Re: [asterisk-users] Mailing List Future

[John Novack]

Frank Vanoni wrote:

On Mon, 2023-12-04 at 08:00 -0400, Joshua C. Colp wrote:


To that end, we’ve decided to discontinue the mailing lists effective
February 1st, 2024.

That's a very sad news! :-(


Agree. Yet another giant step backward.
Interesting that they will continue to send e-mails when postings to the (UGH) 
forum happen though.

John Novack



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[asterisk-users] voip.ms ( was Re: Problems solved )

[John Novack]

aster...@phreaknet.org wrote:

voip.ms is also the only major VoIP provider that supports IAX2, so if you do 
anything else you'll probably have to use SIP.


voip.ms works so well and is certainly affordable, so why would anyone want to 
use anything else?

Even my really cheap friends use it!
I have used it as my PSTN provider for more than 10 years, with only one 
hacking issue with voip.ms, which they fixed fairly quickly. I see no reason to 
change to a protocol that ( it seems )
every thief in the world is banging away on 24/7!!

JMO

John Novack


On 5/27/2023 10:23 AM, Steve Matzura wrote:


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Re: [asterisk-users] Global variables in global variables

[John Novack]

You have posted the same message several times in the last few days!!

I would assume no one has an answer to your question, at least on this list.
It seems most have migrated to another (UGH!) venue, so the few that are left 
can't help.

JMO

John Novack


Antony Stone wrote:

Hi.

I have a very old dialplan (ie: a dialplan for a very old version of Asterisk)
which I've just transferred to Asterisk 16.28.0

The [globals] section of that dialplan includes:

Kphones=SIP/KC470IP/KSnom870
Sphones=SIP/SYealinkT38G/SGC610IP
Allphones=${Kphones}&${Sphones}

In the old system, this results in ${Allphones} containing:

SIP/KC470IP/KSnom870/SYealinkT38G/SGC610IP

I can use this in a dial() command.

On the new system, the variable ${Allphones} ends up containing:

${Kphones}&${Sphones}

(ie: the unexpanded variable names, not the content of those previously-
defined variables.)

This fairly obviously does not work in a dial() command.


a) is this a deliberate backward incompatiblity at some stage in the
development of Asterisk?

b) if not, is this a known bug?

c) is there some other way I'm supposed to be doing this now, to be able to
define a global variable including the value of another global variable?

d) if not, is there some workaround?


Thanks,


Antony.



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Re: [asterisk-users] I found an excellent guide: Configure Asterisk VoIP IP PBX SIP Server with Cisco IP Phones

[John Novack]

The 7940 has 2 line buttons, the 7960 has 6 buttons

There are other versions similar, including the 7970 with a color screen

Not really sure you will have access to the SIP firmware without some 
relationship with Cisco.

Better check before poring money down a rat hole!

Also, you will need a TFTP server working on your Asterisk box


Good luck


JN



Turritopsis Dohrnii Teo En Ming wrote:

I am planning to get a cheap and used Cisco 7960 IP phone for testing. The 
excellent guide I found says that the configuration and setup is extremely well 
documented for this model of Cisco IP phone. I also read that I can google 
search for SIP firmware and download them?

Is Cisco 7960 better and more advanced than Cisco 7940?

On 2020-12-18 22:41, John Novack wrote:

When purchasing these phones, make sure they are SIP, as these were
available with several different firmware loads

You may end up with one used with Call Manager, and struggle to get
the SIP firmware for it.

In addition there were several versions of SIP firmware.

None of the code will be available to you without proper credentials
from Cisco

John Novack

Turritopsis Dohrnii Teo En Ming wrote:


Subject: I found an excellent guide: Configure Asterisk VoIP IP PBX
SIP Server with Cisco IP Phones

Good day from Singapore,

Today 18 December 2020 Friday, I found an excellent guide on
configuring Asterisk VoIP IP PBX SIP server with Cisco IP phones. I
think the author explains very well and very clearly. The guide is
certainly very detailed and well written.

Title of Guide: Configure Asterisk with Cisco IP Phones
Author: Tyler Winfield
Link: http://docshare02.docshare.tips/files/6706/67061980.pdf
Original website: http://minded.ca/ (no longer accessible)

I am going to buy a cheap and used Cisco 7940 or 7960 IP phone for
about SGD$20 and configure it to work with my FreePBX 15 and
Asterisk 16 PBX appliance by following this guide. I will provide
feedback and my own custom guide after I have done so.

I just want to share this very excellent guide.

Another excellent website is https://www.voip-info.org/

On the front page, it says:

"Welcome to VOIP-info: a reference guide to all things VOIP."

Sharing is caring.

Thank you.

Mr. Turritopsis Dohrnii Teo En Ming, 42 years old as of 18 December
2020 Friday, is a TARGETED INDIVIDUAL (TI) living in Singapore. He
is presently an IT consultant with a System Integrator (SI)/computer
firm in Singapore. He is an IT enthusiast.


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Re: [asterisk-users] I found an excellent guide: Configure Asterisk VoIP IP PBX SIP Server with Cisco IP Phones

[John Novack]

When purchasing these phones, make sure they are SIP, as these were available 
with several different firmware loads

You may end up with one used with Call Manager, and struggle to get the SIP 
firmware for it.

In addition there were several versions of SIP firmware.

None of the code will be available to you without proper credentials from Cisco


John Novack


Turritopsis Dohrnii Teo En Ming wrote:

Subject: I found an excellent guide: Configure Asterisk VoIP IP PBX SIP Server 
with Cisco IP Phones

Good day from Singapore,

Today 18 December 2020 Friday, I found an excellent guide on configuring 
Asterisk VoIP IP PBX SIP server with Cisco IP phones. I think the author 
explains very well and very clearly. The guide is certainly very detailed and 
well written.

Title of Guide: Configure Asterisk with Cisco IP Phones
Author: Tyler Winfield
Link: http://docshare02.docshare.tips/files/6706/67061980.pdf
Original website: http://minded.ca/ (no longer accessible)

I am going to buy a cheap and used Cisco 7940 or 7960 IP phone for about SGD$20 
and configure it to work with my FreePBX 15 and Asterisk 16 PBX appliance by 
following this guide. I will provide feedback and my own custom guide after I 
have done so.

I just want to share this very excellent guide.

Another excellent website is https://www.voip-info.org/

On the front page, it says:

"Welcome to VOIP-info: a reference guide to all things VOIP."

Sharing is caring.

Thank you.

Mr. Turritopsis Dohrnii Teo En Ming, 42 years old as of 18 December 2020 
Friday, is a TARGETED INDIVIDUAL (TI) living in Singapore. He is presently an 
IT consultant with a System Integrator (SI)/computer firm in Singapore. He is 
an IT enthusiast.









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Re: [asterisk-users] Fwd: Legacy TDM400

[John Novack SCII_U]

AFAIK it requires DAHDI version 2
For unknown reasons, this and many other card drivers were removed in DAHDI 
version 3

Suggest you compile from source, rather than any repository, selecting the last DAHDI version 2 and at least Asterisk 13, though it is EOL or nearly, it still is a good version to 
work with

For learning there isn't any good reason to have the latest of anything

I have a working version of Asterisk 13 with DAHDI and a 4 port T1 card on 
CentOS 6, and support a buddy with a TDM 400 or 410 - no issues

YMMV

John Novack

Roy Kidder wrote:


Hello all,

It's been quite some number of years since I played around with Asterisk and 
I'm just now getting back into it. I think the last version I worked with was 
1.8.

I have a legacy Digium TDM400 PCI card and am wondering if that will still work on newer versions of Asterisk. My initial attempts on Debian 10 and the Debian repository version 
of Asterisk didn't get me very far.


Any pointers would be appreciated.

-Roy




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Re: [asterisk-users] How to DIY/Setup An Open Source IP PBX Appliance/Server?

[John Novack]

JMO

AstLinux installed on an HP Thin Client is a good choice for someone with 
limited knowledge of Linux who wants a less steep learning curve.

YMMV


John Novack


Turritopsis Dohrnii Teo En Ming wrote:

Subject: How to DIY/Setup An Open Source IP PBX Appliance/Server?

Good day from Singapore,

After reading recent reviews, I gather that Asterisk is the gold standard when 
it comes to open source VoIP systems and it is the most famous open source PBX 
out there.

Article: Compare the Top 10 Best Open Source PBX Software of 2020
Link: https://www.voipreview.org/business-voip/best-open-source-pbx-software

Article: Top 10 Free Open Source PBX Software Solutions
Link: https://getvoip.com/blog/2016/09/23/best-open-source-pbx-software/

The following is an excerpt from Wikipedia:

"Asterisk is a core component in many commercial products and open-source 
projects. Some of the commercial products are hardware and software bundles, for 
which the manufacturer supports and releases the software with an open-source 
distribution model.

AskoziaPBX, a fork of the m0n0wall project, uses Asterisk PBX software to 
realize all telephony functions.

AstLinux is a "Network Appliance for Communications" open-source software 
distribution.[15]

FreePBX, an open-source graphical user interface, bundles Asterisk as the core 
of its FreePBX Distro[16]

LinuxMCE bundles Asterisk to provide telephony; there is also an embedded 
version of Asterisk for OpenWrt routers.

PBX in a Flash/Incredible PBX and trixbox are software PBXes based on Asterisk.

Elastix previously used Asterisk, HylaFAX, Openfire and Postfix to offer PBX, 
fax, instant messaging and email functions, respectively, before switching to 
3CX.

Issabel is an open-source Unified Communications software which uses Asterisk 
for telephony functions. It was forked from the open-source versions of Elastix 
when 3CX acquired it.

*astTECS uses Asterisk in its VoIP and mobile gateways."

Link: https://en.wikipedia.org/wiki/Asterisk_(PBX)

I would like to DIY/setup an IP PBX appliance/server using free open source 
projects.
Which free open source project, mentioned in the list and links above, would 
you recommend to DIY my IP PBX appliance/server?

Should I buy a desktop computer or get one of those appliances listed in the 
link below to serve as my IP PBX appliance/server?

Link: 
https://www.lazada.sg/products/pfsense-iron-metal-case-fanless-intel-celeron-j1800-dual-core-mini-pc-firewall-soft-router-with-ddr3l-msata-ssd-4-gigabit-lan-rj45-com-port-i449270007-s1196780479.html?spm=a2o42.searchlist.list.89.100857d22PjCYx=1

Please also refer me to very good, detailed and well explained 
guides/tutorials/manuals on setting up open source IP PBX appliances/servers.

Lastly, please recommend a cheap and affordable IP phone (suggest brand and 
model) to go along with my DIY open source IP PBX appliance/server.

Mr. Turritopsis Dohrnii Teo En Ming, 42 years as of 1st December 2020 Tuesday, 
is a TARGETED INDIVIDUAL (TI) living in Singapore.

Thank you very much.








-BEGIN EMAIL SIGNATURE-

The Gospel for all Targeted Individuals (TIs):

[The New York Times] Microwave Weapons Are Prime Suspect in Ills of
U.S. Embassy Workers

Link: 
https://www.nytimes.com/2018/09/01/science/sonic-attack-cuba-microwave.html



Singaporean Targeted Individual Mr. Turritopsis Dohrnii Teo En Ming's Academic
Qualifications as at 14 Feb 2019 and refugee seeking attempts at the United 
Nations Refugee Agency Bangkok (21 Mar 2017), in Taiwan (5 Aug 2019) and 
Australia (25 Dec 2019 to 9 Jan 2020):

[1] https://tdtemcerts.wordpress.com/

[2] https://tdtemcerts.blogspot.sg/

[3] https://www.scribd.com/user/270125049/Teo-En-Ming

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Re: [asterisk-users] svnview.digium.com down?

[John Novack]

Works for me from Comcast!


John Novack



Doug Lytle wrote:

I'm currently reviewing the Digium wiki on migrating from chan_sip to res_pjip 
and I'm trying to access the script that is provided to help with conversion.

https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip

It would appear that said server hosting the script is no responding or the 
link is no longer valid.

Doug



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Re: [asterisk-users] error compiling dahdi for recent kernels

[John Novack]

Too bad.

LOTS of users will still want to continue to use these cards, example the OP!


Good news it probably suppresses prices on used cards!


John Novack



Malcolm Davenport wrote:

Howdy,

That is correct.

The list of supported cards is in the README file (not the -complete package 
README, but the dahdi-linux README)

Cheers

On Thu, Jun 6, 2019 at 2:52 PM John Novack SCII_U mailto:jnov...@comcast.net>> wrote:

Doesn't DAHDI 3.0 remove support for a bunch of older cards, including the 
TDM400 and 410?


    John Novack



Greg Woods wrote:



On Thu, Jun 6, 2019 at 12:17 PM Malcolm Davenport mailto:malco...@sangoma.com>> wrote:

Howdy,

There's a dahdi-linux-complete-3.1.0-rc1+3.1.0-rc1.tar.gz.

Try that.


I noticed that was there, but I didn't try it originally because it's 
obviously a beta version. However, I did download it and try it. It does 
compile, but doesn't work correctly. For one thing, it thinks my Digium card is 
an Ethernet controller:

# lspci | grep Digium
07:01.0 Ethernet controller: Digium, Inc. Wildcard TDM410 4-port analog 
card (rev 11)

Attempting to start the dahdi service results in:

Short version:Jun 06 13:11:38 worldsys.gregandeva.net 
<http://worldsys.gregandeva.net> sh[1026]: using 
'/etc/dahdi/assigned-spans.conf'
Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net> 
sh[1026]: DAHDI_CHANCONFIG failed on channel 1: Invalid argument (22)
Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net> 
sh[1026]: Selected signaling not supported
Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net> 
sh[1026]: Possible causes:
Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net> 
sh[1026]:         FXO signaling is being used on a FXO interface (use a FXS signaling 
variant)
Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net> 
sh[1026]:         RBS signaling is being used on a E1 CCS span
Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net> 
sh[1026]:         Signaling is being assigned to channel 16 of an E1 CAS span
Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net> 
systemd[1]: dahdi.service: Main process exited, code=exited, status=1/FAILURE
Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net> 
systemd[1]: dahdi.service: Failed with result 'exit-code'.
Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net> 
systemd[1]: Failed to start The DAHDI drivers allow you to use your linux computer to 
accept incoming data and voice interfaces.

(The assigned-spans.conf file has nothing in it but comments)

Long version:
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> 
asterisk[1333]: [Jun  6 13:11:42] WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable 
to specify channel 1: Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> 
asterisk[1333]: [Jun  6 13:11:42] ERROR[1333]: chan_dahdi.c:12002 mkintf: Unable to 
open channel 1: Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> 
asterisk[1333]: [Jun  6 13:11:42] ERROR[1333]: chan_dahdi.c:17367 build_channels: 
Unable to register channel '1'
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> 
asterisk[1333]: [Jun  6 13:11:42] WARNING[1333]: chan_dahdi.c:17574 process_dahdi: 
Channel '1' failure ignored: ignore_failed_channels.
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> 
asterisk[1333]: [Jun  6 13:11:42] WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable 
to specify channel 2: Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> 
asterisk[1333]: [Jun  6 13:11:42] ERROR[1333]: chan_dahdi.c:12002 mkintf: Unable to 
open channel 2: Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> 
asterisk[1333]: [Jun  6 13:11:42] ERROR[1333]: chan_dahdi.c:17367 build_channels: 
Unable to register channel '2'
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> 
asterisk[1333]: [Jun  6 13:11:42] WARNING[1333]: chan_dahdi.c:17574 process_dahdi: 
Channel '2' failure ignored: ignore_failed_channels.
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> 
asterisk[1333]: [Jun  6 13:11:42] WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable 
to specify channel 3: Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> 
asterisk[1333]: [Jun  6 13:11:42] ERROR[1333]: chan_dahdi.c:12002 mkintf: Unable to 
open channel 3: Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net&

Re: [asterisk-users] error compiling dahdi for recent kernels

[John Novack SCII_U]

Doesn't DAHDI 3.0 remove support for a bunch of older cards, including the 
TDM400 and 410?


John Novack



Greg Woods wrote:



On Thu, Jun 6, 2019 at 12:17 PM Malcolm Davenport mailto:malco...@sangoma.com>> wrote:

Howdy,

There's a dahdi-linux-complete-3.1.0-rc1+3.1.0-rc1.tar.gz.

Try that.


I noticed that was there, but I didn't try it originally because it's obviously a beta version. However, I did download it and try it. It does compile, but doesn't work 
correctly. For one thing, it thinks my Digium card is an Ethernet controller:


# lspci | grep Digium
07:01.0 Ethernet controller: Digium, Inc. Wildcard TDM410 4-port analog card 
(rev 11)

Attempting to start the dahdi service results in:

Short version:Jun 06 13:11:38 worldsys.gregandeva.net 
<http://worldsys.gregandeva.net> sh[1026]: using 
'/etc/dahdi/assigned-spans.conf'
Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net> 
sh[1026]: DAHDI_CHANCONFIG failed on channel 1: Invalid argument (22)
Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net> 
sh[1026]: Selected signaling not supported
Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net> 
sh[1026]: Possible causes:
Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net> 
sh[1026]:         FXO signaling is being used on a FXO interface (use a FXS signaling 
variant)
Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net> 
sh[1026]:         RBS signaling is being used on a E1 CCS span
Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net> 
sh[1026]:         Signaling is being assigned to channel 16 of an E1 CAS span
Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net> 
systemd[1]: dahdi.service: Main process exited, code=exited, status=1/FAILURE
Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net> 
systemd[1]: dahdi.service: Failed with result 'exit-code'.
Jun 06 13:11:38 worldsys.gregandeva.net <http://worldsys.gregandeva.net> systemd[1]: Failed to start The DAHDI drivers allow you to use your linux computer to accept incoming 
data and voice interfaces.


(The assigned-spans.conf file has nothing in it but comments)

Long version:
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> asterisk[1333]: [Jun  6 13:11:42] WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable to specify channel 
1: Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> asterisk[1333]: [Jun  6 13:11:42] ERROR[1333]: chan_dahdi.c:12002 mkintf: Unable to open channel 1: 
Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> asterisk[1333]: [Jun  6 13:11:42] ERROR[1333]: chan_dahdi.c:17367 build_channels: Unable to register 
channel '1'
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> asterisk[1333]: [Jun  6 13:11:42] WARNING[1333]: chan_dahdi.c:17574 process_dahdi: Channel '1' failure 
ignored: ignore_failed_channels.
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> asterisk[1333]: [Jun  6 13:11:42] WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable to specify channel 
2: Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> asterisk[1333]: [Jun  6 13:11:42] ERROR[1333]: chan_dahdi.c:12002 mkintf: Unable to open channel 2: 
Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> asterisk[1333]: [Jun  6 13:11:42] ERROR[1333]: chan_dahdi.c:17367 build_channels: Unable to register 
channel '2'
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> asterisk[1333]: [Jun  6 13:11:42] WARNING[1333]: chan_dahdi.c:17574 process_dahdi: Channel '2' failure 
ignored: ignore_failed_channels.
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> asterisk[1333]: [Jun  6 13:11:42] WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable to specify channel 
3: Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> asterisk[1333]: [Jun  6 13:11:42] ERROR[1333]: chan_dahdi.c:12002 mkintf: Unable to open channel 3: 
Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> asterisk[1333]: [Jun  6 13:11:42] ERROR[1333]: chan_dahdi.c:17367 build_channels: Unable to register 
channel '3'
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> asterisk[1333]: [Jun  6 13:11:42] WARNING[1333]: chan_dahdi.c:17574 process_dahdi: Channel '3' failure 
ignored: ignore_failed_channels.
Jun 06 13:11:42 worldsys.gregandeva.net <http://worldsys.gregandeva.net> asterisk[1333]: [Jun  6 13:11:42] WARNING[1333]: chan_dahdi.c:4058 dahdi_open: Unable to specify channel 
4: Invalid argument
Jun 06 13:11:42 worldsys.gregandeva.net <ht

Re: [asterisk-users] Paging systems?

[John Novack]

All you need then is a loop start trunk port. Any SIP ATA designed to connect 
to an ANALOG PSTN line.

The doc you sent doesn't show any rj11, so someone probably installed one to 
the first pair out of the 25 pair cable, blue/white

If you were to plug any old phone into that you should at least hear a live 
connection and dial tone from the box

You will need to send dtmf tones to the box for specific zones, or all call. That may be 
one reason you think but don't get an "auto answer" IF you are using an FXO ATA 
then it is sitting there waiting for some DTMF digits 1-9 for the zones, and zero for all 
call

This is also a "talk back" system. Does the customer expect to use the talk 
back feature?

You will need to put that into your dialplan

All easy to do in Asterisk

Similar but quite different than what I have. I announce incoming call numbers 
through the system, as well as quarterly time announcements through call files 
and a cron job. I use an FXO port off a channel bank, but an ATA with an FXO 
port should do the trick


Enjoy

John Novack

Michael Munger wrote:

Excellent point.

This is it: https://www.valcom.com/pdf/v-1109rthf.pdf

Get Outlook for Android <https://aka.ms/ghei36>




On Thu, Mar 21, 2019 at 7:22 PM -0400, "John Novack" mailto:jnov...@comcast.net>> wrote:



Michael Munger wrote:


Does anyone have an (overhead) paging system that they like that works with 
SIP?

We’ve got a client with an old paging system that (supposedly) just takes 
an rj11 POTS connection, but when we put an SPA Cisco adapter on it, it doesn’t 
auto-answer the call, so paging never happens.



Does it expect to see a POTS line with battery on it?
Then a Cisco or other ATA that would work to supply service to a POTS phone 
should work
OR:
Does it expect to see a POTS connection from a PBX trunk, and supply 
battery TO the trunk?
Then you would need a Cisco or other ATA with an FXO connection.

Both types of paging systems have been made and both styles of connections 
have existed through the last 30 + years, and since you haven't revealed the 
brand and model of paging system it makes troubleshooting difficult.
Using the existing system can be made to work
I use a very old Harris PagePak VS that was used with a Western Electric 
Horizon system back in the dark ages with Asterisk

John Novack

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Re: [asterisk-users] Paging systems?

[John Novack]

Michael Munger wrote:


Does anyone have an (overhead) paging system that they like that works with SIP?

We’ve got a client with an old paging system that (supposedly) just takes an 
rj11 POTS connection, but when we put an SPA Cisco adapter on it, it doesn’t 
auto-answer the call, so paging never happens.



Does it expect to see a POTS line with battery on it?
Then a Cisco or other ATA that would work to supply service to a POTS phone 
should work
OR:
Does it expect to see a POTS connection from a PBX trunk, and supply battery TO 
the trunk?
Then you would need a Cisco or other ATA with an FXO connection.

Both types of paging systems have been made and both styles of connections have 
existed through the last 30 + years, and since you haven't revealed the brand 
and model of paging system it makes troubleshooting difficult.
Using the existing system can be made to work
I use a very old Harris PagePak VS that was used with a Western Electric 
Horizon system back in the dark ages with Asterisk

John Novack

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Re: [asterisk-users] Dropped calls when all DAHDI lines in use

[John Novack]

Andrew Martin wrote:

- Original Message -

From: "John Novack SCII_U" 
To: "Asterisk Users Mailing List, Non-Commercial Discussion" 
, "Andrew Martin"

Sent: Monday, October 8, 2018 4:29:41 PM
Subject: Re: [asterisk-users] Dropped calls when all DAHDI lines in use
Have you given any thought to moving to at least a current supported version 13?
Asterisk 11 has been EOL for some time now
I doubt you will get a resolution to a version no longer supported.
Moving to the latest version 13 should be relatively quick and painless, and if
the issue persists you might find more assistance.

John Novack


Andrew Martin wrote:

Hello,

I am running Asterisk 11.17 with DAHDI 2.9.0 and an OpenVox A800P with 8x analog
POTS lines coming into my Asterisk server from the phone company. Internally, I
have about 180 SIP clients defined in sip.conf. What appears to be happening is
that if existing calls are consuming all 8 external lines and a new SIP client
attempts to make a call, an existing call gets dropped. The asterisk log simply
shows this as a normal hangup, so I am not able to easily distinguish between a
normal hangup and this type of dropped call. In testing, I am able to get a new
SIP client to report "service unavailable" when all 8 lines are consumed, yet
still drops are reported.

I have been unable to find any configuration settings pertaining to prioritizing
existing calls over new calls. What else can I look for to attempt to debug and
fix this so that existing calls are not dropped?

Thanks,

Andrew


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John,

Thanks for the reply. Yes, I am planning on moving to version 13 but need to 
find a
solution in the interim. If there are any configuration options that pertain to
which actions to take with existing calls when new calls come in, I think it is 
likely
that they would be shared between both versions (and I want to make sure I have 
the
correct settings when I switch to version 13 too). Can you advise on any 
tunables
related to handling existing vs new calls?

Thanks,

Andrew


I really can't help with your existing issue(s)
I suggest you make the switch to the latest version 13, which should go fairly 
smoothly, and you may find that you no longer have an issue.

JN

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Re: [asterisk-users] Dropped calls when all DAHDI lines in use

[John Novack SCII_U]

Have you given any thought to moving to at least a current supported version 13?
Asterisk 11 has been EOL for some time now
I doubt you will get a resolution to a version no longer supported.
Moving to the latest version 13 should be relatively quick and painless, and if 
the issue persists you might find more assistance.

John Novack


Andrew Martin wrote:

Hello,

I am running Asterisk 11.17 with DAHDI 2.9.0 and an OpenVox A800P with 8x analog
POTS lines coming into my Asterisk server from the phone company. Internally, I
have about 180 SIP clients defined in sip.conf. What appears to be happening is
that if existing calls are consuming all 8 external lines and a new SIP client
attempts to make a call, an existing call gets dropped. The asterisk log simply
shows this as a normal hangup, so I am not able to easily distinguish between a
normal hangup and this type of dropped call. In testing, I am able to get a new
SIP client to report "service unavailable" when all 8 lines are consumed, yet
still drops are reported.

I have been unable to find any configuration settings pertaining to prioritizing
existing calls over new calls. What else can I look for to attempt to debug and
fix this so that existing calls are not dropped?

Thanks,

Andrew



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Re: [asterisk-users] Spontaneous reboot due to MySQL lookups ?

[John Novack]

As others have said, clearly it ISN'T "just working" or you would not have 
posted the question

To state again, I am using Version 13, though a few minor revisions behind, 
with MySql, on CentOS 6 and have no rebooting or other MySql related issues

Clearly you need to state in more detail what issues remain, once you migrate 
to AT LEAST 13.xx, and state your OS after becoming current with Asterisk, 
MySql and the OS

I use MySql on every incoming call, and also maintain call detail records in 
MySql for every call, and it just simply works, and has for some time.

Although I may be using it quite differently that you, it simply works.
Is this a newly developing issue, or has it persisted for some time
What if any changes have been made to the dialplan etc?

Have you considered a strictly hardware issue? Memory? HD? MB??

The crystal ball is very cloudy on this one!

John Novack


Jonas Kellens wrote:


Hello

thank you for your answer.

If I read your (and others) reaction correctly I can conclude that this is an 
Asterisk problem and not a problem of MySQL or dialplan logic ?


You should know that the MySQL database is heavily questioned :


mysql> show status like '%onn%';
+--++
| Variable_name    | Value  |
+--++
| Aborted_connects | 469    |
| Connections  | 132762 |
| Max_used_connections | 8  |
| Ssl_client_connects  | 0  |
| Ssl_connect_renegotiates | 0  |
| Ssl_finished_connects    | 0  |
| Threads_connected    | 3  |
+--++
7 rows in set (0.00 sec)



I stick to 1.8 because it just works. I had some issues with version 11 and 13 
in the past.


Regards

Jonas.


Op 04-10-18 om 17:49 schreef John Novack:

Woefully out of date.
You really need to put your efforts into at least a modest upgrade
I use version 13 with MySql queries built into the dialplan on CentOs 6 and 
have NO such issues, either performance or any restart or reboot. It simply 
works

I never used either 1.6 or 1.8, going from 1.4 to version 11, which did require 
some syntax changes to the dialplan.

Given that even version 11 is EOL, you really need to put your efforts into 
doing the migration rather than tracking this one down

JMO

John Novack



Jonas Kellens wrote:


Hello

using Asterisk 1.8.32.

I notice that there is a spontaneous reboot of the Asterisk system from time to 
time.

When I look in the logs (verbose file) I noticed that every time this occurs 
it's at a moment that there is a MySQL action, be it a lookup or an 
insert/update/delete.

I must say I do have some MySQL queries that occur in my dialplan when a call 
comes in, to look up different actions to perform on this call.


An idea how to overcome this problem ? Seems a "performance" issue, no ?!

Is it better to have these MySQL queries to be done by an external script (like 
a php script that I call with the System()-command or a SHELL()-command) ?


Here are some examples from the verbose file.



[Aug 22 15:19:10] VERBOSE[2977] pbx.c: [Aug 22 15:19:10] -- Executing [s@sub-GetAlertInfo:3] 
MYSQL("SIP/SipAgenT01-317d", "Connect connid localhost myuser mypwd myDB") 
in new stack
[Aug 22 15:19:10] VERBOSE[2977] pbx.c: [Aug 22 15:19:10] -- Executing [s@sub-GetAlertInfo:5] 
MYSQL("SIP/SipAgenT01-317d", "Query resultid 1 SELECT uri, callinfo FROM 
distringtone WHERE onoff='1'") in new stack
[Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18]   == Parsing 
'/etc/asterisk/logger.conf': [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 
15:19:18] == Found
[Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18]   == Parsing 
'/etc/asterisk/asterisk.conf': [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 
22 15:19:18] == Found
[Aug 22 15:19:18] VERBOSE[3306] manager.c: [Aug 22 15:19:18]   == Manager 
registered action DataGet
[Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18]   == Parsing 
'/etc/asterisk/codecs.conf': [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 
15:19:18] == Found
[Aug 22 15:19:18] VERBOSE[3306] loader.c: [Aug 22 15:19:18]  Asterisk Dynamic 
Loader Starting:
[Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18]   == Parsing 
'/etc/asterisk/modules.conf': [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 
15:19:18] == Found
[Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18]   == Parsing 
'/etc/asterisk/res_config_mysql.conf': [Aug 22 15:19:18] VERBOSE[3306] 
config.c: [Aug 22 15:19:18]   == Found
[Aug 22 15:19:18] VERBOSE[3306] res_config_mysql.c: [Aug 22 15:19:18]   == 
MySQL RealTime driver loaded.
[Aug 22 15:19:18] VERBOSE[3306] loader.c: [Aug 22 15:19:18]  res_config_mysql.so 
=> (MySQL RealTime Configuration Driver)



[Aug 22 16:23:25] VERBOSE[24283] pbx.c: [Aug 22 16:23:25] -- Executing 
[s@sub-GetSipAccountdetails:3] MYSQL("SIP/SipAgenT01-4184", 

Re: [asterisk-users] Spontaneous reboot due to MySQL lookups ?

[John Novack]

Woefully out of date.
You really need to put your efforts into at least a modest upgrade
I use version 13 with MySql queries built into the dialplan on CentOs 6 and 
have NO such issues, either performance or any restart or reboot. It simply 
works

I never used either 1.6 or 1.8, going from 1.4 to version 11, which did require 
some syntax changes to the dialplan.

Given that even version 11 is EOL, you really need to put your efforts into 
doing the migration rather than tracking this one down

JMO

John Novack



Jonas Kellens wrote:


Hello

using Asterisk 1.8.32.

I notice that there is a spontaneous reboot of the Asterisk system from time to 
time.

When I look in the logs (verbose file) I noticed that every time this occurs 
it's at a moment that there is a MySQL action, be it a lookup or an 
insert/update/delete.

I must say I do have some MySQL queries that occur in my dialplan when a call 
comes in, to look up different actions to perform on this call.


An idea how to overcome this problem ? Seems a "performance" issue, no ?!

Is it better to have these MySQL queries to be done by an external script (like 
a php script that I call with the System()-command or a SHELL()-command) ?


Here are some examples from the verbose file.



[Aug 22 15:19:10] VERBOSE[2977] pbx.c: [Aug 22 15:19:10] -- Executing [s@sub-GetAlertInfo:3] 
MYSQL("SIP/SipAgenT01-317d", "Connect connid localhost myuser mypwd myDB") 
in new stack
[Aug 22 15:19:10] VERBOSE[2977] pbx.c: [Aug 22 15:19:10] -- Executing [s@sub-GetAlertInfo:5] 
MYSQL("SIP/SipAgenT01-317d", "Query resultid 1 SELECT uri, callinfo FROM 
distringtone WHERE onoff='1'") in new stack
[Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Parsing 
'/etc/asterisk/logger.conf': [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 
15:19:18]   == Found
[Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Parsing 
'/etc/asterisk/asterisk.conf': [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 
22 15:19:18]   == Found
[Aug 22 15:19:18] VERBOSE[3306] manager.c: [Aug 22 15:19:18] == Manager 
registered action DataGet
[Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Parsing 
'/etc/asterisk/codecs.conf': [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 
15:19:18]   == Found
[Aug 22 15:19:18] VERBOSE[3306] loader.c: [Aug 22 15:19:18] Asterisk Dynamic 
Loader Starting:
[Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Parsing 
'/etc/asterisk/modules.conf': [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 
15:19:18]   == Found
[Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Parsing 
'/etc/asterisk/res_config_mysql.conf': [Aug 22 15:19:18] VERBOSE[3306] 
config.c: [Aug 22 15:19:18]   == Found
[Aug 22 15:19:18] VERBOSE[3306] res_config_mysql.c: [Aug 22 15:19:18]   == 
MySQL RealTime driver loaded.
[Aug 22 15:19:18] VERBOSE[3306] loader.c: [Aug 22 15:19:18] res_config_mysql.so 
=> (MySQL RealTime Configuration Driver)



[Aug 22 16:23:25] VERBOSE[24283] pbx.c: [Aug 22 16:23:25] -- Executing 
[s@sub-GetSipAccountdetails:3] MYSQL("SIP/SipAgenT01-4184", "Connect connid 
localhost myuser mypwd myDB") in new stack
[Aug 22 16:23:25] VERBOSE[24283] pbx.c: [Aug 22 16:23:25] -- Executing [s@sub-GetSipAccountdetails:4] 
MYSQL("SIP/SipAgenT01-4184", "Query resultid 1 SELECT SIPusername, currstatus, available 
FROM tbl_SIP WHERE ID="800"") in new stack
[Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 16:23:32] == Parsing 
'/etc/asterisk/logger.conf': [Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 
16:23:32]   == Found
[Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 16:23:32] == Parsing 
'/etc/asterisk/asterisk.conf': [Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 
22 16:23:32]   == Found
[Aug 22 16:23:32] VERBOSE[24309] manager.c: [Aug 22 16:23:32]   == Manager 
registered action DataGet
[Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 16:23:32] == Parsing 
'/etc/asterisk/codecs.conf': [Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 
16:23:32]   == Found
[Aug 22 16:23:32] VERBOSE[24309] loader.c: [Aug 22 16:23:32] Asterisk Dynamic 
Loader Starting:
[Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 16:23:32] == Parsing 
'/etc/asterisk/modules.conf': [Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 
22 16:23:32]   == Found
[Aug 22 16:23:32] VERBOSE[24309] config.c: [Aug 22 16:23:32] == Parsing 
'/etc/asterisk/res_config_mysql.conf': [Aug 22 16:23:32] VERBOSE[24309] 
config.c: [Aug 22 16:23:32]   == Found
[Aug 22 16:23:32] VERBOSE[24309] res_config_mysql.c: [Aug 22 16:23:32]   == 
MySQL RealTime driver loaded.
[Aug 22 16:23:32] VERBOSE[24309] loader.c: [Aug 22 16:23:32] res_config_mysql.so 
=> (MySQL RealTime Configuration Driver)



[Oct  4 10:11:25] VERBOSE[4944] pbx.c: [Oct  4 10:11:25] -- Executing [s@sub-settings:16] 
MYSQL("SIP/SipAgenT01-08cb", "Connect connid localhost m

Re: [asterisk-users] Recommended Linux version or how to compile DAHDI on Fedora?

[John Novack]

CentOS 6 is still supported, and does have a 32 bit version
I believe version 6 is supported until 2020, which may give you enough time to 
update your hardware!

I am running CentOS 6.9 32 bit version, and had a problem with the ( then ) 
latest Dahdi 2.11
I believe the issues have now been fixed, and am currently building a 64 bit 
version on another machine

YMMV

John Novack


Ira wrote:

Hi,

I’m currently using 32 CENTOS 5 but it’s now unsupported. I only
have a 32 bit processor and CENTOS no longer supports 32 bit so
I need to move on. I’ve installed the current version of 32 bit
Fedora and I can’t get the latest Dahdi to build. Even tried
downloading the early release DAHDI from github with no luck.

Any recommendation for which 32 bit LINUX to use going forward?

Or optionally, how to compile Dahdi on the most current Fedora.

The error was something to do with xxx.timer.timer.xxx = xxxtimer_xxx

I don’t have the error at hand, because I had to put the old
drive back in to get our phones working again.

Thanks, Ira




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Re: [asterisk-users] how to make International calls from asterisk PBX

[John Novack]

As others have said MUCH more information is needed.
Assume you are using some VOIP provider for international calls
From where to where - gmail gives no clue as to where in the world you are.
Does this provider allow blocking of out of country calls - Do they even 
provide it?
WHICH version of 13?
Care to share a portion of your dialplan?
With your CLI verbosity set high, what are the error messages?
If much of the above is above your paygrade, then perhaps you need to post to 
the biz list and purchase some help?
You will not find any mind readers there either though!


John Novack


Uzma Anjum wrote:

Hello...

I'm running asterisk-13 and international calls are not working in it.How can I 
make it work.Can anyone please tell me.




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Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

[John Novack]

Jonathan H wrote:

On 16 June 2017 at 08:38, J Montoya or A J Stiles
<asterisk_l...@earthshod.co.uk> wrote:


It's hardly Digium's fault, if Google have decided that playing nicely with
syntactically-valid messages doesn't fit their business model

Not really Gmail's fault, either.  Someone above said they had the
same problem with Comcast.net.

Gmail complies with the relevant RFCs just fine. It's most likely
simply because most people who use email, use Gmail.

In addition, gmail properly implement SPF and DMARC checking.

There's over 1 billion gmail account as of 2016, so that's why most
people who are bouncing would be gmail.


Correct Had another one yesterday
Am on several other mailing lists that have no such issue.
Something related to the mailer Digium uses or their ISP


John Novack

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Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

[John Novack]

Not just gmail
Happening as well with Comcast.net

My Comcast address is set to forward to another domain, as Comcast seems to now block 
sending mail with a non Comcast "from" address. they turned that on a couple 
years ago with no  notice.

John Novack


Jonathan H wrote:

Me too, also gmail. I emailed the list owner a couple of days ago, but no reply.

Is everyone else affected also forwarding to another email address
(gmail or not)?

Could be wrong, but I'm guessing there may be an incorrect DMARC
policy somewhere - although this is the only fail I could find in the
headers.

boun...@lists.digium.com;
dmarc=fail (p=NONE sp=NONE dis=NONE) header.from=gmail.com



On 12 June 2017 at 09:12, Steve Davies <davies...@gmail.com> wrote:

I am also getting this, three or four times in the last month after years of
no problems.

I agree that Gmail is the likely common factor, but I would love to have
access to these bounce messages to know whether it is actually an
overly-paranoid list server!

Steve

On Mon, 12 Jun 2017 at 09:09 Andrew Furey <andrew.fu...@gmail.com> wrote:

Ditto; a Gmail issue?

Andrew

On 12 June 2017 at 16:00, Marcelo Terres <mhter...@gmail.com> wrote:

It is happening the same with me.

Regards,
Marcelo H. Terres <mhter...@gmail.com>
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres


On 12 June 2017 at 08:07, Olivier <oza.4...@gmail.com> wrote:

Hello,

I'm a faithful reader of this mailing list, for several years now.

Lately, I'm receiving emails asking me to re-enable my list
subscription due
to "excessive bouncing".

What does this exactly mean and why am I receiving this ?
Beside re-enabling my subscription, what can I do to improve things ?

Regards

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Re: [asterisk-users] Commit dialplan & other config. in memory to disk?

[John Novack]

Nathan Anderson wrote:

'lo,

So yesterday, one of our clients had the misfortune of having the disk that 
their Asterisk config (*.conf) was stored on take a dirt nap.  Of course, 
Asterisk was still running at the time, and everything continued to work 
(except for voicemail, which was stored on the same disk) right up until I shut 
down Asterisk to investigate what was going on.  Because the disk was dead, 
though, I couldn't start Asterisk back up after that, and OF COURSE the backups 
were not firing off correctly so now we are faced with regenerating the config 
again (including dialplan) from scratch.

In the future, if I were to ever run into a similar situation, is there any way 
to request or instruct Asterisk to write the current dialplan that is in memory 
and other important config files (e.g., users.conf) to disk in a *different* 
location than where it originally read them from when it started up?  I could 
have saved myself a crap-ton of work if this were possible...

Thanks,

-- Nathan


Isn't this a task for Linux and a cron job rather than asterisk?
Simplest thing would be to copy to another machine even off site, all the confs 
and whatever else would help you resurrect a machine more quickly.
Backups not "firing off correctly"  means what?

When changes are made to the dialplan a copy off site or at least off machine 
is in order

I run a cron job every early morning to do just that as well as the MySql data 
files so if need be I can recreate a machine and have call records and whatever 
not more than a day stale.


John Novack

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Re: [asterisk-users] Alphabet character in destination number (CDR)

[John Novack]

J Montoya or A J Stiles wrote:

On Thursday 30 Mar 2017, Ikka Tirtawidjaja wrote:

Dear all,

I have PBX with asterisk 13.x

a couple of IPPhone that connect to that asterisk PBX send an alphanumeric
dialed phone number.

for example, in my CDR table, field DST, it show dialed phone number like
- 0C81318304632C  (it should be 081318304632)
- 08D11157112 (it should be 0811157112).

Why it's happening ? and how can I prevent it to happen ?

A, B, C and D are actually valid DTMF digits  (they belong in a column to the
right of 3, 6, 9 and # respectively, and have the "high" frequency 1633 Hz).
TTBOMK they were never actually used for anything in practice, they just keep
kicking around like a vestigial organ  (compare how computer software for the
UK financial industry still includes code to deal with mediaeval pounds,
shillings and pence).

Is it possible for a 1633 Hz tone, loud enough to swamp the "high" frequency
of a dialled digit, to be finding its way somehow into the microphone of the
affected phone and confusing it when digits are dialled?



A,BC & D were used in the US in the AutoVon Military system, and telephones 
that have that keypad often bring pretty good money.

Some external Voice mail systems use one or more of the codes as well ( Toshiba 
systems for one ) Probably because they are an easy way to prevent most users 
from messing around in the system
It was and in some cases still used for network signalling.
Not that computer telephony designers believe in standards but the 16 button 
tone pad is an ITU-T standard.  I believe in the UK it is known as MF-4

Asterisk DTMF detection leaves a lot to be desired. It certainly is not 
Exchange grade. There may be a way to diddle the code to remove the false 
detection but this should only happen with in-band signalling.I thought all SIP 
data was normally sent as data though sip.conf does allow in-band


John Novack

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Re: [asterisk-users] Realtime SIP peers do not register any more after upgrade to Asterisk 13

[John Novack]

Remove yourself

READ - Included with every message -

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Dario Estupinan wrote:


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Re: [asterisk-users] rasberry pi

[John Novack]

A J Stiles wrote:

On Wednesday 06 Jul 2016, John Novack wrote:

AstLinux can be remotely managed with the GUI,
which unlike other Asterisk GUI's the conf files are not modified by the
GUI and can be edited "by the book" AstLinux will NOT work with a Pi
though. It is not for the ARM processor.

What stops it from building properly on armhf architecture?


Check with the developers on prebuilt AstLinux

There is also a custom build environment, but it reportedly doesn't support ARM 
processors either

Well above my paygrade

John Novack

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Re: [asterisk-users] rasberry pi

[John Novack]

Another good choice for SOHO applications is an older HP Thin Client, such as a 
5720. Using AstLinux on it's flash memory, 512K or 1 Gig, with 512K memory. The 
HP thin clients are available used, often quite inexpensive, and are already 
packaged.
AstLinux can be remotely managed with the GUI, which unlike other Asterisk GUI's the conf 
files are not modified by the GUI and can be edited "by the book"
AstLinux will NOT work with a Pi though. It is not for the ARM processor.

In a telephone collectors network, we have more than 30 nodes running, some for 
many years, using AstLinux and various thin clients. The HP's area nice package 
that can be configured and mounted to a wall and simply forgotten.
Sometimes one can be found with the PCI expansion chassis that would allow one 
PCI card to be added for analog or even a T1 interface.
Search AstLinux for much more information

John Novack

Thufir wrote:

ok, that's really all I need to know. Of course, if anyone else wants to throw 
in their two cents, don't let me stop you :)


-Thufir

On Wed, Jul 6, 2016 at 1:36 AM, Frank Vanoni <mailingl...@linuxista.com 
<mailto:mailingl...@linuxista.com>> wrote:

I'm currently using Asterisk 11.7.0 on a Raspberry Pi 2 Model B with
Ubuntu Server 14.04.

Works fine! :-)

Frank

On Wed, 2016-07-06 at 01:10 -0700, Thufir wrote:
> I'm debating between a cloud PBX or, perhaps, rasberry pi.  For a
> SOHO, maybe three hardphones, rasberry pi would suffice?  I would be
> amazed, but, if so, great.




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Re: [asterisk-users] Nube question: where is chan_sip.so?

[John Novack]

Was this compiled from source?

If a package install, you may be at the mercy of the package creator

In version 11, sip is installed by default, unless a noload statement is active 
in modules.conf

All of this may have changed with the transition to pjsip though.


John Novack

Peter Wallis wrote:

I am having real trouble getting started.  A definitive "hello world" is 
certainly missing from the official site and the ones out there are dated or broken.

I am beginning to think something went wrong with the install.  It was a fresh 
install of an  Ubuntu server, and a fresh install of 13.7.0 - Should be Okay no?

A question.  Am I expecting to find chan_sip.so in /usr/lib/asterisk/modules/ ?

If so, it is not there - no idea how I would have lost it.  We do have a new 
cat - perhaps she ate it.

If I ought not expect to find it there, why does
[modules]
require =chan_sip.so
and/or
require = res_pjsip.so
cause asterisk to fail to start? The documentation makes no mention of an 
alternate method of loading modules.

best wishes,

P






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Re: [asterisk-users] Best Asterisk Platform

[John Novack]

Are you trying to start a religious argument?

CentOS /RedHat appear to be the most trouble free when compiling from source
JMO

John Novack


er ic wrote:

What is the best asterisk platform to use? What are you guys using?

I am looking for something to host either in our data center or at the customer 
prem where I have the control over the unit and not through a contractor.

I dont mind paying a license fee for a front end interface but still would 
rather not have to pay.

Thanks,
--Eric




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Re: [asterisk-users] Best Asterisk Platform

[John Novack]

Agreed.
I have built close to 40 AstLinux systems, mostly on HP Thin Clients
And the excellent GUI is constantly improving.

John Novack


Michael Keuter wrote:

Am 23.12.2015 um 14:31 schrieb er ic <email.eherr9...@gmail.com>:


What is the best asterisk platform to use? What are you guys using?

I am looking for something to host either in our data center or at the customer 
prem where I have the control over the unit and not through a contractor.

I dont mind paying a license fee for a front end interface but still would 
rather not have to pay.

Thanks,
--Eric

Hi Eric,

it depends on what you need :-).
For smaller installations (<100), and if you don't need a fancy GUI,
I can recommend AstLinux (Open Source), a complete communication plattform in 
one box (or VM):

http://www.astlinux.org

Michael

http://www.mksolutions.info







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Re: [asterisk-users] Windows Asterisk Help

[John Novack]

Murthy Gandikota wrote:



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To: asterisk-users@lists.digium.com
From: webaccounts...@jgoettgens.de
Date: Wed, 29 Jul 2015 16:11:31 +0200
Subject: Re: [asterisk-users] Windows Asterisk Help



Downloaded latest version of Asterisk from www.asteriskwin32.com 
http://www.asteriskwin32.com and installed on Windows 7.

Here  is my sip.conf

[general]
context = demo  ;  Default context for incoming calls
bindport = 5060  ;  UDP Port to bind to (SIP standard port is 
5060)
bindaddr = 0.0.0.0  ;  IP address to bind to (0.0.0.0 binds to 
all)
srvlookup = yes  ;  Enable DNS SRV lookups on outbound calls
context=incoming
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=g723
externip=72.220.28.226
localnet=192.168.0.0
nat=yes
maxexpiry=15
minexpiry=14
;rtautoclear=no
;autofallthrough=yes

register =16194077214:password@69.59.234.67:5060/202

[authentication]
[3000]
type = friend
context = default
username = 3000
host = dynamic
mailbox = 3000
dtmfmode = rfc2833
[3001]
type = friend
context = default
username = 3001
host = dynamic
mailbox = 3001
dtmfmode = rfc2833

[3002]
type = friend
username = 3002
context = default
host = dynamic
mailbox = 3002
dtmfmode = rfc2833

[vonage-out]

username=16194077214

type=friend

secret=password

port=5061

nat=yes

host=69.59.234.67

fromuser=16194077214

fromdomain=69.59.234.67

dtmfmode=rfc2833

auth=md5

[vonage202]

username=16194077214

;type=friend
type=peer
;type=user

secret=password

port=5061

nat=yes

insecure=port,invite

host=69.59.234.67

fromuser=16194077214

fromdomain=69.59.234.67

;dtmfmode=inband

context=from-pstn

canreinvite=no

;auth=md5
disallow=all
allow=ulaw
;allow=alaw
;allow=g729
;allow=g723

Here is my extensions.conf

[from-pstn]
;exten = 16194077214,1,verbose(0, hello)
exten = 16194077214,1,Answer;
exten = 16194077214,n,SayUnixTime()
exten = 16194077214,n,Hangup


I am able to connect with Asterisk on the first try after fresh load, but 
not on the subsequent tries.
I have to re-reload sip.conf and extensions.conf to connect with Asterisk. 
Looking at the logs, it seems like a registration issue.  So I set minexpirty 
and maxexpirty that seems to have no effect.  can post the logs, if someone 
wants me to.

Your kind help is appreciated.

Best regards
murthy




www.asteriskwin32.com http://www.asteriskwin32.com hosts only a very very old 
version of Asterisk (1.2.something). What speaks against setting up a small virtual 
machine to host a recent version of Asterisk?

jg

You have a point. My SIP provider at the moment is Vonage which I can't access 
from work (some security issue:)
So I am confined to testing from home and I don't have any other machine to 
spare. If there is no other way
to trouble-shoot the problem, I will have to do what you suggest.

Thanks  Regards
murthy



For very little $$$ you could obtain an HP thin client, load a modern version 
of Asterisk using AstLinux, and leave your Win 7 machine to do what it does 
best ( which is certainly NOT Asterisk )
Once installed, it can be completely controlled and configured remotely over 
your home LAN, consumes very little power, has a universal power supply, 
consumes little power and no noisy fans.
HP5720 units can be had off eBay for $20-30 US. Even with shipping to your 
country, really low cost solution much more in the mainstream.
AstLinux uses standard Asterisk confs. The GUI is used for management and 
editing, and doesn't  use the difficult to troubleshoot  and quirky overlays of 
a TrixBox or FreePBX
Check out the astlinux website for more details

John Novack

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Re: [asterisk-users] CDR in an MySQL-Database

[John Novack]

Luca Bertoncello wrote:

Hi list!

I'd like to save all information about calls (CDR) in a MySQL-Database.
I created the DB and a user for Asterisk on a separate server, then I
configured my cdr_mysql.conf so:

[global]
hostname=192.168.10.3
dbname=asterisk
table=cdr
password=MYSECRET
user=asterisk
port=3306

and my cdr.conf so:

[general]
enable=yes
unanswered = yes
safeshutdown=yes

[mysql]
usegmtime=no
loguniqueid=yes
loguserfield=yes
accountlogs=yes

I created the table in the DB so:

CREATE TABLE IF NOT EXISTS `cdr` (
   `id` int(11) unsigned NOT NULL AUTO_INCREMENT,
   `calldate` datetime NOT NULL DEFAULT '-00-00 00:00:00',
   `clid` varchar(80) COLLATE utf8_bin NOT NULL DEFAULT '',
   `src` varchar(80) COLLATE utf8_bin NOT NULL DEFAULT '',
   `dst` varchar(80) COLLATE utf8_bin NOT NULL DEFAULT '',
   `dcontext` varchar(80) COLLATE utf8_bin NOT NULL DEFAULT '',
   `lastapp` varchar(200) COLLATE utf8_bin NOT NULL DEFAULT '',
   `lastdata` varchar(200) COLLATE utf8_bin NOT NULL DEFAULT '',
   `duration` float unsigned DEFAULT NULL,
   `billsec` float unsigned DEFAULT NULL,
   `disposition` enum('ANSWERED','BUSY','FAILED','NO ANSWER','CONGESTION')
COLLATE utf8_bin DEFAULT NULL, `channel` varchar(50) COLLATE utf8_bin DEFAULT
NULL, `dstchannel` varchar(50) COLLATE utf8_bin DEFAULT NULL,
   `amaflags` varchar(50) COLLATE utf8_bin DEFAULT NULL,
   `accountcode` varchar(20) COLLATE utf8_bin DEFAULT NULL,
   `uniqueid` varchar(32) COLLATE utf8_bin NOT NULL DEFAULT '',
   `userfield` float unsigned DEFAULT NULL,
   `answer` datetime NOT NULL,
   `end` datetime NOT NULL,
   PRIMARY KEY (`id`),
   KEY `calldate` (`calldate`),
   KEY `dst` (`dst`),
   KEY `src` (`src`),
   KEY `dcontext` (`dcontext`),
   KEY `clid` (`clid`)
) ENGINE=InnoDB DEFAULT CHARSET=utf8 COLLATE=utf8_bin AUTO_INCREMENT=1 ;

Then I restarted Asterisk (core restart now).
Unfortunately it does not work, since I get on boot:

[Jul  6 21:20:39] WARNING[9735]: res_config_mysql.c:1468 load_mysql_config: 
MySQL RealTime: No database user found, using 'asterisk' as default.
[Jul  6 21:20:39] WARNING[9735]: res_config_mysql.c:1474 load_mysql_config: 
MySQL RealTime: No database password found, using 'asterisk' as default.
[Jul  6 21:20:39] WARNING[9735]: res_config_mysql.c:1480 load_mysql_config: 
MySQL RealTime: No database host found, using localhost via socket.
[Jul  6 21:20:39] WARNING[9735]: res_config_mysql.c:1486 load_mysql_config: 
MySQL RealTime: No database name found, using 'asterisk' as default.
[Jul  6 21:20:39] WARNING[9735]: res_config_mysql.c:1492 load_mysql_config: 
MySQL RealTime: No database port found, using 3306 as default.
[Jul  6 21:20:39] WARNING[9735]: res_config_mysql.c:1509 load_mysql_config: 
MySQL RealTime: No database socket found (and unable to detect a suitable path).

And of course:

OpenWrt*CLI cdr show status

Call Detail Record (CDR) settings
--
   Logging:Enabled
   Mode:   Simple
   Log unanswered calls:   Yes

* Registered Backends
   ---
 cdr-custom

Asterisk 1.8 runs on an OpenWRT-Switch.
Any idea?

Thanks
Luca Bertoncello
(lucab...@lucabert.de)


Been doing this with MySql for the last 10 years, though not on an openWrt 
machine
MySql is on the Asterisk machine.
Also have additional database tables to block by callerId and name
Did have some issues with the dialplan syntax when moving from 1.4 to 11, but 
it just works
I assume OpenWRT is a pre compiled Asterisk package?
You may not have the proper configuration to use MySql
Your error message(s) seem to say it expects to find the MySql server on 
localhost but you say it is on a different machine!!
perhaps you need to fix that first?


John Novack




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Re: [asterisk-users] FXS Solutions for modems and other non jitter tolerant devices

[John Novack]

Jon Pounder wrote:




snip

Fax is really the only need recently, and even that has alternatives like 
emailing scans that most people prefer now.


The legal and medical communities still seem to prefer faxing, in the ( 
mistaken? ) belief that it is more secure. In fact the medical community is 
fearful of the legal beagles.

These groups are really slow to change.
At least in the USA

John Novack


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Re: [asterisk-users] small homebrew pbx

[John Novack]

James Cass wrote:

I picked up a cheap JS200-FX on ebay: http://x100p.com/products/js200fx.php for 
$30, and it works great for a home install.  Very low power draw as well.

James Cass http://goog_987864563
jcas...@gmail.com mailto:jcas...@gmail.com

The JS-200 runs a very old ( 1.4 ) version of Asterisk

I have set up more than 30 nodes using the HP thin clients, many using the 
available cheap T5720 units. Install the latest AstLinux in the flash, and 
follow the advice for a PSTN provider. I prefer voip.ms here in the US, and 
they also will deliver via IAX, which I prefer as SIP has so many hacking 
attempts I just don't want to deal with it.
AstLinux in our private peer to peer network, along with many also having a 
PSTN connection, is easy to set up, easy to support remotely, and with a flash 
based system very reliable.
also Astlinux has a built in facility for an in place upgrade. It also doesn't 
have the PITA configuration of a PIAF. Standard Asterisk conf files are used
The HP 5720's also have a 120-240 volt power supply, so it should work almost 
worldwide
Somewhat larger than a Pi, but in a decent case that could easily be mounted  
on a wall somewhere and connected to the LAN

Other newer units with multiple NIC ports and AstLinux can also be your router 
/firewall

Unless one is running a 100 seat call center, no need for one of those huge 
juice hogs anymore.

John Novack

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Re: [asterisk-users] New Asterisk build

[John Novack SCII]

Ira wrote:

   Hello John,
  
Friday, March 6, 2015, 12:34:42 PM, you wrote:
  

Find a HPT5720 with expansion chassis on eBay for under $50,
load AstLinux ( instructions at AstLinux.org ) Move your
Digium card and your confs , fix up any differences from your

But given that means buying an old computer, why change at all?
I already have a very low power one that works fine. Is
AstLinux better than Centos 5 running Asterisk 13?
  
-- Ira

Better?
Depends on how you define better

Since you haven't revealed what you are currently using, really hard to say, but running 
a box without a spinning hard drive and fans to die, certainly is better
An OS that fits in  1Gig might very well be better than a bloated CentOS 5, 6 
or what have you
If what you have works for you, then why even ask?
If it works, leave it alone.
You will certainly find 1000 opinions on the list, if any decide to take the 
bait


JN





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Re: [asterisk-users] New Asterisk build

[John Novack SCII]

Find a HPT5720 with expansion chassis on eBay for under $50, load AstLinux ( instructions at AstLinux.org ) Move your Digium card and your confs , fix up any differences from your 
older version of Asterisk to the fairly current version 11 currently available with AstLinux.

Use the GUI to edit and mage the system, as AstLinux has a somewhat different 
directory structure than you may be familiar with
You should be up and running in a couple of hours, have a low power  20 watts, 
fanless flash based system that will just work in a real case.
The Pi is OK for a playtoy and some testing, but I much prefer the HP thin 
clients for a robust installation.
I assume you are not doing any fancy call center or heavy database work. For a 
home or home office it is a really good solution.
AstLinux is also used with other embedded installations on computers with 
multiple Ethernet ports, acting as router and firewall in addition.
I prefer the component solution personally, which makes the HP thin clients the 
way to go.


John Novack


I have built more than 30 of these systems on various HP Thin Clients, used off 
of eBay with no failures

Ira wrote:

   Hello Asterisk,
  
   Back in 2009 I built a small Intel Atom based computer running

   Centos 5 for my asterisk system. 5 phones, 2 people 1 POTs
   line and six or so SIP numbers. So basically no load. I'm
   feeling like it's time to build another machine. It's probably
   silly, but it's been six years and I can't upgrade the OS
   which is falling behind. I'd likely just put it on a Raspberry
   Pi or something like that, but I need the one POTS line and
   all I have for that at the moment is a Digium card for a PCI
   slot.

   Are there any current thoughts on this?
  
-- Ira





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Re: [asterisk-users] LAN sip-to-sip

[John Novack]

It looks as if that is more of a question/issue with your router, rather than 
Asterisk.

I have SIP devices working on my LAN, all hardwired, and have no need to open 
any ports or have the router address SIP in any way
My switch is not managed, and the router ports on the LAN side are all unmanaged, just a 
huge Ethernet wirenut
You SHOULD be able to communicate between devices on the LAN without any 
firewall issue.
I have also found with some routers that the DMZ isn't what one expects, and 
can get in the way, depending on the firware.
Does this router have any SIP ALG setting? turn it off!
As an aside, I would caution you to not have SIP 5060 exposed to the public 
Internet, or you will soon regret it.

I am sure others will have much better information though

John Novack

thufir wrote:

I'm reading the O'Reilly Asterisk the definitive guide, 4th ed, with a
starfish on it.  In some ways, astonishing that it's not really that
definitive, it's more general -- and it only clocks in at one ream of
paper!

In any event, I'm having some port problems on my home network:

http://security.stackexchange.com/questions/81752/

I need to open ports for Asterisk to work even on a local level.



so I'm just asking in general.  For SIP to SIP peer calling, and by that
I just mean ring or beep, some sort of ping, basically, just
configure the two softphones to use the IP address for the Asterisk box?


also:


tleilax:~ #
tleilax:~ # asterisk -V
Asterisk 1.8.32.1-vici
tleilax:~ #
tleilax:~ # asterisk -rm
Asterisk 1.8.32.1-vici, Copyright (C) 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer marks...@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for
details.
This is free software, with components licensed under the GNU General
Public
License version 2 and other licenses; you are welcome to redistribute it
under
certain conditions. Type 'core show license' for details.
=
log and verbose output currently muted ('logger mute' to unmute)
Connected to Asterisk 1.8.32.1-vici currently running on tleilax (pid =
3062)
Verbosity is at least 21
tleilax*CLI
tleilax*CLI sip show peer babytel


* Name   : babytel
Secret   : Set
MD5Secret: Not set
Remote Secret: Not set
Context  : default
Subscr.Cont. : Not set
Language : en
AMA flags: Unknown
Netborder CPD: No
Transfer mode: open
CallingPres  : Presentation Allowed, Not Screened
Callgroup:
Pickupgroup  :
MOH Suggest  : default
Mailbox  :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit   : 0
Max forwards : 0
Dynamic  : Yes
Callerid :  
MaxCallBR: 384 kbps
Expire   : -1
Insecure : no
Force rport  : Yes
ACL  : No
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: 4294967295
DirectMedia  : No
PromiscRedir : No
User=Phone   : No
Video Support: No
Text Support : No
Ign SDP ver  : No
Trust RPID   : No
Send RPID: Yes
TrustIDOutbnd: Legacy
Subscriptions: Yes
Overlap dial : No
DTMFmode : rfc2833
Timer T1 : 500
Timer B  : 32000
ToHost   : sip.babytel.ca
Addr-IP : 198.38.7.11:5060
Defaddr-IP  : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 1private
SIP Options  : (none)
Codecs   : 0x4 (ulaw)
Codec Order  : (ulaw:20)
Auto-Framing : No
Status   : UNREACHABLE
Useragent:
Reg. Contact :
Qualify Freq : 6 ms
Sess-Timers  : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine   : asterisk
Parkinglot   :
Use Reason   : No
Encryption   : No

tleilax*CLI
tleilax*CLI sip show peers
Name/username Host Dyn Forcerport ACL Port Status
201/201   (Unspecified) D   N 0UNKNOWN
babytel/1private 198.38.7.11  D N
  5060 UNREACHABLE
gs102/gs102   (Unspecified) D   N 0UNKNOWN
3 sip peers [Monitored: 0 online, 3 offline Unmonitored: 0 online, 0
offline]
tleilax*CLI




thanks,

Thufir




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Re: [asterisk-users] JITTERBUFFER function

[John Novack SCII]

Google is your friend!!!

http://searchunifiedcommunications.techtarget.com/definition/jitter-buffer
http://www.voiptroubleshooter.com/problems/jitterbuffer.html
http://www.voip-info.org/wiki/view/Asterisk+new+jitterbuffer
http://www.webopedia.com/TERM/J/jitter_buffer.html


Peg Leg O'Brien

amertel wrote:

WTF is a jitterbuffer?


Sent from my Verizon Wireless 4G LTE smartphone


 Original message 
From: Matthew Jordan mjor...@digium.com
Date: 01/29/2015 10:41 AM (GMT-05:00)
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] JITTERBUFFER function

On Thu, Jan 29, 2015 at 4:56 AM, Torbjorn Abrahamsson
torbjorn.abrahams...@gmail.com wrote:
 Hello!



 I am going to use the JITTERBUFFER function in a SIP (and local channels)
 only setup, but have some questions of how to use it:



 1.   Do I need to activate jbenable in sip.conf? Or is it enough to call
 the JITTERBUFFER function?

You only need to use the JITTERBUFFER function.

The jbenable option will enable a jitter buffer on every channel
created for that peer (or, if global, for every peer in the system).
Depending on the version of Asterisk, it will also place the jitter
buffer on the write side of the channel, which is often not what you
want.

 2.   What is the preferred way to invoke this function? Say I have
 channel A which is not in need of buffering, while channel B do need it. If
 A calls B and I do Set(JITTERBUFFER(fixed)=default), my guess is that it
 will be attached to channel A:s read side. This is not the desired outcome,
 as I would like to have it on B:s read side. How should I invoke this to
 make the buffer belong to channel B? Maybe using b option to Dial? So that
 when a JB-enabled device (B) calls out one just calls JITTERBUFFER from the
 normal dialplan flow, and if there is a call to the device (B) one need to
 use b option? Sound correct?


Invocation examples are on the wiki:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_JITTERBUFFER

The JITTERBUFFER function only affects the channel it is placed on,
and not any channel it may be bridged with. That means you have to
place it on the correct channel and not expect some magicry inside
Asterisk to try and manipulate things for you (which is almost always
a bad implementation decision). If you need it on an outbound channel,
that means using one of the pre-dial handlers
(https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers) to
place the jitter buffer on the outbound channel after its creation.

Example:

[default]

exten = set_up_outbound,1,NoOp()
same = n,Set(JITTERBUFFER(adaptive)=default)
same = n,Return()

exten = outbound_dial,1,NoOp()
same = n,Dial(PJSIP/Alice,,b(default^set_up_outbound^1))
...

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

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Re: [asterisk-users] Strange Issue: asterisk deleted

[John Novack]

Question remains, how was it compromised?
In the original install ?
A fresh install perhaps from another source?

Best you determine HOW before spending more time going down another rabbit hole!

John Novack

Antoine Megalla wrote:

Hi

Thank you for your support.
The server is actually compromised, I discovered that after making a deep trace 
using the audit daemon and looking for the kill signal (SIGKILL) that 
terminates asterisk.

snipped to please the mailing list 

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Re: [asterisk-users] Upgraded to 13 and now Mailbox is empty in sip show peers

[John Novack]

Eric Wieling wrote:

I doubt the person cares, if you don't like people top posting then stop reading their 
messages.  If someone top posts, nothing you do will make them stop top posting.   
Complaining about something you cannot change just wastes everyone's time.I have a 
rule which deletes messages with top post in them so I don't usually see 
these silly messages.

Some might even say that the constant complaining about top posting, expecting 
different results, is a definition of insanity!

Some of the same folks who constantly complain about top posting will leave many many 
footers from the list in place, which makes the neatly flowing conversation 
nearly impossible.

Peg Leg O'Brien


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles
Sent: Thursday, November 20, 2014 11:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Upgraded to 13 and now Mailbox is empty in sip 
show peers

**  THIS IS NOT WHERE YOUR REPLY BELONGS  **

Which part of THIS IS NOT WHERE YOUR REPLY BELONGS do you not understand?




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Re: [asterisk-users] quoting arguments to System command in dialplan

[John Novack]

Eric Cooper wrote:

On Wed, Jul 02, 2014 at 05:00:47PM -0700, John Kiniston wrote:

How about using the FILTER function to strip out anything you don't like from
the CALLERID variables?
Set(CIDNAME=${FILTER(A-Z,${CALLERID(NAME)})})
Set(CIDNUM=${FILTER(0-9,${CALLERID(NUM)})})

Thanks for the suggestion; I'll try it.  I'm still bothered that I
can't figure out how to reliably quote the characters rather than just
stripping them out.


I have found that this varies from version to version
The parser in 1.4 is quite different than in 1.8 or 11
I ran into this with MSQL commands within the dialplan
Escapes (\) single and double quotes are not uniform from 1.4 through 11.
YMMV

John Novack

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Re: [asterisk-users] Channel is answered by FXO card before callee answered the phone(pick up phone)

[John Novack SCII]

A J Stiles wrote:

On Thursday 05 Jun 2014, Mojtaba wrote:

My scenario is (2)

After doing some tests with my own hardware, I'm now convinced that this is
actually normal behaviour:  As far as Asterisk is concerned, a call is deemed
answered as soon as the hardware seizes the line.  It is only not answered
if the line is not available.

Which makes sense, because an analogue line has no D-channel.  Once the trunk
is acquired successfully, there is no way for a machine to know the state of
the call beyond then.  Such supervisory information as there is -- a regular
cadence during ringing, possibly a burst of white noise and then a human voice
-- is geared towards interpretation by human beings.

Moerover, since the tones are different in every country  (and sometimes,
between different telephone exchanges in the same country; at one time, the UK
was using three sets of supervisory tones depending whether you were on an
old-fashioned clicky-clicky exchange, an intermediate-generation analogue
electronic exchange or System X)  it would not be a trivial task to make sense
of them.


I think if you want full supervisory information, you are going to need to use
some sort of digital telephony technology  (ISDN or GSM).


This is well known behavior for many years, since the inception of 
Asterisk/Zaptel
I wonder why tests had to be run!
The OP issue was answered several days ago
His issue was obvious and well stated until another poster confused the issue!

John Novack

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Re: [asterisk-users] Channel is answered by FXO card before callee answered the phone(pick up phone)

[John Novack]

Mojtaba wrote:

Hello Experts.
Im working with Asterisk PBXand freeswitch PBX.
I have a challenge with FXO card in Asterisk and i could not solve it yet.
I hope you could guide me in this regards.
When i want route the call to FXO channels, Before the callee answer
the phone (pick up phone), The channel is answered with FXO card. How
can change this treat so that the callee dont answer the phone, the
channel dont answered with FXO card. I have this challenge with FXO
gateway too.
This challenge is more important when using callfile to generate call
using DAHDI/g0/
When using it,All attempts to generate call are answered in CDR
field(disposision=answered).

Im waiting for your replying and Thanks.
With Regards.Mojtaba


I believe that is a well known behavior of Zaptel/DAHDI, given that many analog 
lines that would be attached to an FXO card supply little or no indication of 
an answered call, calls are considered answered once dialing is complete.

You will need to use an alternate technology if you want to know if or when the 
caller has answered.

John Novack

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Re: [asterisk-users] Possible Dahdi compile problem

[John Novack]

Possible hardware issue?
If you are not using DAHDI, why are you installing it?


Did you intend to add some information to this post that was made earlier today?

John Novack

Doug wrote:


I am having problems with system crashes in a dahdi/asterisk compile.  The 
is on a Beaglebone Black running Debian 3.8.13 kernel.

Dahdi is patched and asterisk is also custom so this must be compiled from 
source. I believe I have the proper header and source files for the running 
kernel. Both dahdi and asterisk compile and install. When run asterisk runs for 
anywhere from 10 minutes to a half hour or more and crashes. This is a complete 
system crash requiring a reboot. There are no indications of why it crashed. No 
messages and analyzing memory and other parameters before the crash show 
nothing unusual.

 This same package arrangement works fine on several other platforms so I 
discount a problem in the code.

On thing that may give a clue is that when dahdi is installed it creates 
and puts its modules in  /lib/modules/3.8.13   The system modules are in 
/lib/modules/3.8.13-bone43   - the bone43 is the patched 3.8.13 kernel for the 
Beagle Black. The patches are installed in the sources but dahdi insists on 
putting the modules in 3.8.13  and they are not found there.

I have to mv  /lib/modules/3.8.13/dahdi  to /lib/modules/3.8.13-bone43  and 
then do a depmod -a 3.8.13-bone43

Then dahdi installs. I do not use any dahdi modules so no modules install.

In asterisk I get the following -

beaglebone*CLI dahdi show status
Description Alarms IRQbpviol CRC4

beaglebone*CLI dahdi show channels
   Chan Extension  Context Language MOH Interpret
 pseudodefault default


If dahdi installs and at least appears to work with asterisk does that mean 
that it is compiled correctly? I am trying to figure out what might be causing 
this problem.

Doug
WA3DSP







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Re: [asterisk-users] Asterisk CLI Banner

[John Novack]

Richard Kenner wrote:

Modifying a program you have legitimately acquired is Fair Dealing.
The Law of the Land gives you the right to do that, even if the
vendor restricts your exercise of that right in practice by
withholding the Source Code.

That is false.  Modifying a program is creating a derivative work.
As purchaser of a copyrighted item, you normally *do not* have that right.

And this certainly may vary from jurisdiction to jurisdiction.  For a
(quite dated at this point) discussion of this issue from a US perspective,
see

http://www.law.berkeley.edu/php-programs/faculty/facultyPubsPDF.php?facID=346pubID=157

The author is a recognized expert in software IP law.


Of course, any good attorney will never commit to anything. They will never say 
it is alright to do X, unless X is do nothing

Patent  copyright attorneys seem especially non committal, at least in the US. 
probably because if any case ever goes to court, the decision and possible 
punishment is up to the whims of the judge and/or jury, and every law is up to 
interpretation, which can vary from moment to moment.

Law is not physics!

John Novack

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Re: [asterisk-users] Need more meetme users -- hitting some limit

[John Novack]

Steve Edwards wrote:

On Fri, 21 Mar 2014, Adrian Serafini wrote:


Upgrade to 1.4?  hehe, I thought you were the self proclaimed 1.2 luddite? I'm 
a big fan of older releases with 1 year plus of uptime.


Yep, that's me :)

I'm trying to make the leap from 1.2 to 11.8.1


That is a HUGE leap
Watch out for whiplash!

John Novack

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Re: [asterisk-users] Replying to Posts

[John Novack]

Don Kelly wrote:


 On 13/3/14 6:27 pm, Eric Wieling wrote:
  This is an example of why I top post.   Who wrote what?

 Of course, if you use a mail client that's capable of quoting correctly,
 it all works beautifully.


Kevin Larson sez:

Outlook can quote correctly, but it is an all or nothing setting it would appear. Lotus 
Notes actually handles it better as there is a Reply option for normal email and a Reply 
With Internet-Style History that I use for this list. I don't have any problems following 
the rules of the list, but I am fully on the side of the Replies should go at the 
top group and would vote for a change in the rules.

I’ll vote again for top posting, and expect my vote to be recognized 
“internationally” about as much as the Crimean referendum.

--Don


As an interesting aside, the oft quoted rule #5 didn't exist for many years, 
until one of these diatribes took place. Then, and only then, was it added and 
the contention made that it was always there.
Many of the same who continue to carp on top posting are the worst offenders 
when it comes to trimming the footers that arrive with each message, forcing  
the reader to wade through many of these to ( sometimes ) find a reply or 
maybe, just maybe, an answer. Often it isn't worth the effort to scroll through 
all the crap to find the pony.

John Novack

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Re: [asterisk-users] stopping unwanted attempts

[John Novack]

Changing from 5060 is very effective.
Sure, someone with the knowledge could try all the ports IF they know you are 
even running SIP, but it certainly will stop most of these idiots .

That along with fail2ban, not using numbers for device user names all will help.

Using IAX where possible also can be very effective

John Novack
Steve Murphy wrote:




On Sat, Jan 18, 2014 at 3:59 PM, Steve Edwards asterisk@sedwards.com 
mailto:asterisk@sedwards.com wrote:

On Sat, 18 Jan 2014, Jerry Geis wrote:

I see MANY of these in my log files:

[Jan 15 03:06:12] NOTICE[14129] chan_sip.c: Registration from '202 
sip:202@X:5060' failed for '37.8.12.147:26832 http://37.8.12.147:26832' - Wrong 
password

What is the correct way to block these idiots so they
don't even get this far.


Use iptables to allow packets from your legitimate users, block everybody 
else.

If you are dealing with a mobile user base or an extensive geographic area, 
at least block the countries where you do not expect traffic -- North Korea, 
China, xxxistan, etc.

Drop these at the front door (90% of the problem) and use fail2ban to pick 
off the rest.


I see a problem here; firstly that it is no longer so simple to determine
the IP ranges of countries. Things have been fractured quite a bit; you
might have to hire out a service to determine true geographic origination.
Even then, if your service is a little behind, you might occasionally
feel the displeasure of users unable to talk to your servers. How will you
handle this, with a white-list? How much effort will you end up committing
to keeping your whitelist up to date?

Nextly, the well-financed operations running such probes need not use
machines in their native countries. There are plenty of US-based
machines that can be ( and are ) compromised.


In other words, don't forget the fail2ban part!

Here's another idea! How about changing your port from 5060 to something
different, maybe 7067 or some other number that is not popularly being used?
You'll provision your phones to use this port, and the scanners will not
find you. Seems a much simpler solution... but there are some drawbacks...
can anyone think of them? And will these drawbacks matter to you? And, given
this solution, will the odds that a scanner might find your machine be so low,
that it is not worth using something like fail2ban to override them? Food
for thought!

murf

--

Steve Murphy
ParseTree Corporation
57 Lane 17
Cody, WY 82414
?  murf at parsetree dot com
? 307-899-5535






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Re: [asterisk-users] Integration with NEC DSX - help with dial line

[John Novack]

Stephen More wrote:

I am trying to setup an extension in asterisk which dials an extension
on the NEC DSX. i.e. If an asterisk user dials 402 I want it to
connect to the NEC DSX @ 192.168.1.57 and connect to extension 402. (
404 would be the NEC DSX sip account that I have the credentials for
).

[402]
deny=0.0.0.0/0.0.0.0
secret=pass1
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/404:pass2@192.168.1.57/402
mailbox=402@device
permit=0.0.0.0/0.0.0.0
callerid=device 402
callcounter=yes
faxdetect=no


[Nov 14 10:35:45] VERBOSE[13117][C-0010] app_dial.c: -- Called
SIP/404:pass2@192.168.1.57/402
[Nov 14 10:35:45] VERBOSE[11623][C-0010] chan_sip.c: -- Got
SIP response 480 Temporarily not available back from
192.168.1.57:5060
[Nov 14 10:35:45] VERBOSE[13117][C-0010] app_dial.c: --
SIP/192.168.1.57-0019 is circuit-busy
[Nov 14 10:35:45] VERBOSE[13117][C-0010] app_dial.c:   == Everyone
is busy/congested at this time (1:0/1/0)


I tried IAX2, but then I just get ring no answer.


What am I missing in the config - I simply want a one to one mapping.


-Thanks


The NEC DSX does NOT support IAX2.
Sorry to say that, but since IAX2 was never submitted to the rigors of becoming 
an accepted standard, there have been few adopters.
Digium chose back in 2005, I believe, perhaps earlier, not to put the effort 
into whatever it took to get IAX2 accepted as an international standard.

I have Asterisk send a page to my NEC DSX every 15 minutes during waking hours 
that announces the time, as well as some other information.
First, make sure your DSX has the latest firmware. This is easily done through 
the system administrator
Of course you will need the IP daughter board installed
Note that I use a non standard port, as I have my system linked to another DSX 
system over the Internet, and in addition to firewall settings, we chose to 
move the sip control port to discourage hackers
Obviously, if this is ONLY within your LAN, that isn't necessary.
You will also need to set up an extension number in the 4XX range BEYOND the IP 
boards possible extensions, why we chose 421
I do NOT use this to place a SIP call TO Asterisk from the DSX
Hope this helps. If you need more information, feel free to contact me off list
John Novack

Here is my sip.conf:
register=421:password@172.16.0.235:6065;
;
[421]
allow=ulaw
context=internal;
type=friend
username=421
secret=password;
port=6065
host=172.16.0.235 ; DSX IP address
fromuser=421
fromdomain=bigjohnnovack.ckts.info ;
dtmfmode=inband
qualify=yes
accountcode=NECDSX
maxexpirey=3600
defaultexpirey=160
srvlookup=no
canreinvite=yes

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Re: [asterisk-users] issue after install dahdi

[John Novack]

A VERY OLD and beyond EOF version.
If you MUST, due to some driver issue, use Asterisk 1.4, then please use 1.4.44
Otherwise I suggest you move to something more current, either version 
1.8.current or beyond.
Also, CLI says 1.4.43, your message says 1.4.32 ???

Some examination of chan_dahdi and your dialplan would help someone give you 
some assistance.
Is this a fresh install, or one that has been working for years?

What Digium card?

John Novack

Salaheddine Elharit wrote:

i need your help regarding some issue related to the outband calls

i have installed asterisk 1.4.32 with dahdi and i have 1 card diguim with 2 
ports
when i try to call my phone number all time i receive message  busy number

this error just with g1.

with g2 there is no problem i can call without issue

can anyone see the CLI and tell me what is the problem

thanks and regards

  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.4.43.18495-AheevaCCS-3.2.12 currently running on 
SRVRADIO (pid = 4147)
Verbosity is at least 3
-- Executing [0661049303@agents:1] Set(SIP/223-0021, CALLERID(number) 
 =520460587) in new stack
-- Executing [0661049303@agents:2] Dial(SIP/223-0021, DAHDI/g1/066104 
 9303|30) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/0661049303
-- Moving call (DAHDI/3-1) from channel 3 to 2.
[Oct 21 17:43:27] WARNING[4264]: chan_dahdi.c:9438 pri_fixup_principle: Can't 
mo  ve call (DAHDI/3-1) 
from channel 3 to 2.  It is already in use.
[Oct 21 17:43:27] WARNING[4264]: chan_dahdi.c:9558 pri_find_fixup_principle: 
Spa  n 1: PRI requested 
channel 1/2 is not available.
-- Hungup 'DAHDI/3-1'
  == Everyone is busy/congested at this time (1:0/0/1)
-- Executing [0661049303@agents:3] Hangup(SIP/223-0021, ) in new 
sta  ck
  == Spawn extension (agents, 0661049303, 3) exited non-zero on 
'SIP/223-0021'
-- Executing [h@agents:1] GotoIf(SIP/223-0021, 0?3:2) in new stack
-- Goto (agents,h,2)
-- Executing [h@agents:2] AHEventsProxy(SIP/223-0021, MSG_TYPE_TERMIN 
 ATE_CALL1382377407) in new stack
 AHEventsProxy: Channel [SIP/223-0021]. Data 
[MSG_TYPE_TERMINATE_CALL138  2377407]
-- chan is SIP/223-0021
 AHEventsProxy: Send To CtiServer: socket:[89]. 
message:[41,1382377407stcrpb  x^~]
-- Executing [h@agents:3] Hangup(SIP/223-0021, ) in new stack
  == Spawn extension (agents, h, 3) exited non-zero on 'SIP/223-0021'
-- SIP/224-0020 is ringing
SRVRADIO*CLI
Disconnected from Asterisk server
Executing last minute cleanups







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Re: [asterisk-users] is g729 codec free? or under license???

[John Novack]

Darryl Moore wrote:

Thank you Steve, and I read a bit more on the web on this subject
including your own well reasoned page at
http://www.soft-switch.org/patents/index.html

However, despite wide acceptance of the patentability of such codecs
(unfortunately), whether they are in fact software patents or not
appears to be a matter of opinion. The FSF and Fedora both refer to
codec patents as being software patents.

http://endsoftpatents.org/2011/02/usa-patent-reform-not-enough/
http://fedoraproject.org/wiki/Software_Patents

A quick google search of both terms will show that there are a great
many people who see codec patents as software patents, so I don't think
I am alone there.

snip

Law is ALWAYS open to interpretation, so that is not surprising.
See if you can get any lawyer, and especially a patent attorney, to give you a 
definitive answer! You will not get one.
Seldom will you ever get an eggspurt legal opinion Any good lawyer will tell you 
maybe, or if there is any doubt don't do it!
Law is not precisely measurable. No meter or O'scope to assist here.
Any A**hole can sue anyone for the filing fee, and the results are up to the 
opinion of a judge or jury.
The lawyers want it that way, so it isn't ever going to be any different.

John Novack

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Re: [asterisk-users] proper use of Internal Timing

[John Novack]

And here I thought I was back in the dark ages using 1.4.44!!

You had better consider moving up to a more current version before you get bit 
real hard!

John Novack

Comp Aholic wrote:

Hi All,

Could anyone tell me the real use of  internal_ timing=yes option on 
asterisk.conf file? I am using asterisk 1.4.22.

As per my understanding if we don't have any TDM card installed with 
appropriate driver, we use internal_timing = yes to get the timing from ztdummy 
/ztDahdi.

Is there any advantage on enabling  internal_timing=yes even if we are 
proving timing from TDM card?


I would really appreciate your feedback.


Thanks
Sam


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Re: [asterisk-users] Set ringtone by dialed number

[John Novack]

A J Stiles wrote:

On Tuesday 23 July 2013, Josh Hopkins wrote:

The reason for this is we have one primary company office but there are two
entities and if someone call the Denver number it would be for 1
organization and would ring differently helping our staff remember how to
answer the phone for the Denver organization rather than for the Colorado
Springs number which is a different entity.   A lot of times people are
rushing to answer the phone and do not look at the callerID this would
give them and auditory reminder of how they need to answer the phone.

But that is not what you asked the first time around.

Well, not exactly.
He asked how to change the ring TONE


If you want to change the ringing tone *at the far end*

More properly called ringBACK for those who are knowledgeable in telephony.

depending on who is
calling, that's different.  That is just ordinary distinctive ringing, and
Asterisk most certainly supports it  (in fact, even analogue phones on an FXS
card can be given different ringing envelopes; the usual ring-ring, the
French-style riing, or even a ring-ring-ring).

But, does Asterisk support distinctive ringing on SIP phones? Isn't that 
governed within the SIP phone itself

Let's not confuse different ringing patterns on analog circuits with SIP.

John Novack


Your Asterisk at the far end just has to be able to pick up on the caller ID
so it knows where the call is coming from, and  execute a SipAddHeader()
statement to set the appropriate ringing tone.


How would I go about setting up telling the phone to change the ring tone
in the SIP header?

Read the documentation for your phones.  For Digium D40s, look here:

https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=21463877#XMLConfiguration1.1.x-
RingtonesElement

  


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Re: [asterisk-users] recommendations for RJ-11 surge supressors?

[John Novack]

A J Stiles wrote:

On Thursday 27 June 2013, Eric Cooper wrote:

I'd like to protect my expensive Digium FXO cards from spikes on my
three incoming PSTN lines.  Does anyone have any recommendations?

Does your telco not fit surge suppressors to the NTE as a matter of standard
practice?  Perhaps we are spoiled in the UK .


Generally telcos in the US provide protection for THEIR equipment, NOT the 
subscriber.
Of course, dial tone providers via cable provide none other than the RF cable.

Always best to provide additional protection regardless.

John Novack

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Re: [asterisk-users] Joining an astablished call

[John Novack]

In the telephony world that is known as barge-in and is a programmable option 
granting that right to specific extension(s) in systems that normally have automatic 
privacy. Not all electronic key and hybrid systems have automatic privacy, though most do.

John Novack


neo haux wrote:

Hi,

I don't know how to call this functionality, but what I want to do is join an 
already established communication between PSTN---FXS_connected_phone using my 
SIP phone (I have an asterisk v11 with digium TDM400P at home)

Is it possible? What I don't want is using the conference sound and menu 
It's just a normal call between to channels that I have to  join for few 
minutes.

Regards





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Re: [asterisk-users] What would cause a drop between two asterisk systems?

[John Novack]

Carlos Alvarez wrote:



On Tue, Mar 5, 2013 at 2:32 PM, Hose hose+aster...@bluemaggottowel.com 
mailto:hose+aster...@bluemaggottowel.com wrote:

We have an asterisk frontend terminating all our SIP phones to, and an
asterisk backend with a wildcard PRI card in it connecting to the PTSN.
The frontend handles 99% of dialplan logic and just hands off anything
outgoing to the backend via IAX2, which dials out on one of the open
channels.


IAX is buggy.  We've never seen a reliable system using it.  We've given up on 
it.

I have seen this assertion from time to time, but never any real details

There is a world wide network of users who communicate using IAX, and many with 
PSTN service from providers using IAX. with no complaints
Can someone please provide meaningful details on what buggy really means? 
Rather than such a sweeping condemnation. If it is so buggy, why isn't it either fixed or 
discontinued?

It certainly is much less prone to hacking and abuse than SIP. Probably not due 
to the protocol design as much as it isn't as universal

John Novack


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Re: [asterisk-users] fail2ban filter issue

[John Novack]

eherr wrote:


Not sure if this has been answered but I cannot find a solution.

I am running Asterisk 1.4.26.3


Very VERY old. If you need to continue in 1.4, you should really up to the last 
1.4.44
Many MANY changes and broken changes between 1.4.26 and 1.4.44

John Novack


I am seeing the following lines in my log files:

A: [2013-03-05 13:54:27] NOTICE[6928] chan_sip.c: Failed to authenticate user 
sip:192.210.138.12;tag=DmVIjOlfYiiL

B: [2013-03-05 12:20:00] NOTICE[6928] chan_sip.c: Failed to authenticate user 
101sip:1...@my.asterisk.server.ip;tag=eec630f1

C: [2013-03-03 05:15:02] NOTICE[31158] chan_sip.c: Registration from 
'101381sip:101...@my.asterisk.server.ip' failed for '85.25.23.129' - No 
matching peer found

Now Two Part Question:

Part 1:

I understand that line C is from some soft phone like Xlite, IP phone, or 
program trying to register extension 101381 to my server and the user exten 
does not exist. I don't understand the method for A or B. I don't understand 
what generates that error message. Can someone explain?

Part 2:

Fail2ban blocks line C as per the regex in filters/asterisk.conf. What I don't 
understand is why doesn't lines A or B have a built in regex line? This goes 
back to not knowing the method that generates the error message in part 1.

Also, can I update the regex in asterisk.conf,

From: NOTICE.* .*: Failed to authenticate user .*@HOST.* 
mailto:.*@%3cHOST%3e.*

To:  NOTICE.* .*: Failed to authenticate user .*

It should ban both A and B, along with the original Regex line that I modified.

Question is, would this present a problem under normal circumstances? I know 
when the line comes up with my.asterisk.server.ip it will get ignored because I 
am in the ignoreip list but I want to make sure it will be OK to adjust.

Thanks community!

-E



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Re: [asterisk-users] GSM Sip Gateway

[John Novack]

From the Freq. list given on eBay, I don't thinkthey are. The listed freqs. are 
worldwide GSM since the mid 90's, but not 4G

John Novack

Hans Witvliet wrote:

Are these 4G comaptible 


-Original Message-
From: Frank fr...@efirehouse.com
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] GSM Sip Gateway
Date: Sun, 24 Feb 2013 07:40:19 -0500

USA, this will be use with a 4G network.

On Feb 24, 2013, at 5:24 AM, longst longst...@gmail.com wrote:


where are you from by the way

Sent from Shitian Long


On Feb 24, 2013, at 1:54 AM, Frank fr...@efirehouse.com wrote:


Hi all,

Anyone ever used GoIP GSM SIP Gateways ?
If yes, what was your experience with those ?

I'm looking at this:
http://www.ebay.com/itm/HOT-GSM-VOIP-GoIP-Gateway-SIP-Trunk-to-Asterisk-iP-PBX-/280736774012?pt=US_VoIP_Business_Phones_IP_PBXhash=item415d37377c

If anyone has any (good) experience with another brand, I'll take the names and 
models.

Thanks

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Re: [asterisk-users] GSM Sip Gateway

[John Novack]

Check with your provider. I believe these gateways will be usable for quite some 
time to come. In the US ATT and T-Mobile are the National GSM networks. Since 
these gateways are voice only with not data, I just bet they will continue to work 
just fine for many years. 3G/4G is mostly used for data and text? so it may be a 
non issue

John Novack

Frank wrote:

So anyone would know a gateway working on 3G/4G network ?
I remember a website called XY something (I cant find it anymore. I don t 
remember if it was xywireless.com , or xytelecom.com , or something else) where 
they seemed to have good gateways, but I can't find it anymore.

On 2/24/13 9:15 AM, John Novack wrote:

 From the Freq. list given on eBay, I don't thinkthey are. The listed
freqs. are worldwide GSM since the mid 90's, but not 4G

John Novack

Hans Witvliet wrote:

Are these 4G comaptible 


-Original Message-
From: Frankfr...@efirehouse.com
Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] GSM Sip Gateway
Date: Sun, 24 Feb 2013 07:40:19 -0500

USA, this will be use with a 4G network.

On Feb 24, 2013, at 5:24 AM, longstlongst...@gmail.com  wrote:


where are you from by the way

Sent from Shitian Long


On Feb 24, 2013, at 1:54 AM, Frankfr...@efirehouse.com  wrote:


Hi all,

Anyone ever used GoIP GSM SIP Gateways ?
If yes, what was your experience with those ?

I'm looking at this:
http://www.ebay.com/itm/HOT-GSM-VOIP-GoIP-Gateway-SIP-Trunk-to-Asterisk-iP-PBX-/280736774012?pt=US_VoIP_Business_Phones_IP_PBXhash=item415d37377c

If anyone has any (good) experience with another brand, I'll take the names and 
models.

Thanks

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Re: [asterisk-users] How to connect a POTS robo alert dialer to asterisk with email notification

[John Novack]

David Smiley wrote:

Hi John,

On Feb 2, 2013, at 6:52 PM, John Novack jnov...@stromberg-carlson.org wrote:

Sounds like a nice application for an HP Thin Client, AstLinux and an FXS ATA
Even simpler would be an ATA and a VOIP service, though I don't think much of 
SIP exposed to the Internet. I suppose since this will only be placing calls 
from the ATA though.

I can't seem to find a VOIP provider that's pay-per-call (what I want) vs 
pay-per-month (what I don't want).  At least I don't want to pay more than a 
few bucks/month for how little I'll use it.

voip.ms is a prepaid service with a very low monthly fee for their value 
service. $12-$24 per year plus minutes depending on the rate center. I assume 
US.
Certainly a better deal than Magicjack.
OOMA is more up front, and I believe they now have some monthly fee?
Also, do you not ever go to this location and want phone service during that 
time?

But I think I just came up with my solution.  A cheap ~$60 temperature monitor 
that uses standard phone service, coupled with Magic Jack.  The Magic Jack Plus 
is $70 up-front and $30/year; not bad.  I'll look some more for other options 
first.


What would you do IF there is an alert?Do you have someone who could respond 
that is nearby and is able to fix whatever is wrong?

Yes; I have a winter care-taker who happens to live next-door.

~ David



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Re: [asterisk-users] How to connect a POTS robo alert dialer to asterisk with email notification

[John Novack]

David Smiley wrote:

Hello!

I'm totally new to the world of Asterisk, and so my apologies in advance if my 
question has been asked before or is in the manual because I'm too new to even 
know what to search for.

I own a property far-away that isn't inhabited year-round.  I want to come up 
with a low-temperature alert system so I can be notified that there is 
insufficient heat for whatever reason (e.g. failed boiler, or ... ?).  There 
are some systems in the several hundreds of dollars price range that could 
either hook up to my WIFI to a monitoring service (sometimes with monthly 
fees), or a cell-phone based one that sends a text message.  Then there are 
inexpensive ones for about $60 that can hook up to a plain old telephone jack 
and dial a number with an automated voice to alert the receiver of the problem. 
 But I don't want to buy phone service to this place just for this device.

So I'm wondering if I could buy an adapter of some sort with a phone port and ethernet 
port.  An ATA?  But then I'm sure it'd need to talk to some sort of VOIP 
service.  Where I live year-round I have an underpowered build-your-own HTPC computer 
that stays on the internet all the time and occasionally I access it remotely.  Perhaps I 
could install asterisk there.  But then I have no idea.  Ultimately I want to get 
notified somehow (e.g. email) that this phone dialer sent an alert.  Maybe this is more 
trouble than its worth :-)

~ David

Sounds like a nice application for an HP Thin Client, AstLinux and an FXS ATA
Even simpler would be an ATA and a VOIP service, though I don't think much of 
SIP exposed to the Internet. I suppose since this will only be placing calls 
from the ATA though.

What would you do IF there is an alert?Do you have someone who could respond 
that is nearby and is able to fix whatever is wrong?


John Novack


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Re: [asterisk-users] Paging for Praying

[John Novack]

Shaun Ruffell wrote:

On Fri, Dec 28, 2012 at 06:41:38PM -0800, Steve Edwards wrote:

On 12/28/2012 08:13 PM, Steve Edwards wrote:


Please don't top-post. If you don't know what that means, please
consult Google.

On Fri, 28 Dec 2012, jon pounder wrote:


Please stop saying don't top post, some of us prefer it that way.

Besides being my preference, it is the documented rule of the
mailing list:

http://www.asterisk.org/community/discuss/

Note Mailing List Rules, #5.

For a walk down memory lane on top vs bottom posting on the Asterisk
mailing lists:

http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/254997


I would add that the rule # 5 was added long after the first 4, by someone in 
charge after one of the many times this subject has popped up.

Many of the same complainers  routinely do not remove the multi line footers, 
sometimes MANY of them, forcing those who really want to read a reply to wade 
through  the mess. Seems some can't be bothered to delete them

I would also add that rules are made to be broken!

Peg Leg O'Brien


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Re: [asterisk-users] Need qualifications of SIP trunk providers

[John Novack]

Daniel - Asterisk wrote:

Thank you Carlos,
What does mean 'por-out'?
I'm expecting 1 min/month in  out.
Elder


PORT out = Port or move the number away from a provider

Seems this provider is unaware that one may have moved the number to another 
provider, and continues to charge when they no longer have the number.

One has to wait until the port is complete and successful to cancel the 
previous providers account. It seems the user needs to cancel the account with 
the mentioned provider

John Novack



On Thu, Nov 29, 2012 at 5:50 PM, Carlos Alvarez car...@televolve.com 
mailto:car...@televolve.com wrote:


On Thu, Nov 29, 2012 at 3:22 PM, Daniel - Asterisk earohua...@gmail.com 
mailto:earohua...@gmail.com wrote:

Hello List,
Since I'm looking for a new VoIP provider for US 
origination/termination, I will very appreciate if you can chare your 
experience with Flowroute, Vitelity and Voip.ms


Vitelity is reliable and decent, but no phone support.  Have not used the 
others.

Oh also if you lose a number on Vitelity to a port-out, they won't know and 
won't stop billing you for it.

What's your expected volume in/out?

-- 
Carlos Alvarez

TelEvolve
602-889-3003



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Re: [asterisk-users] Need qualifications of SIP trunk providers

[John Novack]

Several in our group use voip.ms and have no complaints at all

We had a few hickups when Sandy rolled through NYC ( we are all on the NYC 
server ) but voip.ms responded quickly to mirror to Seattle and there was 
little downtime, and what was lasted a very short time on one day.
voip.ms was very responsive during this time

We all also use the IAX protocol supported by voip.ms and have no complaints

John Novack

Daniel - Asterisk wrote:

Hello List,
Since I'm looking for a new VoIP provider for US origination/termination, I 
will very appreciate if you can chare your experience with Flowroute, Vitelity 
and Voip.ms
Thanks in advance!
Elder D. Arohuanca
dCAP 1497
Lima - Peru


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Re: [asterisk-users] dahdi firmware for centos 6

[John Novack]

Up one level are the DAHDI directories

http://downloads.asterisk.org/pub/telephony/

JN

Justin Killen wrote:


Those are asterisk downloads, not dahdi downloads

Justin Killen

--

*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Miguel Oyarzo
*Sent:* Wednesday, November 14, 2012 4:28 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] dahdi firmware for centos 6

Did you see this URL?

http://downloads.asterisk.org/pub/telephony/asterisk/

On Wed, Nov 14, 2012 at 4:24 AM, Justin Killen jkil...@allamericanasphalt.com 
mailto:jkil...@allamericanasphalt.com wrote:

In http://packages.digium.com/centos/ there is not yet a centos 6 branch (Nor 
is there a RHEL 6 branch).  Centos 6.0 was release in July of 2011 -- is this 
something that Digium is planning on supporting?  Or is there a different URL 
that I'm not aware of for firmware packages?

-Justin Killen


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http://au.linkedin.com/in/mikeaustralia
Melbourne, Australia

==



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Re: [asterisk-users] My digium card die?

[John Novack]

PedroTron wrote:

2012/9/3 Andrew Colin a...@syrex.cc:

Can you confirm Dahdi is loaded correctly

What does the output of dmesg show?


dmesg output

[   22.183501] dahdi: Telephony Interface Registered on major 196
[   22.183504] dahdi: Version: 2.5.1
[   22.275018] Invalid/unknown operating mode 'COLUMBIA' specified.
Please choose one of:
[   22.275021]   FCC
[   22.275022]   TBR21
[   22.275023]   ARGENTINA
[   22.275024]   AUSTRALIA
[   22.275025]   AUSTRIA
[   22.275025]   BAHRAIN
[   22.275026]   BELGIUM
[   22.275027]   BRAZIL
[   22.275028]   BULGARIA
[   22.275028]   CANADA
[   22.275029]   CHILE
[   22.275030]   CHINA
[   22.275031]   COLOMBIA

snip

Spell COLOMBIA correctly! in the conf file
It might then begin to work, or on to the next error!!

John Novack

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Re: [asterisk-users] Asterisk on Dynamic IP to a SIP extension

[John Novack]

Doug wrote:

Verizon has put another good third party DSL supplier out of the DSL business. 
Their mindset is to kill the competition and then kill DSL and copper 
althogether in FIOS areas.

So am soon losing my static IP and I need to prepare for the change. I 
currently have Asterisk running using, besides local extensions, a remote SIP 
extension in another state. In the new configuration both Asterisk and the 
remote extension will be behind dynamic IP.

I will be running dyndns or equivalent and likely ddclient to update IP's.  
Will there be any issues running in this way? Will Asterisk ride through an IP 
change without a restart? If there is a definitive wiki topic on this please 
pass me the link.


I have used Asterisk and dyndns with ddclient for years, with no issues. I 
can't speak to off premiss SIP phones, but if it will work with a URL rather 
than just an IP address, all should go well.

You could always provision an embedded Asterisk thin client at the remote 
location, using IAX, and avoid any issues with SIP if there are any concerns 
with security.

ddclient is configurable to do any restarts or changes that might be necessary 
should an IP address change.

I am told that Comcast, which I am hoping to get, has sticky dynamic IP 
meaning the IP addresses rarely if ever change. If that is the case then this is pretty 
much a non issue. I think they use the router mac address to assign an IP address.


True enough. Comcast will change your IP if the mac address of the router 
changes, but since most routers allow Mac address cloning, even this is a 
minimal issue. As Comcast in many areas is growing, sometimes the IP address 
changes as they reconfigure their network. Mine hasn't changed now for about 18 
months though. Plus their service is reliable. Not quite as robust as a 
telephone company in the 1980's, but pretty good.

Also the version of Asterisk I am running is old - 1.2.35 - yes I know it's old 
but it works and does what I need. Are there differences in versions on how the 
above would work?


I would suggest you move at least to the last 1.4 version . though 1.4 is also 
EOL, the last version is fairly robust, and obviously you don't need any of the 
newer features added later on. Some changes will be needed in your dialplan, 
but not nearly as many as if you were to become current.

I use my Asterisk in a worldwide private network, as well as an interface to 
voip.ms, and it just works.
I also use IAX for all external connections with good results.

Best of luck

John Novack




Thanks, Doug


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Re: [asterisk-users] VOIP PBX replacement suggestions?

[John Novack]

Daniel Seagraves wrote:

The boss wants to move from landline service to VOIP service as a cost-cutting 
measure. We have one voice line and one fax line. The telco is billing over 
$100 a month for the two. We're using Hylafax for faxing and a PBX for the 
voice line.

Our existing PBX is an Intertel Axxess box with the old v5 processor. The 
management and voicemail computer died years ago (PSU burned up). I'm worried 
that it's going to die before too much longer. We have the IPRC and several IP 
Phone+ devices. It's my understanding that the IP Phone+ speaks only a 
proprietary Intertel protocol and can never be used with any non-Intertel 
equipment. I would like to dump the entire Intertel box and move to Asterisk 
instead, but my budget for this project is exactly $0. I can't afford to buy 
new devices.

The boss is leaning toward getting digital voice service from the local cable 
monopoly. They want to charge us $30 a month per line to start, and we will 
have to sign a 3 year contract. The monopoly in question has a reputation for 
very poor service, but they are a monopoly so my boss sees them as the only 
alternative. My worry is that if we sign that contract, we are trapped with 
both the intertel and the cable monopoly, and if I exceed the capacity of the 
intertel (or it just dies) I am SOL.

My questions then are as follows:

1) Is there a way I don't know about to get Asterisk to talk to either the IPRC 
or the IP Phone+ directly in such a way that gets calls from one to the other?
   

No Intertel made sure of that long ago!

2) Are there any reputable VOIP providers that provide business service at a 
rate less than $30 per line per month? The boss is adamant that we need 
unlimited minutes.

   

Doubtful
voip.ms provides excellent service, but not unlimited minutes.
it can even work into an ATA outputting an analog line, then you could 
go to the input of the Intertel and if/when it dies completely move to 
another analog system or single line phones


You have been given an unreasonable charge. No budget but obtain the moon!

You or your boss will live to regret getting into any contract, and with 
a company that already has a bad reputation even more so.


One wonders how viable this business can even be, with one line, one 
fax, and no budget to replace an aged telephone system.


I do hope you are either independently wealthy or have other prospects 
for employment.


Just one old fart's opinion. Worth what you paid for it

John Novack

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Re: [asterisk-users] Caller ID : FSK ETSI or FSK US

[John Novack]

Hopeless:
Do you know that your provider is delivering CLID?
In the US, conventional providers charge extra for CLID and even more 
for CLID with name


Have you ever determined WHICH delivery system is used in your as yet 
undefined country?
Most systems are coded into Asterisk, but require Asterisk to be told 
which one to use

Some locations may not be covered

There should be no need for external hardware.

Quit thrashing about and resolve this issue in a methodical manner


Peg Leg O'Brien


Satria Anamarta wrote:

Thanks Mitul :)

The patch on the link is so old (2006-2007) so I think it's already 
implemented in the newest version. Honestly to say, I already try any 
combitions but still the caller id doesn't work :(


cidsignalling=bell,dtmf,v23
cidstart=ring,polarity,dtmf

with some parameter if we set it to dtmf

Hopeless :((

On Sun, Jun 3, 2012 at 4:51 PM, Mitul Limbani mi...@enterux.in 
mailto:mi...@enterux.in wrote:


Welcome to da Matrix :)

Look at this issue : https://issues.asterisk.org/view.php?id=6683

And try different combinations suggested over there, you might get
lucky :)

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 mailto:mi...@enterux.in
DID: +91-22-61447605 tel:%2B91-22-61447605
Cell: +91-9820332422 tel:%2B91-9820332422




On Sun, Jun 3, 2012 at 3:08 PM, Satria Anamarta
anam.satri...@gmail.com mailto:anam.satri...@gmail.com wrote:

Hello, All :)

Regarding to incoming caller ID on PSTN line, which one is
best supported by asterisk: is it FSK ETSI or FSK US?
I bought some caller ID converter hardware (convert DTMF to
FSK and vice versa) but still asterisk can not detect it.
The converter has a switch FSK ETSI or FSK US

This is what I put in /etc/asterisk/chan_dahdi.conf
...
cidsignalling=bell
cidstart=ring
...

If after buying this converter hardware and upgrade to dahdi
2.6.1 still not solve my caller id problem, I really dont know
what to do and feel hopeless :(

Thanks,
Anam.


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Re: [asterisk-users] How to stop ringing when incoming PSTN call is answered externally?

[John Novack]

Roger Burton West wrote:

On Tue, May 22, 2012 at 11:32:19PM -0400, ft...@mindspring.com wrote:
   

The calls are routed just fine, but when a call is answered at one of
the extensions or externally (by a home telephone) the asterisk
extensions continue to ring one more time.  Is there a way to have
Asterisk drop an incoming PSTN call as soon as it's answered?
 

I have the same problem, and earlier discussion here suggests it's
insoluble.

   

Nonsense
It is easily solved.
Simply answer through a supervised extension ( asterisk )
Don't expect analog lines to work properly with Asterisk and a POTS 
phone bridged, with the call answered on a POTS phone

Technically that is a poor method of connection.
Simply have the PSTN line go into Asterisk, and ALL analog phones in the 
home or business become extensions off of Asterisk


There ARE other ways, but it requires some coding and testing to get 
things to work, with no real gain
I am doing this to capture and screen caller ID numbers, but it is not 
the best solution


John Novack





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Re: [asterisk-users] Asterisk forward call

[John Novack]

Call forwarding ALWAYS sends a partial ring to the line. This is by 
design from ESS 1 days, and serves to remind the party that the line is 
forwarded.
Asterisk certainly can be configured to ignore the first ring in any 
number of ways.


John Novack

Guy Gold wrote:

On Tue,May 15 02:00:PM, motty.cruz wrote:
   

Hello All,
My Asterisk server is working fine except that at night I forward my number
to another phone number, however my asterisk server still rings once before
call is forward. My Local Phone provider is ATT and they said that there is
not way around it, I'm always going to get a partical ring.

Any suggestions how to stop the Asterisk from rining once before forward to
another number?
 

Hi Motty,
I'm assuming that you're forwarding calls unconditionally, yes ?

I haven't tested it for a while, but, I'm pretty sure that if
your PBX is not told to ring a device before forwarding the call, it
should not do so.
I do recall having worked in  a non-PBX office , and when we
performed CFWD, the local phone would ring once and then get
forwarded, but, that's because the local phone never took control
of the call coming from the carrier. In your case , the PBX can
take over the call, never produce a ring , and then dial the CFWD
number. I guess a trace of this instance can be useful .


Guy Gold

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Re: [asterisk-users] looking for solid state like PC suitable for Asterisk

[John Novack]

I use HP Thin Clients with AstLinux installed.
HP 5720's are available on eBay for not much money, or there are many 
small boards available new if you don't or can't use used.  10 watts, no 
fan, no HD


Not sure what might be available in your part of the world, but there 
are Sockris and ALIX flash based boards. AstLinux has special 
configurations for these.
I have 20-30 AstLinux on thin clients working without a belch on a 
private collectors network


John Novack


Bart Coninckx wrote:

Hi all,

for smaller (or maybe even bigger) sites I'm looking for a smaller, 
appliance-type like PC, preferably solid state and fanless PC.
Since it's only going to run Asterisk for a couple of extensions I 
don't think CPU and RAM need to be maxed out.


Does anyone have inspiration/experience for/about such a model?

thx!!

BC

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Re: [asterisk-users] looking for solid state like PC suitable for Asterisk

[John Novack]

Correct. I have never been accused of being a good speller!

JN


Bart Coninckx wrote:
That's Soekris I suppose. Never heard of them, but it looks mighty 
interesting.


Cheers,

BC


On 05/10/12 13:35, John Novack wrote:

I use HP Thin Clients with AstLinux installed.
HP 5720's are available on eBay for not much money, or there are many 
small boards available new if you don't or can't use used.  10 watts, 
no fan, no HD


Not sure what might be available in your part of the world, but there 
are Sockris and ALIX flash based boards. AstLinux has special 
configurations for these.
I have 20-30 AstLinux on thin clients working without a belch on a 
private collectors network


John Novack


Bart Coninckx wrote:

Hi all,

for smaller (or maybe even bigger) sites I'm looking for a smaller, 
appliance-type like PC, preferably solid state and fanless PC.
Since it's only going to run Asterisk for a couple of extensions I 
don't think CPU and RAM need to be maxed out.


Does anyone have inspiration/experience for/about such a model?

thx!!

BC

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Re: [asterisk-users] Looking for IAX trunk/DID to replace Junction Networks

[John Novack]

Joseph wrote:

On 04/24/12 16:57, Russell Horn wrote:

I received an email today from Junction Networks that they are
substantially increasing their monthly fee to the point that I'd be
cheaper getting a line from my local phone company. I'm now looking
for a replacement US carrier that supports IAX2.

I'm a home user and only need a single DID - preferably a local number
and only use a couple of hundred minutes of calls a month. All I need
is reasonably priced inbound and outbound IAX2 trunking, preferably
with the ability to set my CID.

If anyone has a low cost recommendation, I'd love to hear it.

Thanks,

Russell.


I'm using:
https://www.voip.ms/m/login.php?logout=true

it supports iax2, $5/month.


Voip.ms is high quality, handles number ports and supports both IAX2 and SIP
2 different pricing plans, and their costs range from 4.95 to 7.95 per 
month depending on the rate center for one plan, and less with no free 
incoming minutes for their value plan


Check out their excellent web site. Everyone I have suggested has been 
very pleased.


Several different server locations as well.
US and Canada
Prepaid  service
Not associated with voip.ms

John Novack

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Re: [asterisk-users] Installing Dahdi, libpri of different versions in one pc

[John Novack]

Why would you want to even bother testing EOL products, such as 1.4x and 
1.6.x.x?


Although I am a 1.4 Luddite, I really don't quite understand why you 
can't test with 1.8.x or 10, where you mihgt have a hope of getting 
something fixed if there is a problem, unless you already KNOW there is 
an issue with later versions.


JMO

John Novack


Gopalakrishnan N wrote:

Hi,

I would like to install Dahdi, libpri and Asterisk of different 
versions in one machine. Lets say, Asterisk 1.6.X, Asterisk 1.8.x and 
Asterisk 1.4.x to be installed in one machine, this can be done using 
prefix while building configure.


For dahdi, libpri can it be done in same way? Because I need to test 
telephony cards (PRI, BRI, GSM  Transcoding) with different versions 
of Asterisk, libpri and Dahdi, I can't remove and install again of 
each versions since it is time consuming, sicne there are lot of 
versions available.


Any comments would be appreciated.

Thanks.


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Re: [asterisk-users] asterisk distributions

[John Novack]

A J Stiles wrote:

On Thursday 01 March 2012, Ralph Green wrote:
   

Howdy,
   I have tried all of these and a few more.  PBXinaFlash gave me the
best results, by far.  AsteriskNow produced a basic working system.  I
could not get any of the others configured to work at all.  I should
tell you my restrictions.  I was evaluating these distros to see which
one I could use to teach at a local computer group.  I wanted to do
very little configuration through the command line,
 

And *that* is where you were going wrong.

Look, the command line is a fact of life.  Microsoft have spent a fortune
telling you that you're not smart enough to use it.  You do not have to fall
for that.  Are you going to sit back and let them call you stupid?

Think of trying to make yourself understood in a foreign country by pointing
and gesturing.  There comes a point where you will actually have expended
*more* effort than if you had just bitten the bullet and learned the language
in the first place.

   

since my goal was
not just to get a working system, but to have something I could easily
show others how to setup.
 

Again, this is where the command line excels.  Irrespective of how the user of
the computer has set up the GUI -- what icon theme they have selected or how
they have arranged the menus, which GUI tools are present and so forth -- the
command line method will always be the same.

You really aren't doing your students any favours if you are teaching them
blindly to avoid what is basically the most powerful feature of a GNU/Linux
system.

   
AstLinux is another great one. It has an easy to use GUI, but allows and 
requires editing of the Asterisk confs to build a working system without 
the drawbacks of the others mentioned
It also runs on small platforms, in 1 Gig ( or less ) of flash, no hard 
drive needed, and would allow the student to work with Asterisk without 
the overhead of learning Linux


Building a system from scratch and source code is certainly best however

John Novack


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Re: [asterisk-users] Same provider - IAX sounds bad, SIP sounds great

[John Novack]

Just to stir the pot a bit, I am a member of a worldwide private network 
of Asterisk and AstLinux users. the network uses IAX exclusively, and we 
have no issues relating to audio quality with a large variety of 
providers, routers, host machines, and expertise in configuration of the 
specific nodes
Many ( in the US and Canada ) use a PSTN connection as well as the 
private network using voip.ms with equally stellar quality using IAX


IAX was chosen as the default network protocol because of the many 
issues with SIP, routers, and ( later ) the many attempts at break-ins.


As an aside, didn't the manufacture of the Yugo die with the death of 
Yugoslavia?  Most of the Yugo's shortly thereafter? If anyone has one 
now it may be close to the value of a Delorian



No replies necessary or even desired.


John Novack

Carlos Alvarez wrote:

On Tue, Feb 28, 2012 at 6:14 PM, Alejandro Imassa...@p2ee.org  wrote:
   

Please expand as to how you set-up a SIP ATA behind a common home
router set-up, without port redirection and/or use of a SIP proxy
and/or STUN server? Unless the ATA has some sort of magic (e.g. VPN
support) it _cannot_ be done.
 

Go buy a WRT-54G or nearly any consumer-class router and just plug in
a SIP device.  Done.  It works.  We *never* work on customer routers
and very rarely have to tell them to reconfigure their router at all.
My own home configuration is an Airport Extreme with zero
configuration.  So either these are very old routers you're having a
problem with, or buggy SIP devices, or something else.

   

You should care.
 

Hmm, let me check the reading...

http://i1-win.softpedia-static.com/screenshots/Care-Meter_1.png

   

don't drive a Yugo but if I did I could easily be offended by the
pejorative use of the brand.
 

It's a piece of junk and everyone knows it, including the owners, so who cares?

   


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Re: [asterisk-users] SDP Issue

[John Novack]

Phil has been using his pseudonym for years, and Alex and his 
painful/painstaking posting is the only one I have seen even raising the 
issue.


Says even more about Alex than Phil


Peg Leg O'Brien


Alex Balashov wrote:

I wasn't so much poking fun at the substance of your post as the fact that 
you're the only person on this mailing list that posts with a pseudonym, and at 
that, one evocative of online gaming or forum environments.  It just doesn't 
fit with the culture or the relatively serious, substantive and adult-oriented 
tenor of this type of list.  Do you not notice that?

At the risk of being rude, --[ UxBoD ]-- is something that belongs in WoW or 
a phpBB board full of spotty adolescents.

If your real name is Phil, why not post as such?  Okay, so maybe you don't want to give out your 
surname for one reason or another--fair enough.  So, post as Phil, or Phil 
D., if your full name were Phil Deleterious.

There's no rule saying you have to.  However, the survival of most human social 
institutions, including those devoted to the exchange of knowledge, is upheld 
in part by adherence to some conventions of self-presentation and deportment.  
These conventions help delineate the identity and character of the venue to 
outsiders, and assist in self-knowledge and affirmation of that character 
internally.

Everyone else here posts with their full name because it communicates: I am a real, 
adult person solving real-world technical problems related to Asterisk. It is, at 
least in part, an affirmation of the fact that real personalities--real humans, real 
identities--underlie participation in Internet forums, especially specialised ones.  It 
is also a nod to the benificent academic origins of the Internet.  There are reasons for 
these conventions.  They encapsulate our creation mythos, and they tell us what kind of 
people we are, as a community.

Quite frankly, your From: display name spits on the pedigree, on the storied 
heritage of how this open-source community came to be.  It is not deferential 
to the accrued wisdom of Internet-focused technical specialists in areas such 
as Asterisk or IP telephony, and it does not hallow the ground on which we 
tread.  It says that the ROFLcopter has landed!!!111 and lol p0wned teh n00bs.

Except, you're being the n00b.  Come on, Phil.  Self-awareness is important.  I know I am 
being a self-important ass pontificating on this to you.  Are you okay with an ASCII art 
pseudonym that says, I'm a 14 year old playing WoW on a delapidated, slightly 
yellowed Windows tower draped in dirty underwear?  If not for you, why not for us?  
Please post with a real name.

--
This message was painstakingly thumbed out on my mobile, so apologies for 
brevity, errors, and general sloppiness.

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/

On Jan 24, 2012, at 6:02 AM, --[ UxBoD ]--ux...@splatnix.net  wrote:

   

LOL :) that really made me chuckle this morning; and very apt for the fact I 
did not post any fundamental details about the issue.  All points duly noted!
--
Thanks, Phil

- Original Message -

 

Is --[ UxBoD ]-- a first-person shooter gang--er, clan--name? Like,
one of those who rocket-jumps onto the platform and camps with the
grenade launcher, trying to stop the reds from capturing the blue
flag? I hate how the health and the ammo takes so long to respawn.
Is there any way to fix that in deathmatch?
   
 

--
This message was painstakingly thumbed out on my mobile, so apologies
for brevity, errors, and general sloppiness.
   
 

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/
   
 

On Jan 24, 2012, at 2:10 AM, --[ UxBoD ]--  ux...@splatnix.net
wrote:
   
 

--[ UxBoD ]--
 
   

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Re: [asterisk-users] asterisk 1.8.8 - caller ID not working.

[John Novack]

Joseph wrote:
snip

but not it is not working again.
I wish they stop screwing up with that Asterisk, they keep introducing 
new version and more bugs :-/
I just keep waisting my time hunting and fixing features that stop 
working.



Easy fix.
STOP using the latest and greatest or even near latest.

Revert to a version that works for your specific application, and leave 
it alone.


I am in a community that uses mostly 1.4, some 1.2, even a couple that 
still use 1.0, with a very few using 1.6 or 1.8
I remember 1.4 went through many versions with certain technologies 
broken, then fixed, then broken again


John Novack


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Re: [asterisk-users] Is Asterisk 1.4 compatible with 1.8.7 ?

[John Novack -W7P]

Asterisk 1.4 after, or beginning with, 1.4.26 DOES know about call 
tokens, so this must be upset about something else


John Novack


Danny Nicholas wrote:

Change requirecalltoken from auto to no.  1.4 has no knowledge of this
parameter so turning it on in 1.8 creates an incompatibility (IMO).

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Sunday, December 25, 2011 4:42 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Is Asterisk 1.4 compatible with 1.8.7 ?

After upgrading one of my server to asterisk 1.8.7.2  (the older is running
1.4.39)

When I try to dialin on asterisk-1.4.39 I get an error:
NOTICE[2414]: chan_iax2.c:9541 socket_process: Rejected connect attempt from
192.168.141.8, requested/capability 0x2/0x703 incompatible with our
capability 0xc.
NOTICE[2417]: chan_iax2.c:9541 socket_process: Rejected connect attempt from
192.168.141.8, requested/capability 0x2/0x703 incompatible with our
capability 0xc.

On asterisk-1.8.7 I get:
   WARNING[4277]: chan_iax2.c:10666 socket_process: Call rejected by
192.168.141.1: Unable to negotiate codec


I'm using ulaw / alaw code; why don't they communicate?

iax.conf (1.4.39)
[home_server]
disallow=all
allow=ulaw
allow=alaw

iax.conf (1.8.7)
[clinic_server]
disallow=all
allow=ulaw
allow=alaw
requirecalltoken=auto

--
Joseph

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Re: [asterisk-users] TE122

[John Novack -W7P]

Jerry Geis wrote:

 I have had a couple thunderstorms take out a card, again last night.
The card with dahdi show status still report OK both times.

When calling into the card I get all circuits are busy.

However, simply replacing the card did the trick. Before that
I stopped asterisk, restarted DAHDI, rebooted all those... The card
always reported OK with show status. Only after replacing the card
did it start to work again.

I am running 1.4.42 asterisk, 2.4.1.2 DAHDI and libpri 1.4.12.

So I am surprised that asterisk was not reporting the status as 
something other
that OK - also that 2 cards have gone bad seemingly related to 
thunderstorms.


Any thoughts?

Jerry


Clearly you need to provide better protection on your incoming circuit(s)

Keep in mind that the telco protection is to protect their equipment, 
not yours


This is the case regardless of the circuit provided.

John Novack


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Re: [asterisk-users] Delay before ringing from PSTN`s call

[John Novack]

You need to make sure your (DAHDI or ZAPTEL ) is set up properly for your 
country's CLID protocol
In the US CLID is sent between the first and second rings, and with a proper 
configuration Asterisk waits a ring before processing the call
Other parts of the world use different methods and protocols
You will need to dig into that first.

John Novack


neo haux wrote:

Hi

I am testing a degium TDP400P (2fxo+2fxs) on my asterisk

I configured incoming calls from pstn to ring my SIP phone (extension : 100)

cat  extensions.conf
...
[from-pstn]
exten = s,1,Dial(SIP/100,10)
 same = n,VoiceMail(100,u)




root@PC-debian:/etc/asterisk# cat dahdi-channels.conf
...
...
...
;;; line=1 WCTDM/0/0 FXSKS  (EC: MG2 - INACTIVE)
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel = 1
callerid=
group=
context=default
...
...
...

What I don`t understand is why the SIPphone rings after 3 secondes after 
Astererisk detects the incoming call. Moreover, after hanging off the external 
caller the SIPphone continue to ring for 3 seconds.

I did those modifications in the file  /etc/asterisk/chan_dahdi.conf without 
improuvement ( After restarting Asterisk)

[channels]
cidstart=ring
immediate=yes
faxdetect=no
usecallerid=no




Here is the debug from Asterisk console

*CLI -- Starting simple switch on 'DAHDI/1-1'
-- Executing [s@from-pstn:1] Dial(DAHDI/1-1, SIP/100,10) in new stack
  == Using SIP RTP CoS mark 5
-- Called SIP/100
-- SIP/100-0001 is ringing
  == Spawn extension (from-pstn, s, 1) exited non-zero on 'DAHDI/1-1'
-- Hanging up on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'


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Re: [asterisk-users] PSTN connectivity

[John Novack]

michael k wrote:

Thanks for the update. but how do i resolve this issue ? can you help me please 
?



Can you PLEASE take this to the FreePBX support group?

It seems obvious to most that therein lies the problem
You are thinking you wish to dial out through the X100, but Asterisk is 
attempting to dial out on a non existent SIP connection
Something isn't right in your dialplan, created by FreePBX


Also, no echo canceller on the X100 card isn't wise, but you will not realize 
that until you are able to use it!

John Novack



On Thu, Sep 29, 2011 at 1:00 PM, Sam Govind govoi...@gmail.com 
mailto:govoi...@gmail.com wrote:

Actually its easier. I haven't worked on FreePBX lately so what I remember 
is here: You've created a Zap trunk. Focus on the Dial Rule - You can keep it 
empty as well. Then you've created an outbound route its dial-rule is important.

But the funny thing which I didn't mention before is that you've ZAP 
defined in FreePBX but actually its DAHDI so I remember they've this cute 
parameter in amportal.conf which tells FreePBX to convert ZAP into DAHDI.


snip

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Re: [asterisk-users] Anybody using BinFone Telecom?

[John Novack]

I use Voip.ms and have a friend who uses BinFone. We both use IAX

He has had some issues lately, but it is unclear if it is binfone or his ISP. 
Losing internet connection and BinFone seems to fail to reconnect when his 
connection returns.

I have had no complaints with voip.ms
they have an excellent website with many options and easy configuration


John Novack



ft...@mindspring.com wrote:

Does anyone have any experience with BinFone for IAX termination?

They good look on the website, but I'm looking for any comments.

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

[John Novack]

jon pounder wrote:

On 09/25/2011 04:41 PM, Alex Balashov wrote:

Sometimes people get such swelled heads they need a slap back to reality - I 
completely agree with him the changes were idiotic.

Obviously the comments touched a nerve with you or you would not have replied.




And let's not even THINK about mentioning the version number sequence changes!!!

Peg Leg O'Brien




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.




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Re: [asterisk-users] Inter-astersik dialling encounteres no audio

[John Novack]

Lee, John (Sydney) wrote:


I have been deploying Asterisk (open source PABX) in the company which I work.

Sofar, all the Asterisk servers do not really talk to each other.  Recently, I 
am experimenting to dial from one Asterisk server to another through the WAN 
and I encountered a no-audio problem although the callee's phone can ring.

I understand that the no-audio means that SIP traffic (TCP/UDP 5060) is allowed 
to go through but not RTP (UDP 16384-32767).

Case A

==

This is a simplified diagram of how I am testing the dialling between 2 subnets.

In this case, phone A is registered in Asterisk A and phoneBis registered in 
Asterisk B.

Phone A -- Asterisk A -- Router A == WAN == Router B -- Asterisk B 
-- Phone B

Case B

==

However, before I have tested successfully using this kind of connection.

In this case, phone B1 and B2 are registered in Asterisk B although they are on 
different subnets.

Both phone B1 and B2 can ring and audio is allowed to pass through.

Phone B1 -- Router A == WAN == Router B -- Asterisk B -- Phone B2

I am mystified why audio is allowed go through in case B but not case A.

Can someone be kind enough to help me to understand why I have this problem?

If the router is blocking RTP traffic, then why is that I have no audio problem 
in case B?

Thanks in advance.



Why not use IAX

John Novack

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

[John Novack]

Voip.ms has excellent support if you need it, which many do not.
You log in to your account, then you can change from SIP to IAX, and if you 
click on the correct link they will give you your sample with your account 
information
You need to set up a registration line in IAX, then a context in IAX that 
points to a context in extensions.conf
Registration takes care of voip.ms finding you
their web site setup is about as complete a site as I have seen, with many more 
options than I would ever need
The only somewhat confusing issue is when using IAX they will not show you as 
registered
Your Asterisk will, though.

John Novack


naren wrote:


That's what I am hoping to do as well. Could you share some insight on how you set up the 
DID on the voip.ms http://voip.ms web site to forward to Asterisk using IAX? In 
particular I am trying to find out where you set the url / ip address of your asterisk 
installation on the voip.ms http://voip.ms web site.

Thanks!

On Tue, Sep 13, 2011 at 4:19 PM, Robert-iPhone rhuddles...@gmail.com 
mailto:rhuddles...@gmail.com wrote:

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

Sent from my iPhone



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

[John Novack]

naren wrote:


Ok that makes sense. I will take a look at my set up and see why it is not 
registering with voip.ms http://voip.ms.


Understand that with IAX, voip.ms will not show you as registered.
Your Asterisk should show you as registered from the CLI
CLI iax2 show registry
XX.XX.XX.XXX:4569 N   Y  ZZ.ZZ.ZZZ.ZZZ:4569 60  
Registered

X= Voip.ms server
Y=your account number
Z=Your IP address

In IAX general section:

register = Y:passw...@newyork.voip.ms ; change this to your server 
specified

this is shown in their example, with your data filled in

then your specific section for voipms:

[voipms];
;
type=friend
username=Y
secret=PASSWORD
context=from-voipms ; this points to your inbound context in extensions
host=newyork.voip.ms
disallow=all
allow=ulaw ;Codec 1 supported
allow=gsm  ; Codec 2 supported
insecure=port,invite ; from voip.ms example
requirecalltoken=no ; required after 1.4.26

Hope this helps

JN


I opened a ticket with voip.ms http://voip.ms as well about an hour ago. I do 
like their service as well, that is why I want to try and get it working with them.

Thanks John.

On Tue, Sep 13, 2011 at 5:29 PM, John Novack jnov...@stromberg-carlson.org 
mailto:jnov...@stromberg-carlson.org wrote:

Voip.ms has excellent support if you need it, which many do not.
You log in to your account, then you can change from SIP to IAX, and if you 
click on the correct link they will give you your sample with your account 
information
You need to set up a registration line in IAX, then a context in IAX that 
points to a context in extensions.conf
Registration takes care of voip.ms http://voip.ms finding you
their web site setup is about as complete a site as I have seen, with many 
more options than I would ever need
The only somewhat confusing issue is when using IAX they will not show you 
as registered
Your Asterisk will, though.

John Novack


naren wrote:


That's what I am hoping to do as well. Could you share some insight on how you set up 
the DID on the voip.ms http://voip.ms web site to forward to Asterisk using IAX? In 
particular I am trying to find out where you set the url / ip address of your asterisk 
installation on the voip.ms http://voip.ms web site.

Thanks!

On Tue, Sep 13, 2011 at 4:19 PM, Robert-iPhone rhuddles...@gmail.com 
mailto:rhuddles...@gmail.com wrote:

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

Sent from my iPhone



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

[John Novack]

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 http://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 
mailto:naren.sa...@gmail.com wrote:

Hi,

I am trying to set up my asterisk 1.8.5 with voip.ms http://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 they have on their website?

I have pretty much the same settings. When I dial out, the destination 
rings, but I can't hear a ringback tone from on the source side ( I am using a 
PAP2T router with a phone). I have set up outgoing with actionvoip before and 
that is working fine, so I am thinking my router settings for my ports are 
correct - but I am no expert.

I would really appreciate it if you could post the relevant section of your 
sip.conf for me.

Thanks!
Naren


On Thu, Jun 9, 2011 at 3:22 PM, Steve Edwards asterisk@sedwards.com 
mailto:asterisk@sedwards.com wrote:

On Thu, 9 Jun 2011, John Novack wrote:

I use voip.ms http://voip.ms and have no issues using IAX and 
Asterisk 1.4.xx


'slam-dunk.'


Though they suggest SIP, I chose IAX and have 4569 UDP open in my 
firewall


a

Their on line config samples just work!


is


Suggest you check your firewall and your configs, and above all 
post some more information


IAX


If you really want to upset some, top post as I have just done!


Agreed.


The real issue is communication, top bottom or in the middle


Sometimes, it's just about being considerate to 'the next guy.'

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-
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Voice: +1-760-468-3867 tel:%2B1-760-468-3867 PST
Newline  Fax: +1-760-731-3000 
tel:%2B1-760-731-3000


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Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system

[John Novack]

If you really want to go that route, you should also look at AstLinux and 
install it on an HP thin client such as a 5720. No Hard Drive spinning, and 
something like 30 watts. No fan either. All the asterisk files can be edited 
either through SSH or a web interface.

I have a bunch out working for a private VOIP  network of telephone collectors. 
Many also integrate PSTN lines through various means
5720's can be had on eBay for a LOT less money

John Novack


linux guy wrote:


Great discussion, all of it.  Thanks, people.

How much power does the home asterisk box need ?

I'm using Asus Eee Box (1012Ps) as Myth front ends in another project.  About 
$280 with 320 Gb hard drive and 2 GB RAM.  Atom 510 processor.  Built in Wifi.  
Nearly silent.  Runs F15 nicely.  Would one of them suffice ?

FWIW, I'm typing this email on one now because my main system is down.

It looks like I am going to need an ATA for the fax machines.  Two.  My wife 
informed me yesterday she wants her own in her office.  VOIP handles fax 
machines, right ?

I'm wondering what phones everyone is using.   Should I stick with analog 
wireless handsets or are there some good SIP wireless phones out there that I 
don't know about ???

Thanks.



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Re: [asterisk-users] Looking for ideas for nice home phone system

[John Novack]

C F wrote:

Everything you mention for the NEC
system is available with the 824
but the Lan/Wan programming. In fact on
the 824 every port (to a
system max of 24) or analog as well as
proprietary. Support upto 4
doors with a chime and external relay
(switch cameras etc).

Not quite - In addition to remote programming, the DSX series has single pair 
phones, and built in VM on a CF card. The 824 is an additional box.
Also the DSX has a SIP gateway option that in addition to supporting remote SIP 
phones, with latest software supports SIP trunks
The 824 also requires additional card(s) for CLID
The analog/prop ports are a nice feature, and also allows both a prop Panasonic 
and POTS phone on a single extension
I believe the 824 is also now MD, but some decent buys can be found in the used 
market.

Any of the mentioned products would be a better solution for a newbee. Hang on 
the wall, do some quick programming, and forget it for the next 20 years, 
barring a power or lightning hit.
No steep learning curve required. Certainly less expensive hardware

John Novack


On 8/22/11, John Novackjnov...@stromberg-carlson.org  wrote:

NEC-DX-40 is another best buy
Single pair phones
2 analog station ports
door box ports
AND remote programming via LAN or WAN
Voice mail available with or without email notification
SIP gateway option

Far superior to the TA-824 and Partners


John Novack


C F wrote:

Panasonic KX-TA824
Or the Panasonic KX-TAW848
Or the Avaya Partner ACS 8.0


On Mon, Aug 22, 2011 at 4:11 PM, Linuxguy123linuxguy...@gmail.com
wrote:

I'm looking for ideas for building a innovative, powerful home phone
system.

Something that does voicemail well, integrates cell phones into the
house system, etc.

I know there are a lot of details that need to be discussed, but lets
leave it at that for now.

What is everyone doing ?

Thanks !



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Re: [asterisk-users] Looking for ideas for nice **Asterisk** home phone system

[John Novack]

Linuxguy123 wrote:

My original post didn't mention it, but I would like my home system to
be Asterisk based.

So you really aren't looking for a nice home phone system, as stated in your 
original post.

If you want to learn and have a test bed and you live alone, then set yourself 
up with a system that is based on Asterisk, and have at it.
Learn, play, fail then succeed, partially, then progress.
Keep in mind that you will probably spend more in hardware, and if you have 
other household members they will not be happy with you messing around with 
their telephones.If you have little or no experience in telephony you will have 
a more difficult time, and may create a system that is more difficult for many 
users. Asterisk doesn't yet do a really good job of recreating the user 
interface of many modern hybrid key/pbx systems

You need to start by reading asterisk, the future of telephony A rather 
presumptuous title, but a good reference none the less.
If you search the recent archives of this list, you will see the trials and 
failures of Asterisk and Google

If this is to be used by you while you attempt to make a living on your own, I 
would say to you that is most unwise.
Good luck and revisit when you have something up and working and have some 
serious questions

John Novack


Has anyone figured out how to minimize cell charges when on the road via
making calls via the home phone system ?

Does anyone have their cell phone forwarded to their home phone system
and have it do their messaging ?

Is anyone using Google Phone capabilities in conjunction with Asterisk ?

Thanks !

On Mon, 2011-08-22 at 14:11 -0600, Linuxguy123 wrote:

I'm looking for ideas for building a innovative, powerful home phone
system.

Something that does voicemail well, integrates cell phones into the
house system, etc.

I know there are a lot of details that need to be discussed, but lets
leave it at that for now.

What is everyone doing ?

Thanks !




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Re: [asterisk-users] Looking for ideas for nice home phone system

[John Novack]

C F wrote:

On Tue, Aug 23, 2011 at 5:21 PM, John Novack
jnov...@stromberg-carlson.org  wrote:

snip

What do you mean by MD?




MD is a common telephony term for Manufacture Discontinued

John Novack

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Re: [asterisk-users] Looking for ideas for nice home phone system

[John Novack]

NEC-DX-40 is another best buy
Single pair phones
2 analog station ports
door box ports
AND remote programming via LAN or WAN
Voice mail available with or without email notification
SIP gateway option

Far superior to the TA-824 and Partners


John Novack


C F wrote:

Panasonic KX-TA824
Or the Panasonic KX-TAW848
Or the Avaya Partner ACS 8.0


On Mon, Aug 22, 2011 at 4:11 PM, Linuxguy123linuxguy...@gmail.com  wrote:

I'm looking for ideas for building a innovative, powerful home phone
system.

Something that does voicemail well, integrates cell phones into the
house system, etc.

I know there are a lot of details that need to be discussed, but lets
leave it at that for now.

What is everyone doing ?

Thanks !



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Re: [asterisk-users] Answering machine answers after pickup a phone.

[John Novack]

Completely normal operation.
You need to read and understand more basic telephony and analog lines to 
understand why that won't work.

Asterisk needs to be in control, and once someone answers a phone not under 
Asterisk control, or the call is abandoned there is little you can do.
Sounds like a task for a simple answering machine from Wal-Mart
All you other phones should be connected to FXS ports, or you need to be 
smarter in your dialplan.
Once you answer, Asterisk is behaving normally


John Novack



Jorge Barreiro wrote:

Hi again,

thanks for your answer, but it didn't solve the problem. That Dial command
returns inmediately, so I don't even have the delay.

I'll try to explain myself better. The PBX has only one FXO card, connected to
the PSTN. There is no other phones connected to the PBX nor SIP extensions.
There are analog phones connected to the same PSTN.

What I try to do is that, when there is an incoming call from the ouside, if
someone answers on a phone, then the PBX won't answer.


Thanks.

O Venres, 5 de Agosto de 2011 00:04:02 Ruben Rögels escribiu:

Hi,

your concept using Wait() won't work here.
Try it like this:

[incoming]
exten =  s,1,Dial(DAHDI/1234,30) ; This will ring the phone 30s
exten =  s,n,BackGround(wellcome-message)
exten =  s,n,Voicemail(1234)
exten =  #,1,Hangup()

So, of you answer the call within 30s, you'll get the call on your
phone. After 30s, the Voicemail will answer the phone.


regards,
Ruben

Am 04.08.2011 21:39, schrieb Jorge Barreiro:

Hello,

I'm configuring an Asterisk PBX to use as an answering machine. I have a
FXO card connected to the line, and other analog telephones connected to
the same line. The PBX answers and redirects you to the voicemail after
a delay.

The problem is that even if I pickup any analog phone in the line before
the PBX does, it answers after the delay anyway. And I couldn't find how
to prevent this, or even if this is supposed to happen.

My FXO card is a cheap X100P (source of problems, I know), and I'm using
the Asterisk version included in Debian Squeeze (1.6.2.9).
My dial plan looks like this:

[incoming]
exten =  s,1,Wait(8)
exten =  s,2,Answer
exten =  s,3,BackGround(wellcome-message)
exten =  s,4,Voicemail(1234)
exten =  #,1,Hangup

I don't know if this is related, but I'm in Spain and I had to add:
hanguponpolarityswitch=yes
to the chan_dahdi.conf file so that asterisk detects the remote hangup.
I also added:
answeronpolarityswitch=yes
but this didn't help. It seems to be used just to detect the answer when
you are calling, not when receiving a call.


I'd appreciate any help you could provide.

Thanks!

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Re: [asterisk-users] Answering machine answers after pickup a phone.

[John Novack]

Jorge Barreiro wrote:

O Venres, 5 de Agosto de 2011 17:42:28 Shaun Ruffell escribiu:

On Fri, Aug 05, 2011 at 01:14:58PM +0200, Jorge Barreiro wrote:

Hi,

thanks for your time!

O Venres, 5 de Agosto de 2011 12:35:05 escribiches:

Completely normal operation.
You need to read and understand more basic telephony and analog lines
to understand why that won't work.

I definitely have a lot to learn yet.


Asterisk needs to be in control, and once someone answers a phone not
under Asterisk control, or the call is abandoned there is little you
can do.

What I pretend is that asterisk detects that it's not under control and
gets out of the way. The same way it detects a remote hangup and stops
the dialplan, it could detect that someone else answered (the line is
not ringing anymore) and discard it the same way it does when the remote
part hangup.

I've read comments in forums and tutorials that seem to imply that this
happens, but I couldn't find any confirmation (and indeed, it's not
happening to me).

When I first installed Asterisk in my home I used it in the way that you
described: as a glorified answering machine to email to me any voice mail.

I think what you want is the WaitForRing()[1] dial plan application.  This
function will wait x number of seconds, then look for *another* ring to
come in. If someone answered the phone before the timeout to that function
Asterisk would stop processing the dial plan.

[1] https://wiki.asterisk.org/wiki/display/AST/Application_WaitForRing

I ran into a couple of issues with WaitForRing(). The first being if
someone answered the phone and then quickly hung up *and* a new phone call
came in within the timeout period, Asterisk wouldn't know that the line
was ringing due to a new call. The second problem was I never got the dial
tone detection working so that if I tried to *place* a call from Asterisk
while someone was on the house line I would aggravate my wife.

Since coming to work for Digium I've seen in the data sheets for the FXO
interfaces that there is a capability to detect when a parallel device on a
line goes off hook. This would allow Asterisk to have a better sense of the
state of the line (like it currently can detect when a port is unplugged
and there is not battery by generating a red alarm.) but I haven't looked
into getting that information off the hardware and up into Asterisk.

Hope this helps,
Shaun


That application looks like a good solution. I can't test it until Monday, but
I'll try it and let you know. The drawbacks you mention doesn't seem too
inconvenient in my case.

Anyway, I started with this cause I thought it was an easy first step, if it
gets so complicated I think I'll go forward and put all phones under the
control of the PBX.

Thank you everybody for your help.

the situation gets more complex if Caller ID is sent and one wants to act upon 
it, and Asterisk doesn't handle the call

In the US the data is sent between the first and second ring, and if the call 
isn't answered by Asterisk, it thinks that another call has arrived WITHOUT the 
information, and the situation falls apart rapidly or one needs to have some 
logic to figure out if this is the third ring on call #1 or a new call

Though the chipset in the X100P supports looking at the output port, I don't 
believe the driver fully supports it

using a TDM 4xx or even better a single port T1 card with a channel bank gives 
a lot more  ports for the cost

John Novack

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Re: [asterisk-users] Compact, affordable x86 devices?

[John Novack]

Just about any of the HP thin clients, either new or used off eBay, with 
AstLinux installed do a wonderful job, especially if you are not going 
to need a PCI card.
The older units will need a larger flash. Transcend has several 
different sizes that are direct replacements


Looks like some of the Neoware units will also do the job.

John Novack


Gilles wrote:

Hello

I'd like to build a compact, affordable, fanless x86 solution to
handle my home landline.

I know about the following two platforms:

1. www.pcengines.ch/alix.htm
alix1d + case 100€

Does Availability500 mean that it's just not possible to buy just
one item?

2. www.soekris.com/products.html?limit=all
net4501-30 Board and Case $175.00

Is the net4501 powerful enough to run Asterisk, considering that I'll
use an external VoIP gateway to connect it to my landline?

Are there other manufacturers I should know about?

Thank you.


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Re: [asterisk-users] Compact, affordable x86 devices?

[John Novack]

HP still does make Thin Clients, often with XP Embedded, though I have 
had very good results with many older used ones sold on eBay. With a new 
larger flash from Transcend, they simply work. Consider the used units 
not worn out but simply ones with more experience that probably won't 
fail. New units are always subject to infant mortality!!
The T5720 often comes with a large enough flash that replacement isn't 
needed, and reflashing with AstLinux can be done a number of ways, 
beyond the scope of this list. AstLinux has a web server for 
configuration, not sure about SQlite.

Check out their site for more details
there are other low cost solutions around as well.
the ALIX boards I have seen do not impress me. I think they are somewhat 
overpriced. Jut one opinion


John Novack

Gilles wrote:

On Mon, 18 Jul 2011 08:04:31 -0400, John Novack
jnov...@stromberg-carlson.org  wrote:
   

Just about any of the HP thin clients, either new or used off eBay, with
AstLinux installed do a wonderful job, especially if you are not going
to need a PCI card.
The older units will need a larger flash. Transcend has several
different sizes that are direct replacements

Looks like some of the Neoware units will also do the job.
 

Thanks for the tip. I'd like to buy the unit new: Are those devices
still manufactured? How easy is it to reflash them to run as a
stand-alone Linux host? Which device would you recommend to Asterisk
and a couple of other apps (small web server, SQLite, etc.)?

Thank you.


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Re: [asterisk-users] Conference feature

[John Novack]

Steve Edwards wrote:

On Sun, 26 Jun 2011, Rafael dos Santos Saraiva wrote:


I am referring to 3-way conference


With a little reading, you would discover that meetme can handle lots 
of participants.


For those who know Telephony, 3 way conference and meet me conference 
are NOT the same.


Someone needs to RTM on telephony!

John Novack

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

[John Novack]

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

[John Novack]

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

[John Novack]

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




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