaya.org   || GPG:
 http://otheronepercent.blogspot.com || http://kandalaya.org || CC68
 It is the mind that moves   || http://schizoid.in   || D17F
 
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Re: [asterisk-users] MySql Custom CDR issues

[Robert-IPhone]

Are you using FreePBX or another packaged Asterisk?

Sent from my iPhone 4S

On Dec 12, 2011, at 9:23 AM, silent sayz silent.s...@gmail.com wrote:

 hello ,
  
 I have been working hard to solve the issue of custom CDR in the Asterik with 
 Mysql but in vain.
  
 I searched google for complete 2 hours but in vain.
  
 What i want to achieve is CDR(customcolumn)=anyvaluealthough we can 
 achieve it through other ways like making a script that runs when a call ends 
 and modify the cdr and insert in custom value BUT is there any way to make 
 this work ?
  
 Thank you in advance
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Re: [asterisk-users] Becoming a CLEC

[Robert-IPhone]

Agreed. And facilities based CLEC even scarier.
Regulatory / billing / PUC legals etc ugh


Sent from my iPhone 4S

On Nov 14, 2011, at 8:33 PM, Alex Balashov abalas...@evaristesys.com wrote:

 Worst reason to become a CLEC: improved cost structure.  Or, to be precise, 
 it is a counterfactual reason, because it does not result in improved cost 
 structure.
 
 This idea is driven by an incomplete understanding of what being a CLEC 
 entails, or, for the less critically thoughtful, the free lunch fallacy.  
 There is no free lunch.  There is no such thing as an easy-peasy regulatory 
 reclassification that gets you the same stuff you were paying before, but 
 more cheaply.
 
 Becoming a CLEC is a totally different business model than the one you're in, 
 and it entails magnitudinally more technological and regulatory complexity.  
 It's really almost a different vertical.  You should become a CLEC only if 
 you want to become a CLEC, not if you want to be an ITSP with a lower cost 
 basis, because you won't be.  It is a very capital-intensive, non-trivial 
 endeavour with high barriers to entry for a good reason.  There will be 
 people out there who will tell you that those barriers are low;  they are on 
 the bridge of failing CLECs, treading water.
 
 -- 
 Alex Balashov - Principal
 Evariste Systems LLC
 260 Peachtree Street NW
 Suite 2200
 Atlanta, GA 30303
 Tel: +1-678-954-0670
 Fax: +1-404-961-1892
 Web: http://www.evaristesys.com/
 
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Re: [asterisk-users] Becoming a CLEC

[Robert-IPhone]

Wow so I left before the end of resale Verizon UNE then.
We ran Lucent 5E and Nortel DMS and provided facilities voice and DSL.
Having a large SONET fibre infrastructure helped too.


Sent from my iPhone 4S

On Nov 14, 2011, at 8:53 PM, Alex Balashov abalas...@evaristesys.com wrote:

 On 11/14/2011 08:36 PM, Robert-IPhone wrote:
 
 Agreed. And facilities based CLEC even scarier.
 
 I'm curious what sort of thing would be considered a non-facilities based 
 CLEC, since UNE-P was cancelled in 2003.
 
 There are some non-interconnected CLECs out there that exist for the sole 
 purpose of leveraging rights of way and stuff like that, but there's not too 
 many things you can do switchless, muxless, DACS-less and not interconnected 
 these days.
 
 -- 
 Alex Balashov - Principal
 Evariste Systems LLC
 260 Peachtree Street NW
 Suite 2200
 Atlanta, GA 30303
 Tel: +1-678-954-0670
 Fax: +1-404-961-1892
 Web: http://www.evaristesys.com/
 
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Re: [asterisk-users] Cisco AS5400XM

[Robert Huddleston]

Also used for calling card platforms :)

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andreas
Sikkema
Sent: Thursday, October 06, 2011 5:39 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cisco AS5400XM

On 10/6/11 11:25 PM, Kyle Sexton wrote:
 I'm looking at the Cisco AS5400XM to convert some incoming T1s to SIP
 signaling.  Has anyone had any experience with these devices?  The
 feature cards that Cisco sells can be a little confusing.  I'm
 thinking something like below is what I need.
 
 (1) AS5400XM, AS5400XM Starter Kit (inc Chassis, MB, Def Mem)
 (1) AS54-AC-RPS-PWR, AS5400 AC Redundant Power Supply
 (1) AS54-DFC-8CT1, AS5400 OCTAL T1/PRI DFC Card
 (2) AS54-DFC-108NP, AS5400 108 Voice/Universal Port Feature Card
 
 Any thoughts would be appreciated.  Thanks.

I've used them in the past and still use the little brother (AS5350XM).
I have no experience with T1s, but I used them to convert EuroISDN E1s
to SIP. They were very stable (I don't think I've ever seen one crash)
but can be a pain when you want to set them up.

These machines were originally designed as modembanks for internet
access so the default config has an interface for every B channel. That
is a pain to browse through the configuration. Grouping them solves this.

Make sure you understand how to route calls using dialpeers, and make
sure you understand this before putting them in service. These are very,
very capable machines with lots of useful configuration options.

Make sure you buy enough DSP channels to cover all simultaneous calls
that need transcoding, we generally bought enough DSP cards so we could
transcode all simultaneous calls. If you add it all up we were actually
buying more DSP channels than E1 channels were available, for some
reason Cisco designed the machine like this, perhaps to cover for slow
call teardowns occupying DSPs too long.


-- 
Andreas Sikkema

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Re: [asterisk-users] Who is the creative mind behind changing Asterisk commands at CLI?

[Robert-iPhone]

I am adding dickish to my dictionary - thats a hot one!


Sent from my iPhone

On Sep 25, 2011, at 4:41 PM, Alex Balashov abalas...@evaristesys.com wrote:

 On 09/25/2011 02:23 PM, Bruce B wrote:
 
 Stop wishing for that. I like Asterisk and I will raise a voice
 when I feel uncomfortable with changes.
 
 You won't get an audience if the way you go about it is dickish.
 
 You're being a dick, and you know you're being a dick.  You're just unwilling 
 to admit it or intellectually engage with that.
 
 If you were earnest and sincere about your desire to contribute constructive 
 criticism and effectuate change, you wouldn't start the thread with a 
 sarcastic subject line like Who is the 'creative' mind behind changing 
 Asterisk commands at CLI?  That has a mocking, derisive inflection, and you 
 know it has a mocking, derisive inflection.
 
 If you expect to be taken seriously, you need to align your behaviour with 
 your stated objective--unless that's not actually your objective, and in fact 
 your objective is to be an inflammatory jerk.
 
 -- 
 Alex Balashov - Principal
 Evariste Systems LLC
 260 Peachtree Street NW
 Suite 2200
 Atlanta, GA 30303
 Tel: +1-678-954-0670
 Fax: +1-404-961-1892
 Web: http://www.evaristesys.com/
 
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Re: [asterisk-users] VoIP Abuse to Twitter (real time VoIP Abuse)

[Robert Huddleston]

Sounds like a great idea.. Hopefully the page/account never gets hacked and
bad IP's published.. I could see a great hack of 

127.0.0.1  

192.168.0.0/16 

10.0.0.0/8 

getting up there somehow and next thing you know - BAM!

 

But I haven't RTFM - I'm guessing there is probably a white list that
supersedes the naughty list.

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Thursday, September 22, 2011 11:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] VoIP Abuse to Twitter (real time VoIP Abuse)

 

very cool!

On Thu, Sep 22, 2011 at 10:37 AM, J. Oquendo aster...@tormenting.net
wrote:


Apologies for cross posting but some of us aren't on the other list
(vice/versa) and thought both groups would benefit.

For those familiar with the VoIP Abuse Project, no need to explain the
gist of this. I got tired of parsing through the alerts (lists) I
receive via email daily. They're long and sometimes I don't have the
time to post them all. So for now, posting VoIP Abuse addresses straight
to Twitter.

So, anyone trying to compromise a pbx, is now autoposted on an hourly
basis to Twitter. Still working on pulling, have about 4 machines linked
up now, will mop em up during the week.

http://twitter.com/#!/voipabuse

Now, you can concoct a quick script off of it, e.g.:

links -dump http://twitter.com/voipabuse;|awk '/attacker/{print
iptables -A INPUT -s $2 -j DROP| sort -u}'

Will get a quickie soon from my Acme's, nCites, etc. when I have time.

For those NOT familiar with it, please Google it as I don't feel like
typing anymore ;) (sorry)



--

=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
J. Oquendo
SGFA, SGFE, C|EH, CNDA, CHFI, OSCP, CPT, RWSP, GREM

It takes 20 years to build a reputation and five minutes to
ruin it. If you think about that, you'll do things
differently. - Warren Buffett

42B0 5A53 6505 6638 44BB  3943 2BF7 D83F 210A 95AF
http://pgp.mit.edu:11371/pks/lookup?op=get
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x2BF7D83F210A95AF
search=0x2BF7D83F210A95AF


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Re: [asterisk-users] SNMP problem

[Robert Thomas]

Did you copy the asterisk-mib.txt and digium-mib.txt to the proper folder on
your distro?

I see people forgetting about that step.

On Wed, Sep 14, 2011 at 2:18 AM, Ishfaq Malik i...@pack-net.co.uk wrote:

 Hi

 I'm using Asterisk 1.8.3.2 and am trying to configure snmp using this as
 my resource

 http://ofps.oreilly.com/titles/9780596517342/asterisk-Monitoring.html

 when I execute the following command

 snmpwalk -On -v2c -c public 127.0.0.1 .1.3.6.1.4.1.22736

 I get the following response

 .1.3.6.1.4.1.22736 = No Such Object available on this agent at this OID

 The contents of my /etc/snmp/snmpd.conf is exactly like the book
 instructs

 com2sec notConfigUser  default   public

 group   notConfigGroup v1   notConfigUser
 group   notConfigGroup v2c   notConfigUser

 view allincluded  .1
 view system included  .iso.org.dod.internet.mgmt.mib-2.system

 access  notConfigGroup   any   noauthexact  allnone none

 master agentx
 agentXSocket /var/agentx/master
 agentXPerms 0660 0775 nobody root

 sysObjectID .1.3.6.1.4.1.22736.1

 Does anyone have any pointers as to where I'm going wrong?

 Thanks in advancde

 Ish
 --
 Ishfaq Malik
 Software Developer
 PackNet Ltd

 Office:   0161 660 3062


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Re: [asterisk-users] Question about voip.ms service.

[Robert-iPhone]

I'm using them for inbound and outbound on Asterisk and FreeSwitch

Sent from my iPhone

On Sep 13, 2011, at 5:14 PM, Danny Nicholas da...@debsinc.com wrote:

 That’s what this part of extensions.conf should do:
 
 ; inbound context example for your DID numbers, do not add the number 1 in 
 front
 
  
 
 [voipms-inbound]
 
 exten = 7863643011,1,Answer() ;your DID
 
  
 
  
 
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of naren
 Sent: Tuesday, September 13, 2011 4:09 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Question about voip.ms service.
 
  
 
 Yup, that part I got. What I am not clear about is how to set up the DID to 
 go to my URI. When I select manage DIDs and click on the one I want to 
 change, I see the following options for routing the DID
 
  
 
 x SIP/IAX - [main account] IAX2/10 - with my account number
 
 x SIP URI - SIP:mysi...@myuri.com:5060
 
 x System - Hangup
 
  
 
 There are several other options but they are not selectable for me because I 
 have not set up to use them.
 
  
 
 I used to have the routing set to SIP URI where I was able to specify my URI 
 where the call was routed to. But with the SIP/IAX option I do not have that 
 ability. 
 
  
 
 I am missing something fundamental here. My asterisk has the iax.conf and 
 extensions.conf entries ready to receive calls from voip.ms, but I don't know 
 how to tel voip.ms to send the calls to my asterisk with the IAX protocol. 
 
  
 
 I understand this is probably a question for the voip.ms folks, but since a 
 couple of people mentioned earlier that they were rocking with IAX, I thought 
 it would be an easy question for them to point me in the right direction.
 
  
 
 Thanks. 
 
 On Tue, Sep 13, 2011 at 3:32 PM, Dave Aibel dai...@pervasivetelecom.com 
 wrote:
 
 I was lurking in this conversation and I went to look more carefully
 at the voip.ms site. I found sample files at
 http://wiki.voip.ms/article/PBXs#Asterisk_.28SIP.29
 
 Hope that helps.
 
 
 
 On Tue, Sep 13, 2011 at 3:59 PM, naren naren.sa...@gmail.com wrote:
  I see the section you are talking about. It is on the home page if I am not
  logged in. I see the Authentication section and the text IAX/SIP
  registration, but it doesn't seem to be a link. I am not sure how I can
  find the page that has the details about the IAX/SIP registration. I see in
  the wiki there is a page that has the configuration info for iax.conf and
  extensions.conf.
  Thanks for your help.
  naren
 
  On Tue, Sep 13, 2011 at 2:25 PM, Danny Nicholas da...@debsinc.com wrote:
 
  Did you read the “IAX/SIP registration” section (under Authentication) on
  voip.ms?
 
 
 
  From: asterisk-users-boun...@lists.digium.com
  [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of naren
  Sent: Tuesday, September 13, 2011 2:22 PM
  To: John Novack
  Cc: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] Question about voip.ms service.
 
 
 
  Ok... this is probably a dumb question but I can't figure out how to set
  voip.ms to use IAX for my DID... with SIP I was able to specify the URI so 
  I
  pointed it to my asterisk installation, but with IAX I don't have that
  option. Is that supposed to work some other way?
 
 
 
  Thanks a bunch!
 
  On Mon, Sep 12, 2011 at 11:18 PM, naren naren.sa...@gmail.com wrote:
 
  I am novice with Asterisk, I had to piece together a lot of bits of info
  from lots of internet searches to get my very basic setup working. I
  probably shouldn't say that because it seems like Nat is not a very basic
  setup with Asterisk.
 
 
 
  The reason for wanting to stay with SIP is because I have my setup working
  with that protocol with an incoming and an outgoing line. I just wanted to
  add a second outgoing with voip.ms.
 
 
 
  But, I have come so far, so well why not... I will give IAX a shot, and
  see what traps I need to wade through :)
 
 
 
  Thanks
 
 
 
  On Mon, Sep 12, 2011 at 11:09 AM, John Novack
  jnov...@stromberg-carlson.org wrote:
 
  Never have had a problem with their IAX service.
 
  And ( for now ) a little hedge against the hackers.
 
  Since Asterisk is involved, why not use IAX anyway?
 
 
  John Novack
 
 
  naren wrote:
 
 
 
  I also found this... seems like voip.ms outbound is broken for now!
 
 
 
  http://pbxinaflash.com/forum/showthread.php?t=10735
 
 
 
 
 
  On Sun, Sep 11, 2011 at 10:34 PM, naren naren.sa...@gmail.com wrote:
 
  Hi,
 
 
 
  I am trying to set up my asterisk 1.8.5 with voip.ms. I had no problem
  with the incoming, but my outgoing is not working. If at all possible, I
  would like to stick with SIP. Since the original poster (Glen) had 
  mentioned
  that he had gotten outgoing working, I was wondering if you would be kind
  enough to post some thoughts on that. Were you able to get it working with
  just the default example sip.conf / extensions.conf settings that 

Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?

[Robert Huddleston]

I personally would never install a GUI o/s. By doing so you always open
yourself up to more security concerns.. Packages / ports / etc.

 

Course one might argue - it's behind a firewall..

 

In my professional experience with running numerous ISP and VoITSPs the rule
has always been install the minimum needed software to accomplish the goal.

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of linux guy
Sent: Monday, September 12, 2011 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3
or init level 5 ?

 

I'm about to start building my asterisk server and I can't seem to find
anything that discusses the pros and cons of installing the OS (Fedora 15)
as console only or GUI, ie install KDE as well.

So, other than a bit of disk space, is there any reason why I shouldn't
install KDE when I set it up ?

Is there any great disadvantage to running the server in init level 5 (ie
KDE, xorg, etc) running in the background, but not being logged in, versus
init level 3 ? (Or whatever they call these things these days..., ie F15
uses systemd...)

FWIW, my server hardware will sit on a server rack in the utility room.  I
might drag a display and keyboard down there once in a while to troubleshoot
and/or do maintenance, but mostly I'd ssh in and probably use a remote
desktop app to work on it.

FWIW, I'm OK doing things via the CLI, but sometimes its really nice to have
graphical tools.

I look forward to your input.

Thanks

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Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?

[Robert Huddleston]

Well you are correct - I did not include a discussion on performance impacts
including disk I/O etc.

 

It is true that by installing a GUI o/s additional init.d (startup) services
will fire.. Additional libraries will be inclusive etc.

 

This is why I say minimal is always better.

 

Also take for example risk mitigation with security aspects. If you minimize
the number of libraries (think windows DLL's) you have installed you also
thus minimize your potential exposure.

 

Again - this is just my recommendation and experience. Firewalls are great
at blocking things and in theory - sure you could nmap your box and look for
open ports and conceal them.

 

I remember a Solaris engineer we had once - he bragged and bragged about his
qualifications on Sun Solaris. Just to find out that he installed a bunch of
GUI tools just so that he could install Oracle drivers. Further he didn't
remove or lock down that exposure.

 

Start minimal and work your way up. Now for my poke / razz - GUI's in server
grade operating systems have made people a little to reliant on them.

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of linux guy
Sent: Monday, September 12, 2011 2:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init
level 3 or init level 5 ?

 

 

On Mon, Sep 12, 2011 at 12:24 PM, Robert Huddleston rhuddles...@gmail.com
wrote:

I personally would never install a GUI o/s. By doing so you always open
yourself up to more security concerns.. Packages / ports / etc.

 

Course one might argue - it's behind a firewall..

 

In my professional experience with running numerous ISP and VoITSPs the rule
has always been install the minimum needed software to accomplish the goal.


Thanks for the reply.  I was worried the list would find it a trite and
irritating question.

I was expecting someone to tell me that even with the GUI component running
in the background, the graphical processes have the potential to mess up the
streams.  I guess I should confess that I'm always a bit surprised to
remember that asterisk doesn't require a real time OS !

Have you really exposed much more if you install the GUI components and
normally run at init 3 ?

Thanks again. 

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Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?

[Robert-iPhone]

Asterisk is a company? This is news to me

Sent from my iPhone

On Sep 12, 2011, at 5:35 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote:

 
 
 On Mon, Sep 12, 2011 at 5:12 PM, Steve Totaro stot...@asteriskhelpdesk.com 
 wrote:
 See comments inline.
 
 On Mon, Sep 12, 2011 at 2:21 PM, linux guy linuxguy...@gmail.com wrote:
 I'm about to start building my asterisk server and I can't seem to find 
 anything that discusses the pros and cons of installing the OS (Fedora 15) as 
 console only or GUI, ie install KDE as well.
 
 
 If you want an OS that is going to be supported a year from now, don't use 
 Fedora.
 
 Go for CentOS which is essentially Red Hat Enterprise, Fedora is pretty much 
 beta RHEL.  It's EOL is one year from my understanding.
 
 You want to install the very minimum as most people would agree, why do you 
 think you need a GUI.
 
 Best practice is to only install the bare minimum on a server.
  
 So, other than a bit of disk space, is there any reason why I shouldn't 
 install KDE when I set it up ?
 
 It has and will cause issues.  I have installed KDE or whatever but booted to 
 init 3 for a couple of machines.  I could go to init 5 if I had to, but I 
 never did had to.  I don't see a single pro, but there are many cons.
 
 What benefit do you get from KDE?  Why do you want it.  Is this just going to 
 be an asterisk server or a desktop?
  
 
 Is there any great disadvantage to running the server in init level 5 (ie 
 KDE, xorg, etc) running in the background, but not being logged in, versus 
 init level 3 ? (Or whatever they call these things these days..., ie F15 uses 
 systemd...)
 
 FWIW, my server hardware will sit on a server rack in the utility room.  I 
 might drag a display and keyboard down there once in a while to troubleshoot 
 and/or do maintenance, but mostly I'd ssh in and probably use a remote 
 desktop app to work on it.   
 
 How does remote desktop help you over an SSH CLI?
  
 FWIW, I'm OK doing things via the CLI, but sometimes its really nice to have 
 graphical tools.
 
 
 Ok, I can understand, I used to be like this for a while.  I am a huge fan of 
 Webmin for a GUI.  It allows for almost everything and for me, it is better 
 than KDE or anything else.  It is just a webpage with tools attached.  No big 
 potential problem there.
  
 I look forward to your input.
 
 Thanks
 
 
 I have been using Vyatta (paid for with phone support.)
 
 It makes for the most powerful Asterisk platform you can imagine.  There is a 
 learning curve but I love what I have put together.  There are howtos 
 everywhere and if you buy licenses, you get excellent support and online 
 training courses.
 
 It is a very firewall/Router.  It handles everything from OpenVPN, awesome 
 security features, IPS, and even QoS, wireshark.
 
 I put webmin and NTOP on these machines as well.  Vyatta has become my new 
 platform for Asterisk.
 
 Check it out http://www.vyatta.org/documentation
 
 There is very little you cannot do, but don't have to use the features if you 
 don't want to.
 
 Vyatta is also a company like Asterisk.  Vyatta is the baby of former bigtime 
 corporate Cisco guys.  Asterisk is the baby of former Adtran execs.
 
 Thanks,
 Steve T
 
 Thanks,
 Steve T
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Re: [asterisk-users] Reporting for Asterisk Call Center

[Robert Huddleston]

www.buildityourself.org

:)

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Friday, September 09, 2011 2:14 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Reporting for Asterisk Call Center

Hi All;

Anyone advise for a free (open source) reporting to be used for asterisk
call center?

Regards
Bilal

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Re: [asterisk-users] Phone numbers and asterisk

[Robert-iPhone]

what do you mean? Like speed dial or directory?

Sent from my iPhone

On Sep 4, 2011, at 6:47 PM, neo haux neo.h...@gmx.com wrote:

 Hi,
 
 It is possible to save all the phones numbers on asterisk servers instead of 
 doing so manually in each VoIP device ?
 
 Does SIP take care of such configuration ?
 
 Thanks
 
 
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[asterisk-users] CDR dialed digits missing

[robert boardman]

Hi

I'm using asterisk 1.6.2.18.1

I'm having a problem where only the first four digits are collected in the
cdr when the call is dialed overlap but if the call is dialed en-block the
whole dialed digits are recorded

chan_dahdi.conf

[trunkgroups]

[channels]
language=uk
switchtype=euroisdn
pridialplan=unknown
prilocaldialplan=unknown
nationalprefix=0
internationalprefix=00
echocancel=yes
echocancelwhenbridged=no
overlapdial=yes
inbanddisconnect=yes
priindication=inband
relaxdtmf=no
switchtype=qsig
context=incomming
group=1
signalling=pri_cpe
channel =1-15,17-31

pridialplan=unknown
prilocaldialplan=unknown
nationalprefix=unknown
internationalprefix=unknown
echocancel=yes
echocancelwhenbridged=no
;overlapdial=incoming
overlapdial=yes
inbanddisconnect=yes
priindication=inband
relaxdtmf=yes
switchtype=euroisdn
context=dialednum
group=2
signalling=pri_net
channel =32-46,48-62

the dial command is

exten = _X.,1,dial(DAHDI/g1/${EXTEN})

Thanks for your help

Robb
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Re: [asterisk-users] Prompt for PIN After dialing

[Robert Huddleston]

Have you looked at pin sets in freepbx / trixbox / elastix? I haven't tested
it myself - but I know the feature is present there

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps
Sent: Friday, September 02, 2011 11:15 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Prompt for PIN After dialing

Hello All,

We would like to change our dialplan a bit so that after a user dials a
number (any number, including domestic, international, internal) Asterisk
firsts prompts the user for a PIN before actually allowing the call to go
through.

I know I could setup an IVR that would accomplish this but I'd prefer not to
have the users first call an internal extension before they dial out.  I
want them to be able to dial the destination number directly, have asterisk
intercept and prompt for password, then either allow the call or play a .gsm
file and hangup if the PIN is incorrect.

We are using AELs, and not the exten,x,x format.

Thanks in advance,

Brandon

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Re: [asterisk-users] Avaya to Asterisk Voice mail

[Robert Huddleston]

Search the forum - I believe I remember a recent exchange on this subject

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dustin fails
Sent: Tuesday, August 30, 2011 10:44 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Avaya to Asterisk Voice mail

 

Has anyone have Avaya setup to ring to Asterisk voice mail over an analogue
line. The issue I am having is Avaya is sending the originating caller id
not the station id so Asterisk see that originating id so I can't route the
call correctly in Asterisk. 

Thanks!

Dustin

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Re: [asterisk-users] 1.8.5 Voicemail duration incorrect

[Robert Huddleston]

https://issues.asterisk.org/jira/browse/ASTERISK-16981

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karsten
Wemheuer
Sent: Thursday, August 25, 2011 3:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 1.8.5 Voicemail duration incorrect

Hi,

Am Mittwoch, den 24.08.2011, 13:18 -0400 schrieb Robert Huddleston:
 Anyone else seen this?
 
  
 
 I saw a jira but was in feedback status..

I just checked with a voicemail of 60 seconds. It was reported
in .txt-file with a duration of 19 seconds. So there is a bug. Do You
have a link to the Jira issue?

Karsten




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[asterisk-users] 1.8.5 Voicemail duration incorrect

[Robert Huddleston]

Anyone else seen this?

 

I saw a jira but was in feedback status..

 

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Re: [asterisk-users] Assistance sending mass sms to cellphones

[Robert Huddleston]

This is off topic...

Asterisk will not provide you with the ability to SMS random cell phones.

Being able to transport the SMS yourself is a grewling process.. Look at
software like Kamel...

Basically you have three options:
( a ) cheat and use the email method - i.e. determine everyone's carrier and
use the email address equivalent
( b ) utilize a 3rd party to transmit the sms for you (cost) and they might
end up doing ( a ) above without you knowing
( c ) spend lots of money and headaches transporting sms yourself.

Either way it's off-topic and not related to Asterisk.


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy
Sent: Friday, August 05, 2011 11:42 AM
To: asterisk
Subject: [asterisk-users] Assistance sending mass sms to cellphones

Hello.

I would like to know if is possible to send mass sms with an php agi script
through asterisk?

For example: I have about 50 cellphone numbers I would like to text whenever
theres a meeting, I should load the numbers from a database and send a
message via web with php and have asterisk send it.

I've been googling about it but, I get a lot of providers that already do
this but, I would like to learn how to do it myself since my budget is very
minimum.

Thanks in advanced for your help and time.


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Re: [asterisk-users] Assistance sending mass sms to cellphones

[Robert Huddleston]

Seriously Again?

This is off topic...

Asterisk will not provide you with the ability to SMS random cell phones.

Being able to transport the SMS yourself is a grewling process.. Look at
software like Kamel...

Basically you have three options:
( a ) cheat and use the email method - i.e. determine everyone's carrier and
use the email address equivalent ( b ) utilize a 3rd party to transmit the
sms for you (cost) and they might end up doing ( a ) above without you
knowing ( c ) spend lots of money and headaches transporting sms yourself.

Either way it's off-topic and not related to Asterisk.


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy
Sent: Friday, August 05, 2011 12:42 PM
To: asterisk
Subject: [asterisk-users] Assistance sending mass sms to cellphones

Hello.

I would like to know if is possible to send mass sms with an php agi script
through asterisk?

For example: I have about 50 cellphone numbers I would like to text whenever
theres a meeting, I should load the numbers from a database and send a
message via web with php and have asterisk send it.

I've been googling about it but, I get a lot of providers that already do
this but, I would like to learn how to do it myself since my budget is very
minimum.

Thanks in advanced for your help and time.


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Re: [asterisk-users] Assistance sending mass sms to cellphones

[Robert Huddleston]

When you say expensive... You are talking about pennies per SMS... Again -
if you want to cheat and go the email route - that would be free... It's
unreliable and requires some thought...

If you want more information / consulting contact me off-list.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy
Sent: Friday, August 05, 2011 12:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Assistance sending mass sms to cellphones

Robert.

Thanks for replying.

--- On Fri, 8/5/11, Robert Huddleston rhuddles...@gmail.com wrote:

 From: Robert Huddleston rhuddles...@gmail.com
 Subject: Re: [asterisk-users] Assistance sending mass sms to cellphones
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
 Date: Friday, August 5, 2011, 11:50 AM
 This is off topic...
 
 Asterisk will not provide you with the ability to SMS
 random cell phones.

We actually have a group of people belonging to a rotary club and we wanted
to automate the sms process... is not random cell phones.

 
 Being able to transport the SMS yourself is a grewling
 process.. Look at
 software like Kamel...
 
 Basically you have three options:
 ( a ) cheat and use the email method - i.e. determine
 everyone's carrier and
 use the email address equivalent
 ( b ) utilize a 3rd party to transmit the sms for you
 (cost) and they might

Looks like this is the easiest option but, very expensive for what we really
want to do.

 end up doing ( a ) above without you knowing
 ( c ) spend lots of money and headaches transporting sms
 yourself.
 
 Either way it's off-topic and not related to Asterisk.
 

Sorry, didn't think this wasnt an asterisk related question.

 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com]
 On Behalf Of Landy Landy
 Sent: Friday, August 05, 2011 11:42 AM
 To: asterisk
 Subject: [asterisk-users] Assistance sending mass sms to
 cellphones
 
 Hello.
 
 I would like to know if is possible to send mass sms with
 an php agi script
 through asterisk?
 
 For example: I have about 50 cellphone numbers I would like
 to text whenever
 theres a meeting, I should load the numbers from a database
 and send a
 message via web with php and have asterisk send it.
 
 I've been googling about it but, I get a lot of providers
 that already do
 this but, I would like to learn how to do it myself since
 my budget is very
 minimum.
 
 Thanks in advanced for your help and time.
 
 
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    http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
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    http://lists.digium.com/mailman/listinfo/asterisk-users
 

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[asterisk-users] T38 Fax

[Robert Huddleston]

Anyone have any testing experience with T38 and HT-502 Grandstream?

 

I just want to confirm that t.38 is working on this device.

 

Thanks

 

 

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Re: [asterisk-users] T38 Fax with Grandstream HT-502

[Robert Huddleston]

My apologies - yes.. Grandstream HT-502...

Apparently finding a t.38 provider is also another struggle...

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Monday, August 01, 2011 1:05 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] T38 Fax with Grandstream HT-502

On 08/01/2011 12:02 PM, Robert Huddleston wrote:
 Anyone have any testing experience with T38 and HT-502 Grandstream?

 I just want to confirm that t.38 is working on this device.

You'd be more likely to get relevant responses if you had included the 
information about the HT-502 in your message subject :-)

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] T38 Fax

[Robert Huddleston]

Thanks - and did you find a provider with T.38 DIDs? I don't see many pay as
you go providers with T.38

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Larry Moore
Sent: Monday, August 01, 2011 4:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] T38 Fax

 

On 2/08/2011 1:02 AM, Robert Huddleston wrote: 

Anyone have any testing experience with T38 and HT-502 Grandstream?

 

I just want to confirm that t.38 is working on this device.

 

Thanks

 

 


Yes, it works.

I currently have latest firmware installed and it still works in T.38. I am
using UDP transport for this device as I seem to encounter problems with TCP
or TLS.

I am currently running Asterisk 1.8.5.0.


Product Model: 

  HT-502 V1.1C 


Software Version: 

  Program-- 1.0.5.5Bootloader-- 1.0.0.9Core-- 1.0.5.2Base--
1.0.5.2


Some settings I have set and you may wish to check for the FXS port are;


Force INVITE: 

  (X) No ( ) Yes (Always refresh with INVITE instead of UPDATE)


Send Re-INVITE After Fax: 

  ( ) No (X) Yes 

 


VAD: 

  ( ) No   (X) Yes 


Symmetric RTP: 

  (X) No   ( ) Yes 


Fax mode: 

  (X) T.38 (Auto Detect)   ( ) Pass-Through 


Fax tone detection mode: 

  ( ) Caller   (X) Callee   ( ) Caller or Callee


Jitter buffer type: 

  (X) Fixed   ( ) Adaptive 


Jitter buffer length: 

  (X) Low ( ) Medium   ( ) High 



You will need to ensure you are using redundancy mode instead of FEC.

I am able to send a fax via my voice provider seemingly without errors even
though ECM is not enabled, this is because redundancy mode is working as
expected on the outbound communication.

Unfortunately my voice provider only sends one data item in the incoming
UDPTL hence the occasional missed line.

Here is an extract from my sip.conf

[general]
.
.
t38pt_udptl=yes,redundancy,maxdatagram=400
.
.
[906]
; Grandstream HT502 FXS Port
; Analogue FAX Modem attached
type=friend
defaultuser=906
md5secret=c5bca943c9b0cc303c496fbf9d48a48e
call-limit=1
disallow=g722
transport=udp
qualify=yes
directmedia=no
host=dynamic
context=FAX-T38
faxdetect=no
deny=0.0.0.0/0.0.0.0
permit=10.0.0.0/255.0.0.0
permit=172.16.0.0/255.240.0.0
permit=192.168.0.0/255.255.0.0

Larry.

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Re: [asterisk-users] sip attacks

[Robert-iPhone]

hard to equate sip attack to ping performance.. Run mtr for a bit.
Also try tcpdump or wireshark or tethereal.
If you are really paranoid recycle all your passwords

Sent from my iPhone

On Jul 31, 2011, at 7:04 PM, Dave George dgeo...@teletoneinc.com wrote:

 My asterisk server is getting bogged down every 5 minutes.  My ping time is
 going from 60ms to 800 ms and the call quality is bad.
 
 I have fail2ban running and I am using iptables.  I have two ip connections
 to the box.
 
 How can I tell if the poor performance is due to sip attacks?   I don't see
 any reg attempts in my asterisk cli.  I use to get frequent attacks but
 fail2ban seems to be taking care of that.
 
 See how ping time gets worst in a short space of time and server performance
 at the time:
 
 
 64 bytes from 4.2.2.1: icmp_seq=6 ttl=55 time=87.8 ms
 64 bytes from 4.2.2.1: icmp_seq=7 ttl=55 time=99.8 ms
 64 bytes from 4.2.2.1: icmp_seq=8 ttl=55 time=107 ms
 64 bytes from 4.2.2.1: icmp_seq=9 ttl=55 time=115 ms
 64 bytes from 4.2.2.1: icmp_seq=10 ttl=55 time=120 ms
 64 bytes from 4.2.2.1: icmp_seq=11 ttl=55 time=122 ms
 64 bytes from 4.2.2.1: icmp_seq=12 ttl=55 time=123 ms
 64 bytes from 4.2.2.1: icmp_seq=13 ttl=55 time=126 ms
 64 bytes from 4.2.2.1: icmp_seq=14 ttl=55 time=122 ms
 64 bytes from 4.2.2.1: icmp_seq=15 ttl=55 time=142 ms
 64 bytes from 4.2.2.1: icmp_seq=16 ttl=55 time=142 ms
 64 bytes from 4.2.2.1: icmp_seq=17 ttl=55 time=137 ms
 64 bytes from 4.2.2.1: icmp_seq=18 ttl=55 time=186 ms
 64 bytes from 4.2.2.1: icmp_seq=19 ttl=55 time=255 ms
 64 bytes from 4.2.2.1: icmp_seq=20 ttl=55 time=310 ms
 64 bytes from 4.2.2.1: icmp_seq=21 ttl=55 time=387 ms
 64 bytes from 4.2.2.1: icmp_seq=22 ttl=55 time=445 ms
 64 bytes from 4.2.2.1: icmp_seq=23 ttl=55 time=514 ms
 64 bytes from 4.2.2.1: icmp_seq=24 ttl=55 time=583 ms
 64 bytes from 4.2.2.1: icmp_seq=25 ttl=55 time=650 ms
 64 bytes from 4.2.2.1: icmp_seq=26 ttl=55 time=715 ms
 64 bytes from 4.2.2.1: icmp_seq=27 ttl=55 time=783 ms
 64 bytes from 4.2.2.1: icmp_seq=28 ttl=55 time=821 ms
 64 bytes from 4.2.2.1: icmp_seq=29 ttl=55 time=810 ms
 64 bytes from 4.2.2.1: icmp_seq=30 ttl=55 time=832 ms
 64 bytes from 4.2.2.1: icmp_seq=31 ttl=55 time=812 ms
 64 bytes from 4.2.2.1: icmp_seq=32 ttl=55 time=821 ms
 64 bytes from 4.2.2.1: icmp_seq=33 ttl=55 time=826 ms
 64 bytes from 4.2.2.1: icmp_seq=34 ttl=55 time=815 ms
 64 bytes from 4.2.2.1: icmp_seq=35 ttl=55 time=821 ms
 64 bytes from 4.2.2.1: icmp_seq=36 ttl=55 time=824 ms
 
 top - 19:02:38 up 4 days, 11:26,  4 users,  load average: 0.36, 0.75, 0.82
 Mem:   4051312k total,  1062964k used,  2988348k free,   167004k buffers
 Swap:  6094840k total,0k used,  6094840k free,   680144k cached
 
  PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEMTIME+  COMMAND
 4245 root  15   0  791m  86m  10m S 39.6  2.2   1192:32 asterisk
 18280 root  15   0  3812  600  516 S  2.0  0.0   0:59.00 pppoe
 2582 root  15   0  5912  628  504 S  0.3  0.0   2:02.19 syslogd
 18978 root  15   0 12744 1096  812 R  0.3  0.0   0:00.02 top
1 root  15   0 10352  700  588 S  0.0  0.0   0:01.14 init
2 root  RT  -5 000 S  0.0  0.0   0:00.01 migration/0
3 root  34  19 000 S  0.0  0.0   0:31.90 ksoftirqd/0
4 root  RT  -5 000 S  0.0  0.0   0:00.00 watchdog/0
5 root  RT  -5 000 S  0.0  0.0   0:00.01 migration/1
6 root  34  19 000 S  0.0  0.0   0:08.43 ksoftirqd/1
7 root  RT  -5 000 S  0.0  0.0   0:00.00 watchdog/1
8 root  RT  -5 000 S  0.0  0.0   0:00.13 migration/2
9 root  34  19 000 S  0.0  0.0   2:40.56 ksoftirqd/2
   10 root  RT  -5 000 S  0.0  0.0   0:00.00 watchdog/2
   11 root  RT  -5 000 S  0.0  0.0   0:00.05 migration/3
   12 root  34  19 000 S  0.0  0.0   0:44.56 ksoftirqd/3
   13 root  RT  -5 000 S  0.0  0.0   0:00.00 watchdog/3
   14 root  10  -5 000 S  0.0  0.0   0:00.02 events/0
   15 root  10  -5 000 S  0.0  0.0   0:00.00 events/1
   16 root  10  -5 000 S  0.0  0.0   0:00.00 events/2
   17 root  10  -5 000 S  0.0  0.0   0:00.00 events/3
   18 root  10  -5 000 S  0.0  0.0   0:00.00 khelper
   55 root  10  -5 000 S  0.0  0.0   0:00.00 kthread
   62 root  10  -5 000 S  0.0  0.0   0:00.07 kblockd/0
   63 root  10  -5 000 S  0.0  0.0   0:00.01 kblockd/1
   64 root  10  -5 000 S  0.0  0.0   0:00.00 kblockd/2
   65 root  10  -5 000 S  0.0  0.0   0:00.00 kblockd/3
   66 root  17  -5 000 S  0.0  0.0   0:00.00 kacpid
  166 root  17  -5 000 S  0.0  0.0   0:00.00 cqueue/0
  167 root  18  -5 000 S  0.0  0.0   0:00.00 cqueue/1
 
 
 
 Dave
 
 
 
 --
 _
 -- 

Re: [asterisk-users] MoH - conversion command

[Robert Huddleston]

Personally I like to just hook up an old ghetto blaster / boombox to the
line in port on my sound card :)

Kidding aside - I think audio quality for MoH is not always going to sound
as nice as you might want.

I mostly stream online radio over my MoH and the quality is not the
greatest.

Maybe it's my SIP provider - or maybe just the notion of streaming audio
from an internet stream.


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Thursday, July 28, 2011 11:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MoH - conversion command

On Thu, 28 Jul 2011, Mike wrote:

 I?ve got a hold of Royalty-free Classical music (a safe choice for 
 most of my customers) and I`ve been trying to convert them to the 
 normal telephony/Asterisk format using sox.  Unfortunately, it sounds 
 really bad.

I convert files using:

 sox ${INPUT} -c 1 -s -w -r 8000 /tmp/$$.wav

What does your sox command line look like?

Can you post a link to 'before' and 'after' files?

--
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] Stun Server

[Robert Huddleston]

I like Xen. It's free and rock solid. VMWare is great but their money
greedy.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Wednesday, July 27, 2011 9:30 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Stun Server

 

We have been running a windows stun server for 5 years now and I would like
to change to either a linux of freebsd based unit to phase out the old XP
box in our datacenter.   What should I look at that would be a good
replacement.  The windows box has worked but the hardware is at end of life
and I want to move it to a vm and I don't want Windows. 

Any advise is apperciated. 

Thanks
zktech

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Re: [asterisk-users] Lightning and thunder

[Robert Huddleston]

Maybe write a cludgy init.d script / bash. I.e. make a file somewhere called
rebootflag. set it to 1 after you issue a shutdown -r -n now. then check it
in init.d script.

 

Pseudo code

 

In init.d / startup scripts

If /etc/manualreboot = 0 or file not found

echo 1  /etc/manualreboot

/sbin/shutdown -r -n now

end if

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Claude Hayn
Sent: Wednesday, July 27, 2011 9:44 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Lightning and thunder

 

We are frequently losing power during lightning storms.  (Yes we have UPS,
but often by the time power comes back up the UPS has run out of juice)

 

We are using Asterisk with a T1/PRI card as a front end connected to our
PBX.  Whenever there is a power outage both the Asterisk box and the PBX
automatically reboot when power returns.

 

The Asterisk box automatically reconnects to the ITSP SIP Peer, and the PBX
to the T1/PRI Card Asterisk box.  

 

Incoming calls connect, but outbound calls will not complete until the
Asterisk box is manually rebooted again.

 

Does anyone know of a solution for this issue?  Having to get up in the late
night to manually reboot the Asterisk box is getting old!

 

Thank you,

 

Claude

 

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Re: [asterisk-users] Lightning and thunder

[Robert Huddleston]

I agree - using powerchute or another ups software clean shutdown is great.

 

My response was a scripted way to resolve the reboot issue based on what the
writer asked for.

 

Additionally loop wouldn't happen. That's why I wrote echo 1  some file
and if check that file.

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, July 27, 2011 10:14 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Lightning and thunder

 

This is the right idea - have your UPS write power loss shutdown when it
has to stop the machine, then check for that when you come back up and
reboot when you see it (of course you would need to log something else to
prevent a loop of reboots).

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert
Huddleston
Sent: Wednesday, July 27, 2011 9:05 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Lightning and thunder

 

Maybe write a cludgy init.d script / bash. I.e. make a file somewhere called
rebootflag. set it to 1 after you issue a shutdown -r -n now. then check it
in init.d script.

 

Pseudo code

 

In init.d / startup scripts

If /etc/manualreboot = 0 or file not found

echo 1  /etc/manualreboot

/sbin/shutdown -r -n now

end if

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Claude Hayn
Sent: Wednesday, July 27, 2011 9:44 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Lightning and thunder

 

We are frequently losing power during lightning storms.  (Yes we have UPS,
but often by the time power comes back up the UPS has run out of juice)

 

We are using Asterisk with a T1/PRI card as a front end connected to our
PBX.  Whenever there is a power outage both the Asterisk box and the PBX
automatically reboot when power returns.

 

The Asterisk box automatically reconnects to the ITSP SIP Peer, and the PBX
to the T1/PRI Card Asterisk box.  

 

Incoming calls connect, but outbound calls will not complete until the
Asterisk box is manually rebooted again.

 

Does anyone know of a solution for this issue?  Having to get up in the late
night to manually reboot the Asterisk box is getting old!

 

Thank you,

 

Claude

 

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Re: [asterisk-users] Lightning and thunder

[Robert-iPhone]

gerbals

Sent from my iPhone

On Jul 27, 2011, at 5:32 PM, Hans Witvliet h...@a-domani.nl wrote:

 On Wed, 2011-07-27 at 09:44 -0400, Claude Hayn wrote:
 We are frequently losing power during lightning storms.  (Yes we have
 UPS, but often by the time power comes back up the UPS has run out of
 juice)
 
 snip
 
 Does anyone know of a solution for this issue?  Having to get up in
 the late night to manually reboot the Asterisk box is getting old!
 
 
 Perhaps an other suggestion...
 Re-install asterisk on a other piece of hardware.
 There are small boxes that consume less than 5 Watt.
 If you put that on your UPS, it will last longer.
 
 Other one, ever thought of an alternative power source?
 Either solar of conventional?
 
 hw
 
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Re: [asterisk-users] NAT yes

[Robert Huddleston]

Also consider the setting localnet in sip.conf

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent: Tuesday, July 26, 2011 9:24 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] NAT yes

On 07/26/2011 09:19 AM, Flavio Miranda wrote:

 In a no natted environment if I letnat=yes on sip.conf it would
 cause some thing bad or it is irrelevant ? Anybody know ?

There is no harm unless the endpoint you are dealing with does not do 
symmetric RTP.  The nat=yes option assumes that it is okay to send RTP 
back to the source port from which it originated, irrespectively of 
what's in the SDP.  This will cause one-way audio if the endpoint 
happens to want to receive RTP on a different port than the one it is 
sending it from.

Almost all endpoints these days do symmetric RTP, though, so it's not 
a huge concern.

That said, from a methodological and aesthetic perspective, it is 
better not to break standard RFC-compliant behaviour unnecessarily. 
Thus, I would not enable nat=yes unless there really is no direct 
network and transport-layer reachability to the endpoint.

-- 
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

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Re: [asterisk-users] Securing Asterisk

[Robert-iPhone]

Such a pointless argument. The same problem can happen on any voip platform 
including freeswitch.
Again it's a knowledge thing.
BTW if you were paying attention to your logs or practiced good admin skills 
you would have seen the attacks and stopped them.
I swear by fail2ban and other hardening techniques. If you honestly think you 
can just run the box out in the open after running a yum / apt or
rpm command you are in the wrong position.
Know this is going to sound harsh but you deserve the pay cut if not 
termination.


Sent from my iPhone

On Jul 23, 2011, at 2:13 PM, Danny Nicholas da...@debsinc.com wrote:

 Simple economics tells me that we can't pay enough guys $X U.S. to stop the
 problem when we are competing with multiple folks working for $0.X US.
 Asterisk isn't the problem, it's just another limb on the victim tree.
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger
 Sent: Saturday, July 23, 2011 1:10 PM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Securing Asterisk
 
 On 11-07-23 01:38 PM, CDR wrote:
 I beg to differ. Digium is hiding from the real world and somebody is 
 going take the software and run with it. My customers lost in excess 
 of $50.000 and cut my pay in half, because of hackers. The hackers 
 figured out how to scan every asterisk for weak passwords or open 
 ports, and bang them real good. We need two things: a) disable in 
 sip.conf the reply for INVITES that have wrong user information, and 
 also, b) disable any response to any REGISTER packet altogether. Can 
 somebody please write  patch? Or should we go broke trying to stop the 
 flood of criminals coming from abroad?
 Federico
 
 I'm not sure I understand your statement.  Because your customer was hacked
 for $50,000 and your pay was cut in half, it is a result of Digium (or the
 Asterisk project) 'hiding from the real world'?
 
 Your previous point aside, may I ask how your client solved the problem? 
  I'm assuming they are still operating an Asterisk box without the patches
 you have requested.
 
 --
 Paul Belanger
 Digium, Inc. | Software Developer
 twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at:
 http://digium.com  http://asterisk.org
 
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Re: [asterisk-users] My Asterisk Box was hacked

[Robert Huddleston]

When I get hacked I typically run a rootkit checker
http://www.chkrootkit.org/

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chad Wallace
Sent: Thursday, July 21, 2011 2:18 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] My Asterisk Box was hacked

On Thu, 21 Jul 2011 13:29:09 +0800
Malvin Rito mr...@mail.altcladding.com.ph wrote:

 My asterisk box was hacked! Can anyone help on how do I secure my 
 asterisk box, currently my box is installed with 2 NIC. 1st NIC is
 for LAN access and 2nd NIC has a public IP which is registered to our
 VoIP Provider.


Seven Steps to Better SIP Security with Asterisk
http://blogs.digium.com/2009/03/28/sip-security/


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Re: [asterisk-users] Multiple Asterisk Sessions on same machine

[Robert Huddleston]

I prefer

 How do we do that? Isn't Asterisk a SIP Proxy ;)?

That's a good question...

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Tuesday, July 19, 2011 2:18 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Multiple Asterisk Sessions on same machine

On 07/19/2011 01:16 PM, Alex Balashov wrote:
 On 07/19/2011 02:15 PM, Kevin P. Fleming wrote:

 Actually, you can do this with one installation of Asterisk, and a
 separate set of config files and data directories. When the Asterisk
 executable is started, the '-C' option can be used to point to an
 asterisk.conf file; that file can then tell it where all the other
 config files and the data directories are located.

 If you are using one of the init scripts, then yes, that would need to
 be duplicated and modified.

 How, do you suppose, would the complexity of that compare to chrooting
 two installations?

They are probably equal in terms of complexity and effort required; just 
different methods.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] Requires

[Robert Huddleston]

Boy if only it was Enron :)

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent: Monday, July 18, 2011 8:27 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Requires

First they came and said that instead of offices, doors and hallways, 
we should have massive, open-plan seating or grungy, industrial 
cubicle farms, because open spaces mean open companies!

It's safe to say the advice did not fall on deaf ears.  Now, we're 
ready to take openness to the next level.  Is asterisk-users ready to 
be copied on all internal company correspondence?

Challenge accepted.

-- 
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

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Re: [asterisk-users] Requires

[Robert Huddleston]

Alex you are my role model... Next time I'm in Atlanta - let's do lunch!

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent: Monday, July 18, 2011 9:08 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Requires

On 07/18/2011 09:00 AM, Robert Huddleston wrote:

 Boy if only it was Enron :)

Baby steps.  Success is not built overnight; you have to work your way 
up the totem pole of fleecing people.  Start small: persistently ask 
basic, RTFM-grade newbie questions while assigning yourself pompous, 
self-aggrandising titles like Asterisk Engineer.

Keep it up, and you'll be crashing national economies with 
fraudulently constructed multi-billion dollar securitised debt 
tranches in no time.

-- 
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

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Re: [asterisk-users] Requires

[Robert-iPhone]

wrong address - but I can come Monday if you like ;)

Sent from my iPhone

On Jul 16, 2011, at 8:58 AM, mahesh katta maheshka...@flexydial.com wrote:

 Dear Ashirwad,
 
 Please make ready below things for demo in pune .MONDAY needs to be ready for 
 test in our office.
 1. PRI card single span
 2. PRI cable
 3. Server
 4. SIM cards 4 with recharge. 
 
 
 Best Regards, 
 
 Mahesh Katta
 BUZZWORKS Business Services Private Limited
 BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) 
 Mumbai 400069
 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
 Web http://www.buzzworks.com
 
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Re: [asterisk-users] Asterisk binaries on CentOS version 6

[Robert Huddleston]

I stand amused that people want to experiment with VoIP and Asterisk - but
aren't willing to:
( a ) Read wiki / manuals / faqs
( b ) demand packages for their o/s

This ain't windows folks :)

./configure
make
make install

Is really simple :)

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles
Sent: Thursday, July 14, 2011 12:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk binaries on CentOS version 6

On Thursday 14 Jul 2011, Kaushal Shriyan wrote:
 Hi,

 Any time line of availability of Asterisk binaries on CentOS version 6.

Yeah .  as soon as someone compiles them  :)

Really, building packages from source *IS* *NOT* *HARD*, and it doesn't even

take long anymore  (on any target system with the grunt to run Asterisk).  
The only thing to beware of is, if configure complains that you need a 
package that you already have, then you need the corresponding -devel 
package.

Go on, live a little!  Just because you're using CentOS, doesn't mean you
have 
to be boring  ;)

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] skype for asterisk usage in the future

[Robert Rawlinson]

On 07/12/2011 08:26 AM, Kevin P. Fleming wrote:

 It is unknown whether it will continue to be usable after that period;
 Skype has the ability to disable SFA from accessing the Skype network
 if they feel that is what they want to do. Since it won't get any
 updates between now and then, it is very likely to be obsolete (from a
 'Skype protocol' point of view) in two years and it seems quite likely
 that they won't want it accessing the network any more. It would be
 best to plan for it being non-functional after the two year support
 period is over.

Is there a project to replace Skype with a free software?
Bob Rawlinson


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Re: [asterisk-users] CDRs

[Robert Huddleston]

Read the wiki / manuals

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of deeps backup
Sent: Tuesday, July 12, 2011 11:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] CDRs

 

Hi

 

Like we can define cdr field format for csv, is it possible to define if
cdrs are stored in a database?

Also, what will be size limit for database CDR storage ?

 

 

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Re: [asterisk-users] New VirtualBox Beta Has PCI Pass-Through Support

[Robert-iPhone]

+1 for Xen
-1 for VB


Sent from my iPhone

On Jul 8, 2011, at 10:00 PM, Doug Lytle supp...@drdos.info wrote:

 Warren Selby wrote:
 Not trying to start a war here,
 
 
 That may be, but I have experience with VB.
 
 Doug
 
 
 -- 
 Ben Franklin quote:
 
 Those who would give up Essential Liberty to purchase a little Temporary 
 Safety, deserve neither Liberty nor Safety.
 
 
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[asterisk-users] HDLC Overrun with Chan SS7

[Robert Thomas]

Hi,

I'm running an 8E1 setup to an SS7 carrier. The setup works but when we
start hitting the 80 active calls mark the link became unstable.

I found a lot of the following messages afecting my d channel

Jun 25 15:49:55 ostional kernel: [385661.368857] dahdi: HDLC Receiver
overrun on channel TE4/0/1/31 (master=TE4/0/1/31)

Im using TE410P

dahdi-linux-2.4.1.2.tar.gz
dahdi-tools-2.4.1.tar.gz

-- 
Robert
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Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk

[Robert Huddleston]

If memory serves isn't that support contract include broken phones / parts
too?

 

I thought I read that if my phone Is broken - it is covered

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Monday, June 20, 2011 9:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk

 

On Mon, Jun 20, 2011 at 6:10 PM, Robert-iPhone rhuddles...@gmail.com
wrote:

You are supposed to go via cisco and support contract BUT Google is your
friend (JFGI)


The support contract from Cisco is only US $8.99 on CDW

I really hate to link to my own blog, but I do have a post on there that
details how to setup a 79x1 phone using SIP firmware with asterisk.  Click
the link in my signature and go to the Blog and you should be able to easily
find the relevant post.  

-- 
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--Warren Selby, dCAP
http://www.SelbyTech.com

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Re: [asterisk-users] SMS with Asterisk

[Robert Huddleston]

Hahahah Baltimore and SE DC. How about Philly too J

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro
Sent: Tuesday, June 21, 2011 2:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SMS with Asterisk

 

 

On Mon, Jun 20, 2011 at 2:48 PM, Warren Selby wcse...@selbytech.com wrote:

On Mon, Jun 20, 2011 at 3:52 AM, Steve Totaro stot...@asteriskhelpdesk.com
wrote:

Two requests, not from me but the community.

1.  Don't top post


*cough*
 

2.  When you find your solution, reply to this thread so others will be
(silver) spoon fed the answers and blindly accept them without trying things
and going through a learning curve and experimentation when they find your
post in Google.


I hear some people are actually deploying their asterisk solutions in war
zones and are taking heavy fire while they're looking for answers - seems
like it would make their life a whole lot easier (and safer!) if people
posted simple responses on this list when suggestions worked for them...

-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com


LOL at the haters.

1.  It was joke for those with senses of humor and know me (Randy got it),
but I top post when others do.  I bottom post when others do.  I just go
with the flow.  I am not uptight about it.

2.  I have never heard that but it may be true.  

Personally, I have been shot at on top the Iraqi Government building in the
IZ from the Red Zone.  I was setting up and troubleshooting the Motorola
Canopy WiFi system.  Just a few 7.62x39 rounds, nothing I would call heavy
fire.

The only Heavy Fire I took was standing on top of one of the buildings at
the FOB trying to trace a cable and the ricochets from the firing range were
landing all over the place.  That happens when 30 guys are training with AKs
and a T-Wall as the backstop.

I have deployed Asterisk systems in war zones many times, in West African
countries, Iraq, Baltimore and South East DC.  I would certainly seek
shelter/defensive position if there was gun play.  LOL, you can wish
yourself into a gun fight but you cannot wish yourself out. 


It would also be a whole lot easier for someone to physically feed me so my
hands could be free to work in hostile environments, maybe an LN can bring
me a portable toilet and make sure it is fresh, that would make everything
so easy and easy is what we all want.

Heck, I could just set it up at the FOB and then deploy it.

At any rate, I asked the guy to post his success, so I am not sure why you
posted, but thanks.  It only takes 10% truth to make a legend.

Thanks,
Steve T 

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Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk

[Robert-iPhone]

wow I think someone needs to just spend some time reading and playing. Getting 
these phones working is not rocket science and there are similarities with how 
to do firmware / config pushes.

Not to sound mean but RTFM

Sent from my iPhone

On Jun 21, 2011, at 7:45 PM, Warren Selby wcse...@selbytech.com wrote:

 On Tue, Jun 21, 2011 at 5:35 PM, bilal ghayyad bilmar...@yahoo.com wrote:
 Dear Warren;
 
 Please, keep all discussions to the list.  There's no need to email me 
 personally about this. 
 
 snip
  
 cmterm-7942_7962-sip.9-2-1.cop.sgn (which is written that it is SIP IP Phone 
 load) and cmterm-7942_7962-sip.9-2-1.zip which is written that it is SIP IP 
 Phone firmware files only. So what is the difference between them the load 
 and the firmware?
 
 The .sgn file is basically just a zip container that the Cisco Call Manager 
 uses.  You'll want to grab the zip file, extract the contents of the file 
 into your tftp root directory.  The latest firmware that I've used was 8.5.2, 
 in which most everything I needed worked for me.  I don't know specifics 
 about the later versions of Cisco's SIP releases.
  
 Now, when I need to do the upgrade for the Phone, then I have to determine in 
 the xml files the needed firmware?
 
 You should have, at least with firmware 8.5.2, the following files in your 
 tftproot directory after unzipping the zip file:
 
 apps41.8-5-2TH1-9.sbn
 cnu41.8-5-2TH1-9.sbn
 cvm41sip.8-5-2TH1-9.sbn
 dsp41.8-5-2TH1-9.sbn
 jar41sip.8-5-2TH1-9.sbn
 SIP41.8-5-2S.loads
 term41.default.loads
 term61.default.loads
 XMLDefault.cnf.xml
 SEP[_MAC-ADDR_].cnf.xml
 
 I provide samples of the last two files on the blog post mentioned earlier.  
 The last file, that starts with SEP, should contain the actual mac address of 
 the phone you are trying to provision.  So, for example, it would be 
 SEP0003C9DD5624.cnf.xml, if the mac address of your phone was 0003.C9DD.5624. 
  The example files are pretty much all you need, just go through them and 
 change any location specific variables (such as _USER_, _IPADDR_, or 
 _PASSWD_) to the proper values for your environment.
 
 Once you've got your tftp server setup properly with all of the appropriate 
 config files, plug your phone in and follow the instructions at the bottom 
 part of my blog post that explain how to get the phone reflashed to the SIP 
 image and registered to your asterisk server.
 
 
 -- 
 Thanks,
 --Warren Selby, dCAP
 http://www.SelbyTech.com
 
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Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk

[Robert Huddleston]

I'm using the sip firmware.. It's alright.. I feel like I'm not receiving
all the features I should.. But MWI works and multiple call appearance..

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Monday, June 20, 2011 2:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk

 

On Mon, Jun 20, 2011 at 5:38 AM, bilal ghayyad bilmar...@yahoo.com wrote:

Dears;


snip

Have you thought about perhaps just flashing the phones to use the SIP
firmware?

-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com

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Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk

[Robert-iPhone]

You are supposed to go via cisco and support contract BUT Google is your 
friend (JFGI)

Sent from my iPhone

On Jun 20, 2011, at 6:44 PM, bilal ghayyad bilmar...@yahoo.com wrote:

 If I need to use SIP, from where to get the suitable firmware for these Cisco 
 IP Phones 7942G?
 
 Where do u download the SIP firmware usually for your Cisco IP Phones?
 
 Your kindly help is highly appreciated.
 Regards
 Bilal
 
 ---
 
 I'm using the sip firmware.. It's alright.. I feel like I'm
 not receiving
 all the features I should.. But MWI works and multiple call
 appearance..
 
 
 
 On Mon, Jun 20, 2011 at 5:38 AM, bilal ghayyad bilmar...@yahoo.com
 wrote:
 
 Dears;
 
 
 snip
 
 Have you thought about perhaps just flashing the phones to
 use the SIP
 firmware?
 
 -- 
 Thanks,
 

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Re: [asterisk-users] Bridged Digital call

[robert boardman]

both show transfercapability DIGITAL

Regards
Robb

On 16 June 2011 23:40, Richard Mudgett rmudg...@digium.com wrote:

  Hi All
 
  Just upgraded from 1.6? to 1.8.4.1
 
 
  I ised to be able to get a digital call working across a bridged isdn
  channel in 1.6 and 1.4 using the following;-
 
 
  exten = _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?5:)
  exten = _X.,2,dial(DAHDI/g1/${EXTEN})
  exten = _X.,3,Noop(${CHANNEL})
  exten = _X.,4,hangup
  exten = _X.,5,(Set(CHANNEL(transfercapability)=DIGITAL)
  exten = _X.,6,dial(DAHDI/g1/${EXTEN})
  exten = _X.,7,hangup
 
 
  this still dials and aswers in 1.8 but no frames are passed and the
  call times out and drops
 
  I have also tried
 
  exten = _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?5:)
  exten = _X.,2,dial(DAHDI/g1/${EXTEN})
  exten = _X.,3,Noop(${CHANNEL})
  exten = _X.,4,hangup
  exten = _X.,5,Noop
  exten = _X.,6,dial(DAHDI/g1d/${EXTEN})
  exten = _X.,7,hangup
 
  with exactly the same outcome,

 Both of these methods should work after doing a quick look a the code.

 Does the outgoing call SETUP indicate digital capability?

 Richard

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Re: [asterisk-users] Bridged Digital call

[robert boardman]

any reason why this would happen, should I report a bug on the issue
tracker?

Robb

On 17 June 2011 19:55, Richard Mudgett rmudg...@digium.com wrote:

Hi All
   
Just upgraded from 1.6? to 1.8.4.1
   
   
I ised to be able to get a digital call working across a bridged
isdn
channel in 1.6 and 1.4 using the following;-
   
   
exten = _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?5:)
exten = _X.,2,dial(DAHDI/g1/${EXTEN})
exten = _X.,3,Noop(${CHANNEL})
exten = _X.,4,hangup
exten = _X.,5,(Set(CHANNEL(transfercapability)=DIGITAL)
exten = _X.,6,dial(DAHDI/g1/${EXTEN})
exten = _X.,7,hangup
   
   
this still dials and aswers in 1.8 but no frames are passed and the
call times out and drops
   
I have also tried
   
exten = _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?5:)
exten = _X.,2,dial(DAHDI/g1/${EXTEN})
exten = _X.,3,Noop(${CHANNEL})
exten = _X.,4,hangup
exten = _X.,5,Noop
exten = _X.,6,dial(DAHDI/g1d/${EXTEN})
exten = _X.,7,hangup
   
with exactly the same outcome,
  
   Both of these methods should work after doing a quick look a the code.
  
   Does the outgoing call SETUP indicate digital capability?
 
  both show transfercapability DIGITAL

 Could be a problem in the media stream handling not being setup for digital
 mode.

 Richard

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[asterisk-users] Bridged Digital call

[robert boardman]

Hi All

Just upgraded from 1.6? to 1.8.4.1


I ised to be able to get a digital call working across a bridged isdn
channel in 1.6 and 1.4 using the following;-


exten = _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?5:)
exten = _X.,2,dial(DAHDI/g1/${EXTEN})
exten = _X.,3,Noop(${CHANNEL})
exten = _X.,4,hangup
exten = _X.,5,(Set(CHANNEL(transfercapability)=DIGITAL)
exten = _X.,6,dial(DAHDI/g1/${EXTEN})
exten = _X.,7,hangup

this still dials and aswers in 1.8 but no frames are passed and the call
times out and drops

I have also tried

exten = _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?5:)
exten = _X.,2,dial(DAHDI/g1/${EXTEN})
exten = _X.,3,Noop(${CHANNEL})
exten = _X.,4,hangup
exten = _X.,5,Noop
exten = _X.,6,dial(DAHDI/g1d/${EXTEN})
exten = _X.,7,hangup
with exactly the same outcome,

I wondered if I'm missing something in 1.8, has anyone got this working?

Regards

Robb
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[asterisk-users] CDRs in 1.8

[robert boardman]

I'm using ISDN30 for a bridged application

in all the old versions of asterisk the time slot number is shown in the
channels and dstchannel fields of the cdr

I understand this has chaned in 1.8,is there a way of getting the time slot
information stored somewhere at the end of the call so this can be
interigated?

Thanks in advance

Regards

Robb
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[asterisk-users] Ground Start ATA / VOIP Gateway

[Robert Huddleston]

Anyone have recommendations for a gateway / ATA for business that can do
GroundStart? Preferably with an rj-21 - but okay if not..

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Re: [asterisk-users] [asterisk-biz] Ground Start ATA / VOIP Gateway

[Robert Huddleston]

I only need 4 fxs. I looked at the IAD2431 but it uses T1/E1 as WAN. If I
could assign Fast Ethernet to WAN that would be great. Budget is not that
great

 

From: asterisk-biz-boun...@lists.digium.com
[mailto:asterisk-biz-boun...@lists.digium.com] On Behalf Of Sum Ding Wong
Sent: Tuesday, June 14, 2011 3:23 PM
To: Commercial and Business-Oriented Asterisk Discussion
Cc: asterisk-users@lists.digium.com
Subject: Re: [asterisk-biz] Ground Start ATA / VOIP Gateway

 

Cisco Gateways can do ground start signaling. What is your budget and port
density need?

On Tue, Jun 14, 2011 at 1:19 PM, Robert Huddleston rhuddles...@gmail.com
wrote:

Anyone have recommendations for a gateway / ATA for business that can do
GroundStart? Preferably with an rj-21 - but okay if not..


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Re: [asterisk-users] Ground Start ATA / VOIP Gateway

[Robert Huddleston]

Ya - customer is on a nice NEC SV8100.. The card is a ground start card..
they are currently being fed by a Cisco IAD2431 w/ RJ-21 punchdown
cross-connect.

 

But that IAD2431 uses T1/E1 as WAN.. They are doing away with the T1 and
want to use Ethernet for wan.

 

So IAD2431 would be great - but if it only allows T1/E1 for WAN - I'm shot.

 

From: John Novack [mailto:jnov...@stromberg-carlson.org] 
Sent: Tuesday, June 14, 2011 3:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Robert Huddleston
Subject: Re: [asterisk-users] Ground Start ATA / VOIP Gateway

 


Robert Huddleston wrote: 

Anyone have recommendations for a gateway / ATA for business that can do
GroundStart? Preferably with an rj-21 - but okay if not..

 


I don't know of any ATA that will do GS
An RJ-21 is the designation for a 66 block with 25 pair connector on the
side
GS is available with many channel banks though a T1 card and channel bank
might be overkill for your application.
Is this to go into a legacy switch?
Most have line cards that can be easily switched to Loop 

Is this in the US, or ???
John Novack





-- 
 
Dog is my Co-pilot
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Re: [asterisk-users] Ground Start ATA / VOIP Gateway

[Robert Huddleston]

I'll have to look at that then - as I thought the card actually said Ground
Start on it.. I may have missed or it was scratched off the word loop start

 

From: John Novack [mailto:jnov...@stromberg-carlson.org] 
Sent: Tuesday, June 14, 2011 5:20 PM
To: Robert Huddleston
Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Ground Start ATA / VOIP Gateway

 

The SV8100 can do either ground or loop
Assuming you can access the system it can easily be changed.

Programming manual here:

http://www.telecomcepts.com/downloads/SV8100/SV8100 Programming Manual_1.pdf

the original installer may have locked it down, but it CAN be changed.

John Novack


Robert Huddleston wrote: 

Ya - customer is on a nice NEC SV8100.. The card is a ground start card..
they are currently being fed by a Cisco IAD2431 w/ RJ-21 punchdown
cross-connect.

 

But that IAD2431 uses T1/E1 as WAN.. They are doing away with the T1 and
want to use Ethernet for wan.

 

So IAD2431 would be great - but if it only allows T1/E1 for WAN - I'm shot.

 

From: John Novack [mailto:jnov...@stromberg-carlson.org] 
Sent: Tuesday, June 14, 2011 3:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Robert Huddleston
Subject: Re: [asterisk-users] Ground Start ATA / VOIP Gateway

 


Robert Huddleston wrote: 

Anyone have recommendations for a gateway / ATA for business that can do
GroundStart? Preferably with an rj-21 - but okay if not..

 


I don't know of any ATA that will do GS
An RJ-21 is the designation for a 66 block with 25 pair connector on the
side
GS is available with many channel banks though a T1 card and channel bank
might be overkill for your application.
Is this to go into a legacy switch?
Most have line cards that can be easily switched to Loop 

Is this in the US, or ???
John Novack






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Dog is my Co-pilot





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Re: [asterisk-users] Ground Start ATA / VOIP Gateway

[Robert-iPhone]

considering providing the sip trunking nyself via asterisk.
the sip trunking looks expensive - card and licenses from nec.


Sent from my iPhone

On Jun 14, 2011, at 6:06 PM, John Novack jnov...@stromberg-carlson.org wrote:

 that system can also handle IP trunks, though the equipment might not be 
 available to you or outside your budget window
 
 How does this relate to Asterisk, or does it?
 
 John Novack
 
 
 Robert Huddleston wrote:
 
 I’ll have to look at that then – as I thought the card actually said “Ground 
 Start” on it.. I may have missed or it was scratched off the word loop start
  
 From: John Novack [mailto:jnov...@stromberg-carlson.org] 
 Sent: Tuesday, June 14, 2011 5:20 PM
 To: Robert Huddleston
 Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Ground Start ATA / VOIP Gateway
  
 The SV8100 can do either ground or loop
 Assuming you can access the system it can easily be changed.
 
 Programming manual here:
 
 http://www.telecomcepts.com/downloads/SV8100/SV8100 Programming Manual_1.pdf
 
 the original installer may have locked it down, but it CAN be changed.
 
 John Novack
 
 
 Robert Huddleston wrote:
 Ya – customer is on a nice NEC SV8100.. The card is a ground start card.. 
 they are currently being fed by a Cisco IAD2431 w/ RJ-21 punchdown 
 cross-connect.
  
 But that IAD2431 uses T1/E1 as WAN.. They are doing away with the T1 and 
 want to use Ethernet for wan…
  
 So IAD2431 would be great – but if it only allows T1/E1 for WAN – I’m shot.
  
 From: John Novack [mailto:jnov...@stromberg-carlson.org] 
 Sent: Tuesday, June 14, 2011 3:47 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Cc: Robert Huddleston
 Subject: Re: [asterisk-users] Ground Start ATA / VOIP Gateway
  
 
 Robert Huddleston wrote:
 Anyone have recommendations for a gateway / ATA for business that can do 
 GroundStart? Preferably with an rj-21 – but okay if not..
  
 
 I don't know of any ATA that will do GS
 An RJ-21 is the designation for a 66 block with 25 pair connector on the side
 GS is available with many channel banks though a T1 card and channel bank 
 might be overkill for your application.
 Is this to go into a legacy switch?
 Most have line cards that can be easily switched to Loop 
 
 Is this in the US, or ???
 John Novack
 
 
 
 
 -- 
  
 Dog is my Co-pilot
 
 
 -- 
  
 Dog is my Co-pilot
 
 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Ground Start ATA / VOIP Gateway

[Robert-iPhone]

exactly my other concern - can just drop sip card in and put on the net - would 
also have to get an sbc - which would be more than an ATA.
considering just using a cisco router (low end XM) and throwing a high density 
voice card in it

Sent from my iPhone

On Jun 14, 2011, at 6:48 PM, John Novack jnov...@stromberg-carlson.org wrote:

 Agreed NEC isn't cheap. Their products are generally pretty good and robust 
 though. I have an earlier one still working for 18 years and counting
 Of course, when one considers the asterisk machine, configuration time, 
 firewall and the rise in sip hacking  sip trunking can easily turn into a 
 PITA.
 
 John Novack
 
 
 Robert-iPhone wrote:
 
 considering providing the sip trunking nyself via asterisk.
 the sip trunking looks expensive - card and licenses from nec.
 
 
 Sent from my iPhone
 
 On Jun 14, 2011, at 6:06 PM, John Novack jnov...@stromberg-carlson.org 
 wrote:
 
 that system can also handle IP trunks, though the equipment might not be 
 available to you or outside your budget window
 
 How does this relate to Asterisk, or does it?
 
 John Novack
 
 
 Robert Huddleston wrote:
 
 I’ll have to look at that then – as I thought the card actually said 
 “Ground Start” on it.. I may have missed or it was scratched off the word 
 loop start
  
 From: John Novack [mailto:jnov...@stromberg-carlson.org] 
 Sent: Tuesday, June 14, 2011 5:20 PM
 To: Robert Huddleston
 Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Ground Start ATA / VOIP Gateway
  
 The SV8100 can do either ground or loop
 Assuming you can access the system it can easily be changed.
 
 Programming manual here:
 
 http://www.telecomcepts.com/downloads/SV8100/SV8100 Programming 
 Manual_1.pdf
 
 the original installer may have locked it down, but it CAN be changed.
 
 John Novack
 
 
 Robert Huddleston wrote:
 Ya – customer is on a nice NEC SV8100.. The card is a ground start card.. 
 they are currently being fed by a Cisco IAD2431 w/ RJ-21 punchdown 
 cross-connect.
  
 But that IAD2431 uses T1/E1 as WAN.. They are doing away with the T1 and 
 want to use Ethernet for wan…
  
 So IAD2431 would be great – but if it only allows T1/E1 for WAN – I’m shot.
  
 From: John Novack [mailto:jnov...@stromberg-carlson.org] 
 Sent: Tuesday, June 14, 2011 3:47 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Cc: Robert Huddleston
 Subject: Re: [asterisk-users] Ground Start ATA / VOIP Gateway
  
 
 Robert Huddleston wrote:
 Anyone have recommendations for a gateway / ATA for business that can do 
 GroundStart? Preferably with an rj-21 – but okay if not..
  
 
 I don't know of any ATA that will do GS
 An RJ-21 is the designation for a 66 block with 25 pair connector on the 
 side
 GS is available with many channel banks though a T1 card and channel bank 
 might be overkill for your application.
 Is this to go into a legacy switch?
 Most have line cards that can be easily switched to Loop 
 
 Is this in the US, or ???
 John Novack
 
 
 
 
 -- 
  
 Dog is my Co-pilot
 
 
 -- 
  
 Dog is my Co-pilot
 
 --
 _
 -- 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
 
 -- 
 
 Dog is my Co-pilot
 
 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
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 -- 
 
 Dog is my Co-pilot
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_
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Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk

[Robert-iPhone]

I also had trouble w/ these phones at first. There was a DHCP option (?81?) 
you'll have to google it.
The phones would not talk to tftp until I set dhcp option.
The console aux cable is easy to build and VERY useful


Sent from my iPhone

On Jun 13, 2011, at 8:31 PM, Mark Engelhardt ma...@intuitiveengineering.com 
wrote:

 Bilal,
 
 I suggest you turn on logging on your tftp server to see what files are 
 actually being requested, and if the the tftp server is dishing them out... 
 Try adding a few v's to your tftp setup:
 
 File: /etc/xinetd.d/tftp
 Line to change: server_args = -s /tftpboot -v -v -v
 
 Look in /var/log/messages for the output. 
 
 Also, I believe your 7942G has a console/aux port which is a serial port, you 
 can learn what is happening as the phone boots up with that too. 
 
 Good Luck! 
 
 Mark
 
 
 On Jun 13, 2011, at 3:02 PM, bilal ghayyad wrote:
 
 Dears;
 
 The Asterisk version is 1.8.3.2
 
 The Cisco IP Phone is 7942G and it is running now skinny.
 
 The used TFTP is tftp-server which is installed in fedora.
 
 I placed the following two files (but look like it was not taken from the 
 TFTP, as nothing appeared in the messages), but I am able to to ping from 
 the asterisk box to the vlan that the Phone is connected, so no problem in 
 the reachability:
 
 
 SEPB8BEBF22AB62.cnf.xml
 xmlDefault.CNF.XML
 
 Are the files name correct? Or the Cisco IP Phone 7942G are not working fine 
 with Asterisk or the tftp-server?
 
 Regards
 Bilal
 
 
 
 Hi All;
 
 Can anyone advise if using Cisco IP Phones
 
 Which model(s) are you planning to use ?
 
 
 in skinny protocol is fine or not? Or it is better to
 use it in SIP
 protocol?
 
 
 --
 
 Hi,
 
 On 06/13/2011 01:04 PM, bilal ghayyad wrote:
 Can anyone advise if using Cisco IP Phones in skinny
 protocol is fine or not? Or it is better to use it in SIP
 protocol?
 
 SCCP works better than SIP in my opinion as there are more
 features.
 Check out http://chan-sccp-b.sourceforge.net/
 
 
 
 
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[asterisk-users] Obtain SIP From and To Tag for CDR

[Robert Thomas]

List,

I'm trying to obtain the Call ID, From tag and To tag of the SIP calls from
Asterisk, to store them on a CDR and be able to conciliate with another CDR
system ( opensips )

I have been able to obtain the SIP Call ID CDR(sip_callid)=${SIPCALLID}

I was wondering if there's any way to obtain the From and to tag.

Asterisk 1.6.2.9 and 1.8.5rc1

Thanks,
-- 
Robert
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[asterisk-users] Asterisk port 5000 open

[Robert Thomas]

Hi,

I have been trying to find out what module is causing asterisk to open port
5000

I have already disabled some ( sccp, mgcp, iax and other modules ) since I
only want sip port opened


/etc/asterisk# netstat -aln --programs  | grep asterisk
tcp0  X.X.X.X:5060 0.0.0.0:*   LISTEN
 22523/asterisk
udp0 X.X.X.X:5000  0.0.0.0:*
22523/asterisk
udp0 X.X.X.X:5060 0.0.0.0:*
22523/asterisk

I have port 5000 blocked with IP tables, but would like better to understand
what is it for.

Not sure if there's a list of known ports used by asterisk.

Thanks,

-- 
Robert
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Re: [asterisk-users] Failover Routing

[Robert Thomas]

What value do you get from the hangup cause, are they different?

I think  can you use a gotoif checking the hangup cause.

On Wed, Mar 2, 2011 at 12:43 PM, Tilghman Lesher tilgh...@meg.abyt.eswrote:

 On Wednesday 02 March 2011 07:06:31 Andrew Thomas wrote:
  It seems like it is a v1.8 only function at present (unless a backport
  is released).
 
  From http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
 
  -
  Asterisk 1.8 will allow to read SIP response codes in the dialplan via
 
   ${HASH(SIP_CAUSE,channel-name)}
 
  Asterisk 1.8 also comes with a 'use_q850_reason' configuration option
  for generating and parsing, if available: -
 
  That will give you what you want if you consider upgrading to v1.8.

 A backport on this is not possible.  It depends upon some core
 functionality introduced in the 1.8 branch.

 --
 Tilghman

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-- 
Robert
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