Re: [asterisk-users] Asterisk 16 Certified 16.8 and MagicJack Incoming Calls

[Chris Dos]

It seems that if there is a pause in the auto attendant longer than a second
this problem occurs.  I have this for an extension in my extensions.conf file:

exten => 2799,1,GotoIf($["${CALLERID(num)}" = "${EXTEN}"]?500)
exten => 2799,2,Dial(PJSIP/${EXTEN},14,tr)
exten => 2799,3,Dial(PJSIP/${EXTEN},1,tr)
exten => 2799,4,BackGround(abandon-all-hope)
exten => 2799,5,BackGround(dial-here-often)
exten => 2799,6,Wait(2)
exten => 2799,7,BackGround(gambling-drunk)
exten => 2799,8,BackGround(you-seem-impatient)
exten => 2799,9,BackGround(nobody-but-chickens)
exten => 2799,10,BackGround(tt-somethingwrong)
exten => 2799,11,BackGround(tt-weasels)
exten => 2799,12,Voicemail(${EXTEN},ug(15))
exten => 2799,13,Voicemail(${EXTEN},bg(15))
exten => 2799,14,Hangup
exten => 2799,500,VoicemailMain(${CALLERID(num)})

If I change the Wait to 1 the MagicJack will hear everything.  If I change it
to 2, nothing is heard from that point on.

    Chris

On 6/1/20 8:43 AM, Chris Dos wrote:
> I upgraded our Asterisk 13 LTS to Asterisk 16 Certified 16.8 yesterday and
> converted form SIP to PJSIP using the python script as a start and then
> mofiying from there.  I ran into an issue when testing that incoming calls
> from MagicJack would go silent after about 10 seconds.  This happened while
> in the automated attendant area.  This problem did not occur with Asterisk
> 13 LTS.  I reverted PJSIP back to SIP and the problem still occurred, so
> that was not it.
>
> We connect to Flowroute for our SIP provider.  I added "force_avp = yes" to
> the Flowroute endpoint section in the pjsip.conf and the problem appeared to
> be solved after I tested it a dozen times.  However, this morning this issue
> has reappeared.  Any thoughts on what might be causing this?
>
> My Flowroute pjsip.conf config:
> [transport-udp]
> type = transport
> protocol = udp
> bind = 0.0.0.0
> tos = cs3
>
> [reg_us-west-wa.sip.flowroute.com]
> type = registration
> retry_interval = 20
> expiration = 120
> transport = transport-udp
> outbound_auth = auth_reg_us-west-wa.sip.flowroute.com
> client_uri = sip:12345...@us-west-wa.sip.flowroute.com
> server_uri = sip:us-west-wa.sip.flowroute.com
>
> [auth_reg_us-west-wa.sip.flowroute.com]
> type = auth
> password = XXZZXXZZXXZZ
> username = 12345678
>
> [reg_us-west-or.sip.flowroute.com]
> type = registration
> retry_interval = 20
> expiration = 120
> transport = transport-udp
> outbound_auth = auth_reg_us-west-or.sip.flowroute.com
> client_uri = sip:12345...@us-west-or.sip.flowroute.com
> server_uri = sip:us-west-or.sip.flowroute.com
>
> [auth_reg_us-west-or.sip.flowroute.com]
> type = auth
> password = XXZZXXZZXXZZ
> username = 12345678
>
> [reg_us-east-nj.sip.flowroute.com]
> type = registration
> retry_interval = 20
> expiration = 120
> transport = transport-udp
> outbound_auth = auth_reg_us-east-nj.sip.flowroute.com
> client_uri = sip:12345...@us-east-nj.sip.flowroute.com
> server_uri = sip:us-east-nj.sip.flowroute.com
>
> [auth_reg_us-east-nj.sip.flowroute.com]
> type = auth
> password = XXZZXXZZXXZZ
> username = 12345678
>
> [reg_us-east-va.sip.flowroute.com]
> type = registration
> retry_interval = 20
> expiration = 120
> transport = transport-udp
> outbound_auth = auth_reg_us-east-va.sip.flowroute.com
> client_uri = sip:12345...@us-east-va.sip.flowroute.com
> server_uri = sip:us-east-va.sip.flowroute.com
>
> [auth_reg_us-east-va.sip.flowroute.com]
> type = auth
> password = XXZZXXZZXXZZ
> username = 12345678
>
> [flowroute]
> type = aor
> contact = sip:12345...@us-west-wa.sip.flowroute.com
>
> [flowroute]
> type = identify
> endpoint = flowroute
> match = 147.75.60.160/28, 34.210.91.112/28, 34.226.36.32/28, 147.75.65.192/28
>
> [flowroute]
> type = auth
> username = 12345678
> password = XXZZXXZZXXZZ
>
> [flowroute]
> type = endpoint
> context = from-trunk
> dtmf_mode = rfc4733
> allow = !all,ulaw
> direct_media = no
> from_domain = us-west-wa.sip.flowroute.com
> tos_audio = ef
> tos_video = af41
> ; Note: "force_avp = yes" fixes issues with calls coming from MagicJack with
> no audio after a few seconds.
> force_avp = yes
> auth = flowroute
> outbound_auth = flowroute
> aors = flowroute
> t38_udptl = yes
> t38_udptl_ec = fec
>
> [anonymous]
> type=endpoint
> context = anonymous
> allow = !all,ulaw
>

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[asterisk-users] Asterisk 16 Certified 16.8 and MagicJack Incoming Calls

[Chris Dos]

I upgraded our Asterisk 13 LTS to Asterisk 16 Certified 16.8 yesterday and
converted form SIP to PJSIP using the python script as a start and then
mofiying from there.  I ran into an issue when testing that incoming calls
from MagicJack would go silent after about 10 seconds.  This happened while in
the automated attendant area.  This problem did not occur with Asterisk 13
LTS.  I reverted PJSIP back to SIP and the problem still occurred, so that was
not it.

We connect to Flowroute for our SIP provider.  I added "force_avp = yes" to
the Flowroute endpoint section in the pjsip.conf and the problem appeared to
be solved after I tested it a dozen times.  However, this morning this issue
has reappeared.  Any thoughts on what might be causing this?

My Flowroute pjsip.conf config:
[transport-udp]
type = transport
protocol = udp
bind = 0.0.0.0
tos = cs3

[reg_us-west-wa.sip.flowroute.com]
type = registration
retry_interval = 20
expiration = 120
transport = transport-udp
outbound_auth = auth_reg_us-west-wa.sip.flowroute.com
client_uri = sip:12345...@us-west-wa.sip.flowroute.com
server_uri = sip:us-west-wa.sip.flowroute.com

[auth_reg_us-west-wa.sip.flowroute.com]
type = auth
password = XXZZXXZZXXZZ
username = 12345678

[reg_us-west-or.sip.flowroute.com]
type = registration
retry_interval = 20
expiration = 120
transport = transport-udp
outbound_auth = auth_reg_us-west-or.sip.flowroute.com
client_uri = sip:12345...@us-west-or.sip.flowroute.com
server_uri = sip:us-west-or.sip.flowroute.com

[auth_reg_us-west-or.sip.flowroute.com]
type = auth
password = XXZZXXZZXXZZ
username = 12345678

[reg_us-east-nj.sip.flowroute.com]
type = registration
retry_interval = 20
expiration = 120
transport = transport-udp
outbound_auth = auth_reg_us-east-nj.sip.flowroute.com
client_uri = sip:12345...@us-east-nj.sip.flowroute.com
server_uri = sip:us-east-nj.sip.flowroute.com

[auth_reg_us-east-nj.sip.flowroute.com]
type = auth
password = XXZZXXZZXXZZ
username = 12345678

[reg_us-east-va.sip.flowroute.com]
type = registration
retry_interval = 20
expiration = 120
transport = transport-udp
outbound_auth = auth_reg_us-east-va.sip.flowroute.com
client_uri = sip:12345...@us-east-va.sip.flowroute.com
server_uri = sip:us-east-va.sip.flowroute.com

[auth_reg_us-east-va.sip.flowroute.com]
type = auth
password = XXZZXXZZXXZZ
username = 12345678

[flowroute]
type = aor
contact = sip:12345...@us-west-wa.sip.flowroute.com

[flowroute]
type = identify
endpoint = flowroute
match = 147.75.60.160/28, 34.210.91.112/28, 34.226.36.32/28, 147.75.65.192/28

[flowroute]
type = auth
username = 12345678
password = XXZZXXZZXXZZ

[flowroute]
type = endpoint
context = from-trunk
dtmf_mode = rfc4733
allow = !all,ulaw
direct_media = no
from_domain = us-west-wa.sip.flowroute.com
tos_audio = ef
tos_video = af41
; Note: "force_avp = yes" fixes issues with calls coming from MagicJack with
no audio after a few seconds.
force_avp = yes
auth = flowroute
outbound_auth = flowroute
aors = flowroute
t38_udptl = yes
t38_udptl_ec = fec

[anonymous]
type=endpoint
context = anonymous
allow = !all,ulaw
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Re: [asterisk-users] Cisco 3950 ip phone

[Chris Knipe]

Hi,

https://community.cisco.com/t5/ip-telephony-and-phones/cp-3905-asterisk/td-p/1995981


The phone does work, you do need to TFTP the configuration files to the
phone though.  Doesn't look like custom firmware is required.

--
Chris.


On Fri, Apr 12, 2019 at 3:29 PM Antony Stone <
antony.st...@asterisk.open.source.it> wrote:

> On Friday 12 April 2019 at 15:24:27, Gokan Atmaca wrote:
>
> > > Please give us a link to a datasheet for that device.
> >
> > Hello
> >
> >
> https://www.cisco.com/c/en/us/products/collateral/collaboration-endpoints/u
> > nified-sip-phone-3905/data_sheet_c78-651588.html
>
> Ah - that explains why I couldn't find a model 3950 phone :)
>
> Well, it uses SIP, so it should work as well as any Cisco phone does on
> Asterisk.
>
>
> Antony.
>
> > On Fri, Apr 12, 2019 at 3:58 PM Antony Stone wrote:
> > > On Friday 12 April 2019 at 14:42:57, Gokan Atmaca wrote:
> > > > Hello
> > > >
> > > > Can I use Cisco 3950 on Asterisk ?
> > >
> > > Please give us a link to a datasheet for that device.
>
> --
> BASIC is to computer languages what Roman numerals are to arithmetic.
>
>Please reply to the
> list;
>  please *don't* CC
> me.
>
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>
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-- 

Regards,
Chris Knipe
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Re: [asterisk-users] Writing CDR's to two database servers

[Chris Bagnall]

On 19/6/17 4:47 pm, Tech Support wrote:

I know that there are probably several solutions to this problem, but
what I am trying to do is provide some redundancy for my customers CDR data.
I know that doing simple backups of MySQL is probably the easiest way to go,
but I'm thinking that there may be some benefit to simultaneously writing
the CDR data to multiple servers at once. However, I'm drawing a blank on
this one. Has anyone else done this before? Any insight at all would be
greatly appreciated.


You could - if you really wanted - use two different cdr_ modules to 
write to, for example, a MySQL and a PostgreSQL database simultaneously.


Having said that, and given nearly every modern DBMS has its own 
replication built-in, you'd be far better off using that. There are good 
instructions online for MySQL and PostgreSQL - and no doubt for other 
DBMSs as well.


Kind regards,

Chris
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Re: [asterisk-users] Call List Campaign to an IVR

[Chris Bagnall]

On 6/2/17 5:24 pm, Tech Support wrote:

Basically,
two calls are made.

...

When the first call is made for
such a short period, the remote end still goes off hook, but the call will
end before it starts to ring. Then, halfway through the first call, a second
call is made. Since the remote end is off hook from the first call, the
second call will get sent to voicemail and the message is played there.


Am I right in thinking call waiting isn't a thing on US mobile networks 
then? In the UK, call waiting is pretty standard, and almost universally 
enabled by default on mobile networks. AIUI the same is true for much of 
Europe.


Kind regards,

Chris
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Re: [asterisk-users] 1000 analogue lines with asterisk

[chris]

+1

spending money to get that many fxs ports is going to negate any savings of
reusing analog phones instead of buying ip phones

1000 analog ports sounds like hell and if it was me I  would be embarrassed
to have a setup like that tied to my name if I was a consultant etc.
Someone will come in after you and ask who set it up and the customer will
say you :)
On Feb 17, 2016 4:23 AM, "A J Stiles"  wrote:

> On Wednesday 17 Feb 2016, Goke Aruna wrote:
> > Hello all,
> > Can someone recommend what hardware to use for a 1000 analogue line
> > capacity asterisk PABX?
> >
> > Regards
>
> A PCI express card with four primary rate ISDN ports, each linked up to a
> channel bank, will give you 120 analogue lines.  So you will need nine such
> cards; and for reasons of simple numbers of slots on a motherboard, they
> will
> have to be split among three or more servers, linked to a gigabit switch.
>
> You might end up getting a better deal if you bought 1000 hardware SIP
> phones.
> (You also would probably increase your personal indispensability factor,
> into
> the bargain .)
>
> --
> AJS
>
> Note:  Originating address only accepts e-mail from list!  If replying off-
> list, change address to asterisk1list at earthshod dot co dot uk .
>
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[asterisk-users] Panic Button SMS Asterisk Integration

[Chris Nighswonger]

Has anyone done any integration of USB, etc. panic buttons and Asterisk?

The basic idea would be to have a USB based panic button[1] along with a
bit of code which would trigger a group SMS or perhaps a pre-recorded call
to a group.

Kind regards,
Chris



[1]http://www.amazon.com/StealthSwitch-Pro-USB-Foot-Pedal/dp/B00MI6K77K
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[asterisk-users] Asterisk and Vitelity's vMobile service

[Chris Gentle]

I'm trying to configure my Asterisk machine to work with Vitelity's
vMobile service.  I can place calls to the vMobile device and it rings
as expected.  However, I have no audio in either direction.  There's
no NAT involved though.  My asterisk machine has a public IP address
with port 5060 and the full range of RTP ports open.  (It does have a
second NIC and some SIP clients on the private side but I'm not using
those for my testing.)

Here's the relevant parts of my SIP.  Anyone see any obvious problems?
 Not much documentation around on vMobile.  I'd love to see a working
config if someone has one.  Thanks!

[general]
nat=no
externaddr=my_public_ip
localnet=192.168.1.0/255.255.255.0
register = username:sec...@inbound23.vitelity.net:5060
bindaddr = 0.0.0.0
srvlookup=yes
alwaysauthreject=yes
transport=udp
directmedia=no
allowguest=no

; Vitelity vmobile SIP user
[my_did_here]
type=friend
dtmfmode=rfc2833
host=my_did_here.mobilet100.sipclient.org
context=vmobile-out
defaultuser=my_did_here
fromuser=my_did_here
trustrpid=yes
sendrpid=no
secret=super_secret_password
disallow=all
allow=ulaw,g726,gsm


-- 
Chris

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Re: [asterisk-users] Phone provisioning template Snoms

[Chris Bagnall]

On 7 May 2015, at 23:45, Tafadzwa Nyabasa tnyab...@gmail.com wrote:
 I am looking for a phone provisioning template for Snom phones, Yealinks and 
 Polycoms. I am always doing deployments of many phones and usually configure 
 each phone one by one for each installation.  Any help will be highly 
 appreciated

There’s some excellent documentation about provisioning on the Snom Wiki:

http://wiki.snom.com/Category:Auto_Provisioning:Configuration_Files

You can set the phones (via DHCP options) a firmware url on a web server under 
your control, grab their MAC addresses, then deliver them custom config 
settings as required.

Easiest way to start is to copy the config file (via the web interface) from a 
phone with factory default settings, then just change the settings you need to 
change, and write something in your scripting language of choice (PHP, Perl, 
Python, etc.) to just send those settings to the phone dependent on MAC 
address. Don’t send *every* available config setting to the phone - only the 
changes from default you need to make.

I suspect the same can be done with Yealink and Polycom phones - I’ve not used 
those so can’t really comment. I have a similar system which seems to work for 
Sipura/Linksys/Cisco phones, though most of my new deployments are exclusively 
Snom.

Kind regards,

Chris
-- 
C.M. Bagnall
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Re: [asterisk-users] Anonymous SIP calls

[Chris Bagnall]

On 27/3/15 8:03 pm, James B. Byrne wrote:

One only accepts VOIP calls from known correspondents.  I
am not clear why this is so other than vague warnings respecting
(admittedly real and serious) security issues.


Because on the whole most people don't *want* to receive calls from 
random strangers :-)



What is it
about incoming SIP calls destined to our internal users that make
those calls so dangerous?  Why cannot incoming anonymous SIP calls not
be treated exactly as incoming PSTN calls


Others have already written far more eloquently than I about the 
security implications, but I think there are other factors at play here.


One of the principal benefits E.164 brought to the table was the ability 
to 'bypass' the telco (and their call charges) and route the call direct 
to the desired endpoint over our respective internet connections. But 
the cost of making calls via the PSTN has reduced to a point where the 
cost of the call is no longer a significant factor in whether to place 
the call. Think back even a few years: the cost of calling another 
country could easily rise above 1 (GBP/USD/whatever) per minute. Now, 
with the exception of a few far-flung locations, there are very few 
destinations to which calls are even a fifth of that cost.


Calls that come via the PSTN are subject to some sort of regulation. 
Bonafide marketing companies are obliged to screen their calls through 
the TPS (in the UK - I presume there's a similar 'do not call' screening 
process in other countries). It's not perfect (international marketers 
aren't effectively covered, for example), but it is marginally better 
than a total free for all.


As for solutions, I think that for direct SIP-to-SIP calling to gain the 
traction originally promised, we need to get to the same level of 
incoming call control as we have with spam filtering on email. So there 
will need to be organisations running distributed RBLs similar to (for 
example) Spamhaus which SIP servers can query in real time to check not 
just for hack attempts, but also those SIP servers from which 
unsolicited marketing calls have originated, etc.


In summary:
1) PSTN calls are now /cheap enough/ that the financial benefits of 
direct SIP-to-SIP calls for most users are negligible.
2) When the cost of calls falls to (effectively) zero, the principal 
beneficiaries are fraudsters and telemarketers, and most people would 
rather not deal with either group.
3) Lack of effective protection - both technical and regulatory - 
against SIP-to-SIP misuse (not just fraud, but unsolicited callers, etc.)


Kind regards,

Chris
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Re: [asterisk-users] BlindXfer Sensitivity

[Chris Bagnall]

On 16/2/15 4:13 pm, Andrew Colin wrote:

The strange thing is its only sometimes my dial string is as follows
exten = s,1, Dial (SIP/200,, tT)
For that particular route but obviously s,3 as have Ringing () first etc.
After she pushes ## 6 times it will go thru sometimes.


Are you sure it's a DTMF sensitivity problem rather than a delay problem?

I've had several sites where the default DTMF timeout of 0.5 seconds is 
too short for users to achieve, and have set featuredigittimeout (in 
features.conf) to 3 seconds to give them more time to press the 
combinations they need to press.


Kind regards,

Chris
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Re: [asterisk-users] Google Voice

[Chris Gentle]

I'm using chan_motif with Asterisk 11.  It still works.  I actually
received an email from google yesterday that there had been no traffic on
my number lately so the number would be reclaimed.  I had switched my
outgoing away from GV several months ago when they were supposed to
discontinue the service.  I switched back to it yesterday and have made
several calls.  No problems.

On Sat, Jan 17, 2015 at 7:35 AM, CDR vene...@gmail.com wrote:

 Does the channel chan_motif and res_xmpp still work?
 I heard that Google had blocked this technology.
 Philip

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-- 
Chris
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Re: [asterisk-users] Smartphone Mobility App?

[chris]

Scott,

Somehow never noticed this setting before, I have tested it and it really
works great for the forwarding to cell situation we described. Thanks alot
for pointing me in the right direction.

chris

On Fri, Dec 19, 2014 at 9:39 AM, Scott Griepentrog sgriepent...@digium.com
wrote:

 The main problem we are trying to solve is when our staff  forward to
 their cell phones they cant distinguish if the call was directed at their
 cell phone or the business DID.


 The easiest way to solve that is to have an audio prompt announce the
 calls that were passed through from the business DID before connecting the
 call through.  That does require using a follow-me approach instead of
 forwarding, but is easily done by just changing the confimration prompt.


 On Fri, Dec 19, 2014 at 8:29 AM, chris tknch...@gmail.com wrote:

 Anyone found any good smartphone apps that connect with their asterisk
 boxes that provides basic mobility features?

 The main problem we are trying to solve is when our staff  forward to
 their cell phones they cant distinguish if the call was directed at their
 cell phone or the business DID.

 We also would like to give user ability to control DND and forwarding of
 their extension from the smartphone.

 I know there are many cloud service providers with a offering like this
 but we are not looking to change our service infrastructure but rather
 looking for just a software product that connects to our existing asterisk
 systems and provides this functionality.

 We would ideally like something for both iphone and android but the
 immediate need is for iPhone

 Curious to hear what people have tried, their experiences, etc.

 We are open to both free/open source as well as commercial software as
 long as it is multitenant or scalable beyond single server.

 TIA,
 chris

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[asterisk-users] Smartphone Mobility App?

[chris]

Anyone found any good smartphone apps that connect with their asterisk
boxes that provides basic mobility features?

The main problem we are trying to solve is when our staff  forward to their
cell phones they cant distinguish if the call was directed at their cell
phone or the business DID.

We also would like to give user ability to control DND and forwarding of
their extension from the smartphone.

I know there are many cloud service providers with a offering like this but
we are not looking to change our service infrastructure but rather looking
for just a software product that connects to our existing asterisk systems
and provides this functionality.

We would ideally like something for both iphone and android but the
immediate need is for iPhone

Curious to hear what people have tried, their experiences, etc.

We are open to both free/open source as well as commercial software as long
as it is multitenant or scalable beyond single server.

TIA,
chris
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Re: [asterisk-users] Query on connecting 3G MSC with Asterisk PBX Server via SIP Interface.

[Chris Bagnall]

On 29/10/14 12:59 pm, A J Stiles wrote:

Imagine what would have happened to the human race if Ugg the Caveman decided
not to share the secret of making fire with everyone freely, but instead went
around demanding shiny beads with menaces from anyone who just wanted to keep
themselves warm .


That's the best analogy I've heard in favour of open development for a 
long time, and something that non-techs can understand.


I thank you sir :-)

Kind regards,

Chris
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Re: [asterisk-users] PBX hacked: why hundred of calls to the same number ?

[Chris Bagnall]

On 3/10/14 6:52 pm, Rainer Piper wrote:

the attacking server changed the destination Number  at 18:53  CEST  and
he is still blocked ... LOL
972597438354  callto:00972597438354


It's pretty much an everyday occurrence for any internet-connected SIP 
system these days...



Oct  3 19:46:20 server /sbin/kamailio[3977]: NOTICE: script: blocking
IP 62.210.149.136 sipcli/v1.8 rm=INVITE aU=null rU=100972597438354


Many of these attacks come from fairly easily recognised user-agent 
strings, so if you fancy doing a bit of packet inspection with your 
firewall, you can block many of these before they get as far as your SIP 
server(s) themselves.


For example, the sipcli scans you listed above can be blocked fairly 
easily with:
iptables -A INPUT -p udp --dport 5060 -m string --algo bm --string 
sipcli -j DROP


(obviously there are overheads to string searching UDP/5060 packets that 
you'll want to consider, and the above won't work if you're using sipcli 
legitimately anywhere on your network)


Kind regards,

Chris
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Re: [asterisk-users] how to strip +1 out of incoming number

[Chris Bagnall]

On 2/10/14 6:52 pm, motty cruz wrote:

Hello, our VoIP send us caller ID +1(area)(number) for instance
+16024224334 is there a way to strip +1 out of caller ID?


${CALLERID(num):1} should do what you're after (or :2 if you need to 
strip the + as well)


Kind regards,

Chris
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Re: [asterisk-users] Playback/background audio from MySQL BLOB

[Chris Bagnall]

On 24/9/14 10:36 am, A J Stiles wrote:

But personally, I'd just store the filenames in the database; and rely on the
unix filesystem for storing the actual file contents.  After all, that's what a
filesystem is for.


This.

Shocking as it might appear, filesystems are remarkably good at storing 
files. They were designed to do it. Why try to shoehorn a database into 
doing something it wasn't designed to do (and isn't particularly good at 
doing)?


Kind regards,

Chris
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[asterisk-users] Call Flow Documentation Tools

[chris]

Hello,

I have been researching software for documenting pbx call flow paths and I
was just wondering if anyone out there is using anything they have found
particularly useful or cool.

I am looking for something preferably visual that the average end user can
follow. So far the best thing I have come up with is making a diagram with
a decision tree in visio but its very time consuming to build this by hand
for every customer. We would like to be able to provide every customer a
diagram so they can easily understand the path that a call takes, what
conditions are checked and what actions are taken based on those conditions.

A large portion of my asterisk installs are for non profit or charitable
organizations so while I'm not completely fixed on a free solution, if it
isn't free the cost needs to be relatively low or at least be a
multi-tenant solution that  could at least be used for multiple customers.

For most of our installs we manage everything via CLI, but for a few orgs
with tech savvy people we have been able to setup freepbx for them and let
them make simple changes. I was thinking with freepbx maybe there could
even be a module that takes the freepbx configuration and generates visuals
based on reading the configuration, this would be really slick although not
a complete answer as we have many installs that do not have freepx.

Anyway, just wanted to get some input from others and put my ideas so far
out there, if you have any recommendations or experiences to share feel
free to reply on or off-list.

Thanks!
chris
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Re: [asterisk-users] Asterisk secure fine tune - stop attack

[Chris Bagnall]

On 4/9/14 4:58 pm, Eric Wieling wrote:

If we don't need to allow access from outside the USA we block access from all 
non-ARIN IP addresses by using iptables.   This takes care of at least 80% of 
attacks.


Likewise here (though RIPE rather than ARIN, since we're the other side 
of the pond).


You can also take it a bit further: if, for example, you know what 
ISP(s) your dynamic clients are using, you can limit connections to the 
IP ranges those ISP(s) use - look up their ranges on he.net's BGP 
looking glass if you need to find out what ranges they're using.


Another thing I've been playing with of late is using iptables' string 
matching functionality to block user agents of known attack vectors: 
'sipcli', 'sipvicious', 'friendly-scanner', etc.


This seems to work remarkably well, though what impact it has on net 
performance under load remains to be seen.


Kind regards,

Chris
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Re: [asterisk-users] Internal timing under load is critical ?

[Chris Bagnall]

On 30/7/14 10:08 am, babak wrote:

I am evaluating some voice broadcasting solutions based on Asterisks for more 
than 1000 simultaneous calls.


As a matter of curiosity, what do people use these voice broadcasting 
solutions for?


I'm genuinely struggling to think of (legal) reasons why you'd want to 
broadcast 1000+ simultaneous calls. Perhaps I'm just not being 
imaginative enough... :-)


Kind regards,

Chris
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Re: [asterisk-users] Limit Asterisk

[Chris Bagnall]

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

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


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



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


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


Kind regards,

Chris
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Re: [asterisk-users] Voicemail message to text

[Chris Bagnall]

On 20 May 2014, at 15:35, Ishfaq Malik i...@pack-net.co.uk wrote:
 I was wondering if anyone has implemented voicemail to text and if so, what 
 package is being used to do so?

With the huge variety of different accents and intonations in human speech 
(even in one country), my experience of all speech-to-text engines has been one 
of poor accuracy at best.

If you need messages-to-text, generally best to use a virtual PA company or 
similar - at least in my experience.

Kind regards,

Chris
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Re: [asterisk-users] cdr viewer for csv

[Chris Bagnall]

On 24 Apr 2014, at 11:36, binary dreamer dreamer.bin...@gmail.com wrote:
 I am running asterisk and all of my CDRs are in the default csv.
 the system is so limited to ram (only 256) and I cannot run MySQL or any 
 other program to give CDRs a fancy view.

As an aside, have you considered running your CDR storage/viewing on a separate 
machine? You don't have to log CDRs on the same box as you run asterisk.

 at the moment the only other software running is nginx for a static webpage 
 with guidance on the system.
 is there a way to present to a webpage the CDRs from the csv, please?

You can almost certainly do this if you want using the standard string handling 
functions in $middleware_of_choice, but the lack of indexing on text files will 
make this *very* slow for search queries etc.. The RAM/CPU requirements 
associated with loading huge chunks of text data into memory, manipulating 
them, then displaying the results will likely exceed that of a DB.

Unless you only want a recent call log, you really want to do this in a 
database.

Kind regards,

Chris
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Re: [asterisk-users] Open Source Asterisk Polling Solution

[Chris Bagnall]

On 22/4/14 11:44 am, A J Stiles wrote:

Firstly, be warned:  Are you sure that is even legal to do in your
jurisdiction?  You could be setting yourself up for a hefty fine!  Check
applicable local laws before proceeding.


This. I'm glad someone else thought it worth mentioning as well :-)

Even if it's letter-of-the-law legal, consider whether you'd be willing 
to be on the receiving end of it. If it's providing a useful service, 
i.e. calling a specific list of individuals who might have medical 
appointments in the coming week to confirm they don't need to 
reschedule, then fair enough, but if you're phoning people who've not 
consented to sell them double glazing, then you aren't doing your 
client's reputation any favours and you're going to mightily annoy those 
you're calling.


Kind regards,

Chris
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Re: [asterisk-users] Old Asterisk Release wanting to upgrade ...

[Chris Bagnall]

On 17/4/14 3:53 am, Lee, John (Sydney) wrote:

I have written a lot of AEL2 script in Asterisk 1.4.x and I am not sure if it 
will still run in 11.


If I'm honest, this is why I still have so many 1.4.x boxes around as 
well. I've been using 11 for new installs, but the thought of having to 
redo all the AEL macros from 1.4 does not fill me with any enthusiasm to 
update those boxes.


The switch to Gosub() does not seem to be an easy drop-in replacement 
for Macro().


Kind regards,

Chris
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Re: [asterisk-users] Live Recording on the Storage Server?

[Chris Bagnall]

On 17 Apr 2014, at 16:14, Paul Belanger paul.belan...@polybeacon.com wrote:
 hi. I would not do that due to network issues.
 My approach is to record everything locally and every hour or so to move
 everything to a storage.
 +1 save yourself the headache and do this.

I'll add another +1 to this. I've never been able to get multi-channel 
recording (even 3 or 4 channels) working reliably over an NFS link to another 
server. I suspect, with some tweaking of nfs options it might be possible, but 
if it ain't broke…

Just a cautionary note if you do use a cron job to move recordings to a storage 
device at regular intervals: make sure you use lsof or similar to check the 
recordings aren't actually open by asterisk at the time, otherwise interesting 
things will happen.

Kind regards,

Chris
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Re: [asterisk-users] Old Asterisk Release wanting to upgrade ...

[Chris Bagnall]

On 17/4/14 4:53 pm, Eric Wieling wrote:

I had little problem converting my AEL scripts from 1.4 to 11


Did they have lots of macros in them?

If so, then you, sir, are a better man than I, and I take my hat off to 
you :-)
(and any hints you might want to share in converting 1.4 AEL macros to 
11 would be gratefully appreciated)


Kind regards,

Chris
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Re: [asterisk-users] Replying to Posts

[Chris Bagnall]

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.


On 13/3/14 5:13 pm, Ron Wheeler wrote:

-1
Prefer top posting.
Easy to see if I want to scroll down to see if it is something
interesting to me.
I get a lot of e-mails each day and scrolling wastes too much time.


You can then also reply to another point here like this: it's as much 
about trimming previous posts as about not top posting. New posts should 
include just enough context to ensure the message isn't meaningless, but 
not quoting a 20+ line irrelevance. That way you see both the question 
and the answer without scrolling.


Whilst we're on the subject of mailing lists, I'd like to add my 
personal pet rant: MTAs that don't add/honour In-Reply-To headers. 
Completely breaks threaded readers.


That is all :-)

Kind regards,

Chris
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Re: [asterisk-users] High Availability with Asterisk

[Chris Bagnall]

On 6/3/14 3:21 pm, Thorolf Godawa wrote:

The idea would be having a HA-cluster of two servers with Xen, each of
them runs one instance of an Asterisk-system in a single VM and on a
failure the VM will be restarted on the other node.
This might result in a much higher load on this node, because is runs
two VMs, but for a short period, until the other node comes back again,
it might be tolerable.


This is basically what we do, though in our case we use KVM rather than 
Xen; we found KVM behaved a great deal better managing timing than Xen, 
but YMMV and Xen may well have come along a great deal since we last 
looked at it.


In fact, it could be argued that even without any need for HA, there's 
still an advantage to running a server in a VM: hardware portability. If 
the machine dies, you can quickly redeploy the VM to a new host without 
having to recompile things and so on because hardware has changed.



Are there other options running two Asterisk-instances parallel on one
system, each binded on it's own IP, maybe s.th. with chroot or similar?


You might be able to do something interesting with containers (LXC), but 
given the ease of setting up KVM and the (relatively) small performance 
overhead, we've tended to just stick with that.


On 6/3/14 3:46 pm, Michelle Dupuis wrote:

A lot of HA tools don't look deeper into Asterisk to see if/how it has failed 
(they only detected catastrophic failures).  What happens when the Asterisk 
process is alive but no longer bridging calls?


In fairness, the tools the OP mentioned (pacemaker/corosync) can be set 
up to detect other failures than whether asterisk is alive - a simple 
one to set up is to try connecting on 5060 UDP and make sure you get an 
acknowledgement. Likewise, you could even set up a call using the 
manager interface to a dummy extension and make sure it completes 
successfully.


FWIW, we tend to use pacemaker with heartbeat rather than corosync, but 
both perform a pretty similar function.


Kind regards,

Chris
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Re: [asterisk-users] G729 Licensing Revisited - I'm Sorry!

[Chris Bagnall]

On 28/2/14 9:04 pm, Jayson Devor wrote:

That being said, will purchasing 23 licenses (one for
each channel that we use), and continue to use the open source g729
sorftware keep us legal?


I know at least half a dozen people who do this so that they can more 
effectively balance their licence commitment over a number of services, 
rather than locking licences down to MAC addresses of specific NICs in 
specific servers. But I'm based in the EU where (as others have said) 
patentability laws are quite different.


If you're worried about whether it's legal in your jurisdiction, you 
really should speak to a qualified legal professional to allay your 
concerns. This list has such an international audience that what's 
perfectly acceptable in one jurisdiction might land you in hot water in 
another.


Kind regards,

Chris
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Re: [asterisk-users] Asterisk NAT

[Chris Bagnall]

On 19/2/14 4:53 am, Gholamreza Sabery wrote:

Hello, a few days ago I sent a question:
http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html
but no one answered me! I just want to know is it possible or not?


I can't help on the can Asterisk detect they're behind the same NAT 
part of the question, but I would caution you that an assumption that 
'each NAT box has a single external IP' is risky - especially if you 
have to deal with the possibility of double-NAT and other such evilness 
(and it's hard to avoid sometimes - how many non-tech people go and buy 
a wireless router to 'extend their WiFi' rather than an access point, or 
don't know how to switch said router into AP-only mode).


You also have to consider users who have multiple LANs (which might not 
necessarily be able to route between themselves) behind a single 
external IP: this one's quite common in shared/managed office 
environments - one external IP and several RFC1918 VLANs internally, 
with no routing between them.


So in summary, unless you have a considerable level of control over your 
endpoints such that you can be sure these (and no doubt other) scenarios 
don't apply, it's probably safest to send RTP traffic through Asterisk 
regardless, otherwise you're potentially opening up a support nightmare 
for yourself.


Kind regards,

Chris
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Re: [asterisk-users] Want Queues to ignore mobile operators voice mails and continue ringing...?

[Chris Bagnall]

On 14/2/14 9:21 am, Gareth Blades wrote:

I would suggest using the 'M' option on the Dial command to run a macro.
The macro can just wait fir a key to be pressed and until it is pressed
the Dial is still effectively ringing. So if it does go to voicemail
then the call wont get put through. You need to make sure you have a
suitable value set to abandon the agent call if its ringing too long.
The callee may also find they are left multiple voicemail messages.


This is the approach we've used in the past: force the recipient to hit 
a button to accept the call, something which their mobile voicemail will 
never be able to do.


The alternative - and it only really applies if you have control of the 
mobiles in question - is to disable the mobile network's voicemail 
service entirely, and manage diverts from the handset. That way you can 
then recreate your own 'mobile voicemail' service on your asterisk 
platform with all the normal asterisk VM benefits such as email 
delivery, etc.


You can then of course detect when those mobiles 'divert' to voicemail 
(since it's now on your system), and kick them out of the queue at that 
point.


Kind regards,

Chris
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Re: [asterisk-users] Want Queues to ignore mobile operators voice mails and continue ringing...?

[Chris Bagnall]

On 14/2/14 10:54 am, Tiago Geada wrote:

How does one detect the 'divert' to voicemail?


If you're using the mobile network's voicemail service, you can't as a 
general rule; you've no reliable way of knowing whether that call was 
answered by the user or their voicemail service.


However, if you're providing the mobile voicemail service yourself from 
your asterisk platform, then you can detect the *incoming* call from the 
mobile device in question to their mailbox and act accordingly.


As I said in my earlier reply though, it depends on you having 
end-to-end control of the mobile devices in question and your mobile 
operator will allow their voicemail service to be completely disabled.


Kind regards,

Chris
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Re: [asterisk-users] IOPS required by Asterisk for Call Recording

[Chris Bagnall]

On 25/1/14 5:26 am, Amit wrote:

How do I derive the requirement? I need to size IO system to record
multiple calls concurrently.


I suspect this might be your problem:

250GB SATA disk (No RAID)



Is there any way to tune / optimize / configure for better write
performance?


Perhaps consider recording to a ramdisk first, then periodically write 
out completed files to HDD at your leisure (e.g. during slack periods)?


Or, given the relatively low cost of 250GB SSDs these days, swap out the 
spinning disc for an SSD.


Kind regards,

Chris
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Re: [asterisk-users] stopping unwanted attempts

[Chris Bagnall]

On 19/1/14 2:57 pm, Ron Wheeler wrote:

fail2ban is so easy to set up, there is no reason not to set it up.


One of the dangers with fail2ban - at least in its default configuration 
- is that a legitimate SIP phone with an incorrect password can quite 
easily send dozens of registration attempts in a couple of minutes, thus 
blocking that IP.


If your end users configure their own phones, you will have to factor in 
the increased support burden when users complain that their phones 
'can't connect' and you need to manually unblock those IPs. This can be 
at least partially mitigated using fail2ban's 'ignoreip' directive for 
IPs you know only your users will be connecting from.


If you've a large number of users, it might be worth splitting them 
across a pair of servers - one for 'trusted' users, i.e. where each SIP 
endpoint is locked down to a specific IP (or at least a range), and you 
can configure your firewall to block SIP connection attempts from 
anything apart from that list; and one for 'untrusted' users, i.e. 
travelling users, home workers without static IPs, etc. on which you run 
fail2ban with a fairly ruthless set of rules/limits.


Unless you know that none of your users travel internationally, I'd be 
wary of imposing countrywide IP blocks, especially in this era of IP 
shortage where IP space is being traded on the open market and GeoIP 
databases may not always keep up to date.


Kind regards,

Chris
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Re: [asterisk-users] Text to Speech Engine

[Chris Bagnall]

On 10/1/14 8:16 pm, Jai Rangi wrote:

Anyone know good quality text to speach engine for building IVRs for
asterisk. Open-source will be nice, but I wont mind paying for thing really
good.


We recently used Ivona for a fairly complex IVR project (multi-lingual, 
including pronunciation of foreign names).


http://www.ivona.com

Not free, but we found the sound quality considerably better than we 
were able to get from either Festival or Cepstral.


Worth bearing in mind that we are based in the UK, so our primary 
concern was for good quality British English voices. I cannot comment on 
other variants such as Australian or American.


Kind regards,

Chris
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Re: [asterisk-users] SIP Mass exodus

[Chris Bagnall]

On 13 Nov 2013, at 18:29, Mike Diehl mdiehlena...@gmail.com wrote:
 I've been seeing some strangeness lately on my 10.2.1 server.  It's
 gotten to the point that a few times each day, I see masses of SIP
 clients becoming unreachable.  They're not all on the same network,
 and we don't see any calls drop.  In a few seconds, they all come
 back.
 I don't think it's a connectivity issue because we don't drop calls,
 and the endpoints aren't on the same networks.  We don't see excessive
 CPU load when it happens.
 It does SEEM to happen most right after someone accesses their voicemail.


We saw this happen on a 1.4 server a couple of years ago shortly after 2am each 
day. It was only after a study of the cron schedule we narrowed it down to a 
number of rsync backup jobs which were run at that time.

As in your case, it wasn't a connectivity or bandwidth issue - in the end we 
put it down to a disk I/O bottleneck. It might be worth running something like 
iostat on your box to see if you see a spike in iowait as voicemail is being 
checked. We resolved it simply by rate limiting our rsync jobs. In your case 
with a busy database, you might want to look at your MySQL indexes and/or cache 
settings - this might be something worth asking about on the respective MySQL 
discussion groups as well as here.

Kind regards,

Chris
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Re: [asterisk-users] What linux distro most popular for Asterisk

[Chris Bagnall]

  Is there a recent survey of that Linux distro and version people are
using for the Asterisk installations?  I recall seeing a pie chart over a
year ago (I think on a wiki but I can't find it again)also hoping for
something more current.


Mix of Gentoo and Ubuntu here (Gentoo mostly on old Via Epia MiniITX 
systems which don't have full i686 instruction set support).


The best Linux distro is usually the one you're most familiar with - 
that way, if/when something goes wrong, you stand a reasonable chance of 
being able to fix it.


Kind regards,

Chris
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Re: [asterisk-users] Using sqlite3 for CDR logging

[Chris Bagnall]

On 3/10/13 5:52 pm, Tech Support wrote:

I was
thinking of using sqlite3 to log CDR's, thinking that would be faster than
using MySQL. Has anyone ever benchmarked this to quantify just how much
faster sqlite3 is? Are there any drawbacks to using it?


Lack of multi-user concurrency is the big one.

At the risk of encouraging database contests on the list, have you tried 
using PostgreSQL instead? It's a gross generalisation, but In my 
experience, PG handles writes better than MySQL, which in turn tends to 
handle reads a little faster than PG - assuming both are in 'out of the 
box' (i.e. unoptimised) conditions.


If you wanted to stick with MySQL, you might want to have a go at 
optimising it - there are quite a few scripts knocking around the web 
which run a set of queries on your data and suggest optimisations to apply.


And others have said, running the DB on a separate host is never a bad 
thing, and ideally on SSDs or RAM storage if you can. Spinning disks are 
often the bottleneck with large data sets.


Kind regards,

Chris
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[asterisk-users] AUTO: Chris Douglas is out of the office (returning 09/16/2013)

[Chris Douglas]

I am out of the office until 09/16/2013.

I will be out of the office and will have minimal access to email and
voicemail.  If you need immediate assistance, please contact the Pioneer
I.S. Help Desk at 316-688-8777, 800-613-9382, or via intercompany dialing
using  the internal directory at http://directory.pioneer.world.  There you
can leave an emergency message for the on-call technician.

Otherwise I will respond as soon as I can.

Thanks,
Chris Douglas
Technical Services Manager
Pioneer Balloon Company
tel. 316-688-8648 fax. 316-691-6901



Note: This is an automated response to your message  asterisk-users
Digest, Vol 110, Issue 14 sent on 9/13/2013 12:00:01 PM.

This is the only notification you will receive while this person is away.


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Re: [asterisk-users] Dialplan MySQL inserted ID

[Chris Bagnall]

On 20/8/13 5:00 pm, A J Stiles wrote:

Why not write an AGI script in your favourite language  (Perl, Python, PHP,
Java all have AGI and MySQL bindings)  to perform the INSERT query for you?


+1. It would also give you somewhere to perform sanity checks on your 
${ARGS} to avoid SQL injection attacks...


Kind regards,

Chris
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Re: [asterisk-users] Am I being hacked?

[Chris Nighswonger]

On Mon, Aug 19, 2013 at 2:40 PM, Patrick Lists 
asterisk-l...@puzzled.xs4all.nl wrote:

 On 08/19/2013 08:10 PM, Eric Wieling wrote:

 One of Asterisk's dirty little secrets is that it does not show the
 source IP when a device or hacker tries sending a call without registering.
  The rejection message in the logs do not show the IP of the attacker.
 Yes it sucks, yes it has been that way for many many years.


 Are you aware of a patch that would show the source IP in the console and
 logs?


I do something like this:

1. turn up the logging
2. add foo like this in my dial plan:

exten = _.,1,NoOp(Received incoming SIP connection from unknown peer to
${EXTEN})
exten = _.,n,Log(NOTICE,Anonymous peer IP: ${CHANNEL(peerip)})
exten = _.,n,Set(DID=${IF($[${EXTEN:1:2}=]?s:${EXTEN})})
exten = _.,n,Goto(s,1)

3. do some bar like this in my fail2ban filter:

VERBOSE.*SIP/HOST-.*Received incoming SIP connection from unknown peer
VERBOSE.* logger.c: -- .*IP/HOST-.* Playing 'ss-noservice' (language '.*')
NOTICE.* .*: Anonymous peer IP: HOST
NOTICE.* .*: Failed to authenticate device .*\s?\sip:.*@HOST\.*

and that handles most of the hacking attempts I see on my system. I think
it may be possible for the second line to catch some false matches, but I
have not seen any issues with our system thus far.

Kind Regards,
Chris

PS. Feel free to comment on what is wrong with this and be sure to include
the right way to do it. :-)
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Re: [asterisk-users] Paltel subscribers as called parties for SIP attacks

[Chris Bagnall]

FWIW, we routinely see dodgy traffic from:

ovh.net
hetzner.de


But since those are 2 of the larger short-term contract dedicated server 
vendors, I'm not surprised about that. It's so frequent that I don't 
even bother reporting it any more - when an abuse report is acted upon 
and the server shut down, another pops up to take its place.



all going to 972-59-* numbers (i.e. Paltel/Jawal mobile customers).


Likewise here. Well, not all, but a sizeable percentage of it. We're 
based in the UK.



Why would an internet subscriber from hadara.ps, for instance, want to call a 
Paltel mobile user via some remotely hacked SIP PBX thousands of miles away 
given than Paltel is partially owned by Hadara Technology Investment Co. (and 
Paltel leases long-haul infrastructure from Hadara anyway)?


Are you perhaps reading too much into it? There are insecure servers and 
computers all over the internet. These are (ab)used and co-opted into 
botnets which are in turn used to compromise SIP servers. I suspect that 
it's probably a financial goal (free calls, or substantial termination 
payouts) rather than a political goal the perpetrators are seeking.



I'd be curious to know what everyone else's experiences have been like, and why 
95% or better of the SIP attacks on my PBX are destined for Paltel mobile 
subscribers.


Perhaps the termination payout on those numbers is particularly good, 
and/or regulation/investigation into abuse isn't so good?


Kind regards,

Chris
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Re: [asterisk-users] include directory with multiple files in it

[Chris Bagnall]

On 5/8/13 2:18 pm, Jonas Kellens wrote:

is it possible to use the #include - syntax to include several
configuration files situated  in one directory ?
Something like :
extensions.conf :
#include extra/*
#include addons/*
Is this possible ?


Yes.

You can also do crafty things like:
#include */extensions.conf

Which will include the file extensions.conf from each subdirectory. Very 
handy if you have a structure like this:

/etc/asterisk/client1
/etc/asterisk/client2
etc.

Kind regards,

Chris
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Re: [asterisk-users] FTP server down?

[Chris Hozian]

 Not sure if this is the right place to mention it, but .

 The server downloads.asterisk.org was refusing FTP connections last
night, and
 still does not seem to be accepting them this morning.

 FTP may not be modern or trendy, but the ability to navigate around
 folders textually is nonetheless extremely handy when using a machine
with no
 X server, from its own console.

 --
 AJS

Thanks for your concern and suggestions. Unfortunately, the decision was
made to only offer our public downloads (e.g. downloads.digium.com,
downloads.asterisk.org) over the HTTP protocol, not the FTP protocol, quite
some time ago. As stated in Kevin Fleming's announcement on July 26th, 2007
to the Asterisk Announce mailing list, this was done primarily for reasons
related to our marketing department (
http://lists.digium.com/pipermail/asterisk-announce/2007-July/85.html).
If your response was misunderstood, please let us know and provide
clarification. Thanks again!

--
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Digium, Inc. | Network and Computer Systems Administrator
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
main: +1 256 428 6000 | fax: +1 256 864 0464
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Re: [asterisk-users] Dongle or extra channel and sip SMS

[Chris Bagnall]

On 15/7/13 3:00 am, bilal ghayyad wrote:

I need to be able to send SMS messages for campaign or for specific users, also 
I need to be able to receive SMS messages and do automatic reply.


In my experience, SMS is something best done out of Asterisk. That's not 
to say that Asterisk can't do it, of course, just that there are 
providers out there who can give you a nice friendly API for easy 
integration into your application. This is especially true if you need 
to send *lots* of messages in a short space of time: simply adding a 
single mobile device with a single SIM isn't going to cut it - you're 
going to need a bunch of them, at least. All of those will likely have 
different numbers, so you're going to have to handle that for receiving 
messages. Then you have to consider that some networks will charge more 
to send messages to numbers on the same network vs. a different network, 
so you might have to separate out your numbers into networks (easy if 
they've never been ported; more tricky if they have).


Based on past projects (in the UK), the cost of multiple SIM contracts, 
the necessary hardware to connect them, development time, etc., is 
usually more than the cost of paying a third party with a suitable API 
x per message to deliver them on your behalf.


Kind regards,

Chris
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Re: [asterisk-users] PoE module

[Chris Bagnall]

On 14/7/13 8:12 pm, bilal ghayyad wrote:

We have a cisco switches but they are not PoE and we need only to have PoE 
device so the cables come for it first to provide the power and then goes to 
the switch (to be like batch panel), is there something like this that can be 
used for the IP Phones?


Have a chat with your usual network equipment supplier for midspan PoE 
units. Phihong make some, and those I've used seem to have been pretty 
reliable. There are no doubt many other suppliers of such things.


Kind regards,

Chris
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Re: [asterisk-users] PoE L3 Switches

[Chris Bagnall]

On 14/7/13 11:45 pm, Gregory Malsack wrote:

I've used a lot of Dlink DES-1228p and 1210-28p. Primarily with polycom phones. 
Seem to have pretty good luck with them for the last 7 years or so.


+1. We've used quite a few DES-1228P units in the past, and apart from 2 
early unit failures, we've not had any failures since (out of a few dozen).


The management interface isn't great, especially if you're used to the 
command line goodness of an HP or Cisco unit, but provided you aren't 
fiddling with it too often, you'll manage.


Kind regards,

Chris
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Re: [asterisk-users] Adjusting confbridge call quality

[Chris Gentle]

ulaw

On Wed, Jul 10, 2013 at 7:40 AM, basteon bast...@gmail.com wrote:
 Hi,
 What codec do you use with yours subscribers?


 On 9 July 2013 23:45, Chris Gentle gent...@gmail.com wrote:

 Is there any way I can improve the audio quality in a confbridge in
 Asterisk 11?  I've changed the internal_sample_rate setting to 44100
 but that doesn't seem to make any difference.  I would also think this
 would make my confbridge recordings 44100 but they all end up as 8000.
  Am I completely missing something?

 --
 Chris

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Re: [asterisk-users] Adjusting confbridge call quality

[Chris Gentle]

On Wed, Jul 10, 2013 at 9:16 AM, Matthew J. Roth mr...@imminc.com wrote:
 The sampling frequency for u-law is 8,000 Hz.  You can't produce a recording
 with higher quality than the source, so you'd have to switch to a wideband 
 codec
 to improve the conferences and recordings [1] [2].

OK, thanks for the info.  I'm perfectly OK with 8,000 Hz except that
I'm feeding the audio into a conference room from a microphone.
chan_alsa actually is the first client to connect to the confbridge
and then others can connect via SIP.  For some reason, when the
speaker says words with S's and F's, they almost sound distorted.  Not
quite static but you can tell the quality has been affected.  May just
be a side-effect of 8,000 Hz.  Just wondered if there way some way to
improve that.

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Re: [asterisk-users] Adjusting confbridge call quality

[Chris Gentle]

OK, thanks for the advice.  No, there's no filter so I'll look into that.

On Wed, Jul 10, 2013 at 3:02 PM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
 On 07/10/2013 06:46 PM, Chris Gentle wrote:
 [snip]

 and then others can connect via SIP.  For some reason, when the
 speaker says words with S's and F's, they almost sound distorted.  Not
 quite static but you can tell the quality has been affected.  May just
 be a side-effect of 8,000 Hz.  Just wondered if there way some way to
 improve that.


 The distorted S and F are prevented by a pop filter in front of the mic. Are
 you using a pop filter? Also if you are using a cheap mic, do yourself a
 favor and invest in a decent mic. It will make a world of difference.

 Regards,
 Patrick


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[asterisk-users] Adjusting confbridge call quality

[Chris Gentle]

Is there any way I can improve the audio quality in a confbridge in
Asterisk 11?  I've changed the internal_sample_rate setting to 44100
but that doesn't seem to make any difference.  I would also think this
would make my confbridge recordings 44100 but they all end up as 8000.
 Am I completely missing something?

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Re: [asterisk-users] I need a second opinion on a new phone system deployment

[Chris Bagnall]

On 15/6/13 7:00 pm, Carlos Alvarez wrote:

Interesting product that I was very interested in, but the licensing has
one huge glaring problem.  Be sure to read the FAQ carefully.  If your
hardware fails and you replace almost anything in the machine, you have to
pay for the product again.


Not to mention that installing Pacemaker/Heartbeat/Corosync or your 
other HA solution of preference isn't particularly difficult, and is 
agreeably free.


Kind regards,

Chris
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Re: [asterisk-users] blocking spammer by callerID name

[Chris Gentle]

Google the number and you can probably find other complaints and
possibly who it is.  Not that it will matter, there's nothing you can
do but block it.

My approach to call filtering is:

Deny All
Allow Some

I have a whitelist of callers I always want to accept that may include
businesses outside my local area code.  If my dialplan doesn't
recognize the incoming number I send them to a voicemail mail where
they have to press 5 to leave a message.  That knocks out the robo
dialers.  Then I google the number and if it's a spammer, I add them
to a blacklist where the call is dropped immediately.  Really no point
in playing funny or cute messages to them or even telling them they
are blacklisted because it's usually an auto-dialer and a real person
doesn't hear it anyway.

On Thu, Jun 13, 2013 at 1:31 PM, Joseph syscon...@gmail.com wrote:
 I have a subroutine to block spammer by CALLERID(number)

 exten = 4,1,GotoIf(${BLACKLIST()}?blacklisted,s,1)
 exten = 4,n,Set(goaway=${CALLERID(number):0:2})
 exten = 4,n,GotoIf($[${goaway} = V4 ]?blacklisted,s,1)
 exten = 4,n,GotoIf($[${goaway} = V3 ]?blacklisted,s,1)

 but I just got another spammer (automated calls) who rotates his callerID
 number that starts with valid area code so blocking by prefix is not
 practical but it seems to me he uses the same (or few same) caller name
 like:

 Brit. Columbia  16047726633
 KHAN SHARON  16042984429
 Brit. Columbia  16042231781

 So I was thinking the same subroutine can be used to block by
 CALLERID(name), isn't it:

 exten = 4,n,Set(goaway2=${CALLERID(name):0:11})
 exten = 4,n,GotoIf($[${goaway2} = Brit. Colum ]?blacklisted,s,1)
 exten = 4,n,GotoIf($[${goaway2} = KHAN SHARON ]?blacklisted,s,1)

 The spammer is soliciting lowering credit card interest charges etc. anybody
 know who it is :-/

 --
 Joseph

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Re: [asterisk-users] blocking spammer by callerID name

[Chris Gentle]

Yeah, probably wouldn't work too well in a business environment where
you actually NEED to answer calls.  I go to a lot of trouble to make
sure people can't get in touch with me.  :)

I keep my blacklist and whitelist in AstDB.  However, I maintain it in
a bash script so that I can update the script and then rebuild the
AstDB very quickly.  If I lose my AstDB I can just rebuild it with the
script.

; Check the Asterisk database for blacklisted number
  exten = 
s,n,GotoIf(${DB_EXISTS(blacklisted/${CALLERID(num)})}?blacklisted,s,1)

Whitelist can be done the same way:

; Check the Asterisk database for whitelisted number
  exten = 
s,n,GotoIf(${DB_EXISTS(whitelisted/${CALLERID(num)})}?voicemail,abc,1)

I have a [screened] context that screens the calls and prompts for pressing 5

[screened] ;{{{
  exten = s,1,Zapateller()
  exten = s,n,Set(TSTAMP=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
  exten = s,n,NoOp(${TSTAMP})
  exten = s,n,Monitor(wav,${TSTAMP}-${CALLERID(num)}-screened,m)
  exten = s,n,Set(COUNT=1)
  exten = s,n(loop),WaitExten(1)
  exten = s,n,Background(privacy-screening-unidentified-calls)
  exten = s,n,WaitExten(.5)
  exten = s,n,Background(press-5)
  exten = s,n,Background(T-to-leave-msg)
  exten = s,n,WaitExten(3)
  exten = s,n,Set(COUNT=$[${COUNT} + 1])
  exten = s,n,GotoIf($[${COUNT} = 3]?loop)
  exten = s,n,PlayBack(goodbye)
  exten = s,n,StopMonitor()
  exten = s,n,Hangup()

  exten = 5,1,NoOp(Pressed 5)
  exten = 5,n,PlayBack(tcg/pls-lv-msg-w-nam-phnnum)
  exten = 5,n,StopMonitor()
  exten = 5,n,GoSub(voicemail,tcg,1)
  exten = 5,n,Hangup()

  exten = i,1,Playback(option-is-invalid)
  exten = i,n,Goto(99,msg)

  ;}}}



On Thu, Jun 13, 2013 at 2:55 PM, Joseph syscon...@gmail.com wrote:
 Thank you for input.
 Good idea, I like your approach with press number to leave a message,
 this will definitely cut the robo-calls voice-mail. Do you use database for
 white-list?
 Can you post a section of your dial plan that deals with blocking?

 This is a medical clinic so white-list, black-list is not a good solution
 but it might be good for home use.

 Thanks,
 --
 Joseph


 On 06/13/13 14:30, Chris Gentle wrote:

 Google the number and you can probably find other complaints and
 possibly who it is.  Not that it will matter, there's nothing you can
 do but block it.

 My approach to call filtering is:

 Deny All
 Allow Some

 I have a whitelist of callers I always want to accept that may include
 businesses outside my local area code.  If my dialplan doesn't
 recognize the incoming number I send them to a voicemail mail where
 they have to press 5 to leave a message.  That knocks out the robo
 dialers.  Then I google the number and if it's a spammer, I add them
 to a blacklist where the call is dropped immediately.  Really no point
 in playing funny or cute messages to them or even telling them they
 are blacklisted because it's usually an auto-dialer and a real person
 doesn't hear it anyway.

 On Thu, Jun 13, 2013 at 1:31 PM, Joseph syscon...@gmail.com wrote:

 I have a subroutine to block spammer by CALLERID(number)

 exten = 4,1,GotoIf(${BLACKLIST()}?blacklisted,s,1)
 exten = 4,n,Set(goaway=${CALLERID(number):0:2})
 exten = 4,n,GotoIf($[${goaway} = V4 ]?blacklisted,s,1)
 exten = 4,n,GotoIf($[${goaway} = V3 ]?blacklisted,s,1)

 but I just got another spammer (automated calls) who rotates his callerID
 number that starts with valid area code so blocking by prefix is not
 practical but it seems to me he uses the same (or few same) caller name
 like:

 Brit. Columbia  16047726633
 KHAN SHARON  16042984429
 Brit. Columbia  16042231781

 So I was thinking the same subroutine can be used to block by
 CALLERID(name), isn't it:

 exten = 4,n,Set(goaway2=${CALLERID(name):0:11})
 exten = 4,n,GotoIf($[${goaway2} = Brit. Colum ]?blacklisted,s,1)
 exten = 4,n,GotoIf($[${goaway2} = KHAN SHARON ]?blacklisted,s,1)

 The spammer is soliciting lowering credit card interest charges etc.
 anybody
 know who it is :-/

 --
 Joseph


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Re: [asterisk-users] Asterisk HA

[Chris Bagnall]

On 6/6/13 4:53 am, Gopalakrishnan N wrote:

Any other HA applications available or the lsyncd with pacemaker is good?


I generally use Pacemaker with Heartbeat, which seems to work pretty well.

Kind regards,

Chris
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[asterisk-users] Confbridge doesn't kick chan_local

[Chris Gentle]

I have a confbridge setup that feeds the conference from the ALSA
microphone input (this is the conference leader) and then uses
app_ices to send the conference audio to icecast.

I start the conference leader like this:

console dial 1000_admin@conferences

I join the ices user to the confbridge with a call file:

Channel:  Local/1000@conferences
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: conferences
Extension: 1000_ices
Priority: 1

This all seems to work great until I need to kill the conference.  My
confbridge user profile sets all users up with end_marked=yes so that
they should be kicked when the leader exits.  The Local channel
doesn't seem to exit properly:

 Hangup on console 
-- Stopped music on hold on Local/1000@conferences-;2
-- Local/1000@conferences-;2 Playing
'custom/thank-you.slin' (language 'en')
-- Executing [1000@conferences:2]
Hangup(Local/1000@conferences-;2, ) in new stack
  == Spawn extension (conferences, 1000, 2) exited non-zero on
'Local/1000@conferences-;2'
  == Spawn extension (conferences, 1000_ices, 4) exited non-zero on
'Local/1000@conferences-;1'
[Jun  3 07:44:27] NOTICE[18237]: pbx_spool.c:402 attempt_thread: Call
completed to Local/1000@conferences

The bridge gets left in an odd state.  Rather than the bridge being
destroyed, a confbridge list shows the conference is still active
with 0 users.  I have to restart asterisk to clear it.

I wonder if someone might take a look at my dialplan snippet below and
see if I'm doing something wrong before I file a bug report.  I'm
using Asterisk 11.4.0.

[conferences] ; {{{
  ; this is where normal callers enter the conference
  exten = 
1000,1,ConfBridge(${EXTEN},testfone_bridge,testfone_user,testfone_user_menu)
  exten = 1000,n,Hangup()

  ; this is where the conf leader enters
  exten = 1000_admin,1,Answer()
  exten = 1000_admin,n,Set(CALLERID(name)=ConfLeader)
  exten = 1000_admin,n,Set(CALLERID(num)=001000)
  exten = 1000_admin,n,Set(CONFBRIDGE(user,admin)=yes)
  exten = 1000_admin,n,Set(CONFBRIDGE(user,marked)=yes)
  exten = 1000_admin,n,Set(CONFBRIDGE(bridge,record_conference)=yes)
  exten = 
1000_admin,n,ConfBridge(1000,testfone_bridge,testfone_user,testfone_user_menu)
  exten = 1000_admin,n,Hangup()

  ; this is the Local channel that connects to app_ices
  exten = 1000_ices,1,Answer()
  exten = 1000_ices,n,Ices(/home/asterisk/asterisk-ices-1000.xml)
 ;exten = 1000_ices,n,Hangup()
;}}}



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Re: [asterisk-users] Confbridge doesn't kick chan_local

[Chris Gentle]

On Mon, Jun 3, 2013 at 11:52 AM, Matthew Jordan mjor...@digium.com wrote:
 (1) Verify that with all 'normal' channel drivers, such as chan_sip,
 that the Conference tears down correctly.

OK, looks like this is the problem.  Taking chan_local out of the
picture, I tested it with an incoming SIP call and an incoming IAX
call and sure enough, the tear down fails.  So if I rely on
end_marked=yes to kick all my conference participants when the leader
exits, it fails like this:

  Hangup on console 
-- Bridge/0x2364be4-input Playing 'confbridge-leave.slin' (language 'en')
-- Stopped music on hold on SIP/gent_2880-0002
-- SIP/gent_2880-0002 Playing 'custom/thank-you.ulaw' (language 'en')
-- Executing [1000@conferences:2] Hangup(SIP/gent_2880-0002,
) in new stack
== Spawn extension (conferences, 1000, 2) exited non-zero on
'SIP/gent_2880-0002'

confbridge list shows this:

Conference Bridge Name   Users  Marked Locked?
 == == 
1000  0  0 unlocked

Now my confbridge is in a bad state.

I tested it on both 11.4.0 and 11.3.0 on two different boxes with the
same results.

 If both (1) and (2) are successful, than there's some impact that the
 Ices application is having on the Local channel that is messing up the
 reference counting inside the ConfBridge. Otherwise, it's an error in
 ConfBridge.

So what do you think?  Should I file a bug?

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Re: [asterisk-users] Issue in transcoding

[Chris Bagnall]

On 2/6/13 2:01 pm, Muhammad Yousuf wrote:

I am trying to use asterisk as transcoder between voipswitch 2.0 and gsm
gateway. Voipswitch supports g723.1 but gsm gateway does not. Now I have
g723.1 codec in my asterisk. call leg from voipswitch is using codec g723.1
and call leg from gsm gateway is using codec gsm. I am having one way audio
and getting below mentioned warning. Asterisk version is 1.8.11.0


Isn't g723.1 considered pretty poor quality these days? Can't you set 
voipswitch to use something apart from that?


Kind regards,

Chris
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[asterisk-users] Help me understand these log messages

[Chris Gentle]

OK, I need a bit of help here.  I'm configuring a new Asterisk 11
system and I accidentally let my firewall rules drop for a day or so.
When I logged in today, I found messages like the ones below on my
asterisk console.  Obviously somebody was trying to take advantage of
my carelessness.  So can someone explain what would cause these types
of messages to show up on my console?

I understand that my iptables would have stopped this but I'm just
trying to understand more about the problem.  What other settings
might have stopped this?  Fail2ban was running but there were no
failed registration type messages that would have triggered it.

[May 31 01:47:40] NOTICE[2544][C-0001] chan_sip.c: Call from ''
(188.161.238.232:28203) to extension '972595595767' rejected because
extension not found in context 'default'.
[May 31 01:47:40] VERBOSE[2544][C-0002] netsock2.c:   == Using SIP
RTP CoS mark 5
[May 31 01:47:40] NOTICE[2544][C-0002] chan_sip.c: Call from ''
(188.161.238.232:28203) to extension '00972595595767' rejected because
extension not found in context 'default'.
[May 31 01:47:41] VERBOSE[2544][C-0003] netsock2.c:   == Using SIP
RTP CoS mark 5
[May 31 01:47:41] NOTICE[2544][C-0003] chan_sip.c: Call from ''
(188.161.238.232:28203) to extension '000972595595767' rejected
because extension not found in context 'default'.
[May 31 01:47:41] VERBOSE[2544][C-0004] netsock2.c:   == Using SIP
RTP CoS mark 5
[May 31 01:47:41] NOTICE[2544][C-0004] chan_sip.c: Call from ''
(188.161.238.232:28203) to extension '011972595595767' rejected
because extension not found in context 'default'.
snip


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Re: [asterisk-users] Help me understand these log messages

[Chris Gentle]

OK, I understand now.  I didn't realize allowguest was on by default.
I guess I should read more closely.  Thanks!

On Fri, May 31, 2013 at 5:15 PM, Yves A. yves...@gmx.de wrote:
 ... an anonyous (not registerted) sip user from 188.161.238.232 was trying
 to initiate a call to
 9725955 and so on...
 you could enable sip tracing to get more information.

 maybe you should change the 'allowguest' option in sip.conf..?

 regards,
 yves

 Am 31.05.2013 23:57, schrieb Chris Gentle:

 OK, I need a bit of help here.  I'm configuring a new Asterisk 11
 system and I accidentally let my firewall rules drop for a day or so.
 When I logged in today, I found messages like the ones below on my
 asterisk console.  Obviously somebody was trying to take advantage of
 my carelessness.  So can someone explain what would cause these types
 of messages to show up on my console?

 I understand that my iptables would have stopped this but I'm just
 trying to understand more about the problem.  What other settings
 might have stopped this?  Fail2ban was running but there were no
 failed registration type messages that would have triggered it.

 [May 31 01:47:40] NOTICE[2544][C-0001] chan_sip.c: Call from ''
 (188.161.238.232:28203) to extension '972595595767' rejected because
 extension not found in context 'default'.
 [May 31 01:47:40] VERBOSE[2544][C-0002] netsock2.c:   == Using SIP
 RTP CoS mark 5
 [May 31 01:47:40] NOTICE[2544][C-0002] chan_sip.c: Call from ''
 (188.161.238.232:28203) to extension '00972595595767' rejected because
 extension not found in context 'default'.
 [May 31 01:47:41] VERBOSE[2544][C-0003] netsock2.c:   == Using SIP
 RTP CoS mark 5
 [May 31 01:47:41] NOTICE[2544][C-0003] chan_sip.c: Call from ''
 (188.161.238.232:28203) to extension '000972595595767' rejected
 because extension not found in context 'default'.
 [May 31 01:47:41] VERBOSE[2544][C-0004] netsock2.c:   == Using SIP
 RTP CoS mark 5
 [May 31 01:47:41] NOTICE[2544][C-0004] chan_sip.c: Call from ''
 (188.161.238.232:28203) to extension '011972595595767' rejected
 because extension not found in context 'default'.
 snip


 --
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Re: [asterisk-users] Auto dialer scripts and software

[Chris Bagnall]

On 22/5/13 10:54 am, A J Stiles wrote:

You do know that sort of thing is against the law -- or at least requires a
permit from the authorities -- in most civilised countries, right?


And it's worth adding that even if it is legal in your country, you're 
almost guaranteed to offend/annoy your target audience. Recorded calls 
always do.


Kind regards,

Chris
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Re: [asterisk-users] Asterisk Log rotate not working

[Chris Bagnall]

On 21/5/13 4:19 pm, Ahmed Munir wrote: Last year, I installed Asterisk 
10.4.2 and enabled logrotate on daily basis

which was working perfect. Now in couple of months back, the logrotate
feature is not working at all but simply appending the logs in 'messages'
file. Listing down down the configuration for logrotate below;


This sounds more like a Linux/logrotate issue rather than 
asterisk-specific. Are your other system logfiles successfully rotating? 
(e.g. /var/log/messages)


If not, it may be something as simple as logrotate's daemon not running. 
You should be able to fix that in your distro's startup scripts.


On Gentoo, you'd do something like /etc/init.d/logrotate start to 
start it now, and rc-update add logrotate default to add it to your 
default runlevel.


Difficult to advise further without knowing the distro in question.

Kind regards,

Chris
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Re: [asterisk-users] Performance Asterisk large installation on Vmware/Xen

[Chris Bagnall]

On 18/5/13 8:09 pm, Mitul Limbani wrote:

Not recommended to run Asterisk on Virualization


I used to share that view, but having done a few medium-sized installs 
recently in virtualised environments and encountering no problems to 
speak of, I'm not sure it's necessarily the case any more.


Things to look into closely:
  - passing hardware devices from bare metal to VMs is at best 
'imperfect', so if you need PSTN connectivity using ISDN or POTS cards, 
you're probably best doing that in physical hardware. IP-only stuff 
seems to be okay.


  - be *very* careful about the load on the host machine. If you have 
total control over the VMs running on each machine, you'll probably be 
okay, but if you have to share a bare metal host with other VMs over 
which you have no control over the load, you'll run into problems.


 - you may come across problems with timing sources for conferences and 
the like, though I understand this has improved considerably in recent 
asterisk versions (i.e. no dependency on dahdi_dummy or ztdummy any longer).


FWIW, I've recently been using KVM as an alternative to both Xen and 
VMware, and I'm very impressed. It's certainly my preferred VM platform 
at the moment (not just for asterisk stuff, but in general).


Kind regards,

Chris
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Re: [asterisk-users] Monitoring SIP trunk status on call by call basis

[Chris Bagnall]

On 14/5/13 4:30 pm, Ishfaq Malik wrote:

I'm using asterisk 1.8.7.0 and adding a fail over trunk in case my
primary goes down. I'm wondering what the best method of checking if the
primary being up is.


Well, the obvious start point might be ChanIsAvail() - that'll at least 
weed out an upstream SIP peer that's unavailable (assuming you're using 
qualify) before you even get as far as Dial().


However, one of the problems you might encounter when sending calls to a 
provider is an inability to distinguish between Congestion and Busy. 
Ideally, of course, you want to route the call to upstream2 if you get 
Congestion from upstream1, but not if the dialled number is Busy. 
There's not always a good way around that.


As others have said, the only real way around it is to send calls 
periodically to verify end to end operation - at least this way you're 
testing both your upstream's SIP connectivity and also their PSTN 
termination.


Kind regards,

Chris
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Re: [asterisk-users] chan_alsa and confbridge

[Chris Gentle]

Answering my own question.  Setting the following in alsa.conf fixed my
problem:

input_device=plughw:0,0
output_device=null

Changing the input device to plughw helped some but didn't completely clear
the audio up.  Setting the output device to null did the trick.  I'm
wondering if there was some kind of interrupt hammering going on here with
my particular hardware.  Even before the audio completely fell apart I
could hear some little pops that sounded like the interrupts were not
being serviced fast enough.



On Mon, May 6, 2013 at 8:31 PM, Chris Gentle gent...@gmail.com wrote:

 OK, somebody may have a much better way of doing what I'm attempting.  If
 so, I'm open to suggestions.

 I am trying to configure confbridge to create a conference room with an
 audio stream coming from my sound card.  The idea is for a group of people
 to be able to call in and listen to someone giving a speech but not
 necessarily interact.  I've got confbridge configured and it seems to work
 when I connect via other SIP phones.

 To get the alsa input into the conference I configured the alsa module and
 did this at the console:

 console dial 100@conferences

 This seems to work, once I got my alsamixer stuff set right.  However,
 within about 10 seconds the audio goes bad.  Lots of distortion, echo,
 etc.  So I recorded a snippet right out of the sound card and loaded it
 into audacity.  The snippet was fine.  No distortion at all.  So the
 problem seems to be something in asterisk.

 Any ideas what I'm missing here?  Is there a better way to do this?
 --
 Chris




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[asterisk-users] chan_alsa and confbridge

[Chris Gentle]

OK, somebody may have a much better way of doing what I'm attempting.  If
so, I'm open to suggestions.

I am trying to configure confbridge to create a conference room with an
audio stream coming from my sound card.  The idea is for a group of people
to be able to call in and listen to someone giving a speech but not
necessarily interact.  I've got confbridge configured and it seems to work
when I connect via other SIP phones.

To get the alsa input into the conference I configured the alsa module and
did this at the console:

console dial 100@conferences

This seems to work, once I got my alsamixer stuff set right.  However,
within about 10 seconds the audio goes bad.  Lots of distortion, echo,
etc.  So I recorded a snippet right out of the sound card and loaded it
into audacity.  The snippet was fine.  No distortion at all.  So the
problem seems to be something in asterisk.

Any ideas what I'm missing here?  Is there a better way to do this?
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Re: [asterisk-users] looking for a way to do appointment reminders

[Chris Bagnall]

On 26/4/13 10:38 am, jg wrote:

they are currently calling patients. I think these calls apply only to a
certain fraction of the patients, who are difficult to contact by other
methods.


I suspect there will be different requirements depending on how 
'helpful' to patients you wish to be. At the very simplest end of the 
scale, you could simply call the patient's number and remind them of 
their appointment on dd hhmm, then disconnect.


However, the OP probably wants something a little more sophisticated 
than that. At the very least, you would want some method of handling 
shared numbers (e.g. a shared dwelling with a single phone), so you 
didn't inadvertently advertise a patient's appointment to someone else 
who answered the phone. So you would at the very minimum want a simple 
IVR that says We are trying to reach Mr. Joe Bloggs. If this is he, 
press 1 now, otherwise please hang up.


Going beyond that, you might want your reception staff, when booking 
appointments, to ask the patient when they would like their reminder 
call - the day before, an hour before, etc. etc. (and if the day before, 
would they prefer it in the morning, afternoon, or evening).


As others have said, the OP might be best advised to request (paid) 
assistance with the project on the [asterisk-biz] list.


Kind regards,

Chris
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Re: [asterisk-users] looking for a way to do appointment reminders

[Chris Bagnall]

On 26/4/13 10:14 am, Hans Witvliet wrote:

Only reasonable option is to send them an SMS.


Given the likelihood that a sizeable percentage of people attending a 
medical establishment are going to be at the upper end of the age scale, 
it's possible they may not have mobile phones, and even if they do, 
might not understand how to read SMS messages on their phone.


Probably would work okay for certain establishments, but I'd be wary of 
exclusively using SMS in a medical context, given the likely patient 
demographic.


Kind regards,

Chris
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Re: [asterisk-users] looking for a way to do appointment reminders

[Chris Bagnall]

On 26/4/13 12:24 pm, jg wrote:

This way the callees would always talk to a human being


If possible, this would definitely be a Good Thing. Many people (myself 
included) will disconnect a call as soon as they realise it's a recorded 
message. It also means the human caller can confirm they really are 
talking to the patient (perhaps by asking their DOB or similar).


It may be possible to outsource something like this to a Virtual PA 
service or similar.


Kind regards,

Chris
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[asterisk-users] E911 Voip Trunking

[Chris Nighswonger]

During the course of a conversation with an member of the IT group who
handles the E911 center for our county, I learned that all of the county's
E911 is voip based. This got me to wondering why we could not just
configure up a SIP or some such trunk directly to the E911 center to handle
our emergency traffic. The county seems interested in exploring the
possibility.

So I'm wondering if anyone else has attempted this.

Kind Regards,
Chris
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Re: [asterisk-users] E911 Voip Trunking

[Chris Nighswonger]

Section 6.5.2 (v4 interface) of NENA's v2 Interim Voip Architecture
Standard shows a ladder diagram of their SIP flow which seems to match
standard SIP. Maybe I'm oversimplifying it?


[1]
http://c.ymcdn.com/sites/www.nena.org/resource/collection/2851C951-69FF-40F0-A6B8-36A714CB085D/NENA_08-001-v2_Interim_VoIP_Architecture_i2.pdf

On Fri, Apr 19, 2013 at 2:51 PM, Terry Brummell te...@brummell.net wrote:

  E911 does not follow the standard SIP RFC.  That would be a good reason
 that they couldn't/wouldn't do it.  Now that I say that I should qualify it
 and say NG911 (or ESINet) does not follow SIP RFC
 http://en.wikipedia.org/wiki/Next_Generation_9-1-1.  That is not saying
 your county is not using standard SIP for E911, it just wouldn't be
 considered NG911.

 --
 *From:* Chris Nighswonger
 *Sent:* Fri 4/19/2013 11:41 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* [asterisk-users] E911 Voip Trunking

   During the course of a conversation with an member of the IT group who
 handles the E911 center for our county, I learned that all of the county's
 E911 is voip based. This got me to wondering why we could not just
 configure up a SIP or some such trunk directly to the E911 center to handle
 our emergency traffic. The county seems interested in exploring the
 possibility.

 So I'm wondering if anyone else has attempted this.

 Kind Regards,
 Chris

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Re: [asterisk-users] E911 Voip Trunking

[Chris Nighswonger]

On Fri, Apr 19, 2013 at 8:59 PM, Nathan Anderson nath...@fsr.com wrote:

 On Friday, April 19, 2013 5:35 PM, Warren Selby wrote:
  There are E911 providers that offer this functionality.  I know off the
  top of my head, 911Enable offers a service like this.  A former client of
  mine that provided hosted PBX services had a contract with them.  I'm
  sure there are other providers out there as well.

 Indeed.  911ETC is who we use, and is another example.  Even if you could
 peer directly with your county's PSAP, in the case of 911, I think it is a
 way better idea to go with one of these specialty SIP-based E911 providers,
 for the simple reason that even if you only sell VoIP service to people
 residing within your county


Actually we are not reselling service and the majority of our phones are
stationary. The few mobile soft phones we run would not need 911 service
since they also carry cell phones, the soft phones being mainly remote
extensions.

So it sounds like it is at least worth pursuing.

Kind Regards,
Chris
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Re: [asterisk-users] Network based transcoding

[Chris Bagnall]

On 12/4/13 4:38 pm, Nick Khamis wrote:

 We were looking more into the lines of a
scalable multi server router like a cisco 3745.


Perhaps it might help to tell the list just how many concurrent calls 
you're looking to transcode?


Kind regards,

Chris
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Re: [asterisk-users] xmpp priority setting and GoogleVoice

[Chris Gentle]

On Sat, Mar 23, 2013 at 10:45 AM, Harley Peters
har...@thepetersclan.com wrote:
 I had it set to 1 originally and it worked fine at first then suddenly
 stopped.
 It drove me crazy until I ran across this link:
 http://iprouteth0.blogspot.com/2013/01/new-thoughts-troubleshooting-google.html

 Set it to 127 and it has worked ever since.

Yep, that's the same article I found and mentioned in original post.
Thanks for posting the link.  While the article was written for
Asterisk 1.8 and Jabber, the same setting works in the xmpp.conf file.
 Very useful stuff.

-- 
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[asterisk-users] xmpp priority setting and GoogleVoice

[Chris Gentle]

I just wanted to send out some information that will hopefully help
others.  I don't know, maybe I'm the only one that's been having
problems with this.  I've been pulling my hair out for a while
wondering why Google would not send my incoming calls to my Asterisk
box.  The calls would just roll to voice mail and no packets ever
reached Asterisk.  This has happened on two separate Asterisk boxes
and three different GoogleVoice numbers.  I've been all through the GV
web page settings but nothing I did changed anything.  I figured it
had to be something simple, and I was right.  I finally ran across an
article that talked about the priority setting in xmpp.conf.  This is
set to 1 by default, which is apparently the lowest priority setting.
Apparently, GV routes calls to whatever session has the highest
priority.  If I understand correctly, being logged into gmail/gtalk
has a higher priority, somewhere around 20.  So Google calls were
probably being routed to my logged-in session but I never saw them
because I don't use Gtalk.  I changed this value to 100, as suggested
in the article, and incoming calls immediately started working for all
three GoogleVoice numbers.

The Asterisk 11 xmpp.conf sample file currently has this to say about
the priority setting:

;priority=1 ; Resource priority

I'm going to go out on a limb here and say that probably could use a
bit more.  This seems like a pretty important setting.

Just my two cents ...

Hope this helps somebody else.

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Chris

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Re: [asterisk-users] Asterisk does not persist callgroup and pickgroup configuration.

[chris]

Freepbx is over writing the conf files ?
On Mar 15, 2013 8:11 AM, Luis H. Forchesatto luisforchesa...@gmail.com
wrote:

 Greetings.

 I'm running asterisk here (elastix) and I have a few extensions configured
 in it. I have here two different callgroup/pickgroup where the extensions
 are configured in, but it doesn't work when I try do pickup a call. Looking
 the config file (sip_additional.conf) I see they are not configured with
 callgroup/pickgroup, the fields are empty.

 Manually inserting callgroup/pickgroup on the extensions worked just fine
 but the next day the configuration just vanished and the extensions was not
 working.

 Has someone a clue of whats going on here?

 --
 Att.*
 ***
 Luis H. Forchesatto


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[asterisk-users] Asterisk 11 GoogleVoice/Motif

[Chris Gentle]

I'm currently running Asterisk 11.2.1 and I've noticed that when asterisk
has been up for a while (usually about a day), outgoing calls through
GoogleVoice fail to complete.  I hear it ringing on my end but the caller
never hears the phone ring.  A simple restart of Asterisk seems to clear it
up for another day or so.  Has anyone else noticed this?

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Re: [asterisk-users] Serviced Office operator panel

[Chris Bagnall]

On 11/3/13 11:07 pm, Andrew Yager wrote:

Basically, if you know of a product, open or closed source, and would like to 
sell it to me and you think it does the job, or you've seen something that 
works, contact me off list ASAP!


Actually, please post *on* the list if you know or have used something 
that meets the above. I suspect many of us would find such a product or 
application useful from time to time.


Kind regards,

Chris
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Re: [asterisk-users] Asterisk 11 GoogleVoice/Motif

[Chris Gentle]

Awesome, thanks.  I'll give it a try.
On Mar 11, 2013 4:56 PM, Joshua Colp jc...@digium.com wrote:

 Chris Gentle wrote:

 I'm currently running Asterisk 11.2.1 and I've noticed that when
 asterisk has been up for a while (usually about a day), outgoing calls
 through GoogleVoice fail to complete.  I hear it ringing on my end but
 the caller never hears the phone ring.  A simple restart of Asterisk
 seems to clear it up for another day or so.  Has anyone else noticed this?


 This is fixed in Asterisk 11.3.0-rc1. The Google XMPP server has become
 prone to disconnecting as of late, which triggers a bug in older versions
 where chan_motif ignores XMPP traffic.

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

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Re: [asterisk-users] asterisk with 1000 extensions

[Chris Bagnall]

On 7/3/13 6:50 am, Bharat Lalcheta wrote:

You can use ATA box with pstn phone to reduce cost.


I would caution against that approach. Analogue to Digital conversions 
often seem to have 'problems' - mostly related to hangup detection 
and/or echo. If you really do want to use analogue phones, then use a 
good quality channel bank to bring the analogue extensions into 
Asterisk, not low-end ATAs.


You also have to consider the value of your time. There's little point 
shaving a few pounds (or dollars, or euros) from the hardware cost if 
it's going to double the configuration time. And using 'cheap' 
components will add to your ongoing support burden for the system.


Cheap != good value for money.

Personally, I'd consider using something like the Snom 710. They aren't 
the cheapest SIP phones by any means, but they do have a very good 
remote provisioning and configuration system, which will substantially 
reduce the work you need to do in configuring handsets.


If your budget won't stretch to the Snom units, the Yealink range as 
suggested by another poster might be worth looking at. I believe their 
cheapest (is it the T18?) SIP endpoint can be had for around 35GBP - I 
don't know what pricing is like in your local currency of course. I 
believe Yealink do also have a fairly reasonable remote provisioning 
system, but unlike the Snom system, I can't claim to have used it in anger.


Kind regards,

Chris
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Re: [asterisk-users] Exiting the queue doesn't work

[Chris Bagnall]

On 4/3/13 12:27 pm, Gertjan Baarda wrote:

After extensively googling the issue, I've found everything (also bug
related), accept my answer. What am I missing here?


It sounds like the call is being caught by a retry cycle on the queue.

Try adding n to your queue command from your dialplan.
Also worth making sure you have retry=0 in your queue config.

Kind regards,

Chris
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Re: [asterisk-users] Asterisk 1.8, Siemens C610IP with 3 handsets: all are ringing on incoming calls

[Chris Bagnall]

On 18/2/13 5:39 pm, Administrator TOOTAI wrote:

on incoming call we have exten =
100,n,Dial(SIP/Handset_102SIP/Handset_103SIP/Handset_104,,)
and always only Handset_102 is ringing, we receive busy back from the
2 others but they are not. Any clue?


It depends which base station you're using - some of the earlier ones 
only supported one or two simultaneous SIP calls (remember dialling 
counts as a call, even if it's not answered).


I seem to recall the N300IP (the one we use) supports 3 concurrent SIP 
calls.


The easiest workaround is probably to create a fourth SIP account called 
'102_103_104' or something that's set to ring all 3 handsets on the 
Gigaset web interface. You can then Dial(SIP/Handset_102_103_104) 
substitute the SIP account you created above from Asterisk instead.


A cautionary note with Gigasets in general: they claim each base station 
will support up to 7 handsets. In my experience, things start to get a 
bit flaky above 3 or 4 handsets (specific handsets not ringing 
periodically, etc.), so I suspect the base station might be CPU limited 
at some point, especially if you're asking it to use an expensive 
(computationally) codec like G.729.


Kind regards,

Chris
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Re: [asterisk-users] Asterisk 1.8, Siemens C610IP with 3 handsets: all are ringing on incoming calls

[Chris Bagnall]

On 17/2/13 5:02 pm, Administrator TOOTAI wrote:

customer102/Handset_102 xxx.yyy.zzz.153  D
N 5062 OK (80 ms)
customer103/Handset_103 xxx.yyy.zzz.153  D
N 5062 OK (70 ms)
customer104/Handset_104  xxx.yyy.zzz.153 D   N 5062
OK (66 ms)


That's perfectly normal with these phones, and shouldn't pose a problem.


As you see, all handsets are identified with the same port, which means
that on incoming call to one handset or when transfering a call with the
asterisk transfer feature, all 3 handsets are ringing :-(


You can specify which SIP account correlates to each handset in the 
Gigaset web interface.


Go to Settings - Telephony - Number Assignment
You want Handset 1 to use Connection 'Handset_102' for outgoing calls 
and for incoming calls (untick everything else except this for incoming 
calls).
Likewise Handset 2 should use Connection 'Handset_103' for outgoing and 
incoming (again, untick everything but this option).


Rinse and repeat for other handsets.

I can confirm it does work properly - we have dozens of clients with 
Gigaset phones and separate SIP registrations per handset.


Kind regards,

Chris
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Re: [asterisk-users] SayDigits

[Chris Bagnall]

On 8/2/13 12:11 pm, Doug Lytle wrote:

Is there a way to slow down or speed up the speed at which SayDigits

So, I'd have to say no.


I suppose potentially you could re-record the sound files to 'say' each 
digit faster (and with shorter rolloff at the end of each word), then 
put those into a separate [language] folder in /var/lib/asterisk/sounds, 
then use those instead in your dialplan.


You might even be able to process the existing recordings using your 
favourite audio editing tool to speed the sound files and reduce the 
rolloff at the end. No guarantees it'll sound any good, mind.


Kind regards,

Chris
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Re: [asterisk-users] Wierd question - Give me your opinion please

[Chris Bagnall]

On 5/2/13 11:45 am, Jared Baxley wrote:

The closest building is 950 ft, the second is 1850 ft. These two buildings
are connected via LRE's using existing 6 Pair, Unfortunately re
cabling isn't an option. Other buildings are even further from the office,
about 15 or so scattered about that only require 1 phone each.


Have you considered running your own SHDSL between the sites, i.e. run a 
small DSLAM in the main building? That'd give you IP connectivity in the 
remote buildings which could be used for both phones and also general 
net access if required. You'd then avoid having to worry about analogue 
phones at all.


A friend did this down the length of a heritage railway as they already 
had cable running the length of their tracks, and I believe it was 
fairly successful.


Kind regards,

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

[Chris Bagnall]

On 3/2/13 4:59 pm, David Smiley wrote:

I finally found the perfect solution for 
me:http://www.amazon.com/La-Crosse-D111-101-E1-WGB-Wireless-Monitor/dp/B0081UR76G/ref=dp_ob_title_def
The device is $69, plus $10/month for alerts.  And I get to monitor the 
temperature online, which is a great bonus.


Working on the assumption that you already have internet access at the 
property in question, I do wonder whether you might be better off with 
something network-based rather than phone based. You should be able to 
pick up a network temperature sensor relatively inexpensively, which you 
could in turn use to fire HTTP requests to a server under your control 
(even if it's at your home). You could then store temperature stats in a 
database and set up your own triggers to do something as and when they 
drop below a certain threshold.


In my $dayjob I've set up a number of similar systems at bird hides in 
various national parks/wetlands here in the UK for similar purposes 
(with the addition of the ability to pull off CCTV images). These are 
running programmable ICs which make a simple HTTP call to a webserver 
running SQL over a 3G data SIM every hour. I doubt it'd be difficult for 
you to knock up something similar for your property.


Kind regards,

Chris
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Re: [asterisk-users] RTP timeout if the asterisk box behind NAT

[Chris Bagnall]

On 3/2/13 11:38 pm, bilal ghayyad wrote:

What should I do?


Given that you said:

This problem was not appearing when Asterisk machine was having static real IP 
address because I was enabling the rtptimeout paramters.


I do believe the solution is simple: put it back on a public IP.

For what it's worth, we have dozens of clients with boxes on RFC1918 IPs 
and we don't see this issue, so I wonder if it's something 'special' 
your NAT router's doing to mess up RTP traffic. It's probably worth 
trying a different router (ideally different make/model) and see if 
that's any better. And it's always worth disabling any SIP ALG present 
in the router - they seem to do nothing but break things.


(as a random aside, has anyone *ever* come across a scenario where a SIP 
ALG in a consumer router has actually helped?)


Kind regards,

Chris
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Re: [asterisk-users] RJ11 x RJ45

[Chris Bagnall]

On 1/2/13 12:41 pm, Luis H. Forchesatto wrote:

Como que se faz um conector RJ45 em uma ponta e RJ11 e outra. Pretendo
conectar a linha de um ATA em uma placa Khomp KFXO IP. A ponta que tem o
conector RJ45 está crimpada com a sequencia 568B e vai ser conectada na
placa Khomp, mas a ponta RJ11 eu não sei como deve ficar.
Li alguns manuais na internet mas não entendi ao certo como tem que ser
feito.


Thanks to Google Translate, this apparently says something like:
How do you make a RJ45 connector on one end and one RJ11. I intend
connect to an ATA line on a plate Khomp KFXO IP. The tip having
RJ45 connector is crimped with the sequence 568B and will be connected 
to the

Khomp plate, but the tip RJ11 I do not know how it should be.
I read some manuals on the internet but did not understand exactly how 
it must be

made.

Generally speaking the line pair are on pins 4+5, usually the blue pair. 
So if all you need is a single line pair, you should be able to just 
wire up the blue pair to the centre pins on your RJ11 connector.


Alternatively, you can cheat, and just use an RJ11 - RJ11 cable - these 
usually fit just fine into an RJ45 socket.


Kind regards,

Chris
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Re: [asterisk-users] AGI command

[Chris Bagnall]

On 20/1/13 4:15 pm, Eric Wieling wrote:

Personally, I use the PHPAGI library and don't worry about all the low level 
stuff.


This. It also gives you a nice logging function you can use to output 
debug information to the asterisk CLI so you don't have to kill and 
start asterisk interactively.


Kind regards,

Chris
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[asterisk-users] Voicemail and recordings storage: best practices

[Chris Bagnall]

Greetings list,

I'm currently building a new cluster to replace our ageing Asterisk 1.4 
infrastructure - it's easier to start from scratch then migrate users 
across than it is to upgrade 1.4 to 1.8 in situ.


Anyway, it got me thinking about audio recordings in a multi-server 
environment and whether there was a better way to do it. On our existing 
1.4 cluster we NFS mount voicemail and recordings directories from 
another server (or more accurately a master/backup pair of servers) into 
each asterisk server. I'd say it's worked 'okay' - but since less than 
10% of our users regularly use call recording, it's never really reached 
a point where I/O throughput has been an issue.


So, since I have the opportunity to build up the new cluster from 
scrach, I thought it was an ideal opportunity to do a quick straw poll 
of the list and see what approaches others are using to store voicemail 
and recordings, and to make those available across a multi-server 
environment.


Let the discussions begin.

Kind regards,

Chris
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Re: [asterisk-users] recrding calls

[Chris Bagnall]

On 19/1/13 1:25 am, Joseph wrote:

I would like to outgoing/icoming calls and email the files.
This is what I have:
exten = _7.,n,Set(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten = _7.,n,Monitor(wav,${CALLFILENAME},m)
How do I email these file?


You probably want to use MixMonitor() instead of Monitor():
http://www.voip-info.org/wiki/view/Asterisk+cmd+Mixmonitor

One of its options allows you to execute a command at the end of 
recording, which you can then use to call a script to handle your 
recordings however you wish.


Kind regards,

Chris
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Re: [asterisk-users] DIDForSale spam

[chris]

+1

On Jan 9, 2013 8:59 PM, Don Kelly d...@donkelly.biz wrote:

 Jai,



 It should not be necessary for me to remove my email address from your
list. It should not be on there to start with—we do not have, and have
never had, a relationship that justified you sending me email.

 --Don

 Don Kelly

 PCF Corp
 People Come First
 651 842-1000
 651 842-1001 fax



 From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of Jai Rangi
 Sent: Wednesday, January 09, 2013 7:50 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] DIDForSale spam



 Guys,
 Since I am attached to did for sale:
 My apology to every one who received the DIDForSale 2012 Achievement
email and you hated it.

 As a asterisk user my question will be.
 If some xyz company send you a so called spam email, what made you think
that you should spam the mailing lists. I am sure we all get lots if spam
emails every day. If you really got some time and talent, why don't you
write some good tips and tricks about asterisk.

 Long story short We have a link where you can unsubscribe your email for
any further communication.
 http://www.didforsale.com/unsubscribe.php  or Send me your email  address
I will personally take care of that and will remove your email. This will
take less than 5 seconds.
 I am sure there will be lot of arguments on why you should that and all.
I will refrain myself on any further unproductive communication.

 Happy new year to you all.


 On Wed, Jan 9, 2013 at 4:39 PM, Mitul Limbani mi...@enterux.in wrote:

 +1 here.

 On Jan 10, 2013 5:50 AM, Steve Totaro stot...@totarotechnologies.com
wrote:

 On Wed, Jan 9, 2013 at 7:03 PM, chris tknch...@gmail.com wrote:
  On Wed, Jan 9, 2013 at 2:02 PM, Doug Lytle supp...@drdos.info wrote:
  What were the senders IP(s)?
 
  Will have to look it up when I get home.
 
  Doug
 
  --
  Ben Franklin quote:
 
  Those who would give up Essential Liberty to purchase a little
Temporary Safety, deserve neither Liberty nor Safety.
 
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  I have gotten hit with this twice so far. in March and Today:
 
  Rohit Dhaka ro...@didforsale.com via mail.bingotelecom.com
  3/8/12
 
  DIDForSale donotre...@didforsale.com via mail.bingotelecom.com
  1/9/13
 
  UGH, when I asked in March where he got my email he said:
 
  Hi Chris,
  We got your contact from the Internet. Let me know the good time to
  talk about this in detail.
  Thank you,
  -Rohit Dhaka
 

 Obviously by harvesting these lists.  I received 2 myself.

 Thanks,
 Steve T

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Re: [asterisk-users] DIDForSale spam

[Chris Bagnall]

On 10 Jan 2013, at 22:09, C. Savinovich c.savinov...@itntelecom.com wrote:

 Unfortunately, there is a fine line between being a forum where people can 
 exchange ideas, and being a forum where people can find asterisk consultants, 
 and both don't seem to co-exist well together.

Isn't this precisely the raison d'être for [asterisk-biz]?


Kind regards,

Chris
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Re: [asterisk-users] DIDForSale spam

[chris]

There is a big difference between publicly posting offering services
to the list and harvesting all the email addresses and them contacting
everyone privately

On Thu, Jan 10, 2013 at 5:32 PM, C. Savinovich
c.savinov...@itntelecom.com wrote:

Isn't this precisely the raison d'être for [asterisk-biz]?

 Oh my goodness!, the asteriz-biz?  nooo, they will kill you if you try to
 post anything offering your services!...  that list ceased to provide any
 value and died a long time ago precisely because its members ran each other
 away from it. A while back, I wrote a nice click-to-call  service and I
 dared put a post indicating that I was offering it for a fee, and in no time
 they called me spammer.  There is really no incentive to reward someone
 else's achievements, unless you tell them that you are given them your code
 for free, then they want it (totally contradicting the meaning of the word
 business).


 Christian Savinovich
 VoIP  Telephony Consultant
 646-982-3572



  Original Message 
 Subject: Re: [asterisk-users] DIDForSale spam
 From: Chris Bagnall aster...@lists.minotaur.cc
 Date: Thu, January 10, 2013 5:17 pm
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com

 On 10 Jan 2013, at 22:09, C. Savinovich c.savinov...@itntelecom.com wrote:

 Unfortunately, there is a fine line between being a forum where people can
 exchange ideas, and being a forum where people can find asterisk
 consultants, and both don't seem to co-exist well together.

 Isn't this precisely the raison d'être for [asterisk-biz]?


 Kind regards,

 Chris
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Re: [asterisk-users] Playing music through VoIP handsets while on hook

[chris]

I've seen this implemented on polycom phones where a secondary extension is
on the phone that is setup to auto answer and they have something on the
PBX side that is configured to call some or all of the secondary extensions
On Jan 10, 2013 8:28 PM, Carlos Alvarez car...@televolve.com wrote:

 This is something I've seen with some key systems and PBXs.  When the
 phones are on-hook, they can play music throughout the office instead of
 having an overhead speaker system do it.  Never heard of it being done with
 VoIP, but figured I'd ask if anyone else has.  I don't see any way to do
 this on any phones I've looked at.


 --
 Carlos Alvarez
 TelEvolve
 602-889-3003


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Re: [asterisk-users] Playing music through VoIP handsets while on hook

[chris]

Lol yes it was all local on a gigE network even :) I also didnt say it
was the most elegant solution but it seemed to work well with them
they even had grouped it into extensions and I'm sure you could even
write some logic to make sure the calls are local

On Thu, Jan 10, 2013 at 9:31 PM, Carlos Alvarez car...@televolve.com wrote:
 On Thu, Jan 10, 2013 at 6:42 PM, Christopher Harrington ch...@acsdi.com
 wrote:


 Wow, that seems wildly bandwidth inefficient. Is it possible to do
 multicast VoIP?


 Depends on whether the phones are local to the server.  Unless you're
 looking at hundreds of phones, a 100MB network running 80k to every phone
 wouldn't even notice.

 --
 Carlos Alvarez
 TelEvolve
 602-889-3003


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Re: [asterisk-users] DIDForSale spam

[chris]

On Wed, Jan 9, 2013 at 2:02 PM, Doug Lytle supp...@drdos.info wrote:
 What were the senders IP(s)?

 Will have to look it up when I get home.

 Doug

 --
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 Safety, deserve neither Liberty nor Safety.

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I have gotten hit with this twice so far. in March and Today:

Rohit Dhaka ro...@didforsale.com via mail.bingotelecom.com
3/8/12

DIDForSale donotre...@didforsale.com via mail.bingotelecom.com
1/9/13

UGH, when I asked in March where he got my email he said:

Hi Chris,
We got your contact from the Internet. Let me know the good time to
talk about this in detail.
Thank you,
-Rohit Dhaka

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Re: [asterisk-users] Asterisk for Razberry Pi

[Chris Gentle]

On Wed, Jan 2, 2013 at 9:03 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:

 On Wed, Jan 02, 2013 at 09:55:44AM -0500, Robert Rawlinson wrote:
  Has anyone ported Asterisk to the Razzberry Pi? If so could you point me
  to info on doing so?

 apt-get install asterisk


Does anyone know of any asterisk 11 packages for the Pi?  I ended up
compiling it myself this weekend.  Took a while.

-- 
Chris
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