Re: [asterisk-users] Block Spam Calls

[Hermann Wecke]

On Tue, Dec 10, 2019 at 11:01 AM Alexander Perkins
 wrote:
> Hi All.  Does anybody know if Google/Android has an API I can sign up for 
> that will allow us to query the caller ID and find out if it is spam or a 
> robocaller?

I don't think that there is a public (free) API. All robocall
protection services are paid ones. You can find several on Twilio from
US$ 0.003 up to US$ 0.06 per query, depending on what you are looking
for.

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Re: [asterisk-users] Digium IP Phones UNREACHABLE after registration

[Hermann Wecke]

On Thu, Apr 12, 2018 at 11:43 AM, Antony Stone
 wrote:
>> A few seconds after registration, the Digium phones will become
>> UNREACHABLE. Right after that, the entire VoIP network (where the
>> Digiums are located) will be also dropped - all other devices
>> (non-Digium) connected will be kicked from the asterisk box. There are
>> ObiHai, Yealink and Linksys at this location - all will be kicked.
>
> Are you by any chance running fail2ban, without the IP address of this
> location in a whitelist?

fail2ban: yes
whitelist: yes

> I'm wondering if some device is misconfigured and failing registration, which
> get spicked up by fail2ban, and the network's public IP gets blocked.

If I remove the Digium phones, after a while the other devices
(brands) will register again.

>> The server remains operational and all other users/peers (not running
>> Digium phones) are up and running.
>
> You mean, all others at other locations, right?

Yes, other locations are OK. This location will be "clogged" (or
"flooded"?) and unavailable.

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[asterisk-users] Digium IP Phones UNREACHABLE after registration

[Hermann Wecke]

I'm trying to solve a mystery for the last couple of days.

I have a mix of D70, D50 and D40 behind NAT. Server is in a
colocation, not a VPS.

For several years, everything was working fine, no issues. A few days
ago I started having problems at one particular site. NO CHANGES have
been made to this office network - same router, switch and internet
provider. No new equipment added or configuration changed (I only
upgraded the firmware and asterisk trying to solve the problem).

A few seconds after registration, the Digium phones will become
UNREACHABLE. Right after that, the entire VoIP network (where the
Digiums are located) will be also dropped - all other devices
(non-Digium) connected will be kicked from the asterisk box. There are
ObiHai, Yealink and Linksys at this location - all will be kicked.

The server remains operational and all other users/peers (not running
Digium phones) are up and running.

Sip debug and tcpdump didn't show any relevant information to solve
the puzzle. I also replaced the router (twice, different models and
firmware versions), PoE switch and cable modem. No success. IP is
dynamic but the provider will only change it once a year. Qualify=yes
(or no) didn't fix. Removing the password (deny/permit IP) also
didn't.

Running Asterisk 13.20.0, firmware 2.6.2.

Any ideas where I should dig further?

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Re: [asterisk-users] Dock-N-Talk with Asterisk, Anyone?

[Hermann Wecke]

Jeng Yu wrote:
 I would like to hear if anyone out there in Asteriskland has used the
 Dock-N-Talk (DNT) box to connect cell phones to Asterisk box.

The only problem I noticed is that after a random amount of time the box
will lost contact/synch with the cell phone. I'm using DockNTalk for
about 2 or 3 years, and this is happening after 1 or 2 weeks. After a
power cycle the DNT will work again.

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Re: [asterisk-users] Asterisk-1.2.22 DeadAGI Hangup

[Hermann Wecke]

Arun Kumar wrote:
 I've upgraded my server to asterisk-1.2.22 from 1.2.10 after that my 
 DeadAGI scripts are not working properly. Like after hangup I used to do 
 some more work now its not working.

Try, at your own risk, this:
http://svn.digium.com/view/asterisk/branches/1.2/res/res_agi.c?r1=71065r2=71656

Original message from lamer on Fri Jul 27, 2007 5:36 am:
This happens due to change in res_agi behaviour. Thus, EXEC DIAL will 
hangup straight away even SIGHUP is ignored as EXEC DIAL works as an 
underlying app since 1.2.20 (and probably 1.4.8).

Dial with 'g' seems to solve half of the problem but there are some side 
effects.

It's currently reported here http://bugs.digium.com/view.php?id=10315

Solution is to revert the change in:
http://svn.digium.com/view/asterisk/branches/1.2/res/res_agi.c?r1=71065r2=71656

Source: http://forum.asterisk2billing.org/viewtopic.php?p=8118#8118

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Re: [asterisk-users] Best wifi IP phone for asterisk

[Hermann Wecke]

Michelle Dupuis wrote:
 We're looking at a large wifi phone deployment, and we're looking for 
 wifi phones that:
  
 1. Are SIP compliant (Asterisk friendly)
 2. Provision capable (ideally TFTP of own MAC address)
 3. Industrial quality (no cheap plastic stuff).
 4. Well documented (and none of the only telco's get documentation crap)

NOT SIP, but AFAIK supported by Asterisk: Cisco IP Phone 7920.
SCCP, basic functions (answer, place a call, transfer) does work 
according to voip-info page.
Price: starting at US$ 150 @ eBay. US$ 345 NIB. If it is a large 
purchase and you are in the US, Cisco does offer a very competitive 
lease financing.

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Re: [asterisk-users] GSM Cards for Asterisk (UK)

[Hermann Wecke]

Matt Brown wrote:

Does anyone have any experience with a GSM card, preferably Quad Span
(4 GSM modules or higher) for use in the UK. I have seen the
Junghanns* version but I am not keen on the limitation of having to
use a BriStuffed version of Asterisk.


I'm buying this one to test:
http://www.portech.com.tw/eweb/MV-372/mv372.htm
made by Portech:
http://www.portech.com.tw/eweb/product_index.htm

Anyone is using this device?
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Re: [asterisk-users] Apple IPhone mobile is released in India?

[Hermann Wecke]

Crazy Boy wrote:
If IPhone is released in India, Can you tell me any Apple authorized 
showroom in Hyderabad (Andhrapradesh, India)?


Oh gosh... another troll... Google IS your friend:
http://www.google.com/search?q=apple+iphone
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Re: [asterisk-users] saydigits in another language

[Hermann Wecke]

Julian Lyndon-Smith wrote:

however, I get no errors, but still get the default Allison sounds
for the digits. Anyone got any clues on what I'm doing wrong ?


1) Create a directory named your_country_iso_code (AR|MX|ES|ETC) [1] 
under the main sounds directory (/var/lib/asterisk/sounds/ ???);

2) Also remember to create the same subdirectory under every other main
directory (letters, digits, phonetic etc);
3) Copy/move the newly recorded messages into these new directories - 
numbers into digits.



exten = foo,1,Set(CHANNEL(language)=custom)
exten = foo,2,SayDigits(1234567890)


Instead of custom use the ISO code. [1]

[1] http://preview.tinyurl.com/btkp
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Re: [asterisk-users] sip_header=value?

[Hermann Wecke]

Rizwan Hisham wrote:

is there anyway i can set SIP_HEADER(To) to the value i like?


If voip-info is correct, you can read, but you can't change.
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header
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Re: [asterisk-users] cisco 7905

[Hermann Wecke]

Khaled Chehab wrote:
How to configure cisco 7905 with asterisk ,if you please can send me 
step by step configuration steps .


This electronic message and its attachments are solely addressed to 
the addressee(s), and contain confidential information protected from

 disclosure belonging to Xplorium.


Sorry, can't help you because of this BS. If you want help, repost
without this crap.
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[asterisk-users] Asterisk x Mera MVTS

[Hermann Wecke]

I'm having trouble to send calls to a Mera MVTS softswitch (with SIPHIT) 
when the asterisk box has a dynamic IP address.


If the Asterisk box has a fixed IP, everything is OK.

Any ideas? I'm looking for a working sample of the sip.conf in this 
case... user.cfg (for MVTS) is also appreciated if any special setting 
should be done there also.

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Re: [asterisk-users] Problem in using Two BRi Cards in Asterisk

[Hermann Wecke]

Farooq Ahmed wrote:

And any idea about the issue on card one... means why outgoing is not working.


Not quite sure if Traverse Technology Netjet ISDN-s will really work.
Last time I had to use a ISDN BRI I bought one with Cologne chipset and 
used bristuff. Worked like a charm...

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Re: [asterisk-users] Cisco 7970 with skinny on * 1.4.1

[Hermann Wecke]

Richard Klingler wrote:

Has any1 got their 7970 to work with * 1.4.x ?


Why don't you use 7970 with SIP firmware? I'm running SIP 8.0.4SR2
without problems (Asterisk 1.2.16). Just remember that 7970 only will
register if your Asterisk is at the same network - no NAT between them - 
check http://preview.tinyurl.com/345fmj

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Re: [asterisk-users] asterisk on debian

[Hermann Wecke]

Josu Lazkano Lete wrote:

I need to download the sources or just with apt-get install is
enought???


apt-get is the easiest way, but won't give you the latest release.
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Re: [asterisk-users] zttool always reports OK on TDM400P

[Hermann Wecke]

Yuan LIU wrote:
Just noticed that no matter what the line condition is, zttool always 
reports OK, so it's pretty useless. (In contrast, I'd get Red alert 
if I unplug the line connecting to an X100P.)


This is the normal behavior. Only X100P will report the real status.
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Re: [asterisk-users] Cisco 7912

[Hermann Wecke]

Matt Putnam wrote:

anything useful any sugestions?


Are they requesting anything via TFTP? Do you have the full tftp files 
ready?

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Re: [asterisk-users] Cisco 7912

[Hermann Wecke]

Tom Lynn wrote:

Do they appear to have failed as a result of Daylight Savings time?


DST for 7905/7912 are set inside the lddefault/gkdefault - or the 
individual config file (ldMAC / gkMAC), but can't be set in advance like 
7940/7960. DST is not the reason here...

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Re: [asterisk-users] Adding 4 more POTS lines

[Hermann Wecke]

Jim Freeze wrote:

I suppose that is my alternative - remove the 4FXO card and add an
8FXO card. But I'm not seeing the prices you list. The Digium
TDM2402B is listed at $837.00. Am I missing something?


Digium is releasing a new 8 FXO/FXS card TDM800P, based on the same 
expansion cards used for TDM2400. This new card has been announced on 
the IT Expo East, which ended last Friday in Fort Lauderdale, FL.

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Re: [asterisk-users] International Provider

[Hermann Wecke]

Carlos Rojas wrote:

Anyone know a good carrier of voip for international calls?


Please use asterisk-biz list
http://lists.digium.com/mailman/listinfo/asterisk-biz
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Re: [asterisk-users] AdvancedVoIP Billing ?

[Hermann Wecke]

Noc Phibee wrote:

after 2 mounth of search, i don't have see a billing solution
for my small business..


Not quite sure as I didn't research very much their product, but did you 
check Aradial?

http://www.aradial.com/voip-billing-radius.html
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Re: [asterisk-users] g729 codec help

[Hermann Wecke]

programming dept wrote:

What happens is that if we terminate calls to carriers who accept
only the g729 codec we get a 503 service unavailable.


are you sure that your carrier will accept g.729? Sometimes they don't 
accept under iax2 and do accept under sip... check your debug for more 
information about this specific error also - maybe it is something not 
related to your g.729 codec/license


codecs are allowed into your sip.conf or iax.conf?

[general]

disallow=all
allow=g729

or under your peer/friend config:

[blahblahblah]
type=...
disallow=all
allow=g729
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Re: [asterisk-users] CDR - mysql with asterisk 1.2.12 not working

[Hermann Wecke]

On Sun, Oct 08, 2006 at 10:04:51PM -0600, Joseph wrote:
 What am I missing?

Maybe your /etc/mysql/my.cnf ?

# Instead of skip-networking you can listen only on
# localhost which is more compatible and is not less secure.
# bind-address  = 127.0.0.1
#skip-networking
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Re: [asterisk-users] CDR - mysql with asterisk 1.2.12 not working

[Hermann Wecke]

On Sun, Oct 08, 2006 at 10:39:26PM -0600, Joseph wrote:
 I have bind-address  = 127.0.0.1 in my.cnf
 the cdr was working find with asterisk 1.0.1 just after upgrade
 something is not connecting.

I don't know if asterisk will use the localhost or the network IP to
connect. Just try to comment your line and see what happens. This is really
a guess... 
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Re: [asterisk-users] Dropping extra frame of G.729 ?

[Hermann Wecke]

Noc Phibee wrote:

anyone know where i can solve this problems ? :


1) By doing a quick google search;
2) By reading previous posts regarding the same issue;
3) By disabling VAD (Voice activity detection) in your device.
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Re: [asterisk-users] Call center reports

[Hermann Wecke]

Technical Support wrote:

Can someone point me to call center reports available from Asterisk?


http://queuemetrics.loway.it/
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Re: [asterisk-users] G729 Replacement Codec - FREE or may ne cheaper than existing one.

[Hermann Wecke]

PLEASE DON'T CROSS POST!

Kannaiyan Natesan wrote:
I heard of a news, that there is a replacement codec available for 
g729 and accept the g729 codec data for decoding. [...] If there is 
any royalty need to pay, is that cheaper than the existing g729 
cost?.


G729 is not royalty free.
http://lists.digium.com/pipermail/asterisk-dev/2004-July/005544.html
http://lists.digium.com/pipermail/asterisk-users/2006-August/162221.html
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Re: [asterisk-users] Sipura SPA3000

[Hermann Wecke]

Michael Strelnikov wrote:
1. I want all incoming calls are redirected from SPA3000 to my 
asterisk server. 2. Asterisk then should direct this call to my SIP 
phones (including Sipura) 3. In case asterisk server is down I want 
that call be directed straight to the handset connected to the Sipura

 Is this configuration possible?


Yes for 1 and 2, never tested the #3.
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[asterisk-users] Cisco 7960G SIP firmware 8.4

[Hermann Wecke]

Cisco released last Aug 23 the latest SIP firmware for Cisco 7960G. Any 
info?


SIP Flash Image for 7940/7960 IP Phone v8.4(0) - Non CallManager
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Re: [asterisk-users] Portuguese sound files available?

[Hermann Wecke]

Ricardo Carvalho wrote:
I've been searching for sound files in Portuguese language to use in 
Asterisk for example for voicemail, but I couldn't find anything...
Does anyone know where I could find them for download, if there is such 
thing already?


Brazilian Portuguese only...
http://www.google.com/search?q=asterisk+sound+files+site%3Avoip-info.org
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Re: [asterisk-users] OT: Changing Cisco tftp root directory

[Hermann Wecke]

Julian Lyndon-Smith wrote:
Is there any way of specifying a directory to load tftp files from 
instead of from the root tftp directory when booting a cisco 7960 phone ?


SIPDefault.cnf:

# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: ./7960/ ; Example:  ./sip_phone/

/etc/dhcpd.conf:

option tftp-server-name 10.0.0.1;

running debian sarge, tftpd 0.17-12
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Re: [asterisk-users] Asterisk time not the same as unit time ?

[Hermann Wecke]

Andre Courchesne - Consultant wrote:

   [EMAIL PROTECTED] tmp]# date
   Mon Aug 14 16:44:15 EDT 2006
 The Linux command line time is connect, but not Asterisk...


just guessing... not sure:
date -u
is showing what?
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Re: [asterisk-users] CDR Variable

[Hermann Wecke]

Abdul wrote:
Could any one tell me how i can change CDR variable value from 
extentions.conf file.
 
for the example i would like to change the src field value different 
that caller phone on the first attempt of call?


exten = blabla,1,Set(CDR(fieldname)=new_value) (for asterisk = 1.2)

http://www.google.com/search?q=set+cdr+site%3Avoip-info.org
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Re: [asterisk-users] Inbound Calls SIP/2.0 404 Not Found

[Hermann Wecke]

Mr. Jones wrote:
I have 20 DIDs, some I want to send to a menu, most directly to an 
extension.


sip debug is (really) your friend. It should give you the [context]
where your DID is being send to and the 404 not found error also.

A particular line to look for: Looking for 
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Re: [asterisk-users] Callback and Asterisks

[Hermann Wecke]

Vic wrote:
 I am in immediate need of configuring an Asterix to act as wake up call 
 system.

Amazing:
http://www.google.com/search?q=asterisk+wake+up+call+site%3Avoip-info.org
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Re: [asterisk-users] HFC-S Cards in the UK

[Hermann Wecke]

Ron Wellsted wrote:

I have been trying all the major distributors but they are all out of
stock with no dates for new stock to be delivered. 


As you are in the UK, why not talking directly to Billion? Maybe they 
can help: http://www.billion.uk.com/contact.htm

I'm also trying to find a new supplier after Solwise ran out of stock...
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Re: [asterisk-users] Japanese Sound Files

[Hermann Wecke]

Nhadie wrote:

Does anyone here have Japanese version of the asterisk sound files?


http://www.google.com/search?q=japanese+sound+files+site%3Avoip-info.org
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Re: [asterisk-users] Setting CALLERID on a residential telco line

[Hermann Wecke]

Andrea Spadaccini wrote:

Is there any hope to change the caller-id on a BRI line?


I guess you can do it within the range assigned to you. If you have 2 
numbers, you can choose between these two numbers. Not tested, as I have 
only 1 number here (and still fighting with the zaphfc: empty HDLC 
frame or bad CRC received error).

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Re: [asterisk-users] Check call duration on active call in CLI?

[Hermann Wecke]

voiplist wrote:

Is there a command to check the call duration of an active call in
the CLI?


show channels verbose
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Re: [Asterisk-Users] ISDN

[Hermann Wecke]

Mimmus wrote:

Could some goodwill man summarize this topic for me before I engage
myself in the rediscovery of warm water?


Read a topic posted a few days ago: ISDN BRI NetJet
You will find good advice there.
If you need to buy a Cologne chipset card, check here:
http://www.solwise.co.uk/isdn.htm
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Re: [Asterisk-Users] Packet8 and Asterisk, do they play nice?

[Hermann Wecke]

Grady Neely wrote:

Has anyone gotten Packet8 setup as a sip trunk for Asterisk?


I have it here. With a TDM400.
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[Asterisk-Users] ISDN BRI NetJet

[Hermann Wecke]

I'm trying to use a Teles (netjet) ISDN BRI card with asterisk 1.2.9.1

Anyone was able to use this card with asterisk? I couldn't find much 
information about it. Any help?

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Re: [Asterisk-Users] instalacion

[Hermann Wecke]

samuel wrote:
I am of Argentina, and I do not speak very well English, I cannot 
install asterisk in red hat 9.


Don't send HTML messages to the list.
Install [EMAIL PROTECTED] Please remember that [EMAIL PROTECTED] will erase all data on 
your HD.

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Re: [Asterisk-Users] FreeBSD Digium g.729 codec seg faults on rev 30652

[Hermann Wecke]

Kim Culhan wrote:

Was running the Digium FreeBSD g.729 codec until recently when the latest
Asterisk bits were obtained via svn:
svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk


MAYBE it is the same problem:

http://lists.digium.com/pipermail/asterisk-users/2006-April/147577.html
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Re: [Asterisk-Users] Is NuFone Really Dead?

[Hermann Wecke]

Andy Jefferson wrote:

Went to their site today. Site claims they are still in biz. What is
the story? What really happened to Nufone anyway?


I'm using them for almost 3 years now. They are having some problems 
with OLD DIDs and toll free numbers, but newly assigned are working 
fine. I ordered one new toll free and I'm waiting to my old one being 
reactivated.


Their support - despite other reports - is working fine. Every message I 
sent were responded. WHAT IS HAPPENING is that they configured their 
mail server wrong, and they are refusing to fix it. So, if your mail 
server refuses emails from other badly configured server, their reply 
will never reach you. Other than this TERRIBLE behavior, it is OK.

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Re: [Asterisk-Users] Now that Nufone is dead...

[Hermann Wecke]

Carlos Chavez wrote:

Now that Nufone is dead, what are other providers of 800 numbers that
work with Asterisk?


Nufone is NOT dead. It is working and I just added more funds into my 
account.
You may also consider Asterlink. I'm a new client there, their support 
is a little slow, sometimes irresponsive (you need to send several 
messages until they notice you), they also have a misconfigured mail 
server but other than these problems, so far so good.

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Re: [Asterisk-Users] Is NuFone Really Dead?

[Hermann Wecke]

Steve Totaro wrote:

In what way is their email server configured badly?


Wrong DNS entries.
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Re: [Asterisk-Users] FW: NuFone Update: DIDs

[Hermann Wecke]

Matt wrote:

Is there more to this story then we know?


No secrets, but at least some information may be found here:
http://www.nufone.net/press/
Latest update April 28.
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Re: [Asterisk-Users] Codec G729 / x86_64 bits.

[Hermann Wecke]

Jefferson Carvalho wrote:

I always used a compiled version for a x86 system
From [...]
Someone could help me on this?


Yes, the folks at Digium will be more than happy to help you.
Visit http://www.digium.com/en/products/voice/g729codec.php and get a 
licensed codec.

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Re: [Asterisk-Users] Cisco 7940/7960 SIP 8.2 Freely Downloadable

[Hermann Wecke]

Alexander Burke wrote:
Just in case anyone here hadn't noticed, Cisco is apparently making 
7940/7960 SIP 8.2 firmware freely downloadable by anyone:


8.2 isn't broken? Any comments?

http://lists.digium.com/pipermail/asterisk-users/2006-March/143501.html
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Re: [Asterisk-Users] Re: Cisco 7960 International

[Hermann Wecke]

Shaun wrote:
Well looks like the phone is sending some data...  I was unable to debug the 
problem however..

 Looking for 9011905326471222 in default (domain 204.10.xxx.xxx)

Do you have a pattern in the default context that will match 
9011905326471222 ?

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Re: [Asterisk-Users] Cisco 7960 International

[Hermann Wecke]

Shaun wrote:
I'm having a problem with my Cisco 7960 phones with the SIP image.  When i 
try to dial a international number i keep getting a busy signal but i dont 
see anything on the asterisk console (-vc) like i do when i dial 
local or long distance numbers.


sip debug peer your-phone-extension-number-here

and check your debug messages for what your phone is sending to asterisk.
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Re: [Asterisk-Users] faxing setup

[Hermann Wecke]

Corne Labuschagne wrote:

How do I setup faxing in asterisk


http://tinyurl.com/qddpf
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Re: [Asterisk-Users] OT: XML Content Manager for Cisco 79XX Phones

[Hermann Wecke]

Corey S. McFadden wrote:

PHP/MySQL based content manager for the Cisco 79XX series IP Phones


Any mailing list available for this project? I have some
questions/updates about this project...
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Re: [Asterisk-Users] Asterisk FXO Panasonic PBX

[Hermann Wecke]

Waldo Rubinstein wrote:
I'm looking for a reliable 2 FXO-port gateway to connect a Panasonic  
PBX to Asterisk. Can anyone recommend a stable and reliable one?


Use 2x Sipura SPA-3000 - and you will also get 2x FXS...
Or use a Digium TDM02B (2x FXO).
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Re: [Asterisk-Users] name that vendor...

[Hermann Wecke]

[EMAIL PROTECTED] wrote:
Well yeah, I had no intention of buying one, I was just wondering what 
the hell it actually was that the seller was trying to hide.


Their supplier?
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Re: [Asterisk-Users] Cisco 7912G through NAT, problems with tones detection.

[Hermann Wecke]

Diego Mariano Velo wrote:

Hi, i have a cisco 7912G with SIP firmware, its connect to the asterisk
through nat. The only problems is in the voice mailasterisk not
detect the tones, therefore i cant access to my voice mail extension.


Check the DTMF settings...

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

dtmfmode: inband | info | rfc2833
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Re: [Asterisk-Users] Database update after hangup

[Hermann Wecke]

Hermann Wecke wrote:
I'm having a little problem to update the database after a call was 
placed. I have several PSTN lines and I need to split the calls between 
them.

[...]

Any idea?


Solution: write to the database BEFORE the dial command. Worked very well.
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Re: [Asterisk-Users] International Dialing Code

[Hermann Wecke]

Innocent Evil wrote:

I am trying to download a list of international dialing codes.
Would anybody please post a link to get it


Google IS your friend. Did you try?

Google: international country code
Wikipedia: http://en.wikipedia.org/wiki/List_of_country_calling_codes
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[Asterisk-Users] Database update after hangup

[Hermann Wecke]

I'm having a little problem to update the database after a call was 
placed. I have several PSTN lines and I need to split the calls between 
them.


The approach I used didn't work:

[sipphone]

include = trunktest
; other rules here blah blah blah

[trunktest]

exten = _1800NXX,1,DBget(LAST=lastused/trunk)
exten = _1800NXX,2,GotoIf($[${LAST} = 1]?20:3)
exten = _1800NXX,3,GotoIf($[${LAST} = 2]?30:4)
exten = _1800NXX,4,GotoIf($[${LAST} = 3]?40:10)
;
exten = _1800NXX,10,Set(used=1)
exten = _1800NXX,11,Dial(${TRK1}/${EXTEN})
exten = _1800NXX,12,DBput(lastused/trunk=1)
exten = _1800NXX,13,Hangup
;
exten = _1800NXX,20,Set(used=2)
exten = _1800NXX,21,Dial(${TRK2}/${EXTEN})
exten = _1800NXX,22,DBput(lastused/trunk=2)
exten = _1800NXX,23,Hangup
[]

; I also tried

exten = h,1,Set(DB(lastused/trunk)=${used})
exten = h,2,Hangup

Any idea?
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[Asterisk-Users] Clipcomm CG-410 and caller-id from PSTN

[Hermann Wecke]

Does anyone know if Clipcomm CG-410 [1] is able to handle caller-id 
information from PSTN and send it to Asterisk? Any trick on asterisk 
side to handle it? I tried several configurations but none worked. TIA


[1] http://tinyurl.com/c6k4f
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Re: [Asterisk-Users] stop asterisk when Idle

[Hermann Wecke]

[EMAIL PROTECTED] wrote:

I need to reboot every day an asterisk box, but I would like to do that
only when asterisk is not doing anything.


I have no idea *why* do you need to reboot the machine every day.
What I do is a full asterisk restart - removing the modules and 
reinstalling them. My boxes uptime are around 320 days and counting.


Check 
http://lists.digium.com/pipermail/asterisk-users/2004-October/068512.html

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Re: [Asterisk-Users] Dial command in extensions

[Hermann Wecke]

Edwin Lam wrote:

is there anyway to make the dial command return and execute
the next line in the dial plan after the channel hangs up?


Try g:

exten = 1234,1,dial(SIP/1234,,g)
exten = 1234,2,do something

g: When the called party hangs up, exit to execute more commands in the 
current context.


http://www.voip-info.org/wiki-Asterisk+cmd+Dial

I'm not sure if this is what you are looking for...
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[Asterisk-Users] Sipura 3000 x special dialling pattern (pin code)

[Hermann Wecke]

I need to place a call using a pin code. To access an external line, 
the host PBX (a Ericsson MD-110) will require that I dial 
*72*pincode#phone_number to complete any (trunk) call.


When I send the number, my Sipura 3000 will reject the call with 
Forbidden - wrong password on authentication for INVITE (see below). 
All other calls sent to the Sipura box without the weird pattern are OK.


Any ideas?

=== PIN CODE ===

-- Executing NoOp(SIP/1022-f773, Call to PSTN - PIN CODE) in 
new stack
-- Executing Dial(SIP/1022-f773, 
SIP/[EMAIL PROTECTED]|90) in new stack

-- Called [EMAIL PROTECTED]
Jul 20 17:02:52 WARNING[7979]: chan_sip.c:6846 handle_response: 
Forbidden - wrong password on authentication for INVITE to 'Line 2 
sip:[EMAIL PROTECTED];tag=as4da311dc'

-- SIP/pstn-spa3k-61d5 is circuit-busy
  == Everyone is busy/congested at this time
-- Executing Busy(SIP/1022-f773, ) in new stack
  == Spawn extension (from-sip, 001888555, 103) exited non-zero on 
'SIP/1022-f773'
-- Got SIP response 481 Call Leg/Transaction Does Not Exist back 
from 10.50.0.20


=== REGULAR CALL ===

-- Executing NoOp(SIP/1022-0568, Call to PSTN number 5359 - 
pt9QAnP8) in new stack
-- Executing Dial(SIP/1022-0568, SIP/[EMAIL PROTECTED]|60|) in 
new stack

-- Called [EMAIL PROTECTED]
-- SIP/pstn-spa3k-d03e is ringing
-- SIP/pstn-spa3k-d03e answered SIP/1022-0568
-- Attempting native bridge of SIP/1022-0568 and SIP/pstn-spa3k-d03e
  == Spawn extension (from-sip, 5359, 2) exited non-zero on 
'SIP/1022-0568'


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Re: [Asterisk-Users] lost g729 lic

[Hermann Wecke]

altus wrote:
We installed a box a long time ago and they bought g729a licenses 
Now we want to upgrade and reinstall,whats going to happen with the

codec,if I give the box the same ip as always will it work?


The Digium g729 license is bonded to the MAC address of all the 
interfaces you have. If you change one NIC, it is gone. The IP address 
is not used for anything. If you reinstall your box, you need to 
re-register the codec.
Digium allows 2 registrations. After that, you need to contact them to 
reset the database.

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Re: [Asterisk-Users] QoS for improvements

[Hermann Wecke]

Kumara Jayaweera wrote:
I want to run VoIP in the same LAN (15 windows clients) which we use for
surfing the Internet.
Some magic words: QoS Asterisk HTB TC. Not easy to find good material 
over the internet, but Google may give you some ideas - how to use them 
is another problem, which you have to figure out alone, as there are a 
few resources to research.

Start here:
http://www.krisk.org/astlinux/misc/astshape
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[Asterisk-Users] TFTP question

[Hermann Wecke]

I'm trying to use TFTP to update the firmware on a Cisco ATA188 but I'm 
receiving this error:

May  1 06:51:50 mail2 in.tftpd[11499]: connect from 192.168.2.2
May  1 06:51:50 mail2 tftpd[11500]: tftpd: trying to get file: 
ata01234567890a
May  1 06:51:50 mail2 tftpd[11500]: tftpd: serving file from /tftpboot
May  1 06:51:50 mail2 in.tftpd[11501]: connect from 192.168.2.2
May  1 06:51:50 mail2 tftpd[11502]: tftpd: trying to get file: 
ata01234567890a
May  1 06:51:50 mail2 tftpd[11502]: tftpd: serving file from /tftpboot
May  1 06:51:50 mail2 in.tftpd[11503]: connect from 192.168.2.2
May  1 06:51:50 mail2 tftpd[11504]: tftpd: trying to get file: 
ata01234567890a
May  1 06:51:50 mail2 tftpd[11504]: tftpd: serving file from /tftpboot
May  1 06:51:55 mail2 tftpd[11500]: tftpd: read: Connection refused
May  1 06:51:55 mail2 tftpd[11502]: tftpd: read: Connection refused
May  1 06:51:55 mail2 tftpd[11504]: tftpd: read: Connection refused

hosts.allow and hosts.deny are empty, directory /tftpboot and files are 
readable by owner/group/others. Running tftpd (0.17-12) on Debian Sarge. 
Similar error message when running atftp.

Ideas?
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[Asterisk-Users] Sipura 2000 x dual g729 channels x other choices?

[Hermann Wecke]

I found a thread [1] last month about the poor/crappy g729 quality on 
Sipura units. Anyone noticed an improvement or the quality is still poor?

If the Sipura firmware/g729 offers no quality yet, who else is offering 
a dual channel g729 ATA? I heard about Uniden, but I have no reports 
about their ATA...

[1] Sipura g729 call quality to PSTN
http://lists.digium.com/pipermail/asterisk-users/2005-February/089309.html
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Re: [Asterisk-Users] Cisco 7960 SIP 7.4

[Hermann Wecke]

Chris Lee wrote:
Has anyone else upgraded to 7.4 and found that the date  time no
longer appears on the phone?
This problem was pointed at the SIPPhoneReleaseNotes7_4.pdf file.
What I noticed is that when the phone lost the internet connection the 
date/time will no longer be present on the phone.
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Re: [Asterisk-Users] Sipura 2000 x dual g729 channels x other choices?

[Hermann Wecke]

William Suffill wrote:
According to the small print in the bottom graphic:
http://www.sipura.com/products/spa2100.htm
The  SPA 2100 would give u 2 ports + 2 RJ45 as well as 2 G729
When I was placing an online order, I found this:
support for two concurrent calls using the G.729 codec (in a firmware 
release expected in the first quareter  of 2005) (sic)

The dual g729 is reality now or is planned for a near future? I found 
that the latest firmware is 2.0.5d, but no info about the features or 
when it was released (release notes only display a copyright notice of 
2003-2005).
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Re: [Asterisk-Users] Reg Asterisk

[Hermann Wecke]

Sys Admin wrote:
couldnt agree with u more !!
And, please, add another one to the list: PLEASE TRIM THE ^*[EMAIL PROTECTED] 
MESSAGE. TIA.

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Re: [Asterisk-Users] FWD to Vonage not working?

[Hermann Wecke]

Brian Dingman wrote:
The FWD - Vonage interconnect has been down for some time now. Vonage
claimed there was a secuity issue and pulled the plug. No word when/if
it will ever be working again.
So I'm guessing that FWD - Packet8 falls into the same problem? Not 
working here for a couple of weeks...

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Re: [Asterisk-Users] Broadvoice alternatives

[Hermann Wecke]

Vicky Shrestha wrote:
I have tried a lot of things to make broadvoice work with asterisk , but I 
failed each time.
I had some problems here, mainly because I was trying to use g729 and 
broadvoice will only accept g711. Other than that, configuration itself 
took about 10~15 minutes with some google search to fix my mistakes...
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Re: [Asterisk-Users] Cisco 7960 SIP boot takes 2 minutes?

[Hermann Wecke]

Tom wrote:
What times are others seeing for the load when you reboot a phone?
About the same here, but I don't care as I never reboot my phone (about 
once every month or two).
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Re: [Asterisk-Users] Cisco 7960 x Asterisk CVS-HEAD-03/13/05 - registration issues

[Hermann Wecke]

Rich Adamson wrote:
Looks like a couple of problems here. I don't believe the Cisco phone
handles md5, so remove that line.
As I told before, tried 3 different approaches:
1) password; md5;
2) password, no md5;
3) no password, no md5.
Only the third one worked. Trying to give SOME security, I added:
deny=0.0.0.0/0.0.0.0
permit=1.2.3.4/255.255.255.255
to the config.
In your sip.conf you have nat=yes, but in the sip show peers it is
saying Nat=N. That would imply that you need to stop asterisk
and restart it after making such changes. Reload does _not_ reread
all such changes, so don't use that until you have a solid understanding
of its use.
The config was reloaded using sip reload and by stoping and restarting 
asterisk. Both returned the same Nat=N. No changes noticed.
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Re: [Asterisk-Users] Cisco 7960 x Asterisk CVS-HEAD-03/13/05 - registration issues

[Hermann Wecke]

C F wrote:
how are you telling the cisco what the password is? TFTP?
TFTP (SIPmacaddress.cnf)
you will not see anything on * CLI unelss you do sip debug
And after sip debug I saw (among other lines):
[...]
Retransmitting #5 (NAT):
SIP/2.0 407 Proxy Authentication Required
[...]
SIP/2.0 401 Unauthorized
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[Asterisk-Users] Cisco 7960 x Asterisk CVS-HEAD-03/13/05 - registration issues

[Hermann Wecke]

After fighting with a Unable to create/find channel [1] [2], I gave up 
on my previous installation and rebuild my asterisk from CVS-Head. I 
guess the Debian package available today is broken somewhere (after a 
previous broken release made with an old libpri package), but now I'm 
having another issue with my 7960 registration (SIP v. 7.1).

The call is being (silent) rejected by asterisk, and the sip debug is 
showing:
[...]
Retransmitting #5 (NAT):
SIP/2.0 407 Proxy Authentication Required
[...]
SIP/2.0 401 Unauthorized

Even with set verbose 9 no message is displayed on console regarding 
invalid context, password, call attempt...

Digging the list, I found a message suggesting to remove the password 
from the sip.conf [3]. I did it and now the calls can be placed (I was 
always able to receive calls, even with the broken debian package I had 
before).

Is there *any* reason to this very strange behavior?
The specific extension sip.conf entry is:
[1234]
type=friend
host=dynamic
qualify=1500
username=1234
secret=yeah
auth=md5
context=cisco
nat=yes
disallow=all
allow=g729
I also tried some different approaches, like removing the auth=md5 tag 
and lately removing the password also. Only when no password is set I 
was able to place calls. I'm sure the password is the same in the phone 
and the sip.conf

In any scenery, I'm always seeing:
 sip show peers
Name/username  HostDyn Nat ACL Mask 
1234/1234  1.2.3.4 D   N  255.255.255.255 
Port Status
63415OK (982 ms)

which, I guess, means that the phone is registered with * and the 
password has been accepted.

Any ideas?
[1] 
http://lists.digium.com/pipermail/asterisk-users/2005-February/090364.html
[2] http://lists.digium.com/pipermail/asterisk-users/2005-March/092083.html
[3]
http://lists.digium.com/pipermail/asterisk-users/2004-September/064998.html
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Re: [Asterisk-Users] Where to get (cheap) VoIP

[Hermann Wecke]

Christian faucher wrote:
I read that, using a modem,I can use a standard phone line, and
convert that as input for Asterisk PBX, right?
Not that simple, not every modem, but yes.
Also, where can I get VOIP phones?
eBay
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Re: [Asterisk-Users] Cisco 7960 x g729 x Unable to create/find channel

[Hermann Wecke]

Guy Decarpentrie wrote:
Try to configure your Cisco type=friend in your sip.conf
It is already type=friend
[1234]
type=friend
username=1234
auth=md5
secret=supersecret
deny=0.0.0.0/0.0.0.0
permit=my_ip/255.255.255.255
canreinvite=no
reinvite=no
host=dynamic
dtmfmode=rfc2833
qualify=1800
mailbox=1234
disallow=all
allow=g729
nat=yes
context=cisco
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[Asterisk-Users] Cisco 7960 x g729 x Unable to create/find channel

[Hermann Wecke]

I'm trying to place a call from my Cisco 7960 and I'm receiving this error:
Mar  1 06:19:44 NOTICE[3060]: chan_sip.c:7399 handle_request: Unable to 
create/find channel
Mar  1 06:19:58 NOTICE[3060]: chan_sip.c:7399 handle_request: Unable to 
create/find channel

I can't place calls, but I can receive them:
mail*CLI sip show channels
Peer User/ANRCall ID  Seq (Tx/Rx)   Format
192.168.3.2  30182168101b16b  00102/0   g729
I tried to find some old messages about this error but I couldn't find 
any clue. Any ideas?
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Re: [Asterisk-Users] Servidor SIP

[Hermann Wecke]

Max wrote:
Pessoal estou querendo montar um servidor SIP para fazer testes [...]
wrong list. For Portuguese mailing list please subscribe to 
http://groups.yahoo.com/group/asteriskbr/

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Re: [Asterisk-Users] Zaptel Red Alarm

[Hermann Wecke]

Matthew Boehm wrote:
Is there a way for asterisk to notify you of this? Send an email? Send a
page? Call you?
Nagios (I believe now is called NetSaint) can do this and much more.
But you must have the power to configure it... after that, Nagios can 
send you an email, a pager, even call you and turn on your coffee 
machine... ;-)

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Re: [Asterisk-Users] Anyone had a Cisco 7970 working with Asterisk?

[Hermann Wecke]

Paul A Brown wrote:
Anyone had a Cisco 7970 working with Asterisk?
As 7970 uses SCCP, you can do it with asterisk. I did it with 7960.
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Re: [Asterisk-Users] Re: FAX

[Hermann Wecke]

Olaf Klein wrote:
Why not just kill yourself, fucking wannabe spammer? DIE DIE DIE
This is *REALLY* offtopic, but Isamar is the founder of Brazilian 
AntiSPAM - http://antispam.org.br/ and later http://spambr.org/
Does it matter here? I don't think so, but calling he (or even me) a 
spammer is really too bad! ;-)

Now can we return to the main issue of this thread? TIA.
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Re: [Asterisk-Users] chan_sip.c:7296 handle_request: Unable to create/find channel

[Hermann Wecke]

Roger Schreiter wrote:
But when dialing a number, I get:
Feb  2 09:44:45 NOTICE[20380]: chan_sip.c:7296 handle_request: Unable to 
create/find channel
After I installed my Digium g729 license, I'm trying to place a call 
from my Cisco 7960 and I'm receiving the same error:

Feb 19 09:47:06 NOTICE[25246]: chan_sip.c:7399 handle_request: Unable to 
create/find channel
Feb 19 09:47:07 NOTICE[25246]: chan_sip.c:7399 handle_request: Unable to 
create/find channel

I can't place calls, but I can receive them:
mail*CLI sip show channels
Peer User/ANRCall ID  Seq (Tx/Rx)   Format
192.168.3.2  30182168101b16b  00102/0   g729
g729 is available for asterisk:
mail*CLI show translation
 Translation times between formats (in milliseconds)
  Source Format (Rows) Destination Format(Columns)
 g723   gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex
   g723 - - - - - - - - - -
gsm - - 2 2 5 2 1101639
   ulaw - 7 - 1 5 2 1101639
   alaw - 7 1 - 5 2 1101639
   g726 - 9 4 4 - 4 3121841
  adpcm - 7 2 2 5 - 1101639
   slin - 6 1 1 4 1 - 91538
  lpc10 - 9 4 4 7 4 3 -1841
   g729 - 8 3 3 6 3 211 -40
  speex - 8 3 3 6 3 21117 -
   ilbc - 9 4 4 7 4 3121841
(I removed ilbc from posting to keep the line feed)
Did I miss something?
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Re: [Asterisk-Users] ATA's

[Hermann Wecke]

Matthew Boehm wrote:
[...] In the meantime, get a Sipura 2100, supports 2 729 calls and
has both WAN/LAN ports.
I was told that the Uniden DTA200 also supports 2 g729 calls. I'm buying 
one to test. Street price around US$ 90.
Another one with dual g729 channels is MTA V102. Street price US$ 100. 
Also will test this one.

I'm still looking for other units with dual g729 channels...
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Re: [Asterisk-Users] Debian way of compiling zaptel kernel modules

[Hermann Wecke]

Tzafrir Cohen wrote:
BTW: did I mention that we have binary packages for standard Debian
Sarge kernels in our apt source?
zaptel is the only package that never worked for me from apt-get. I need 
to download, compile and install the kernel (specially because the 
original debian install is pre 2.4.20), then download all the CVS (or 
whatever) files for asterisk and zaptel, compile-but-not-install the 
asterisk and then compile the zaptel.

Not terrible, but not quite easy for a beginner. Or did I miss something?
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[Asterisk-Users] Help with dial command and h, H and g parameters

[Hermann Wecke]

I'm trying to find some live examples on how to use the h, H and g 
parameters on the dial command 
(http://www.voip-info.org/wiki-Asterisk+cmd+dial)

Any ideas? I was testing with the code below but after pressing * 
nothing happens (only after a long pause the goodye file was played)

[testset]
exten = 1023,1,NoCDR()
exten = 1023,2,Dial(SIP/1023,30,Hg)
exten = h,1,Background(goodbye)
exten = h,2,Hangup
exten = i,1,Hangup
exten = t,1,Hangup
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Re: [Asterisk-Users] Cisco 7960 Beating a Dead Horse

[Hermann Wecke]

Dave Green wrote:
Following a top posted thread is a pain.
not trimming the useless part of a reply is another pain...
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Re: [Asterisk-Users] Regex in number dialed

[Hermann Wecke]

Brian West wrote:
exten = _8001133[12345789]XX.,1,Dial(SIP/france-gateway,60,tr)
or
exten = _8001133[1-57-9]XX.,1,Dial(SIP/france-gateway,60,tr)
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[Asterisk-Users] Softphone x G729 x IAX

[Hermann Wecke]

Is there any winblows softphone available offering g729 *and* IAX?
I couldn't find any http://www.voip-info.org/wiki-VOIP+Phones
The best choice should be dIAX, but it is only GSM.
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Re: IAX long distance... Re: [Asterisk-Users] Asterisk for home office

[Hermann Wecke]

Michael Graves wrote:
[...] Although there have
been a few (very few) times when I've notcied a brief pause after
dialing and found that it had in fact dialed out on the last possible
option.
[...]
The problem of your approach is that if you are out of credit with the 
first provider, your call will be dropped, not trying the next one, 
right? After all, I believe that ChanIsAvail 
(http://www.voip-info.org/wiki-Asterisk+cmd+ChanIsAvail) will only check 
if you can connect to that provider (ip route), not for available funding...

I'm using now something like this:
exten = _91NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:1},45)
exten = _91NXXNXX,2,PlayBack(beep)
exten = _91NXXNXX,3,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:1},45)
exten = _91NXXNXX,4,PlayBack(beep)
exten = _91NXXNXX,5,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:1},45)
exten = _91NXXNXX,6,PlayBack(beep)
exten = _91NXXNXX,7,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:1},45)
exten = _91NXXNXX,8,Playtones(congestion)
exten = _91NXXNXX,9,Wait(3)
exten = _91NXXNXX,10,Hangup
I know after every beep  that I changed the provider (out of credit? 
dialing error? no connection?), and if the call is ringing after 45 
seconds and I hear a beep, I will hangup. Not the best, but I believe is 
the best failover solution (for a small company/home office at least).

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Re: [Asterisk-Users] Asterisk + Satellite connection

[Hermann Wecke]

Eric Wieling aka ManxPower wrote:
What company are you using for your service?
Intelsat. But I'm not using it point-to-point as I'm not the primary 
contractor of this channel - I'm buying internet access.

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[Asterisk-Users] sipgate x asterisk: problems to receive PSTN calls?

[Hermann Wecke]

I noticed that I'm no longer able to receive calls from PSTN to my 
SipGate DID number.

I changed the sip.conf and extension.conf as per 
http://www.voip-info.org/tiki-index.php?page=Sipgate but the problem 
remains...

However, I can receive calls from another sipgate user. The problem is 
only affecting calls from the PSTN (DID). Anyone with the same problem?

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Re: [Asterisk-Users] Asterisk + Satellite connection

[Hermann Wecke]

Federico Gonzalez wrote:
I have an Asterisk with one local Cisco ATA and one remote Cisco ATA
connected to the Asterisk, the remore connection is a satellite link
with an 900ms delay.
This is the same delay I have here. Never less than 900, sometimes over 
1500 ms.
Check 
http://lists.digium.com/pipermail/asterisk-biz/2004-November/001243.html 
also.

Can anybody help me with this ?. Is there a delay parameter to set ?
This is my sip.conf setup for this phone over satellite (a Cisco 7960 to 
one server and a GS-102 to another server):

[1234]
type=friend
username=1234
auth=md5
secret=supers
canreinvite=no
reinvite=no
host=dynamic
dtmfmode=rfc2833
qualify=1200
mailbox=1234
disallow=all
allow=g729
nat=yes
context=mycontext
callerid=My Name 1234
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Re: [Asterisk-Users] Question on IXAy (IAXy actually)

[Hermann Wecke]

nkb wrote:
So, do I still need to have an Asterisk server connected to my IAXy even 
after I've made provision for it?
You can only connect IAXy to an asterisk server. Yours or from a VoIP 
provider.

Like, can I just carry this IAXy 
around(after provision) and just plug into any broadband connection and 
start making voip calls via my asterisk provider server?
Yes, as long as your service provider or your own server supports IAX2 
protocol... Any comments from anyone?

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Re: [Asterisk-Users] Playing reveived message WAV file

[Hermann Wecke]

Joseph wrote:
After somebody records a message asterisk notifies me and encloses the
WAV file.  Though I'm not sure if this is a WAV format. I can not play
it.
How to play received message?
Did you try to use Windows Merdia Player?
In other hand, if you are receiving a .GSM file, you can use the j2 program:
http://www.j2.com/jconnect/twa/page/download
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Re: [Asterisk-Users] Question on IXAy

[Hermann Wecke]

nkb wrote:
I was wondering if I could use IXAy to forward my call via the internet 
to my destination, something of similar function to SIPURA 3000?
The IAXy is similar to the Sipura 1000 or 2000, or the Cisco ATA 18x... 
You can use it to connect to a VoIP server with the IAX2 protocol 
(instead of SIP for the other ATA boxes).

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Re: [Asterisk-Users] Fw: Gift for Mark Spencer

[Hermann Wecke]

Steve Kann wrote:
[...] I've gotten 270 already:  [...]
I've got only 1. But... what is the main issue now? Is this topic just 
another (endless) troll or someone is trying to get some config help for *?

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Re: [Asterisk-Users] ATA186 V2.15.ms

[Hermann Wecke]

Damon Estep wrote:
[...] Contains a link you need for firmware.
Correct URL is http://www.voip-info.org/wiki-Asterisk+phone+cisco+ATA18x
URL:http://www.voip-info.org/wiki-Asterisk+phone+cisco+ATA18x
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