[Asterisk-Users] Number of Calls per Proxy on Cisco 7960G?

[Glenn Powers]

Does anyone know how many simultaneous calls per proxy I can 
recieve/place on a Cisco 7960G?

thanks,
glenn
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Re: [Asterisk-Users] Cisco 7940 Configuration

[Glenn Powers]

from the voip-info.org wiki:
add a file "SEP0.cnf.xml" in your /tftpboot with:

  
  
  
  
  
  2000
  
  SERVER_IP_ADDRESS
  
  
  
  
  
  
  
  
  P0S3-07-3-00
  
  
  

where  is the MAC address of your phone (usually on a small 
sticker on the back of the phone), SERVER_IP_ADDRESS is the server ip 
address.

you'll also want to change:
  
  P0S3-07-3-00
to the match the firmware and phone model that you are using.
cheers,
glenn
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Re: [Asterisk-Users] Reverse phone lookup interface with asterisk

[Glenn Powers]

Joe Presto wrote:
I've had some success setting up a crude "screen pops" using Jabber, and am
pretty happy with it. But I'd love to have it pop up with the full name and
address of the caller.
 

I'm working on screen pop functionality for XRMS, an open source CRM/SFA 
package. I would be very interested to see your jabber implementation.

The easiest way to do this is using ServiceObjects "GeoPhone" web service -
although it's expensive for small shops (free trial, but $50+/mo after
that).  The interface would be pretty simple using Perl and SOAP::Lite, if
one knew Perl.  Which, after a few hours of tinkering, I can safely say I
don't.
 

Where can I find out more about GeoPhone? (Google didn't produce any 
obvious results on the name.)

My question is twofold:
Has anyone written an interface to this by any chance?
Or even better -
Has anyone written an interface to Google's search, which appears to have
free reverse # lookup?
 

Are refering to a Google API implementation or just opening the google 
search page?
If you just want a search page, use 
http://www.google.com/search?q=$phone_number

cheers,
glenn
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Re: [Asterisk-Users] Cisco 7940G

[Glenn Powers]

Could you post your sip.conf and /tfpboot/* files? (wiht passwords removed)
off list would be fine. I'll take a look at them and see if I can find 
anything.
I just got my 7960G working. I love the phone, but they were clearly
designed to be managed in large groups and setting up one is nearly as hard
setting up one hundred of them.

cheers,
glenn
ps, I don't have the URL handy, but "standard" computer headsets can 
used with the 79xxx
phones (if they have a headset jack), you just need to make a cable 
adaptor. (Most stores want
~$100 for a "Cisco-Compatible" headset. $10 "Multimedia" headsets are 
compatible, they just
need a the plug adaptor. No electrical changes are needed.)

Chris TenHarmsel wrote:
I'm getting similar issues.  We have a 7940 that doesn't register with
* even though the config file is right and it gets it via TFTP.  I can
tell that it gets the config file correctly because the display name
of the phone shows up, but the little phone icon has an X next to it
and the phone can't receive incoming calls.  It can, however, place
outgoing calls.  A 'sip show peers' doesn't show the phone as
registered either.
I'd love to get this figured out.
 

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Re: [Asterisk-Users] E911 Testing !

[Glenn Powers]

Thanks, Brett, for the info! I actually /like/ the long winded descriptions.
FYI - In some places, the 911 dispatchers are the same people who answer 
the Sheriff's, Local PD and Fire phone numbers. So, simply calling the 
Sheriff's Dept. and saying that you just installed a new phone system 
and want to test 911 would be a good place to start.

Calling 911 and saying "oh, I'm just testing" would be a bad idea, 
although it happens *a lot* (older people mostly, from what I hear.)

While it's a bad idea, it's better than not testing it at all. 911 
dispatchers are well trained. They know how to handle all sorts of 
calls, including testing, info, "I lost my dog" and "I'm dying." If they 
/are/ busy and you say "testing" they can clear the call in a matter of 
seconds and get back to the emergencies at hand.

Obviously, if you're a CLEC, or someone who's going to be making several 
test calls, you'll want to establish a procedure with the dispatch 
center first.

As other posters have pointed out, it's always far better to test (even 
with bad procedures) than to not test and have the system fail in an 
emergency.

I've done volunteer work for emergency services and disaster agencies 
and the rule of thumb is *always*, "When In Doubt, Call It In!"

When calling _anyone_ involved in emergency services, be brief and to 
the point. And, in most cases, skip introducing yourself, your company, 
what your working, etc. Just say what you want and answer any questions 
directly and briefly. ie, call the Sheriff or local PD and say "I want 
to test 911 on my new phone system." Don't get into a long winded 
introduction. Also, when you're transfer to someone else (this may 
happen more times than you'd like). Always start by saying the same 
thing, "I want to test 911 on my new phone system."

This might sound like I'm stating the obvious, but emergency service 
workers are trained in effective, efficent communication. If you speak 
to them in the same way, you're immediately be considered "professional" 
instead of "someone I have to deal with."

Okay, that's my long winded post of the day. Hopefully, someone will 
find it useful.

cheers,
glenn
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Re: [Asterisk-Users] Cisco 7940G

[Glenn Powers]

[EMAIL PROTECTED] wrote:
Is it somehow possible to enable web-based configuration on them?  I made the
changes in the config file but it still doesn't register with Asterisk.
 

AFAIK, Cisco 79xx phones don't have web-based configuration. They have a 
telnet interface, though
it's enabled/disabled based on the config files the phone gets from the 
TFTP server.

cheers,
glenn
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[Asterisk-Users] Cisco IP Phones

[Glenn Powers]

I'm considering put this on the voip-info.org Wiki, but I thought I'd 
throw it out a few observations here first:

* Cisco IP Phones are designed for enterprise deployments.
They are designed to be provisioned by the hundred or thousand. They are 
not designed to be deployed for a single user or even a small office. 
Sure, they work great in either of these settings, but they require more 
knowledge and infrastructure than most small offices have.

If you're a consultant or reseller, buying one or two and spending an 
afternoon figuring out how to provision them makes sense. Once you know 
how to provision one, provisioning a hundred is not difficult.

If you're an end user or a small office, you're not going to need to 
provision a hundred, so the process for provisioning one is going to 
seem a bit overwhelming.

Other VoIP equipment is clearly designed for at-home installation, with 
web-based interfaces, etc.

I think people should be aware of this when comparing IP Phone options.
cheers,
glenn
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Re: [Asterisk-Users] OT: Headset for the Cisco 7960

[Glenn Powers]

Florian Overkamp wrote:
You can even make your own adapter if it has to be really cheap :)
 

Instructions for making an adaptor:
http://www.mml.uni-hannover.de/einhorn/headset/index_e.html
cheers,
glenn
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Re: [Asterisk-Users] Cisco IP Phones

[Glenn Powers]

Mike Dent wrote:
Hi Glenn,
What do you mean by "provisioning"? 
 

loading the config files, with proxy servers, usernames, passwords, etc.
cheers,
glenn
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Re: [Asterisk-Users] .call file creation

[Glenn Powers]

Dan Adams wrote:
I am curious partly because it has occurred randomly in my asterisk 
system. How does one go about creating a .call file for placing a call 
between two extensions/phones? I know this has been mentioned and is 
probably in one of the wikis somewhere, but I am unsure exactally how 
to go about doing it. Can anyone point me in the right direction.

http://voip-info.org/tiki-index.php?page=Asterisk%20auto-dial%20out
Here's a web interface (click-to-dial) for creating .call files (part of 
XRMS, an open source CRM package):

http://www.asterisk.org/
*
* copyright 2004 Glenn Powers <[EMAIL PROTECTED]>
* Licensed Under the Open Software License v. 2.0
*
*/
/*
*
* If you are using the sip.conf based lookupCID,
* Be sure to add crm_username(s) to your sip.conf file. See below.
*
* IF Asterisk is running on the same server as XRMS,
* MAKE SURE /var/spool/asterisk/outgoing is writable
* by your web server.
*
* IF asterisk is running on another server, use sftp
* to copy the file over.
*
*/
/*
* LookupCID :: ismaeljcarlo
* simple function to lookup extension number from sip.conf
*
* [EMAIL PROTECTED] created this function which looks up
* the value of crm_username from sip.conf and returns the proper extension.
*
*/
function lookupCID($thelookupCID) {
   $lookupCID_sip_array = parse_ini_file("/etc/asterisk/sip.conf", 
true);

   while ($v = current($lookupCID_sip_array)) {
   if (isset($v['crm_username'])){
   if($v['crm_username'] == $thelookupCID) {
   $thelookupCID = key($lookupCID_sip_array);
   return $thelookupCID;
   }
   }
   next($lookupCID_sip_array);
   }
}
/*
* End LookupCID
*/
// include the common files
require_once('../../include-locations.inc');
require_once($include_directory . 'vars.php');
require_once($include_directory . 'utils-interface.php');
require_once($include_directory . 'utils-misc.php');
require_once($include_directory . 'adodb/adodb.inc.php');
require_once($include_directory . 'adodb-params.php');
$con = &adonewconnection($xrms_db_dbtype);
$con->connect($xrms_db_server, $xrms_db_username, $xrms_db_password, 
$xrms_db_dbname);
// $con->debug = 1;

$session_user_id = session_check();
$session_username = $_SESSION['username'];
$msg = $_GET['msg'];
$contact_id = $_GET['contact_id'];
$company_id = $_GET['company_id'];
$phone = $_GET['phone'];
$phone_dial_prefix = "1";
$msg = urlencode(_("Dialing Phone Number: ") . $phone);
// Get contact name
$sql = "SELECT first_names,last_name from contacts
   WHERE contact_id = " . $contact_id . " LIMIT 1";
$rst = $con->execute($sql);
if ($rst) {
   if (!$rst->EOF) {
   $contact_name = urlencode($rst->fields['first_names'] . " "
 . $rst->fields['last_name']);
   }
}
// Get variables from the custom fields of the user's contact id.
$sql = "SELECT custom1, custom2, custom3 from contacts, users
   WHERE  users.user_id = " . $session_user_id . "
   AND contacts.contact_id = users.user_contact_id
   LIMIT 1";
$rst = $con->execute($sql);
if ($rst) {
   if (!$rst->EOF) {
   $channel = $rst->fields['custom1'];
   $extension_to_dial = $rst->fields['custom2'];
   $CID = $rst->fields['custom3'];
   }
}
// $sipCID = lookupCID($session_username);
// This is the file that will be passed to Asterisk
$dial_file_contents = "Channel:$channel$extension_to_dial
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Callerid: $CID
Context: xrms
Extension: $phone_dial_prefix$phone
Priority: 1
";
$filename = $xrms_file_root . "/tmp/outdial-$phone";
  if (!$handle = fopen($filename, 'w')) {
echo "Cannot open file ($filename)";
exit;
  }
  if (fwrite($handle, $dial_file_contents) === FALSE) {
  echo "Cannot write to file ($filename)";
  exit;
  }
  system("mv $filename /var/spool/asterisk/outgoing");
  fclose($handle);
// Create an Activity on Dial
header("Location: ../../activities/new-2.php?user_id=" . $session_user_id
   . "&activity_status=o&activity_type_id=1&contact_id="
   . $contact_id . "&company_id=" . $company_id . "&activity_title="
   . _("Call%20To%20") . $contact_name
   . "&return_url=/contacts/one.php?contact_id=" . $contact_id);
// if you don't want to create an activity on dial, use this instead:
// header("Location: 
$http_site_root/contacts/one.php?contact_id=$contact_id&msg

  if (fwrite($handle, $dial_file_contents) === FALSE) {
  echo "Cannot write to file ($filename)"

Re: [Asterisk-Users] Asterisk on sattelite link

[Glenn Powers]

[EMAIL PROTECTED] wrote:
marius baranescu <[EMAIL PROTECTED]> writes:
 

I have a running Asterisk box . It is running great 
My problem is that I can not get connected to the world :) . 
My only option available here is a satellite connection . 
I was testing different service providers but all of them are doing
firewalling and NAT so SIP, IAX are not working
I desperately need to get connected to the world :)) 
Please recommend me a good ISP for Middle East (permanent 2 way 
connection) , real IP adresses etc
   

I guess it depends on where you are in the Middle East.
 

It doesn't matter /where/ you are. It matters who your satellite 
provider is and what equipment they are using.

I worked at a satellite uplink facility for awhile. I recommend iDirect 
modems/routers (iDirect.net). They don't sell to anyone but large 
operators, but their equipment is very good. Constellation (Hub in 
Sweden; NOC in Traverse City, MI) is probably your best bet.

http://www.constellationnetcorp.com/
You can also try SES-Global ( http://www.ses-global.com/ ). I've worked 
with SES-Americom. I can't speak for SES-Global

Ping times on iDirect are ~600-800ms, low jitter, which for satellite is 
very good. Starband gives ping times around 2000ms on a good day and has 
*terrible* jitter. I've used Cisco ATA-186's with iDirect and many 
people could not tell them apart from landline.

If you have a satellite modem made by Gilat, don't even waste your time 
/trying/ to get VoIP to work...

Iraq was/is a popular destination for _many_ iDirect units.
Bear in mind that getting satellite gear INTO many countries can be 
difficult. Some countries (Jordan) require co-ordination through the US 
Embassy. Some countries have $4,000-$6,000 import duties on VSAT dishes.

cheers,
glenn

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Re: [Asterisk-Users] Re: Bellster - IAX-based interchange -- lets you call anywhere for free

[Glenn Powers]

"It's all fun and games until someone loses a week's salary."
Accidental misconfig can be VERY expensive. For example: someone decides 
to allow "free" 800# calls. Someone then calls 800-CALL-ATT and places 
repeated OPERATOR ASSISTED calls to Nepal. Even with a 30 minute limit, 
that is VERY expensive.

I can call Paris for $0.0145/min ($0.86/HOUR) [voipjet]. Why should I 
risk hundreds or thousands of dollars?

Plus, configuring * for an IAX service provider requires LESS skill than 
configuring BELLSTER.

I love the concept of Bellster, but (IMHO) the risks don't justify the 
benefits.

cheers,
glenn
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Re: [Asterisk-Users] Anyone having problems with LiveVoIP?

[Glenn Powers]

I ordered an 800# from LiveVoIP two days ago. I can register with 
Asterisk just fine, but when I call my 800#, I get a fast busy. I 
emailed support a day and a half ago and have heard NOTHING from them.

VoicePulse Connect and VoipJet both work great for me.
Someone on -users said "you get what you pay for" regarding LiveVoip. 
They couldn't have been more correct!

cheers,
glenn
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[Asterisk-Users] Asterisk auto-dial out deliver message

[Glenn Powers]

It appears that my choices for auto-dial out are one-call-at-a-time or 
all-calls-at-once. Does anyone have a solution for calling 100 people, 
10 at a time?

thanks,
glenn
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Re: [Asterisk-Users] IP Phone for IP PBX

[Glenn Powers]

Jay Wilton wrote:
I just purchased a Cisco 7960 IP phone and I couldn't be
more happier.
It supports 6 lines. You can usually find a good one off
eBay.
   

I thought asterisk could only register once per ip?  guess
i'll go read the wiki for a while longer. :)
 

I have a 7960G. It doesn't have six "lines," it can register with 6 
_proxies_, plus a backup, plus an emergency (the latter two don't have 
buttons.) Each of the six main proxies has it own "button." (You can 
register six times, with six different account, on the same proxy or six 
different proxies.) each proxy can accept two incoming calls, plus an 
outgoing call. You can also conference on the same "button" with the 
phone. You cannot (w/SIP 7.3) conference between proxies.

Before someone jumps up and "does the math" and says "it can handle 12 
incoming calls." Well, no. It can handle as many calls as your asterisk 
server can, but the user needs to use something like the "Flash Operator 
Panel" for Asterisk. Also, one or more of the proxies are often setup as 
"intercom" proxies (ie, they auto-answer and put the call on the speaker).

cheers,
glenn
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Re: [Asterisk-Users] RE: Answering Machine Function?

[Glenn Powers]

   Is this possible with asterisk?  Anyone have a sample dialplan?
-other than the problem outlined below I would try something like 

S,1,wait(20)
S,2,voicemail(uwhatever)
S,3,hangup
 

One issue I ran into with using * as an answering machine is that it 
can't just share a line with other phones. My first attempt at making an 
"Asterisk Answering Machine" was similiar to your approach above. 
However, * would _always_ answer the phone after 20 seconds, even if 
another phone on the line was picked up. I'd love to see a work-around 
for this other than "get a FXS card." FXO cards are cheap, FXS cards are 
not.

This isn't an issue for business uses. This is an issue for people who 
want to try out asterisk on a POTS line, but share that POTS line with 
other people. Even if someone did get an FXS card AND an FXO card, 
"other people" who share the line are NOT likely to be very tolerant of 
dial plan errors, server problems, etc.

This is an important issue for the asterisk community becuase it 
addresses how new users and new developers can start working with 
asterisk. There is no reason that "Carrier Grade" software can't /also/ 
be easy for a home user to setup and use.

cheers,
glenn
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Re: [Asterisk-Users] X100P Clone

[Glenn Powers]

Razvan Turtureanu wrote:
I'm new to asterisk and fror a cupple of days I heave been googleing 
the net for digium "clones", because it's very hard for me to get a 
digium card (X100P).
Does anyone Know another substitute for X100P (I know that intel based 
modem with chip 537/MD3200 is working but I did not find any of those) ?

ebay.com: search for "digium"
cheers,
glenn
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[Asterisk-Users] Re: [Asterisk-Dev] Asterisk Dialplan command "PPPD" released

[Glenn Powers]

Oskar Senft wrote:
Sirrix AG, Saarbrücken (manufacturer of the Sirrix.PCI4S0 4-port ISDN
card for Asterisk) has released the new Asterisk dialplan command PPPD
(app_pppd). It allows to connect a Linux PPP daemon to an arbitrary
digital (ISDN) Asterisk channel to provide RAS dialin and dialout.

This is great news! Thank you -- and your company -- for the efforts!
cheers,
glenn
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Re: [Asterisk-Users] Line sharing

[Glenn Powers]

Christian Schnell wrote:

Hi, I would like to have a VoIP-card share a physical phone line with 
some other analog device. When a call comes in it is supposed to be 
answered by that analog device, only occasionally I'd like the 
VoIP-card to "join" that conversation. Using the VoIP-card, I'd like 
to establish a three-way conference (external party, analog device, 
PC+IP-phone) and then also record that conversation digitally to a PC 
in the LAN. All three parties must be able to hear and speak freely.



I don't think you need special hardware to do this. I think you /may/ 
just need to patch the current Zaptel driver. Currently, using a 
Wildcard X100P, you can Answer() and Hangup() up the call. However, when 
"sharing the line" I've found that the driver will barge into many calls 
at somewhat random times. The driver does not appear to support 
"line-in-use" detection. So, basically you'd need to patch the driver to 
/only/ answer on an Answer() command. It would also be useful to have 
the driver report (and Asterisk support) ring indication, and CallerID 
without answering.


cheers,
glenn

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Re: [Asterisk-Users] external IVR

[Glenn Powers]

mohammad wrote:

 Is there anyway to load voicemail prompts from an external IVR server 
like a Tftp server?



Do you need to load them once or repeatly? All you need to do is to get 
GSM files into your asterisk sounds directory. It doesn't really matter 
(to asterisk) how you do that.


cheers,
glenn

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Re: [Asterisk-Users] How does Vonage support fax machines?

[Glenn Powers]

Deon wrote:


My boss is insisting we support fax, and I keep telling him that Fax over
IP is very unreliable and not recommended and his immediate come-back is
"Vonage does it." and it's very hard to figure out how.

I don't think Vonage does T.38, the Linksys/Sipura units they're using
doesn't support T.38 to my knowledge. 


That means they have to be using G.711Ulaw to send faxes. But how do they
compensate for packet loss/jitter/etc.
 



They don't. I've tried faxing through Vonage. It works *sometimes*. 
Vonage *claims* to do fax, but they don't do it even somewhat reliably. 
If someone thinks differently, have them try it. (Yes, it's just Fax 
over VoIP, ie encode the fax modem in g711u and hope for the best. If 
packets drop, the fax dies.)


cheers,
glenn

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Re: [Asterisk-Users] Calls authentication by IP address

[Glenn Powers]

VoIP Newbie wrote:


This setting is for making calls. How can incoming calls be
authenticated by Callers' IP address? It requires an AGI or similar,
am I right? Any advice to get one?

Thanks for help anyway.
 



Way back in the day, many UNIX systems authenticated users by IP 
address, then SysAdmins figured out it was a bad idea. (IP addresses can 
be spoofed.)


None the less, how are you recieving incoming calls? A guest account? 
host= in sip.conf & iax.conf applies to the client, not direction of 
call, so it should work fine.


cheers,
glenn

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Re: [Asterisk-Users] Any SIP hardphone recommendations?

[Glenn Powers]

Cisco 7960's work well and are highly recommended by many people, 
including myself. They have the qualities you list.


cheers,
glenn


Tim Karl wrote:


Hello,

Can anyone recommend a hardphone that has the following qualities...

Both headset and handset ports
Headset port has amplification built in
Supports SIP using G729

We are switching from a Nortel switch to Asterisk. If anyone is 
familiar with Nortel phones, the Nortel 2216 phone has the features 
we're looking for in a SIP phone. Any help would be greatly 
appreciated. Thank you.


--Timothy Karl
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[Asterisk-Users] monmp3thread: Request to schedule in the past?!?!

[Glenn Powers]

I keep getting this error every five minutes:
Apr  4 13:35:00 NOTICE[20551]: res_musiconhold.c:309 monmp3thread: 
Request to schedule in the past?!?!
Apr  4 13:35:01 NOTICE[20551]: res_musiconhold.c:309 monmp3thread: 
Request to schedule in the past?!?!
Apr  4 13:35:01 NOTICE[20551]: res_musiconhold.c:309 monmp3thread: 
Request to schedule in the past?!?!
Apr  4 13:35:01 NOTICE[20551]: res_musiconhold.c:309 monmp3thread: 
Request to schedule in the past?!?!

Apr  4 13:40:00 NOTICE[20551]: res_musiconhold.c:309 monmp3thread: 
Request to schedule in the past?!?!
Apr  4 13:40:00 NOTICE[20551]: res_musiconhold.c:309 monmp3thread: 
Request to schedule in the past?!?!
Apr  4 13:40:01 NOTICE[20551]: res_musiconhold.c:309 monmp3thread: 
Request to schedule in the past?!?!
Apr  4 13:40:02 NOTICE[20551]: res_musiconhold.c:309 monmp3thread: 
Request to schedule in the past?!?!

I'm running  CVS-v1-0-03/08/05-09:27:38. How can I fix this?
thanks,
glenn
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Re: [Asterisk-Users] Cisco 7960 SIP images

[Glenn Powers]

Sorry for the late followup, but I want to share my lovely Cisco experience.

First, after placing orders for the $8 contracts with both CDW and INSIGHT
and having both orders cancelled a week later (for some "supplier
problem"), I went with the $74 contract from INSIGHT (CDW wanted $84,
IIRC). I actually got that contract.

Then, I tried to "register" the phone, only to find that the
factory-applied serial number wasn't even in Cisco's database. (Another
phone's serial number from the same purchase worked fine.) I actually had
a Cisco customer support person tell me "once you give us a valid serial
number for the phone, we can open a case for the invalid serial number on
the phone." I was speechless.

I never had an issue with who owned the phone. I told Cisco it belonged to
a client (true) and I didn't know who purchased it. They seemed fine with
that.

Upgrading old (Circa 2000) Cisco 7960 phones is a joy in itself. They
don't actually follow any documented self-update procedure AND the
procedure they do follow changes significantly by current firmware
version. Plus, you can't upgrade directly from an old ( As a side note to the above (in the US), the contract reseller is suppose
>  to obtain the phone's serial number. If that serial number is not
> registered to the individual requesting the contract, the contract
> supposedly will not be issued. That process is apparently used to identify
> when used phones are sold via eBay (etc), and essentially says one does
> not have a valid software license therefore it cannot be placed on
> maintenance. (A software license cannot be transferred with the sale of a
> used phone or any of cisco's equipment.) That same process is used for all
> Cisco equipment,
> however some used equipment resellers have been able to find ways around it
> (one way or another).
>
>
> Once a maintenance contract number has been issued (regardless of whether
>  its on a piece of paper or email), that contract number has to be
> entered into a cisco system that tracks the number against a customer
> account. If you don't have a customer account, that process can't be
> completed either. Some resellers will create your account for you and
> others won't.
>
> Once the account has been created and the contract recorded, then the
> customer is granted access to the download sections of their site via their
> login/authentication process.
>
> So the bottom line is the process requires a fair amount of manual labor
> and for $8 (in the US), few resellers have any interest in the sales
> commission resulting from an $8 sale. (Guess that says if you're buying
> 500 contracts, one might receive a different level of reseller interest.)
>
>
> Regardless of whether we like it or not, cisco wrote the license terms
> and asterisk users are not going to change their "machine". It's obviously
>  written to discourage reselling used equipment without paying a
> re-certification fee, and that re-certification re-license process can get
> to be far more costly then simply purchasing their new equipment. Surprise
> surprise!
>
> I don't work for cisco or any of their resellers.
>

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[Asterisk-Users] Asterisk-cvs does not compile on Red Hat 9

[Glenn Powers]

I just got the lastest Asterisk-CVS and it does NOT compile on Red Hat 
9, neither does Asterisk-1.0.1 (or Asterisk-1.0). Asterisk-1.0RC2 
compiles fine. Here is the error:

gcc -c -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes 
-Wmissing-declarations -g  -Iinclude -I../include -D_REENTRANT 
-D_GNU_SOURCE  -O6 -march=i686   -DZAPTEL_OPTIMIZATIONS  
-DASTERISK_VERSION=\"CVS-HEAD-10/27/04-18:27:47\" 
-DASTERISK_VERSION_NUM=99 -DINSTALL_PREFIX=\"\" 
-DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\" 
-DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\" 
-DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" 
-DASTCONFPATH=\"/etc/asterisk/asterisk.conf\" 
-DASTMODDIR=\"/usr/lib/asterisk/modules\" 
-DASTAGIDIR=\"/var/lib/asterisk/agi-bin\" -DBUSYDETECT_MARTIN  
-Wno-missing-prototypes -Wno-missing-declarations   -DZAPATA_PRI   
-DIAX_TRUNKING   -DCRYPTO -fPIC  -o chan_zap.o chan_zap.c
chan_zap.c: In function `handle_init_event':
chan_zap.c:5717: `ZT_EVENT_POLARITY' undeclared (first use in this function)
chan_zap.c:5717: (Each undeclared identifier is reported only once
chan_zap.c:5717: for each function it appears in.)
chan_zap.c: In function `handle_pri_show_span':
chan_zap.c:8257: warning: unused variable `info_str'
make[1]: *** [chan_zap.o] Error 1
make[1]: Leaving directory `/home/glenn/src/asterisk-cvs/asterisk/channels'
make: *** [subdirs] Error 1

[EMAIL PROTECTED] asterisk]$ uname -a
Linux HOST 2.4.20-30.9 #1 Wed Feb 4 20:44:26 EST 2004 i686 i686 i386 
GNU/Linux

Any idea what is wrong?
thanks,
glenn
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Re: [Asterisk-Users] Asterisk-cvs does not compile on Red Hat 9

[Glenn Powers]

Brian West wrote:
Yes it does update your zaptel and your other stuff too and you'll be fine.
bkw
 

Thanks. That worked.
BTW, If anyone has this problem in the future, do what the docs say and:
cd zaptel; make install
just running "make" will NOT work (as in asterisk will not compile).   :\
cheers,
glenn
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Re: [Asterisk-Users] - ACAN - the Asterisk Comprehensive ArchiveNetwork (was RE: GPL thoughts)

[Glenn Powers]

Why not just use voip-info.org for this? It already seems to be the site 
for all things asterisk. Is there a problem with many people 
contributing dialplans, chunchs of dialplans, or AGI scripts to it? 
"Forking" is just as bad on the doc site as it is on the code side.

cheers,
glenn
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Re: [Asterisk-Users] iax.cc and/or Sixtel.net seems like IT IS A SCAM.

[Glenn Powers]

I have an 866# number with iax.cc. It works fine. It did take me a 
couple of days to get it and they did have some problem the first day it 
was active, but I was able to contact support via IM and email. They 
resolved the problems and the service is working fine for me. Although, 
I still haven't gotten the 734 DID that I ordered on 1-28. (The 866 
number was more important to me, so I don't mind the delays on the 734 
DID so much.)

Is IAX.cc / Sixtel.net a scam? No.
Do they have provisioning problems? Yes.
Does the service work once you have the numbers? Yes.
cheers,
glenn
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Re: [Asterisk-Users] FRS / FRS/GMRS 2-way radios as SIP clients

[Glenn Powers]

TC wrote:
Any one know of software that allows 2-way radios as VoIP(SIP) clients,
besides dingotel's usb & mic cable trick ?
http://www.dingotel.com/2way/requirements2way.asp
They might be ok if the SIP client was not hardcode to their own SIP proxy
Has anyone tried any hacks to get the 2-way radio SIP client to regsiter to
a * box.
hmm chan_frsgmfrs anyone? using the usb/mic cable under linux :)
 

*The Asterisk  app_rpt project
The integration of 2-way radio systems and reasonable telephony
*http://www.zapatatelephony.org/app_rpt.html
cheers,
glenn
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Re: [Asterisk-Users] FRS / FRS/GMRS 2-way radios as SIP clients

[Glenn Powers]

I'd also suggest looking into the GNU Radio Project.
It's an open source Software Defined Radio (SDR).
It's a R&D project, but it's making good progress.
http://www.gnu.org/software/gnuradio/
"Software radio is a revolution in radio design due to its ability to 
create radios that change on the fly, creating new choices for users. At 
the baseline, software radios can do pretty much anything a traditional 
radio can do. The exciting part is the flexibility that software 
provides you. Instead of a bunch of fixed function gadgets, in the next 
few years we'll see a move to universal communication devices. Imagine a 
device that can morph into a cell phone and get you connectivity using 
GPRS, 802.11 Wi-Fi, 802.16 WiMax, a satellite hookup or the emerging 
standard of the day. You could determine your location using GPS, 
GLONASS or both."

cheers,
glenn

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Re: [Asterisk-Users] Fedora Core 3?

[Glenn Powers]

Race Vanderdecken wrote:
I am running on an Intel Pentium 3, 1.5 GHz, mother board stuck inside
an old E-machine case and it is very happy... (I only wish I could find
a Okidata B4250 printer driver or a PCL-6 I could understand.)
 

http://www.linuxprinting.org/pipermail/okidata-list/2004q2/000359.html
cheers,
glenn
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Re: [Asterisk-Users] Re: FRS and GMRS via *

[Glenn Powers]

Rich Adamson wrote:
GMRS, FRS and MURS radios may not be interconnected with the PSTN (47 
CFR 95.141). There has been a lot of talk from lobbyists to clarify this 
rule, but as it stands you could conceivably connect a *private* network 
to GMRS or MURS radios (you can't make any plugins or modifications to 
an FRS radio that isn't type accepted with the radio, so connecting a 
phone line or * box would be out). The language is vague, see the 
history at http://www.provide.net/~prsg/ 
   

Would plugging into the headphone jack with a phone-patch-type device
be considered a modification for radios with vox capability?
 

I don't think that would be considered a "modification" (IANAL).
cheers,
glenn
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[Asterisk-Users] FRS & *: an actual business use

[Glenn Powers]

I've noticed a growing number of stores using FRS radios. It would make 
sense to interface (via soundcard/console driver, with the nessacary 
electrical conversion) a VOX FRS radio to asterisk to allow someone in 
the office to page/talk with people on the floor or warehouse. You could 
throw that call (ie, all the radios) into a meetme conference. Then, you 
could have people in the office either dial that extension and/or have 
some of them always in that conference on a speaker phone (muted usually).

While I haven't tried this (yet!), it does seem like it would be a 
useful feature. The restriction against PSTN interconnection would be 
met UNLESS your dialplan allowed outside (PSTN) callers into that 
conference. You /could/ allow remote softphone users into the conference.

cheers,
glenn
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Re: [Asterisk-Users] FRS & *: an actual business use

[Glenn Powers]

Rich Adamson wrote:
I've noticed a growing number of stores using FRS radios. It would make 
sense to interface (via soundcard/console driver, with the nessacary 
electrical conversion) a VOX FRS radio to asterisk to allow someone in 
the office to page/talk with people on the floor or warehouse. You could 
throw that call (ie, all the radios) into a meetme conference. Then, you 
could have people in the office either dial that extension and/or have 
some of them always in that conference on a speaker phone (muted usually).
   

Unless I'm not understanding your comments, the meetme conference isn't
needed (assuming all radios are on the same channel). The radios 
become the meetme for all practical purposes. (When the base radio
transmits, all remote radios listen.)
 

I had intended for the meetme conf to allow more than one /phone/ user 
to communicate with
the radio users at the same time.

cheers,
glenn
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Re: [Asterisk-Users] DyDNS + externip

[Glenn Powers]

Giovanni Powell wrote:
Can i use a domain name instead of an IP address for externip
(sip.conf) Because im using dynamic dns. Not sure what i'm trying to
achieve as yet but, i want to know if it is possible?
 

I do it and it seems to work.
-glenn
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